diff options
Diffstat (limited to 'media')
98 files changed, 11732 insertions, 2889 deletions
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk index 2c0c3a5..96755bb 100644 --- a/media/libmedia/Android.mk +++ b/media/libmedia/Android.mk @@ -53,7 +53,8 @@ LOCAL_SRC_FILES:= \ Visualizer.cpp \ MemoryLeakTrackUtil.cpp \ SoundPool.cpp \ - SoundPoolThread.cpp + SoundPoolThread.cpp \ + StringArray.cpp LOCAL_SRC_FILES += ../libnbaio/roundup.c diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 40ff1bf..9faa497 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -19,18 +19,13 @@ #define LOG_TAG "AudioRecord" #include <sys/resource.h> -#include <sys/types.h> - #include <binder/IPCThreadState.h> -#include <cutils/atomic.h> -#include <cutils/compiler.h> #include <media/AudioRecord.h> -#include <media/AudioSystem.h> -#include <system/audio.h> #include <utils/Log.h> - #include <private/media/AudioTrackShared.h> +#define WAIT_PERIOD_MS 10 + namespace android { // --------------------------------------------------------------------------- @@ -41,7 +36,9 @@ status_t AudioRecord::getMinFrameCount( audio_format_t format, audio_channel_mask_t channelMask) { - if (frameCount == NULL) return BAD_VALUE; + if (frameCount == NULL) { + return BAD_VALUE; + } // default to 0 in case of error *frameCount = 0; @@ -75,8 +72,7 @@ status_t AudioRecord::getMinFrameCount( AudioRecord::AudioRecord() : mStatus(NO_INIT), mSessionId(0), - mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT), - mProxy(NULL) + mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) { } @@ -89,14 +85,15 @@ AudioRecord::AudioRecord( callback_t cbf, void* user, int notificationFrames, - int sessionId) + int sessionId, + transfer_type transferType) : mStatus(NO_INIT), mSessionId(0), mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT), mProxy(NULL) { - mStatus = set(inputSource, sampleRate, format, channelMask, - frameCount, cbf, user, notificationFrames, false /*threadCanCallJava*/, sessionId); + mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, + notificationFrames, false /*threadCanCallJava*/, sessionId, transferType); } AudioRecord::~AudioRecord() @@ -111,11 +108,13 @@ AudioRecord::~AudioRecord() mAudioRecordThread->requestExitAndWait(); mAudioRecordThread.clear(); } - mAudioRecord.clear(); + if (mAudioRecord != 0) { + mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); + mAudioRecord.clear(); + } IPCThreadState::self()->flushCommands(); AudioSystem::releaseAudioSessionId(mSessionId); } - delete mProxy; } status_t AudioRecord::set( @@ -128,8 +127,32 @@ status_t AudioRecord::set( void* user, int notificationFrames, bool threadCanCallJava, - int sessionId) + int sessionId, + transfer_type transferType) { + switch (transferType) { + case TRANSFER_DEFAULT: + if (cbf == NULL || threadCanCallJava) { + transferType = TRANSFER_SYNC; + } else { + transferType = TRANSFER_CALLBACK; + } + break; + case TRANSFER_CALLBACK: + if (cbf == NULL) { + ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL"); + return BAD_VALUE; + } + break; + case TRANSFER_OBTAIN: + case TRANSFER_SYNC: + break; + default: + ALOGE("Invalid transfer type %d", transferType); + return BAD_VALUE; + } + mTransfer = transferType; + // FIXME "int" here is legacy and will be replaced by size_t later if (frameCountInt < 0) { ALOGE("Invalid frame count %d", frameCountInt); @@ -143,6 +166,7 @@ status_t AudioRecord::set( AutoMutex lock(mLock); if (mAudioRecord != 0) { + ALOGE("Track already in use"); return INVALID_OPERATION; } @@ -159,14 +183,16 @@ status_t AudioRecord::set( if (format == AUDIO_FORMAT_DEFAULT) { format = AUDIO_FORMAT_PCM_16_BIT; } + // validate parameters if (!audio_is_valid_format(format)) { - ALOGE("Invalid format"); + ALOGE("Invalid format %d", format); return BAD_VALUE; } mFormat = format; if (!audio_is_input_channel(channelMask)) { + ALOGE("Invalid channel mask %#x", channelMask); return BAD_VALUE; } mChannelMask = channelMask; @@ -200,6 +226,7 @@ status_t AudioRecord::set( size_t minFrameCount = 0; status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask); if (status != NO_ERROR) { + ALOGE("getMinFrameCount() failed; status %d", status); return status; } ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount); @@ -207,6 +234,7 @@ status_t AudioRecord::set( if (frameCount == 0) { frameCount = minFrameCount; } else if (frameCount < minFrameCount) { + ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount); return BAD_VALUE; } @@ -215,7 +243,7 @@ status_t AudioRecord::set( } // create the IAudioRecord - status = openRecord_l(sampleRate, format, frameCount, input); + status = openRecord_l(sampleRate, format, frameCount, input, 0 /*epoch*/); if (status != NO_ERROR) { return status; } @@ -233,7 +261,7 @@ status_t AudioRecord::set( mActive = false; mCbf = cbf; mNotificationFrames = notificationFrames; - mRemainingFrames = notificationFrames; + mRefreshRemaining = true; mUserData = user; // TODO: add audio hardware input latency here mLatency = (1000*mFrameCount) / sampleRate; @@ -244,117 +272,78 @@ status_t AudioRecord::set( mInputSource = inputSource; mInput = input; AudioSystem::acquireAudioSessionId(mSessionId); + mSequence = 1; + mObservedSequence = mSequence; + mInOverrun = false; return NO_ERROR; } -status_t AudioRecord::initCheck() const -{ - return mStatus; -} - -// ------------------------------------------------------------------------- - -uint32_t AudioRecord::latency() const -{ - return mLatency; -} - -audio_format_t AudioRecord::format() const -{ - return mFormat; -} - -uint32_t AudioRecord::channelCount() const -{ - return mChannelCount; -} - -size_t AudioRecord::frameCount() const -{ - return mFrameCount; -} - -audio_source_t AudioRecord::inputSource() const -{ - return mInputSource; -} - // ------------------------------------------------------------------------- status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession) { - status_t ret = NO_ERROR; - sp<AudioRecordThread> t = mAudioRecordThread; - ALOGV("start, sync event %d trigger session %d", event, triggerSession); AutoMutex lock(mLock); - // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed - // while we are accessing the cblk - sp<IAudioRecord> audioRecord = mAudioRecord; - sp<IMemory> iMem = mCblkMemory; - audio_track_cblk_t* cblk = mCblk; + if (mActive) { + return NO_ERROR; + } - if (!mActive) { - mActive = true; + // reset current position as seen by client to 0 + mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); - cblk->lock.lock(); - if (!(cblk->flags & CBLK_INVALID)) { - cblk->lock.unlock(); - ALOGV("mAudioRecord->start()"); - ret = mAudioRecord->start(event, triggerSession); - cblk->lock.lock(); - if (ret == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID, &cblk->flags); - } - } - if (cblk->flags & CBLK_INVALID) { - audio_track_cblk_t* temp = cblk; - ret = restoreRecord_l(temp); - cblk = temp; + mNewPosition = mProxy->getPosition() + mUpdatePeriod; + int32_t flags = android_atomic_acquire_load(&mCblk->flags); + + status_t status = NO_ERROR; + if (!(flags & CBLK_INVALID)) { + ALOGV("mAudioRecord->start()"); + status = mAudioRecord->start(event, triggerSession); + if (status == DEAD_OBJECT) { + flags |= CBLK_INVALID; } - cblk->lock.unlock(); - if (ret == NO_ERROR) { - mNewPosition = cblk->user + mUpdatePeriod; - cblk->bufferTimeoutMs = (event == AudioSystem::SYNC_EVENT_NONE) ? MAX_RUN_TIMEOUT_MS : - AudioSystem::kSyncRecordStartTimeOutMs; - cblk->waitTimeMs = 0; - if (t != 0) { - t->resume(); - } else { - mPreviousPriority = getpriority(PRIO_PROCESS, 0); - get_sched_policy(0, &mPreviousSchedulingGroup); - androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); - } + } + if (flags & CBLK_INVALID) { + status = restoreRecord_l("start"); + } + + if (status != NO_ERROR) { + ALOGE("start() status %d", status); + } else { + mActive = true; + sp<AudioRecordThread> t = mAudioRecordThread; + if (t != 0) { + t->resume(); } else { - mActive = false; + mPreviousPriority = getpriority(PRIO_PROCESS, 0); + get_sched_policy(0, &mPreviousSchedulingGroup); + androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); } } - return ret; + return status; } void AudioRecord::stop() { - sp<AudioRecordThread> t = mAudioRecordThread; - - ALOGV("stop"); - AutoMutex lock(mLock); - if (mActive) { - mActive = false; - mCblk->cv.signal(); - mAudioRecord->stop(); - // the record head position will reset to 0, so if a marker is set, we need - // to activate it again - mMarkerReached = false; - if (t != 0) { - t->pause(); - } else { - setpriority(PRIO_PROCESS, 0, mPreviousPriority); - set_sched_policy(0, mPreviousSchedulingGroup); - } + if (!mActive) { + return; + } + + mActive = false; + mProxy->interrupt(); + mAudioRecord->stop(); + // the record head position will reset to 0, so if a marker is set, we need + // to activate it again + mMarkerReached = false; + sp<AudioRecordThread> t = mAudioRecordThread; + if (t != 0) { + t->pause(); + } else { + setpriority(PRIO_PROCESS, 0, mPreviousPriority); + set_sched_policy(0, mPreviousSchedulingGroup); } } @@ -364,14 +353,11 @@ bool AudioRecord::stopped() const return !mActive; } -uint32_t AudioRecord::getSampleRate() const -{ - return mSampleRate; -} - status_t AudioRecord::setMarkerPosition(uint32_t marker) { - if (mCbf == NULL) return INVALID_OPERATION; + if (mCbf == NULL) { + return INVALID_OPERATION; + } AutoMutex lock(mLock); mMarkerPosition = marker; @@ -382,7 +368,9 @@ status_t AudioRecord::setMarkerPosition(uint32_t marker) status_t AudioRecord::getMarkerPosition(uint32_t *marker) const { - if (marker == NULL) return BAD_VALUE; + if (marker == NULL) { + return BAD_VALUE; + } AutoMutex lock(mLock); *marker = mMarkerPosition; @@ -392,13 +380,12 @@ status_t AudioRecord::getMarkerPosition(uint32_t *marker) const status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) { - if (mCbf == NULL) return INVALID_OPERATION; - - uint32_t curPosition; - getPosition(&curPosition); + if (mCbf == NULL) { + return INVALID_OPERATION; + } AutoMutex lock(mLock); - mNewPosition = curPosition + updatePeriod; + mNewPosition = mProxy->getPosition() + updatePeriod; mUpdatePeriod = updatePeriod; return NO_ERROR; @@ -406,7 +393,9 @@ status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const { - if (updatePeriod == NULL) return BAD_VALUE; + if (updatePeriod == NULL) { + return BAD_VALUE; + } AutoMutex lock(mLock); *updatePeriod = mUpdatePeriod; @@ -416,10 +405,12 @@ status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const status_t AudioRecord::getPosition(uint32_t *position) const { - if (position == NULL) return BAD_VALUE; + if (position == NULL) { + return BAD_VALUE; + } AutoMutex lock(mLock); - *position = mCblk->user; + *position = mProxy->getPosition(); return NO_ERROR; } @@ -427,7 +418,7 @@ status_t AudioRecord::getPosition(uint32_t *position) const unsigned int AudioRecord::getInputFramesLost() const { // no need to check mActive, because if inactive this will return 0, which is what we want - return AudioSystem::getInputFramesLost(mInput); + return AudioSystem::getInputFramesLost(getInput()); } // ------------------------------------------------------------------------- @@ -437,7 +428,8 @@ status_t AudioRecord::openRecord_l( uint32_t sampleRate, audio_format_t format, size_t frameCount, - audio_io_handle_t input) + audio_io_handle_t input, + size_t epoch) { status_t status; const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); @@ -447,7 +439,7 @@ status_t AudioRecord::openRecord_l( } pid_t tid = -1; - // FIXME see similar logic at AudioTrack + // FIXME see similar logic at AudioTrack for tid int originalSessionId = mSessionId; sp<IAudioRecord> record = audioFlinger->openRecord(input, @@ -470,133 +462,138 @@ status_t AudioRecord::openRecord_l( ALOGE("Could not get control block"); return NO_INIT; } - mAudioRecord.clear(); + if (mAudioRecord != 0) { + mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this); + mDeathNotifier.clear(); + } mAudioRecord = record; - mCblkMemory.clear(); mCblkMemory = iMem; audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); mCblk = cblk; - mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); - cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; - cblk->waitTimeMs = 0; + + // starting address of buffers in shared memory + void *buffers = (char*)cblk + sizeof(audio_track_cblk_t); // update proxy - delete mProxy; - mProxy = new AudioRecordClientProxy(cblk, mBuffers, frameCount, mFrameSize); + mProxy = new AudioRecordClientProxy(cblk, buffers, frameCount, mFrameSize); + mProxy->setEpoch(epoch); + mProxy->setMinimum(mNotificationFrames); + + mDeathNotifier = new DeathNotifier(this); + mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this); return NO_ERROR; } status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) { - ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL); + if (audioBuffer == NULL) { + return BAD_VALUE; + } + if (mTransfer != TRANSFER_OBTAIN) { + audioBuffer->frameCount = 0; + audioBuffer->size = 0; + audioBuffer->raw = NULL; + return INVALID_OPERATION; + } - AutoMutex lock(mLock); - bool active; - status_t result = NO_ERROR; - audio_track_cblk_t* cblk = mCblk; - uint32_t framesReq = audioBuffer->frameCount; - uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; - - audioBuffer->frameCount = 0; - audioBuffer->size = 0; - - size_t framesReady = mProxy->framesReady(); - - if (framesReady == 0) { - cblk->lock.lock(); - goto start_loop_here; - while (framesReady == 0) { - active = mActive; - if (CC_UNLIKELY(!active)) { - cblk->lock.unlock(); - return NO_MORE_BUFFERS; - } - if (CC_UNLIKELY(!waitCount)) { - cblk->lock.unlock(); - return WOULD_BLOCK; - } - if (!(cblk->flags & CBLK_INVALID)) { - mLock.unlock(); - // this condition is in shared memory, so if IAudioRecord and control block - // are replaced due to mediaserver death or IAudioRecord invalidation then - // cv won't be signalled, but fortunately the timeout will limit the wait - result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); - cblk->lock.unlock(); - mLock.lock(); - if (!mActive) { - return status_t(STOPPED); - } - // IAudioRecord may have been re-created while mLock was unlocked - cblk = mCblk; - cblk->lock.lock(); - } - if (cblk->flags & CBLK_INVALID) { - goto create_new_record; - } - if (CC_UNLIKELY(result != NO_ERROR)) { - cblk->waitTimeMs += waitTimeMs; - if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { - ALOGW( "obtainBuffer timed out (is the CPU pegged?) " - "user=%08x, server=%08x", cblk->user, cblk->server); - cblk->lock.unlock(); - // callback thread or sync event hasn't changed - result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); - cblk->lock.lock(); - if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID, &cblk->flags); -create_new_record: - audio_track_cblk_t* temp = cblk; - result = AudioRecord::restoreRecord_l(temp); - cblk = temp; - } - if (result != NO_ERROR) { - ALOGW("obtainBuffer create Track error %d", result); - cblk->lock.unlock(); - return result; + const struct timespec *requested; + if (waitCount == -1) { + requested = &ClientProxy::kForever; + } else if (waitCount == 0) { + requested = &ClientProxy::kNonBlocking; + } else if (waitCount > 0) { + long long ms = WAIT_PERIOD_MS * (long long) waitCount; + struct timespec timeout; + timeout.tv_sec = ms / 1000; + timeout.tv_nsec = (int) (ms % 1000) * 1000000; + requested = &timeout; + } else { + ALOGE("%s invalid waitCount %d", __func__, waitCount); + requested = NULL; + } + return obtainBuffer(audioBuffer, requested); +} + +status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, + struct timespec *elapsed, size_t *nonContig) +{ + // previous and new IAudioRecord sequence numbers are used to detect track re-creation + uint32_t oldSequence = 0; + uint32_t newSequence; + + Proxy::Buffer buffer; + status_t status = NO_ERROR; + + static const int32_t kMaxTries = 5; + int32_t tryCounter = kMaxTries; + + do { + // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to + // keep them from going away if another thread re-creates the track during obtainBuffer() + sp<AudioRecordClientProxy> proxy; + sp<IMemory> iMem; + { + // start of lock scope + AutoMutex lock(mLock); + + newSequence = mSequence; + // did previous obtainBuffer() fail due to media server death or voluntary invalidation? + if (status == DEAD_OBJECT) { + // re-create track, unless someone else has already done so + if (newSequence == oldSequence) { + status = restoreRecord_l("obtainBuffer"); + if (status != NO_ERROR) { + break; } - cblk->waitTimeMs = 0; - } - if (--waitCount == 0) { - cblk->lock.unlock(); - return TIMED_OUT; } } - // read the server count again - start_loop_here: - framesReady = mProxy->framesReady(); - } - cblk->lock.unlock(); - } + oldSequence = newSequence; - cblk->waitTimeMs = 0; - // reset time out to running value after obtaining a buffer - cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; + // Keep the extra references + proxy = mProxy; + iMem = mCblkMemory; - if (framesReq > framesReady) { - framesReq = framesReady; - } + // Non-blocking if track is stopped + if (!mActive) { + requested = &ClientProxy::kNonBlocking; + } - uint32_t u = cblk->user; - uint32_t bufferEnd = cblk->userBase + mFrameCount; + } // end of lock scope - if (framesReq > bufferEnd - u) { - framesReq = bufferEnd - u; - } + buffer.mFrameCount = audioBuffer->frameCount; + // FIXME starts the requested timeout and elapsed over from scratch + status = proxy->obtainBuffer(&buffer, requested, elapsed); + + } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); - audioBuffer->frameCount = framesReq; - audioBuffer->size = framesReq * mFrameSize; - audioBuffer->raw = mProxy->buffer(u); - active = mActive; - return active ? status_t(NO_ERROR) : status_t(STOPPED); + audioBuffer->frameCount = buffer.mFrameCount; + audioBuffer->size = buffer.mFrameCount * mFrameSize; + audioBuffer->raw = buffer.mRaw; + if (nonContig != NULL) { + *nonContig = buffer.mNonContig; + } + return status; } void AudioRecord::releaseBuffer(Buffer* audioBuffer) { - ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL); + // all TRANSFER_* are valid + + size_t stepCount = audioBuffer->size / mFrameSize; + if (stepCount == 0) { + return; + } + + Proxy::Buffer buffer; + buffer.mFrameCount = stepCount; + buffer.mRaw = audioBuffer->raw; AutoMutex lock(mLock); - (void) mProxy->stepUser(audioBuffer->frameCount); + mInOverrun = false; + mProxy->releaseBuffer(&buffer); + + // the server does not automatically disable recorder on overrun, so no need to restart } audio_io_handle_t AudioRecord::getInput() const @@ -616,215 +613,304 @@ audio_io_handle_t AudioRecord::getInput_l() return mInput; } -int AudioRecord::getSessionId() const -{ - // no lock needed because session ID doesn't change after first set() - return mSessionId; -} - // ------------------------------------------------------------------------- ssize_t AudioRecord::read(void* buffer, size_t userSize) { - ssize_t read = 0; - Buffer audioBuffer; - int8_t *dst = static_cast<int8_t*>(buffer); + if (mTransfer != TRANSFER_SYNC) { + return INVALID_OPERATION; + } - if (ssize_t(userSize) < 0) { - // sanity-check. user is most-likely passing an error code. - ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", - buffer, userSize, userSize); + if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { + // sanity-check. user is most-likely passing an error code, and it would + // make the return value ambiguous (actualSize vs error). + ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize); return BAD_VALUE; } - mLock.lock(); - // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed - // while we are accessing the cblk - sp<IAudioRecord> audioRecord = mAudioRecord; - sp<IMemory> iMem = mCblkMemory; - mLock.unlock(); - - do { + ssize_t read = 0; + Buffer audioBuffer; - audioBuffer.frameCount = userSize/frameSize(); + while (userSize >= mFrameSize) { + audioBuffer.frameCount = userSize / mFrameSize; - // By using a wait count corresponding to twice the timeout period in - // obtainBuffer() we give a chance to recover once for a read timeout - // (if media_server crashed for instance) before returning a length of - // 0 bytes read to the client - status_t err = obtainBuffer(&audioBuffer, ((2 * MAX_RUN_TIMEOUT_MS) / WAIT_PERIOD_MS)); + status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); if (err < 0) { - // out of buffers, return #bytes written - if (err == status_t(NO_MORE_BUFFERS)) { + if (read > 0) { break; } - if (err == status_t(TIMED_OUT)) { - // return partial transfer count - return read; - } return ssize_t(err); } size_t bytesRead = audioBuffer.size; - memcpy(dst, audioBuffer.i8, bytesRead); - - dst += bytesRead; + memcpy(buffer, audioBuffer.i8, bytesRead); + buffer = ((char *) buffer) + bytesRead; userSize -= bytesRead; read += bytesRead; releaseBuffer(&audioBuffer); - } while (userSize); + } return read; } // ------------------------------------------------------------------------- -bool AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) +nsecs_t AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) { - Buffer audioBuffer; - uint32_t frames = mRemainingFrames; - size_t readSize; - mLock.lock(); - // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed - // while we are accessing the cblk - sp<IAudioRecord> audioRecord = mAudioRecord; - sp<IMemory> iMem = mCblkMemory; - audio_track_cblk_t* cblk = mCblk; + + // Can only reference mCblk while locked + int32_t flags = android_atomic_and(~CBLK_OVERRUN, &mCblk->flags); + + // Check for track invalidation + if (flags & CBLK_INVALID) { + (void) restoreRecord_l("processAudioBuffer"); + mLock.unlock(); + // Run again immediately, but with a new IAudioRecord + return 0; + } + bool active = mActive; - uint32_t markerPosition = mMarkerPosition; - uint32_t newPosition = mNewPosition; - uint32_t user = cblk->user; - // determine whether a marker callback will be needed, while locked - bool needMarker = !mMarkerReached && (mMarkerPosition > 0) && (user >= mMarkerPosition); - if (needMarker) { - mMarkerReached = true; - } - // determine the number of new position callback(s) that will be needed, while locked + + // Manage overrun callback, must be done under lock to avoid race with releaseBuffer() + bool newOverrun = false; + if (flags & CBLK_OVERRUN) { + if (!mInOverrun) { + mInOverrun = true; + newOverrun = true; + } + } + + // Get current position of server + size_t position = mProxy->getPosition(); + + // Manage marker callback + bool markerReached = false; + size_t markerPosition = mMarkerPosition; + // FIXME fails for wraparound, need 64 bits + if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { + mMarkerReached = markerReached = true; + } + + // Determine the number of new position callback(s) that will be needed, while locked + size_t newPosCount = 0; + size_t newPosition = mNewPosition; uint32_t updatePeriod = mUpdatePeriod; - uint32_t needNewPos = updatePeriod > 0 && user >= newPosition ? - ((user - newPosition) / updatePeriod) + 1 : 0; - mNewPosition = newPosition + updatePeriod * needNewPos; + // FIXME fails for wraparound, need 64 bits + if (updatePeriod > 0 && position >= newPosition) { + newPosCount = ((position - newPosition) / updatePeriod) + 1; + mNewPosition += updatePeriod * newPosCount; + } + + // Cache other fields that will be needed soon + size_t notificationFrames = mNotificationFrames; + if (mRefreshRemaining) { + mRefreshRemaining = false; + mRemainingFrames = notificationFrames; + mRetryOnPartialBuffer = false; + } + size_t misalignment = mProxy->getMisalignment(); + int32_t sequence = mSequence; + + // These fields don't need to be cached, because they are assigned only by set(): + // mTransfer, mCbf, mUserData, mSampleRate + mLock.unlock(); - // perform marker callback, while unlocked - if (needMarker) { + // perform callbacks while unlocked + if (newOverrun) { + mCbf(EVENT_OVERRUN, mUserData, NULL); + } + if (markerReached) { mCbf(EVENT_MARKER, mUserData, &markerPosition); } - - // perform new position callback(s), while unlocked - for (; needNewPos > 0; --needNewPos) { - uint32_t temp = newPosition; + while (newPosCount > 0) { + size_t temp = newPosition; mCbf(EVENT_NEW_POS, mUserData, &temp); newPosition += updatePeriod; + newPosCount--; + } + if (mObservedSequence != sequence) { + mObservedSequence = sequence; + mCbf(EVENT_NEW_IAUDIORECORD, mUserData, NULL); } - do { - audioBuffer.frameCount = frames; - // Calling obtainBuffer() with a wait count of 1 - // limits wait time to WAIT_PERIOD_MS. This prevents from being - // stuck here not being able to handle timed events (position, markers). - status_t err = obtainBuffer(&audioBuffer, 1); - if (err < NO_ERROR) { - if (err != TIMED_OUT) { - ALOGE_IF(err != status_t(NO_MORE_BUFFERS), - "Error obtaining an audio buffer, giving up."); - return false; + // if inactive, then don't run me again until re-started + if (!active) { + return NS_INACTIVE; + } + + // Compute the estimated time until the next timed event (position, markers) + uint32_t minFrames = ~0; + if (!markerReached && position < markerPosition) { + minFrames = markerPosition - position; + } + if (updatePeriod > 0 && updatePeriod < minFrames) { + minFrames = updatePeriod; + } + + // If > 0, poll periodically to recover from a stuck server. A good value is 2. + static const uint32_t kPoll = 0; + if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { + minFrames = kPoll * notificationFrames; + } + + // Convert frame units to time units + nsecs_t ns = NS_WHENEVER; + if (minFrames != (uint32_t) ~0) { + // This "fudge factor" avoids soaking CPU, and compensates for late progress by server + static const nsecs_t kFudgeNs = 10000000LL; // 10 ms + ns = ((minFrames * 1000000000LL) / mSampleRate) + kFudgeNs; + } + + // If not supplying data by EVENT_MORE_DATA, then we're done + if (mTransfer != TRANSFER_CALLBACK) { + return ns; + } + + struct timespec timeout; + const struct timespec *requested = &ClientProxy::kForever; + if (ns != NS_WHENEVER) { + timeout.tv_sec = ns / 1000000000LL; + timeout.tv_nsec = ns % 1000000000LL; + ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); + requested = &timeout; + } + + while (mRemainingFrames > 0) { + + Buffer audioBuffer; + audioBuffer.frameCount = mRemainingFrames; + size_t nonContig; + status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); + LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), + "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); + requested = &ClientProxy::kNonBlocking; + size_t avail = audioBuffer.frameCount + nonContig; + ALOGV("obtainBuffer(%u) returned %u = %u + %u", + mRemainingFrames, avail, audioBuffer.frameCount, nonContig); + if (err != NO_ERROR) { + if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { + break; + } + ALOGE("Error %d obtaining an audio buffer, giving up.", err); + return NS_NEVER; + } + + if (mRetryOnPartialBuffer) { + mRetryOnPartialBuffer = false; + if (avail < mRemainingFrames) { + int64_t myns = ((mRemainingFrames - avail) * + 1100000000LL) / mSampleRate; + if (ns < 0 || myns < ns) { + ns = myns; + } + return ns; } - break; } - if (err == status_t(STOPPED)) return false; size_t reqSize = audioBuffer.size; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); - readSize = audioBuffer.size; + size_t readSize = audioBuffer.size; // Sanity check on returned size - if (ssize_t(readSize) <= 0) { - // The callback is done filling buffers + if (ssize_t(readSize) < 0 || readSize > reqSize) { + ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", + reqSize, (int) readSize); + return NS_NEVER; + } + + if (readSize == 0) { + // The callback is done consuming buffers // Keep this thread going to handle timed events and - // still try to get more data in intervals of WAIT_PERIOD_MS + // still try to provide more data in intervals of WAIT_PERIOD_MS // but don't just loop and block the CPU, so wait - usleep(WAIT_PERIOD_MS*1000); - break; + return WAIT_PERIOD_MS * 1000000LL; } - if (readSize > reqSize) readSize = reqSize; - audioBuffer.size = readSize; - audioBuffer.frameCount = readSize/frameSize(); - frames -= audioBuffer.frameCount; + size_t releasedFrames = readSize / mFrameSize; + audioBuffer.frameCount = releasedFrames; + mRemainingFrames -= releasedFrames; + if (misalignment >= releasedFrames) { + misalignment -= releasedFrames; + } else { + misalignment = 0; + } releaseBuffer(&audioBuffer); - } while (frames); + // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer + // if callback doesn't like to accept the full chunk + if (readSize < reqSize) { + continue; + } + // There could be enough non-contiguous frames available to satisfy the remaining request + if (mRemainingFrames <= nonContig) { + continue; + } - // Manage overrun callback - if (active && (mProxy->framesAvailable() == 0)) { - // The value of active is stale, but we are almost sure to be active here because - // otherwise we would have exited when obtainBuffer returned STOPPED earlier. - ALOGV("Overrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); - if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { - mCbf(EVENT_OVERRUN, mUserData, NULL); +#if 0 + // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a + // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA + // that total to a sum == notificationFrames. + if (0 < misalignment && misalignment <= mRemainingFrames) { + mRemainingFrames = misalignment; + return (mRemainingFrames * 1100000000LL) / mSampleRate; } - } +#endif - if (frames == 0) { - mRemainingFrames = mNotificationFrames; - } else { - mRemainingFrames = frames; } - return true; + mRemainingFrames = notificationFrames; + mRetryOnPartialBuffer = true; + + // A lot has transpired since ns was calculated, so run again immediately and re-calculate + return 0; } -// must be called with mLock and cblk.lock held. Callers must also hold strong references on -// the IAudioRecord and IMemory in case they are recreated here. -// If the IAudioRecord is successfully restored, the cblk pointer is updated -status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& refCblk) +status_t AudioRecord::restoreRecord_l(const char *from) { + ALOGW("dead IAudioRecord, creating a new one from %s()", from); + ++mSequence; status_t result; - audio_track_cblk_t* cblk = refCblk; - audio_track_cblk_t* newCblk = cblk; - ALOGW("dead IAudioRecord, creating a new one"); - - // signal old cblk condition so that other threads waiting for available buffers stop - // waiting now - cblk->cv.broadcast(); - cblk->lock.unlock(); - // if the new IAudioRecord is created, openRecord_l() will modify the // following member variables: mAudioRecord, mCblkMemory and mCblk. // It will also delete the strong references on previous IAudioRecord and IMemory - result = openRecord_l(mSampleRate, mFormat, mFrameCount, getInput_l()); + size_t position = mProxy->getPosition(); + mNewPosition = position + mUpdatePeriod; + result = openRecord_l(mSampleRate, mFormat, mFrameCount, getInput_l(), position); if (result == NO_ERROR) { - newCblk = mCblk; - // callback thread or sync event hasn't changed - result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); + if (mActive) { + // callback thread or sync event hasn't changed + // FIXME this fails if we have a new AudioFlinger instance + result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); + } } if (result != NO_ERROR) { + ALOGW("restoreRecord_l() failed status %d", result); mActive = false; } - ALOGV("restoreRecord_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", - result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); - - if (result == NO_ERROR) { - // from now on we switch to the newly created cblk - refCblk = newCblk; - } - newCblk->lock.lock(); + return result; +} - ALOGW_IF(result != NO_ERROR, "restoreRecord_l() error %d", result); +// ========================================================================= - return result; +void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who) +{ + sp<AudioRecord> audioRecord = mAudioRecord.promote(); + if (audioRecord != 0) { + AutoMutex lock(audioRecord->mLock); + audioRecord->mProxy->binderDied(); + } } // ========================================================================= AudioRecord::AudioRecordThread::AudioRecordThread(AudioRecord& receiver, bool bCanCallJava) - : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) + : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) { } @@ -842,10 +928,26 @@ bool AudioRecord::AudioRecordThread::threadLoop() return true; } } - if (!mReceiver.processAudioBuffer(this)) { - pause(); + nsecs_t ns = mReceiver.processAudioBuffer(this); + switch (ns) { + case 0: + return true; + case NS_WHENEVER: + sleep(1); + return true; + case NS_INACTIVE: + pauseConditional(); + return true; + case NS_NEVER: + return false; + default: + LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); + struct timespec req; + req.tv_sec = ns / 1000000000LL; + req.tv_nsec = ns % 1000000000LL; + nanosleep(&req, NULL /*rem*/); + return true; } - return true; } void AudioRecord::AudioRecordThread::requestExit() @@ -859,6 +961,17 @@ void AudioRecord::AudioRecordThread::pause() { AutoMutex _l(mMyLock); mPaused = true; + mResumeLatch = false; +} + +void AudioRecord::AudioRecordThread::pauseConditional() +{ + AutoMutex _l(mMyLock); + if (mResumeLatch) { + mResumeLatch = false; + } else { + mPaused = true; + } } void AudioRecord::AudioRecordThread::resume() @@ -866,7 +979,10 @@ void AudioRecord::AudioRecordThread::resume() AutoMutex _l(mMyLock); if (mPaused) { mPaused = false; + mResumeLatch = false; mMyCond.signal(); + } else { + mResumeLatch = true; } } diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 7eeb4f8..faca054 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -19,31 +19,14 @@ //#define LOG_NDEBUG 0 #define LOG_TAG "AudioTrack" -#include <stdint.h> -#include <sys/types.h> -#include <limits.h> - -#include <sched.h> #include <sys/resource.h> - -#include <private/media/AudioTrackShared.h> - -#include <media/AudioSystem.h> +#include <audio_utils/primitives.h> +#include <binder/IPCThreadState.h> #include <media/AudioTrack.h> - #include <utils/Log.h> -#include <binder/Parcel.h> -#include <binder/IPCThreadState.h> -#include <utils/Timers.h> -#include <utils/Atomic.h> - -#include <cutils/bitops.h> -#include <cutils/compiler.h> - -#include <system/audio.h> -#include <system/audio_policy.h> +#include <private/media/AudioTrackShared.h> -#include <audio_utils/primitives.h> +#define WAIT_PERIOD_MS 10 namespace android { // --------------------------------------------------------------------------- @@ -82,7 +65,9 @@ status_t AudioTrack::getMinFrameCount( // Ensure that buffer depth covers at least audio hardware latency uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); - if (minBufCount < 2) minBufCount = 2; + if (minBufCount < 2) { + minBufCount = 2; + } *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : afFrameCount * minBufCount * sampleRate / afSampleRate; @@ -97,8 +82,7 @@ AudioTrack::AudioTrack() : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT), - mProxy(NULL) + mPreviousSchedulingGroup(SP_DEFAULT) { } @@ -112,16 +96,16 @@ AudioTrack::AudioTrack( callback_t cbf, void* user, int notificationFrames, - int sessionId) + int sessionId, + transfer_type transferType) : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT), - mProxy(NULL) + mPreviousSchedulingGroup(SP_DEFAULT) { mStatus = set(streamType, sampleRate, format, channelMask, frameCount, flags, cbf, user, notificationFrames, - 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); + 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType); } AudioTrack::AudioTrack( @@ -134,27 +118,20 @@ AudioTrack::AudioTrack( callback_t cbf, void* user, int notificationFrames, - int sessionId) + int sessionId, + transfer_type transferType) : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), - mPreviousSchedulingGroup(SP_DEFAULT), - mProxy(NULL) + mPreviousSchedulingGroup(SP_DEFAULT) { - if (sharedBuffer == 0) { - ALOGE("sharedBuffer must be non-0"); - mStatus = BAD_VALUE; - return; - } mStatus = set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags, cbf, user, notificationFrames, - sharedBuffer, false /*threadCanCallJava*/, sessionId); + sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType); } AudioTrack::~AudioTrack() { - ALOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); - if (mStatus == NO_ERROR) { // Make sure that callback function exits in the case where // it is looping on buffer full condition in obtainBuffer(). @@ -165,11 +142,13 @@ AudioTrack::~AudioTrack() mAudioTrackThread->requestExitAndWait(); mAudioTrackThread.clear(); } - mAudioTrack.clear(); + if (mAudioTrack != 0) { + mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); + mAudioTrack.clear(); + } IPCThreadState::self()->flushCommands(); AudioSystem::releaseAudioSessionId(mSessionId); } - delete mProxy; } status_t AudioTrack::set( @@ -184,8 +163,44 @@ status_t AudioTrack::set( int notificationFrames, const sp<IMemory>& sharedBuffer, bool threadCanCallJava, - int sessionId) + int sessionId, + transfer_type transferType) { + switch (transferType) { + case TRANSFER_DEFAULT: + if (sharedBuffer != 0) { + transferType = TRANSFER_SHARED; + } else if (cbf == NULL || threadCanCallJava) { + transferType = TRANSFER_SYNC; + } else { + transferType = TRANSFER_CALLBACK; + } + break; + case TRANSFER_CALLBACK: + if (cbf == NULL || sharedBuffer != 0) { + ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); + return BAD_VALUE; + } + break; + case TRANSFER_OBTAIN: + case TRANSFER_SYNC: + if (sharedBuffer != 0) { + ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); + return BAD_VALUE; + } + break; + case TRANSFER_SHARED: + if (sharedBuffer == 0) { + ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); + return BAD_VALUE; + } + break; + default: + ALOGE("Invalid transfer type %d", transferType); + return BAD_VALUE; + } + mTransfer = transferType; + // FIXME "int" here is legacy and will be replaced by size_t later if (frameCountInt < 0) { ALOGE("Invalid frame count %d", frameCountInt); @@ -199,6 +214,7 @@ status_t AudioTrack::set( ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); AutoMutex lock(mLock); + if (mAudioTrack != 0) { ALOGE("Track already in use"); return INVALID_OPERATION; @@ -228,7 +244,7 @@ status_t AudioTrack::set( // validate parameters if (!audio_is_valid_format(format)) { - ALOGE("Invalid format"); + ALOGE("Invalid format %d", format); return BAD_VALUE; } @@ -281,6 +297,7 @@ status_t AudioTrack::set( mFrameCount = frameCount; mReqFrameCount = frameCount; mNotificationFramesReq = notificationFrames; + mNotificationFramesAct = 0; mSessionId = sessionId; mAuxEffectId = 0; mFlags = flags; @@ -298,7 +315,8 @@ status_t AudioTrack::set( frameCount, flags, sharedBuffer, - output); + output, + 0 /*epoch*/); if (status != NO_ERROR) { if (mAudioTrackThread != 0) { @@ -309,20 +327,21 @@ status_t AudioTrack::set( } mStatus = NO_ERROR; - mStreamType = streamType; mFormat = format; - mSharedBuffer = sharedBuffer; - mActive = false; + mState = STATE_STOPPED; mUserData = user; - mLoopCount = 0; + mLoopPeriod = 0; mMarkerPosition = 0; mMarkerReached = false; mNewPosition = 0; mUpdatePeriod = 0; - mFlushed = false; AudioSystem::acquireAudioSessionId(mSessionId); + mSequence = 1; + mObservedSequence = mSequence; + mInUnderrun = false; + return NO_ERROR; } @@ -330,87 +349,45 @@ status_t AudioTrack::set( void AudioTrack::start() { - sp<AudioTrackThread> t = mAudioTrackThread; - - ALOGV("start %p", this); - AutoMutex lock(mLock); - // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed - // while we are accessing the cblk - sp<IAudioTrack> audioTrack = mAudioTrack; - sp<IMemory> iMem = mCblkMemory; - audio_track_cblk_t* cblk = mCblk; + if (mState == STATE_ACTIVE) { + return; + } - if (!mActive) { - mFlushed = false; - mActive = true; - mNewPosition = cblk->server + mUpdatePeriod; - cblk->lock.lock(); - cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; - cblk->waitTimeMs = 0; - android_atomic_and(~CBLK_DISABLED, &cblk->flags); - if (t != 0) { - t->resume(); - } else { - mPreviousPriority = getpriority(PRIO_PROCESS, 0); - get_sched_policy(0, &mPreviousSchedulingGroup); - androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); - } + mInUnderrun = true; - ALOGV("start %p before lock cblk %p", this, cblk); - status_t status = NO_ERROR; - if (!(cblk->flags & CBLK_INVALID)) { - cblk->lock.unlock(); - ALOGV("mAudioTrack->start()"); - status = mAudioTrack->start(); - cblk->lock.lock(); - if (status == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID, &cblk->flags); - } - } - if (cblk->flags & CBLK_INVALID) { - audio_track_cblk_t* temp = cblk; - status = restoreTrack_l(temp, true /*fromStart*/); - cblk = temp; - } - cblk->lock.unlock(); - if (status != NO_ERROR) { - ALOGV("start() failed"); - mActive = false; - if (t != 0) { - t->pause(); - } else { - setpriority(PRIO_PROCESS, 0, mPreviousPriority); - set_sched_policy(0, mPreviousSchedulingGroup); - } - } + State previousState = mState; + mState = STATE_ACTIVE; + if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { + // reset current position as seen by client to 0 + mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); } + mNewPosition = mProxy->getPosition() + mUpdatePeriod; + int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->flags); -} - -void AudioTrack::stop() -{ sp<AudioTrackThread> t = mAudioTrackThread; + if (t != 0) { + t->resume(); + } else { + mPreviousPriority = getpriority(PRIO_PROCESS, 0); + get_sched_policy(0, &mPreviousSchedulingGroup); + androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); + } - ALOGV("stop %p", this); - - AutoMutex lock(mLock); - if (mActive) { - mActive = false; - mCblk->cv.signal(); - mAudioTrack->stop(); - // Cancel loops (If we are in the middle of a loop, playback - // would not stop until loopCount reaches 0). - setLoop_l(0, 0, 0); - // the playback head position will reset to 0, so if a marker is set, we need - // to activate it again - mMarkerReached = false; - // Force flush if a shared buffer is used otherwise audioflinger - // will not stop before end of buffer is reached. - // It may be needed to make sure that we stop playback, likely in case looping is on. - if (mSharedBuffer != 0) { - flush_l(); + status_t status = NO_ERROR; + if (!(flags & CBLK_INVALID)) { + status = mAudioTrack->start(); + if (status == DEAD_OBJECT) { + flags |= CBLK_INVALID; } + } + if (flags & CBLK_INVALID) { + status = restoreTrack_l("start"); + } + + if (status != NO_ERROR) { + ALOGE("start() status %d", status); + mState = previousState; if (t != 0) { t->pause(); } else { @@ -419,57 +396,85 @@ void AudioTrack::stop() } } + // FIXME discarding status +} + +void AudioTrack::stop() +{ + AutoMutex lock(mLock); + // FIXME pause then stop should not be a nop + if (mState != STATE_ACTIVE) { + return; + } + + mState = STATE_STOPPED; + mProxy->interrupt(); + mAudioTrack->stop(); + // the playback head position will reset to 0, so if a marker is set, we need + // to activate it again + mMarkerReached = false; +#if 0 + // Force flush if a shared buffer is used otherwise audioflinger + // will not stop before end of buffer is reached. + // It may be needed to make sure that we stop playback, likely in case looping is on. + if (mSharedBuffer != 0) { + flush_l(); + } +#endif + sp<AudioTrackThread> t = mAudioTrackThread; + if (t != 0) { + t->pause(); + } else { + setpriority(PRIO_PROCESS, 0, mPreviousPriority); + set_sched_policy(0, mPreviousSchedulingGroup); + } } bool AudioTrack::stopped() const { AutoMutex lock(mLock); - return stopped_l(); + return mState != STATE_ACTIVE; } void AudioTrack::flush() { + if (mSharedBuffer != 0) { + return; + } AutoMutex lock(mLock); - if (!mActive && mSharedBuffer == 0) { - flush_l(); + if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { + return; } + flush_l(); } void AudioTrack::flush_l() { - ALOGV("flush"); - ALOG_ASSERT(!mActive); + ALOG_ASSERT(mState != STATE_ACTIVE); // clear playback marker and periodic update counter mMarkerPosition = 0; mMarkerReached = false; mUpdatePeriod = 0; - mFlushed = true; + mState = STATE_FLUSHED; + mProxy->flush(); mAudioTrack->flush(); - // Release AudioTrack callback thread in case it was waiting for new buffers - // in AudioTrack::obtainBuffer() - mCblk->cv.signal(); } void AudioTrack::pause() { - ALOGV("pause"); AutoMutex lock(mLock); - if (mActive) { - mActive = false; - mCblk->cv.signal(); - mAudioTrack->pause(); + if (mState != STATE_ACTIVE) { + return; } + mState = STATE_PAUSED; + mProxy->interrupt(); + mAudioTrack->pause(); } status_t AudioTrack::setVolume(float left, float right) { - if (mStatus != NO_ERROR) { - return mStatus; - } - ALOG_ASSERT(mProxy != NULL); - if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { return BAD_VALUE; } @@ -490,18 +495,11 @@ status_t AudioTrack::setVolume(float volume) status_t AudioTrack::setAuxEffectSendLevel(float level) { - ALOGV("setAuxEffectSendLevel(%f)", level); - - if (mStatus != NO_ERROR) { - return mStatus; - } - ALOG_ASSERT(mProxy != NULL); - if (level < 0.0f || level > 1.0f) { return BAD_VALUE; } - AutoMutex lock(mLock); + AutoMutex lock(mLock); mSendLevel = level; mProxy->setSendLevel(level); @@ -511,18 +509,17 @@ status_t AudioTrack::setAuxEffectSendLevel(float level) void AudioTrack::getAuxEffectSendLevel(float* level) const { if (level != NULL) { - *level = mSendLevel; + *level = mSendLevel; } } status_t AudioTrack::setSampleRate(uint32_t rate) { - uint32_t afSamplingRate; - if (mIsTimed) { return INVALID_OPERATION; } + uint32_t afSamplingRate; if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { return NO_INIT; } @@ -550,58 +547,44 @@ uint32_t AudioTrack::getSampleRate() const status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) { - AutoMutex lock(mLock); - return setLoop_l(loopStart, loopEnd, loopCount); -} - -// must be called with mLock held -status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) -{ if (mSharedBuffer == 0 || mIsTimed) { return INVALID_OPERATION; } - audio_track_cblk_t* cblk = mCblk; - - Mutex::Autolock _l(cblk->lock); - if (loopCount == 0) { - cblk->loopStart = UINT_MAX; - cblk->loopEnd = UINT_MAX; - cblk->loopCount = 0; - mLoopCount = 0; - return NO_ERROR; - } - - if (loopStart >= loopEnd || - loopEnd - loopStart > mFrameCount || - cblk->server > loopStart) { - ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, " - "user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user); + ; + } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && + loopEnd - loopStart >= MIN_LOOP) { + ; + } else { return BAD_VALUE; } - if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) { - ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " - "framecount %d", - loopStart, loopEnd, mFrameCount); - return BAD_VALUE; + AutoMutex lock(mLock); + // See setPosition() regarding setting parameters such as loop points or position while active + if (mState == STATE_ACTIVE) { + return INVALID_OPERATION; } - - cblk->loopStart = loopStart; - cblk->loopEnd = loopEnd; - cblk->loopCount = loopCount; - mLoopCount = loopCount; - + setLoop_l(loopStart, loopEnd, loopCount); return NO_ERROR; } +void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) +{ + // FIXME If setting a loop also sets position to start of loop, then + // this is correct. Otherwise it should be removed. + mNewPosition = mProxy->getPosition() + mUpdatePeriod; + mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; + mStaticProxy->setLoop(loopStart, loopEnd, loopCount); +} + status_t AudioTrack::setMarkerPosition(uint32_t marker) { if (mCbf == NULL) { return INVALID_OPERATION; } + AutoMutex lock(mLock); mMarkerPosition = marker; mMarkerReached = false; @@ -614,6 +597,7 @@ status_t AudioTrack::getMarkerPosition(uint32_t *marker) const return BAD_VALUE; } + AutoMutex lock(mLock); *marker = mMarkerPosition; return NO_ERROR; @@ -625,9 +609,8 @@ status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) return INVALID_OPERATION; } - uint32_t curPosition; - getPosition(&curPosition); - mNewPosition = curPosition + updatePeriod; + AutoMutex lock(mLock); + mNewPosition = mProxy->getPosition() + updatePeriod; mUpdatePeriod = updatePeriod; return NO_ERROR; @@ -639,6 +622,7 @@ status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const return BAD_VALUE; } + AutoMutex lock(mLock); *updatePeriod = mUpdatePeriod; return NO_ERROR; @@ -649,58 +633,73 @@ status_t AudioTrack::setPosition(uint32_t position) if (mSharedBuffer == 0 || mIsTimed) { return INVALID_OPERATION; } + if (position > mFrameCount) { + return BAD_VALUE; + } AutoMutex lock(mLock); - - if (!stopped_l()) { + // Currently we require that the player is inactive before setting parameters such as position + // or loop points. Otherwise, there could be a race condition: the application could read the + // current position, compute a new position or loop parameters, and then set that position or + // loop parameters but it would do the "wrong" thing since the position has continued to advance + // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app + // to specify how it wants to handle such scenarios. + if (mState == STATE_ACTIVE) { return INVALID_OPERATION; } + mNewPosition = mProxy->getPosition() + mUpdatePeriod; + mLoopPeriod = 0; + // FIXME Check whether loops and setting position are incompatible in old code. + // If we use setLoop for both purposes we lose the capability to set the position while looping. + mStaticProxy->setLoop(position, mFrameCount, 0); - audio_track_cblk_t* cblk = mCblk; - Mutex::Autolock _l(cblk->lock); + return NO_ERROR; +} - if (position > cblk->user) { +status_t AudioTrack::getPosition(uint32_t *position) const +{ + if (position == NULL) { return BAD_VALUE; } - cblk->server = position; - android_atomic_or(CBLK_FORCEREADY, &cblk->flags); + AutoMutex lock(mLock); + // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes + *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : + mProxy->getPosition(); return NO_ERROR; } -status_t AudioTrack::getPosition(uint32_t *position) +status_t AudioTrack::getBufferPosition(size_t *position) { + if (mSharedBuffer == 0 || mIsTimed) { + return INVALID_OPERATION; + } if (position == NULL) { return BAD_VALUE; } - AutoMutex lock(mLock); - *position = mFlushed ? 0 : mCblk->server; + AutoMutex lock(mLock); + *position = mStaticProxy->getBufferPosition(); return NO_ERROR; } status_t AudioTrack::reload() { - if (mStatus != NO_ERROR) { - return mStatus; - } - ALOG_ASSERT(mProxy != NULL); - if (mSharedBuffer == 0 || mIsTimed) { return INVALID_OPERATION; } AutoMutex lock(mLock); - - if (!stopped_l()) { + // See setPosition() regarding setting parameters such as loop points or position while active + if (mState == STATE_ACTIVE) { return INVALID_OPERATION; } - - flush_l(); - - (void) mProxy->stepUser(mFrameCount); - + mNewPosition = mUpdatePeriod; + mLoopPeriod = 0; + // FIXME The new code cannot reload while keeping a loop specified. + // Need to check how the old code handled this, and whether it's a significant change. + mStaticProxy->setLoop(0, mFrameCount, 0); return NO_ERROR; } @@ -719,7 +718,7 @@ audio_io_handle_t AudioTrack::getOutput_l() status_t AudioTrack::attachAuxEffect(int effectId) { - ALOGV("attachAuxEffect(%d)", effectId); + AutoMutex lock(mLock); status_t status = mAudioTrack->attachAuxEffect(effectId); if (status == NO_ERROR) { mAuxEffectId = effectId; @@ -737,7 +736,8 @@ status_t AudioTrack::createTrack_l( size_t frameCount, audio_output_flags_t flags, const sp<IMemory>& sharedBuffer, - audio_io_handle_t output) + audio_io_handle_t output, + size_t epoch) { status_t status; const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); @@ -747,7 +747,8 @@ status_t AudioTrack::createTrack_l( } uint32_t afLatency; - if (AudioSystem::getLatency(output, streamType, &afLatency) != NO_ERROR) { + if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) { + ALOGE("getLatency(%d) failed status %d", output, status); return NO_INIT; } @@ -775,7 +776,10 @@ status_t AudioTrack::createTrack_l( frameCount = sharedBuffer->size(); } else if (frameCount == 0) { size_t afFrameCount; - if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { + status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); + if (status != NO_ERROR) { + ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, + status); return NO_INIT; } frameCount = afFrameCount; @@ -806,17 +810,26 @@ status_t AudioTrack::createTrack_l( // FIXME move these calculations and associated checks to server uint32_t afSampleRate; - if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { + status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); + if (status != NO_ERROR) { + ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, + status); return NO_INIT; } size_t afFrameCount; - if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { + status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); + if (status != NO_ERROR) { + ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); return NO_INIT; } // Ensure that buffer depth covers at least audio hardware latency uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); - if (minBufCount < 2) minBufCount = 2; + ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", + afFrameCount, minBufCount, afSampleRate, afLatency); + if (minBufCount <= 2) { + minBufCount = sampleRate == afSampleRate ? 2 : 3; + } size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" @@ -826,12 +839,9 @@ status_t AudioTrack::createTrack_l( if (frameCount == 0) { frameCount = minFrameCount; } - if (mNotificationFramesAct == 0) { - mNotificationFramesAct = frameCount/2; - } // Make sure that application is notified with sufficient margin // before underrun - if (mNotificationFramesAct > frameCount/2) { + if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { mNotificationFramesAct = frameCount/2; } if (frameCount < minFrameCount) { @@ -881,6 +891,10 @@ status_t AudioTrack::createTrack_l( ALOGE("Could not get control block"); return NO_INIT; } + if (mAudioTrack != 0) { + mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); + mDeathNotifier.clear(); + } mAudioTrack = track; mCblkMemory = iMem; audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); @@ -898,26 +912,38 @@ status_t AudioTrack::createTrack_l( if (trackFlags & IAudioFlinger::TRACK_FAST) { ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); mAwaitBoost = true; + if (sharedBuffer == 0) { + // double-buffering is not required for fast tracks, due to tighter scheduling + if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) { + mNotificationFramesAct = frameCount; + } + } } else { ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); // once denied, do not request again if IAudioTrack is re-created flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); mFlags = flags; - } - if (sharedBuffer == 0) { - mNotificationFramesAct = frameCount/2; + if (sharedBuffer == 0) { + if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { + mNotificationFramesAct = frameCount/2; + } + } } } + mRefreshRemaining = true; + + // Starting address of buffers in shared memory. If there is a shared buffer, buffers + // is the value of pointer() for the shared buffer, otherwise buffers points + // immediately after the control block. This address is for the mapping within client + // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. + void* buffers; if (sharedBuffer == 0) { - mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); + buffers = (char*)cblk + sizeof(audio_track_cblk_t); } else { - mBuffers = sharedBuffer->pointer(); + buffers = sharedBuffer->pointer(); } mAudioTrack->attachAuxEffect(mAuxEffectId); - cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; - cblk->waitTimeMs = 0; - mRemainingFrames = mNotificationFramesAct; // FIXME don't believe this lie mLatency = afLatency + (1000*frameCount) / sampleRate; mFrameCount = frameCount; @@ -928,147 +954,143 @@ status_t AudioTrack::createTrack_l( } // update proxy - delete mProxy; - mProxy = new AudioTrackClientProxy(cblk, mBuffers, frameCount, mFrameSizeAF); + if (sharedBuffer == 0) { + mStaticProxy.clear(); + mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); + } else { + mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); + mProxy = mStaticProxy; + } mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); mProxy->setSendLevel(mSendLevel); mProxy->setSampleRate(mSampleRate); - if (sharedBuffer != 0) { - // Force buffer full condition as data is already present in shared memory - mProxy->stepUser(frameCount); - } + mProxy->setEpoch(epoch); + mProxy->setMinimum(mNotificationFramesAct); + + mDeathNotifier = new DeathNotifier(this); + mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); return NO_ERROR; } status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) { - ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL); + if (audioBuffer == NULL) { + return BAD_VALUE; + } + if (mTransfer != TRANSFER_OBTAIN) { + audioBuffer->frameCount = 0; + audioBuffer->size = 0; + audioBuffer->raw = NULL; + return INVALID_OPERATION; + } - AutoMutex lock(mLock); - bool active; - status_t result = NO_ERROR; - audio_track_cblk_t* cblk = mCblk; - uint32_t framesReq = audioBuffer->frameCount; - uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; + const struct timespec *requested; + if (waitCount == -1) { + requested = &ClientProxy::kForever; + } else if (waitCount == 0) { + requested = &ClientProxy::kNonBlocking; + } else if (waitCount > 0) { + long long ms = WAIT_PERIOD_MS * (long long) waitCount; + struct timespec timeout; + timeout.tv_sec = ms / 1000; + timeout.tv_nsec = (int) (ms % 1000) * 1000000; + requested = &timeout; + } else { + ALOGE("%s invalid waitCount %d", __func__, waitCount); + requested = NULL; + } + return obtainBuffer(audioBuffer, requested); +} - audioBuffer->frameCount = 0; - audioBuffer->size = 0; +status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, + struct timespec *elapsed, size_t *nonContig) +{ + // previous and new IAudioTrack sequence numbers are used to detect track re-creation + uint32_t oldSequence = 0; + uint32_t newSequence; - size_t framesAvail = mProxy->framesAvailable(); + Proxy::Buffer buffer; + status_t status = NO_ERROR; - cblk->lock.lock(); - if (cblk->flags & CBLK_INVALID) { - goto create_new_track; - } - cblk->lock.unlock(); - - if (framesAvail == 0) { - cblk->lock.lock(); - goto start_loop_here; - while (framesAvail == 0) { - active = mActive; - if (CC_UNLIKELY(!active)) { - ALOGV("Not active and NO_MORE_BUFFERS"); - cblk->lock.unlock(); - return NO_MORE_BUFFERS; - } - if (CC_UNLIKELY(!waitCount)) { - cblk->lock.unlock(); - return WOULD_BLOCK; - } - if (!(cblk->flags & CBLK_INVALID)) { - mLock.unlock(); - // this condition is in shared memory, so if IAudioTrack and control block - // are replaced due to mediaserver death or IAudioTrack invalidation then - // cv won't be signalled, but fortunately the timeout will limit the wait - result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); - cblk->lock.unlock(); - mLock.lock(); - if (!mActive) { - return status_t(STOPPED); - } - // IAudioTrack may have been re-created while mLock was unlocked - cblk = mCblk; - cblk->lock.lock(); - } + static const int32_t kMaxTries = 5; + int32_t tryCounter = kMaxTries; - if (cblk->flags & CBLK_INVALID) { - goto create_new_track; - } - if (CC_UNLIKELY(result != NO_ERROR)) { - cblk->waitTimeMs += waitTimeMs; - if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { - // timing out when a loop has been set and we have already written upto loop end - // is a normal condition: no need to wake AudioFlinger up. - if (cblk->user < cblk->loopEnd) { - ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " - "server=%08x", this, cblk->mName, cblk->user, cblk->server); - //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) - cblk->lock.unlock(); - result = mAudioTrack->start(); - cblk->lock.lock(); - if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID, &cblk->flags); -create_new_track: - audio_track_cblk_t* temp = cblk; - result = restoreTrack_l(temp, false /*fromStart*/); - cblk = temp; - } - if (result != NO_ERROR) { - ALOGW("obtainBuffer create Track error %d", result); - cblk->lock.unlock(); - return result; - } + do { + // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to + // keep them from going away if another thread re-creates the track during obtainBuffer() + sp<AudioTrackClientProxy> proxy; + sp<IMemory> iMem; + + { // start of lock scope + AutoMutex lock(mLock); + + newSequence = mSequence; + // did previous obtainBuffer() fail due to media server death or voluntary invalidation? + if (status == DEAD_OBJECT) { + // re-create track, unless someone else has already done so + if (newSequence == oldSequence) { + status = restoreTrack_l("obtainBuffer"); + if (status != NO_ERROR) { + break; } - cblk->waitTimeMs = 0; } + } + oldSequence = newSequence; - if (--waitCount == 0) { - cblk->lock.unlock(); - return TIMED_OUT; - } + // Keep the extra references + proxy = mProxy; + iMem = mCblkMemory; + + // Non-blocking if track is stopped or paused + if (mState != STATE_ACTIVE) { + requested = &ClientProxy::kNonBlocking; } - // read the server count again - start_loop_here: - framesAvail = mProxy->framesAvailable_l(); - } - cblk->lock.unlock(); - } - cblk->waitTimeMs = 0; + } // end of lock scope - if (framesReq > framesAvail) { - framesReq = framesAvail; - } + buffer.mFrameCount = audioBuffer->frameCount; + // FIXME starts the requested timeout and elapsed over from scratch + status = proxy->obtainBuffer(&buffer, requested, elapsed); - uint32_t u = cblk->user; - uint32_t bufferEnd = cblk->userBase + mFrameCount; + } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); - if (framesReq > bufferEnd - u) { - framesReq = bufferEnd - u; + audioBuffer->frameCount = buffer.mFrameCount; + audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; + audioBuffer->raw = buffer.mRaw; + if (nonContig != NULL) { + *nonContig = buffer.mNonContig; } - - audioBuffer->frameCount = framesReq; - audioBuffer->size = framesReq * mFrameSizeAF; - audioBuffer->raw = mProxy->buffer(u); - active = mActive; - return active ? status_t(NO_ERROR) : status_t(STOPPED); + return status; } void AudioTrack::releaseBuffer(Buffer* audioBuffer) { - ALOG_ASSERT(mStatus == NO_ERROR && mProxy != NULL); + if (mTransfer == TRANSFER_SHARED) { + return; + } + + size_t stepCount = audioBuffer->size / mFrameSizeAF; + if (stepCount == 0) { + return; + } + + Proxy::Buffer buffer; + buffer.mFrameCount = stepCount; + buffer.mRaw = audioBuffer->raw; AutoMutex lock(mLock); - audio_track_cblk_t* cblk = mCblk; - (void) mProxy->stepUser(audioBuffer->frameCount); - if (audioBuffer->frameCount > 0) { - // restart track if it was disabled by audioflinger due to previous underrun - if (mActive && (cblk->flags & CBLK_DISABLED)) { - android_atomic_and(~CBLK_DISABLED, &cblk->flags); - ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName); + mInUnderrun = false; + mProxy->releaseBuffer(&buffer); + + // restart track if it was disabled by audioflinger due to previous underrun + if (mState == STATE_ACTIVE) { + audio_track_cblk_t* cblk = mCblk; + if (android_atomic_and(~CBLK_DISABLED, &cblk->flags) & CBLK_DISABLED) { + ALOGW("releaseBuffer() track %p name=%#x disabled due to previous underrun, restarting", + this, cblk->mName); + // FIXME ignoring status mAudioTrack->start(); } } @@ -1078,68 +1100,46 @@ void AudioTrack::releaseBuffer(Buffer* audioBuffer) ssize_t AudioTrack::write(const void* buffer, size_t userSize) { - - if (mSharedBuffer != 0 || mIsTimed) { + if (mTransfer != TRANSFER_SYNC || mIsTimed) { return INVALID_OPERATION; } - if (ssize_t(userSize) < 0) { + if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { // Sanity-check: user is most-likely passing an error code, and it would // make the return value ambiguous (actualSize vs error). - ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", - buffer, userSize, userSize); + ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); return BAD_VALUE; } - ALOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); - - if (userSize == 0) { - return 0; - } - - // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed - // while we are accessing the cblk - mLock.lock(); - sp<IAudioTrack> audioTrack = mAudioTrack; - sp<IMemory> iMem = mCblkMemory; - mLock.unlock(); - - // since mLock is unlocked the IAudioTrack and shared memory may be re-created, - // so all cblk references might still refer to old shared memory, but that should be benign - - ssize_t written = 0; - const int8_t *src = (const int8_t *)buffer; + size_t written = 0; Buffer audioBuffer; - size_t frameSz = frameSize(); - do { - audioBuffer.frameCount = userSize/frameSz; + while (userSize >= mFrameSize) { + audioBuffer.frameCount = userSize / mFrameSize; - status_t err = obtainBuffer(&audioBuffer, -1); + status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); if (err < 0) { - // out of buffers, return #bytes written - if (err == status_t(NO_MORE_BUFFERS)) { + if (written > 0) { break; } return ssize_t(err); } size_t toWrite; - if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { // Divide capacity by 2 to take expansion into account - toWrite = audioBuffer.size>>1; - memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite); + toWrite = audioBuffer.size >> 1; + memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); } else { toWrite = audioBuffer.size; - memcpy(audioBuffer.i8, src, toWrite); + memcpy(audioBuffer.i8, buffer, toWrite); } - src += toWrite; + buffer = ((const char *) buffer) + toWrite; userSize -= toWrite; written += toWrite; releaseBuffer(&audioBuffer); - } while (userSize >= frameSz); + } return written; } @@ -1155,10 +1155,12 @@ status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) AutoMutex lock(mLock); status_t result = UNKNOWN_ERROR; +#if 1 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed // while we are accessing the cblk sp<IAudioTrack> audioTrack = mAudioTrack; sp<IMemory> iMem = mCblkMemory; +#endif // If the track is not invalid already, try to allocate a buffer. alloc // fails indicating that the server is dead, flag the track as invalid so @@ -1174,13 +1176,9 @@ status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) // If the track is invalid at this point, attempt to restore it. and try the // allocation one more time. if (cblk->flags & CBLK_INVALID) { - cblk->lock.lock(); - audio_track_cblk_t* temp = cblk; - result = restoreTrack_l(temp, false /*fromStart*/); - cblk = temp; - cblk->lock.unlock(); + result = restoreTrack_l("allocateTimedBuffer"); - if (result == OK) { + if (result == NO_ERROR) { result = mAudioTrack->allocateTimedBuffer(size, buffer); } } @@ -1197,9 +1195,10 @@ status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, audio_track_cblk_t* cblk = mCblk; // restart track if it was disabled by audioflinger due to previous underrun if (buffer->size() != 0 && status == NO_ERROR && - mActive && (cblk->flags & CBLK_DISABLED)) { + (mState == STATE_ACTIVE) && (cblk->flags & CBLK_DISABLED)) { android_atomic_and(~CBLK_DISABLED, &cblk->flags); ALOGW("queueTimedBuffer() track %p disabled, restarting", this); + // FIXME ignoring status mAudioTrack->start(); } } @@ -1214,12 +1213,8 @@ status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, // ------------------------------------------------------------------------- -bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) +nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) { - Buffer audioBuffer; - uint32_t frames; - size_t writtenSize; - mLock.lock(); if (mAwaitBoost) { mAwaitBoost = false; @@ -1240,86 +1235,181 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) } return true; } - // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed - // while we are accessing the cblk - sp<IAudioTrack> audioTrack = mAudioTrack; - sp<IMemory> iMem = mCblkMemory; - audio_track_cblk_t* cblk = mCblk; - bool active = mActive; - mLock.unlock(); - // since mLock is unlocked the IAudioTrack and shared memory may be re-created, - // so all cblk references might still refer to old shared memory, but that should be benign + // Can only reference mCblk while locked + int32_t flags = android_atomic_and( + ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->flags); - // Manage underrun callback - if (active && (mProxy->framesAvailable() == mFrameCount)) { - ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); - if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { - mCbf(EVENT_UNDERRUN, mUserData, 0); - if (cblk->server == mFrameCount) { - mCbf(EVENT_BUFFER_END, mUserData, 0); - } - if (mSharedBuffer != 0) { - return false; - } + // Check for track invalidation + if (flags & CBLK_INVALID) { + (void) restoreTrack_l("processAudioBuffer"); + mLock.unlock(); + // Run again immediately, but with a new IAudioTrack + return 0; + } + + bool active = mState == STATE_ACTIVE; + + // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() + bool newUnderrun = false; + if (flags & CBLK_UNDERRUN) { +#if 0 + // Currently in shared buffer mode, when the server reaches the end of buffer, + // the track stays active in continuous underrun state. It's up to the application + // to pause or stop the track, or set the position to a new offset within buffer. + // This was some experimental code to auto-pause on underrun. Keeping it here + // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. + if (mTransfer == TRANSFER_SHARED) { + mState = STATE_PAUSED; + active = false; + } +#endif + if (!mInUnderrun) { + mInUnderrun = true; + newUnderrun = true; } } - // Manage loop end callback - while (mLoopCount > cblk->loopCount) { - int loopCount = -1; - mLoopCount--; - if (mLoopCount >= 0) loopCount = mLoopCount; + // Get current position of server + size_t position = mProxy->getPosition(); - mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); + // Manage marker callback + bool markerReached = false; + size_t markerPosition = mMarkerPosition; + // FIXME fails for wraparound, need 64 bits + if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { + mMarkerReached = markerReached = true; + } + + // Determine number of new position callback(s) that will be needed, while locked + size_t newPosCount = 0; + size_t newPosition = mNewPosition; + size_t updatePeriod = mUpdatePeriod; + // FIXME fails for wraparound, need 64 bits + if (updatePeriod > 0 && position >= newPosition) { + newPosCount = ((position - newPosition) / updatePeriod) + 1; + mNewPosition += updatePeriod * newPosCount; + } + + // Cache other fields that will be needed soon + uint32_t loopPeriod = mLoopPeriod; + uint32_t sampleRate = mSampleRate; + size_t notificationFrames = mNotificationFramesAct; + if (mRefreshRemaining) { + mRefreshRemaining = false; + mRemainingFrames = notificationFrames; + mRetryOnPartialBuffer = false; + } + size_t misalignment = mProxy->getMisalignment(); + int32_t sequence = mSequence; + + // These fields don't need to be cached, because they are assigned only by set(): + // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags + // mFlags is also assigned by createTrack_l(), but not the bit we care about. + + mLock.unlock(); + + // perform callbacks while unlocked + if (newUnderrun) { + mCbf(EVENT_UNDERRUN, mUserData, NULL); + } + // FIXME we will miss loops if loop cycle was signaled several times since last call + // to processAudioBuffer() + if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { + mCbf(EVENT_LOOP_END, mUserData, NULL); + } + if (flags & CBLK_BUFFER_END) { + mCbf(EVENT_BUFFER_END, mUserData, NULL); + } + if (markerReached) { + mCbf(EVENT_MARKER, mUserData, &markerPosition); + } + while (newPosCount > 0) { + size_t temp = newPosition; + mCbf(EVENT_NEW_POS, mUserData, &temp); + newPosition += updatePeriod; + newPosCount--; + } + if (mObservedSequence != sequence) { + mObservedSequence = sequence; + mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); } - // Manage marker callback - if (!mMarkerReached && (mMarkerPosition > 0)) { - if (cblk->server >= mMarkerPosition) { - mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); - mMarkerReached = true; - } + // if inactive, then don't run me again until re-started + if (!active) { + return NS_INACTIVE; } - // Manage new position callback - if (mUpdatePeriod > 0) { - while (cblk->server >= mNewPosition) { - mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); - mNewPosition += mUpdatePeriod; - } + // Compute the estimated time until the next timed event (position, markers, loops) + // FIXME only for non-compressed audio + uint32_t minFrames = ~0; + if (!markerReached && position < markerPosition) { + minFrames = markerPosition - position; + } + if (loopPeriod > 0 && loopPeriod < minFrames) { + minFrames = loopPeriod; + } + if (updatePeriod > 0 && updatePeriod < minFrames) { + minFrames = updatePeriod; } - // If Shared buffer is used, no data is requested from client. - if (mSharedBuffer != 0) { - frames = 0; - } else { - frames = mRemainingFrames; + // If > 0, poll periodically to recover from a stuck server. A good value is 2. + static const uint32_t kPoll = 0; + if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { + minFrames = kPoll * notificationFrames; } - // See description of waitCount parameter at declaration of obtainBuffer(). - // The logic below prevents us from being stuck below at obtainBuffer() - // not being able to handle timed events (position, markers, loops). - int32_t waitCount = -1; - if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) { - waitCount = 1; + // Convert frame units to time units + nsecs_t ns = NS_WHENEVER; + if (minFrames != (uint32_t) ~0) { + // This "fudge factor" avoids soaking CPU, and compensates for late progress by server + static const nsecs_t kFudgeNs = 10000000LL; // 10 ms + ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; } - do { + // If not supplying data by EVENT_MORE_DATA, then we're done + if (mTransfer != TRANSFER_CALLBACK) { + return ns; + } + + struct timespec timeout; + const struct timespec *requested = &ClientProxy::kForever; + if (ns != NS_WHENEVER) { + timeout.tv_sec = ns / 1000000000LL; + timeout.tv_nsec = ns % 1000000000LL; + ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); + requested = &timeout; + } - audioBuffer.frameCount = frames; + while (mRemainingFrames > 0) { - status_t err = obtainBuffer(&audioBuffer, waitCount); - if (err < NO_ERROR) { - if (err != TIMED_OUT) { - ALOGE_IF(err != status_t(NO_MORE_BUFFERS), - "Error obtaining an audio buffer, giving up."); - return false; + Buffer audioBuffer; + audioBuffer.frameCount = mRemainingFrames; + size_t nonContig; + status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); + LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), + "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); + requested = &ClientProxy::kNonBlocking; + size_t avail = audioBuffer.frameCount + nonContig; + ALOGV("obtainBuffer(%u) returned %u = %u + %u", + mRemainingFrames, avail, audioBuffer.frameCount, nonContig); + if (err != NO_ERROR) { + if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { + return 0; } - break; + ALOGE("Error %d obtaining an audio buffer, giving up.", err); + return NS_NEVER; } - if (err == status_t(STOPPED)) { - return false; + + if (mRetryOnPartialBuffer) { + mRetryOnPartialBuffer = false; + if (avail < mRemainingFrames) { + int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; + if (ns < 0 || myns < ns) { + ns = myns; + } + return ns; + } } // Divide buffer size by 2 to take into account the expansion @@ -1331,66 +1421,76 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) size_t reqSize = audioBuffer.size; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); - writtenSize = audioBuffer.size; + size_t writtenSize = audioBuffer.size; + size_t writtenFrames = writtenSize / mFrameSize; // Sanity check on returned size - if (ssize_t(writtenSize) <= 0) { + if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { + ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", + reqSize, (int) writtenSize); + return NS_NEVER; + } + + if (writtenSize == 0) { // The callback is done filling buffers // Keep this thread going to handle timed events and // still try to get more data in intervals of WAIT_PERIOD_MS // but don't just loop and block the CPU, so wait - usleep(WAIT_PERIOD_MS*1000); - break; - } - - if (writtenSize > reqSize) { - writtenSize = reqSize; + return WAIT_PERIOD_MS * 1000000LL; } if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { // 8 to 16 bit conversion, note that source and destination are the same address memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); - writtenSize <<= 1; + audioBuffer.size <<= 1; } - audioBuffer.size = writtenSize; - // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for - // 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of - // 16 bit. - audioBuffer.frameCount = writtenSize / mFrameSizeAF; - - frames -= audioBuffer.frameCount; + size_t releasedFrames = audioBuffer.size / mFrameSizeAF; + audioBuffer.frameCount = releasedFrames; + mRemainingFrames -= releasedFrames; + if (misalignment >= releasedFrames) { + misalignment -= releasedFrames; + } else { + misalignment = 0; + } releaseBuffer(&audioBuffer); - } - while (frames); - if (frames == 0) { - mRemainingFrames = mNotificationFramesAct; - } else { - mRemainingFrames = frames; + // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer + // if callback doesn't like to accept the full chunk + if (writtenSize < reqSize) { + continue; + } + + // There could be enough non-contiguous frames available to satisfy the remaining request + if (mRemainingFrames <= nonContig) { + continue; + } + +#if 0 + // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a + // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA + // that total to a sum == notificationFrames. + if (0 < misalignment && misalignment <= mRemainingFrames) { + mRemainingFrames = misalignment; + return (mRemainingFrames * 1100000000LL) / sampleRate; + } +#endif + } - return true; + mRemainingFrames = notificationFrames; + mRetryOnPartialBuffer = true; + + // A lot has transpired since ns was calculated, so run again immediately and re-calculate + return 0; } -// must be called with mLock and refCblk.lock held. Callers must also hold strong references on -// the IAudioTrack and IMemory in case they are recreated here. -// If the IAudioTrack is successfully restored, the refCblk pointer is updated -// FIXME Don't depend on caller to hold strong references. -status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart) +status_t AudioTrack::restoreTrack_l(const char *from) { + ALOGW("dead IAudioTrack, creating a new one from %s()", from); + ++mSequence; status_t result; - audio_track_cblk_t* cblk = refCblk; - audio_track_cblk_t* newCblk = cblk; - ALOGW("dead IAudioTrack, creating a new one from %s", - fromStart ? "start()" : "obtainBuffer()"); - - // signal old cblk condition so that other threads waiting for available buffers stop - // waiting now - cblk->cv.broadcast(); - cblk->lock.unlock(); - // refresh the audio configuration cache in this process to make sure we get new // output parameters in getOutput_l() and createTrack_l() AudioSystem::clearAudioConfigCache(); @@ -1398,68 +1498,47 @@ status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart // if the new IAudioTrack is created, createTrack_l() will modify the // following member variables: mAudioTrack, mCblkMemory and mCblk. // It will also delete the strong references on previous IAudioTrack and IMemory + size_t position = mProxy->getPosition(); + mNewPosition = position + mUpdatePeriod; + size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; result = createTrack_l(mStreamType, mSampleRate, mFormat, mReqFrameCount, // so that frame count never goes down mFlags, mSharedBuffer, - getOutput_l()); + getOutput_l(), + position /*epoch*/); if (result == NO_ERROR) { - uint32_t user = cblk->user; - uint32_t server = cblk->server; + // continue playback from last known position, but + // don't attempt to restore loop after invalidation; it's difficult and not worthwhile + if (mStaticProxy != NULL) { + mLoopPeriod = 0; + mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); + } + // FIXME How do we simulate the fact that all frames present in the buffer at the time of + // track destruction have been played? This is critical for SoundPool implementation + // This must be broken, and needs to be tested/debugged. +#if 0 // restore write index and set other indexes to reflect empty buffer status - newCblk = mCblk; - newCblk->user = user; - newCblk->server = user; - newCblk->userBase = user; - newCblk->serverBase = user; - // restore loop: this is not guaranteed to succeed if new frame count is not - // compatible with loop length - setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); - size_t frames = 0; - if (!fromStart) { - newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; + if (!strcmp(from, "start")) { // Make sure that a client relying on callback events indicating underrun or // the actual amount of audio frames played (e.g SoundPool) receives them. if (mSharedBuffer == 0) { - if (user > server) { - frames = ((user - server) > mFrameCount) ? - mFrameCount : (user - server); - memset(mBuffers, 0, frames * mFrameSizeAF); - } // restart playback even if buffer is not completely filled. - android_atomic_or(CBLK_FORCEREADY, &newCblk->flags); + android_atomic_or(CBLK_FORCEREADY, &mCblk->flags); } } - if (mSharedBuffer != 0) { - frames = mFrameCount; - } - if (frames > 0) { - // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to - // the client - mProxy->stepUser(frames); - } - if (mActive) { +#endif + if (mState == STATE_ACTIVE) { result = mAudioTrack->start(); - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); - } - if (fromStart && result == NO_ERROR) { - mNewPosition = newCblk->server + mUpdatePeriod; } } - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); - ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", - result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); - - if (result == NO_ERROR) { - // from now on we switch to the newly created cblk - refCblk = newCblk; + if (result != NO_ERROR) { + ALOGW("restoreTrack_l() failed status %d", result); + mState = STATE_STOPPED; } - newCblk->lock.lock(); - - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d", result); return result; } @@ -1480,16 +1559,33 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const result.append(buffer); snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); result.append(buffer); - snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); + snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); result.append(buffer); ::write(fd, result.string(), result.size()); return NO_ERROR; } +uint32_t AudioTrack::getUnderrunFrames() const +{ + AutoMutex lock(mLock); + return mProxy->getUnderrunFrames(); +} + +// ========================================================================= + +void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) +{ + sp<AudioTrack> audioTrack = mAudioTrack.promote(); + if (audioTrack != 0) { + AutoMutex lock(audioTrack->mLock); + audioTrack->mProxy->binderDied(); + } +} + // ========================================================================= AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) - : Thread(bCanCallJava), mReceiver(receiver), mPaused(true) + : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) { } @@ -1507,10 +1603,26 @@ bool AudioTrack::AudioTrackThread::threadLoop() return true; } } - if (!mReceiver.processAudioBuffer(this)) { - pause(); + nsecs_t ns = mReceiver.processAudioBuffer(this); + switch (ns) { + case 0: + return true; + case NS_WHENEVER: + sleep(1); + return true; + case NS_INACTIVE: + pauseConditional(); + return true; + case NS_NEVER: + return false; + default: + LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); + struct timespec req; + req.tv_sec = ns / 1000000000LL; + req.tv_nsec = ns % 1000000000LL; + nanosleep(&req, NULL /*rem*/); + return true; } - return true; } void AudioTrack::AudioTrackThread::requestExit() @@ -1524,6 +1636,17 @@ void AudioTrack::AudioTrackThread::pause() { AutoMutex _l(mMyLock); mPaused = true; + mResumeLatch = false; +} + +void AudioTrack::AudioTrackThread::pauseConditional() +{ + AutoMutex _l(mMyLock); + if (mResumeLatch) { + mResumeLatch = false; + } else { + mPaused = true; + } } void AudioTrack::AudioTrackThread::resume() @@ -1531,7 +1654,10 @@ void AudioTrack::AudioTrackThread::resume() AutoMutex _l(mMyLock); if (mPaused) { mPaused = false; + mResumeLatch = false; mMyCond.signal(); + } else { + mResumeLatch = true; } } diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp index 13d47c9..f034164 100644 --- a/media/libmedia/AudioTrackShared.cpp +++ b/media/libmedia/AudioTrackShared.cpp @@ -19,178 +19,664 @@ #include <private/media/AudioTrackShared.h> #include <utils/Log.h> +extern "C" { +#include "../private/bionic_futex.h" +} namespace android { audio_track_cblk_t::audio_track_cblk_t() - : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), - userBase(0), serverBase(0), frameCount_(0), - loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), - mSampleRate(0), mSendLevel(0), flags(0) + : server(0), frameCount_(0), mFutex(0), mMinimum(0), + mVolumeLR(0x10001000), mSampleRate(0), mSendLevel(0), mName(0), flags(0) { + memset(&u, 0, sizeof(u)); } -uint32_t audio_track_cblk_t::stepUser(size_t stepCount, size_t frameCount, bool isOut) +// --------------------------------------------------------------------------- + +Proxy::Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, + bool isOut, bool clientInServer) + : mCblk(cblk), mBuffers(buffers), mFrameCount(frameCount), mFrameSize(frameSize), + mFrameCountP2(roundup(frameCount)), mIsOut(isOut), mClientInServer(clientInServer), + mIsShutdown(false) { - ALOGV("stepuser %08x %08x %d", user, server, stepCount); +} - uint32_t u = user; - u += stepCount; - // Ensure that user is never ahead of server for AudioRecord - if (isOut) { - // If stepServer() has been called once, switch to normal obtainBuffer() timeout period - if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { - bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; - } - } else if (u > server) { - ALOGW("stepUser occurred after track reset"); - u = server; +// --------------------------------------------------------------------------- + +ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize, bool isOut, bool clientInServer) + : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer), mEpoch(0) +{ +} + +const struct timespec ClientProxy::kForever = {INT_MAX /*tv_sec*/, 0 /*tv_nsec*/}; +const struct timespec ClientProxy::kNonBlocking = {0 /*tv_sec*/, 0 /*tv_nsec*/}; + +#define MEASURE_NS 10000000 // attempt to provide accurate timeouts if requested >= MEASURE_NS + +// To facilitate quicker recovery from server failure, this value limits the timeout per each futex +// wait. However it does not protect infinite timeouts. If defined to be zero, there is no limit. +// FIXME May not be compatible with audio tunneling requirements where timeout should be in the +// order of minutes. +#define MAX_SEC 5 + +status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *requested, + struct timespec *elapsed) +{ + if (buffer == NULL || buffer->mFrameCount == 0) { + ALOGE("%s BAD_VALUE", __func__); + return BAD_VALUE; } + struct timespec total; // total elapsed time spent waiting + total.tv_sec = 0; + total.tv_nsec = 0; + bool measure = elapsed != NULL; // whether to measure total elapsed time spent waiting - if (u >= frameCount) { - // common case, user didn't just wrap - if (u - frameCount >= userBase ) { - userBase += frameCount; + status_t status; + enum { + TIMEOUT_ZERO, // requested == NULL || *requested == 0 + TIMEOUT_INFINITE, // *requested == infinity + TIMEOUT_FINITE, // 0 < *requested < infinity + TIMEOUT_CONTINUE, // additional chances after TIMEOUT_FINITE + } timeout; + if (requested == NULL) { + timeout = TIMEOUT_ZERO; + } else if (requested->tv_sec == 0 && requested->tv_nsec == 0) { + timeout = TIMEOUT_ZERO; + } else if (requested->tv_sec == INT_MAX) { + timeout = TIMEOUT_INFINITE; + } else { + timeout = TIMEOUT_FINITE; + if (requested->tv_sec > 0 || requested->tv_nsec >= MEASURE_NS) { + measure = true; + } + } + struct timespec before; + bool beforeIsValid = false; + audio_track_cblk_t* cblk = mCblk; + bool ignoreInitialPendingInterrupt = true; + // check for shared memory corruption + if (mIsShutdown) { + status = NO_INIT; + goto end; + } + for (;;) { + int32_t flags = android_atomic_and(~CBLK_INTERRUPT, &cblk->flags); + // check for track invalidation by server, or server death detection + if (flags & CBLK_INVALID) { + ALOGV("Track invalidated"); + status = DEAD_OBJECT; + goto end; + } + // check for obtainBuffer interrupted by client + if (!ignoreInitialPendingInterrupt && (flags & CBLK_INTERRUPT)) { + ALOGV("obtainBuffer() interrupted by client"); + status = -EINTR; + goto end; + } + ignoreInitialPendingInterrupt = false; + // compute number of frames available to write (AudioTrack) or read (AudioRecord) + int32_t front; + int32_t rear; + if (mIsOut) { + // The barrier following the read of mFront is probably redundant. + // We're about to perform a conditional branch based on 'filled', + // which will force the processor to observe the read of mFront + // prior to allowing data writes starting at mRaw. + // However, the processor may support speculative execution, + // and be unable to undo speculative writes into shared memory. + // The barrier will prevent such speculative execution. + front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront); + rear = cblk->u.mStreaming.mRear; + } else { + // On the other hand, this barrier is required. + rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear); + front = cblk->u.mStreaming.mFront; + } + ssize_t filled = rear - front; + // pipe should not be overfull + if (!(0 <= filled && (size_t) filled <= mFrameCount)) { + ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled); + mIsShutdown = true; + status = NO_INIT; + goto end; + } + // don't allow filling pipe beyond the nominal size + size_t avail = mIsOut ? mFrameCount - filled : filled; + if (avail > 0) { + // 'avail' may be non-contiguous, so return only the first contiguous chunk + size_t part1; + if (mIsOut) { + rear &= mFrameCountP2 - 1; + part1 = mFrameCountP2 - rear; + } else { + front &= mFrameCountP2 - 1; + part1 = mFrameCountP2 - front; + } + if (part1 > avail) { + part1 = avail; + } + if (part1 > buffer->mFrameCount) { + part1 = buffer->mFrameCount; + } + buffer->mFrameCount = part1; + buffer->mRaw = part1 > 0 ? + &((char *) mBuffers)[(mIsOut ? rear : front) * mFrameSize] : NULL; + buffer->mNonContig = avail - part1; + // mUnreleased = part1; + status = NO_ERROR; + break; + } + struct timespec remaining; + const struct timespec *ts; + switch (timeout) { + case TIMEOUT_ZERO: + status = WOULD_BLOCK; + goto end; + case TIMEOUT_INFINITE: + ts = NULL; + break; + case TIMEOUT_FINITE: + timeout = TIMEOUT_CONTINUE; + if (MAX_SEC == 0) { + ts = requested; + break; + } + // fall through + case TIMEOUT_CONTINUE: + // FIXME we do not retry if requested < 10ms? needs documentation on this state machine + if (!measure || requested->tv_sec < total.tv_sec || + (requested->tv_sec == total.tv_sec && requested->tv_nsec <= total.tv_nsec)) { + status = TIMED_OUT; + goto end; + } + remaining.tv_sec = requested->tv_sec - total.tv_sec; + if ((remaining.tv_nsec = requested->tv_nsec - total.tv_nsec) < 0) { + remaining.tv_nsec += 1000000000; + remaining.tv_sec++; + } + if (0 < MAX_SEC && MAX_SEC < remaining.tv_sec) { + remaining.tv_sec = MAX_SEC; + remaining.tv_nsec = 0; + } + ts = &remaining; + break; + default: + LOG_FATAL("%s timeout=%d", timeout); + ts = NULL; + break; + } + int32_t old = android_atomic_dec(&cblk->mFutex); + if (old <= 0) { + int rc; + if (measure && !beforeIsValid) { + clock_gettime(CLOCK_MONOTONIC, &before); + beforeIsValid = true; + } + int ret = __futex_syscall4(&cblk->mFutex, + mClientInServer ? FUTEX_WAIT_PRIVATE : FUTEX_WAIT, old - 1, ts); + // update total elapsed time spent waiting + if (measure) { + struct timespec after; + clock_gettime(CLOCK_MONOTONIC, &after); + total.tv_sec += after.tv_sec - before.tv_sec; + long deltaNs = after.tv_nsec - before.tv_nsec; + if (deltaNs < 0) { + deltaNs += 1000000000; + total.tv_sec--; + } + if ((total.tv_nsec += deltaNs) >= 1000000000) { + total.tv_nsec -= 1000000000; + total.tv_sec++; + } + before = after; + beforeIsValid = true; + } + switch (ret) { + case 0: // normal wakeup by server, or by binderDied() + case -EWOULDBLOCK: // benign race condition with server + case -EINTR: // wait was interrupted by signal or other spurious wakeup + case -ETIMEDOUT: // time-out expired + break; + default: + ALOGE("%s unexpected error %d", __func__, ret); + status = -ret; + goto end; + } } - } else if (u >= userBase + frameCount) { - // user just wrapped - userBase += frameCount; } - user = u; - - // Clear flow control error condition as new data has been written/read to/from buffer. - if (flags & CBLK_UNDERRUN) { - android_atomic_and(~CBLK_UNDERRUN, &flags); +end: + if (status != NO_ERROR) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + } + if (elapsed != NULL) { + *elapsed = total; } + if (requested == NULL) { + requested = &kNonBlocking; + } + if (measure) { + ALOGV("requested %d.%03d elapsed %d.%03d", requested->tv_sec, requested->tv_nsec / 1000000, + total.tv_sec, total.tv_nsec / 1000000); + } + return status; +} - return u; +void ClientProxy::releaseBuffer(Buffer* buffer) +{ + size_t stepCount = buffer->mFrameCount; + // FIXME + // check mUnreleased + // verify that stepCount <= frameCount returned by the last obtainBuffer() + // verify stepCount not > total frame count of pipe + if (stepCount == 0) { + return; + } + audio_track_cblk_t* cblk = mCblk; + // Both of these barriers are required + if (mIsOut) { + int32_t rear = cblk->u.mStreaming.mRear; + android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear); + } else { + int32_t front = cblk->u.mStreaming.mFront; + android_atomic_release_store(stepCount + front, &cblk->u.mStreaming.mFront); + } } -bool audio_track_cblk_t::stepServer(size_t stepCount, size_t frameCount, bool isOut) +void ClientProxy::binderDied() { - ALOGV("stepserver %08x %08x %d", user, server, stepCount); + audio_track_cblk_t* cblk = mCblk; + if (!(android_atomic_or(CBLK_INVALID, &cblk->flags) & CBLK_INVALID)) { + // it seems that a FUTEX_WAKE_PRIVATE will not wake a FUTEX_WAIT, even within same process + (void) __futex_syscall3(&cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, + 1); + } +} - if (!tryLock()) { - ALOGW("stepServer() could not lock cblk"); - return false; +void ClientProxy::interrupt() +{ + audio_track_cblk_t* cblk = mCblk; + if (!(android_atomic_or(CBLK_INTERRUPT, &cblk->flags) & CBLK_INTERRUPT)) { + (void) __futex_syscall3(&cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, + 1); } +} - uint32_t s = server; - bool flushed = (s == user); +size_t ClientProxy::getMisalignment() +{ + audio_track_cblk_t* cblk = mCblk; + return (mFrameCountP2 - (mIsOut ? cblk->u.mStreaming.mRear : cblk->u.mStreaming.mFront)) & + (mFrameCountP2 - 1); +} - s += stepCount; - if (isOut) { - // Mark that we have read the first buffer so that next time stepUser() is called - // we switch to normal obtainBuffer() timeout period - if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { - bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1; - } - // It is possible that we receive a flush() - // while the mixer is processing a block: in this case, - // stepServer() is called After the flush() has reset u & s and - // we have s > u - if (flushed) { - ALOGW("stepServer occurred after track reset"); - s = user; +// --------------------------------------------------------------------------- + +void AudioTrackClientProxy::flush() +{ + mCblk->u.mStreaming.mFlush++; +} + +// --------------------------------------------------------------------------- + +StaticAudioTrackClientProxy::StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, + size_t frameCount, size_t frameSize) + : AudioTrackClientProxy(cblk, buffers, frameCount, frameSize), + mMutator(&cblk->u.mStatic.mSingleStateQueue), mBufferPosition(0) +{ +} + +void StaticAudioTrackClientProxy::flush() +{ + LOG_FATAL("static flush"); +} + +void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount) +{ + StaticAudioTrackState newState; + newState.mLoopStart = loopStart; + newState.mLoopEnd = loopEnd; + newState.mLoopCount = loopCount; + mBufferPosition = loopStart; + (void) mMutator.push(newState); +} + +size_t StaticAudioTrackClientProxy::getBufferPosition() +{ + size_t bufferPosition; + if (mMutator.ack()) { + bufferPosition = mCblk->u.mStatic.mBufferPosition; + if (bufferPosition > mFrameCount) { + bufferPosition = mFrameCount; } + } else { + bufferPosition = mBufferPosition; } + return bufferPosition; +} + +// --------------------------------------------------------------------------- - if (s >= loopEnd) { - ALOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); - s = loopStart; - if (--loopCount == 0) { - loopEnd = UINT_MAX; - loopStart = UINT_MAX; +ServerProxy::ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, + size_t frameSize, bool isOut, bool clientInServer) + : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer), mUnreleased(0), + mAvailToClient(0), mFlush(0), mDeferWake(false) +{ +} + +status_t ServerProxy::obtainBuffer(Buffer* buffer) +{ + if (mIsShutdown) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + mUnreleased = 0; + return NO_INIT; + } + audio_track_cblk_t* cblk = mCblk; + // compute number of frames available to write (AudioTrack) or read (AudioRecord), + // or use previous cached value from framesReady(), with added barrier if it omits. + int32_t front; + int32_t rear; + // See notes on barriers at ClientProxy::obtainBuffer() + if (mIsOut) { + int32_t flush = cblk->u.mStreaming.mFlush; + rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear); + if (flush != mFlush) { + front = rear; + mFlush = flush; + } else { + front = cblk->u.mStreaming.mFront; } + } else { + front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront); + rear = cblk->u.mStreaming.mRear; + } + ssize_t filled = rear - front; + // pipe should not already be overfull + if (!(0 <= filled && (size_t) filled <= mFrameCount)) { + ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled); + mIsShutdown = true; + } + if (mIsShutdown) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + mUnreleased = 0; + return NO_INIT; } + // don't allow filling pipe beyond the nominal size + size_t availToServer; + if (mIsOut) { + availToServer = filled; + mAvailToClient = mFrameCount - filled; + } else { + availToServer = mFrameCount - filled; + mAvailToClient = filled; + } + // 'availToServer' may be non-contiguous, so return only the first contiguous chunk + size_t part1; + if (mIsOut) { + front &= mFrameCountP2 - 1; + part1 = mFrameCountP2 - front; + } else { + rear &= mFrameCountP2 - 1; + part1 = mFrameCountP2 - rear; + } + if (part1 > availToServer) { + part1 = availToServer; + } + size_t ask = buffer->mFrameCount; + if (part1 > ask) { + part1 = ask; + } + // is assignment redundant in some cases? + buffer->mFrameCount = part1; + buffer->mRaw = part1 > 0 ? + &((char *) mBuffers)[(mIsOut ? front : rear) * mFrameSize] : NULL; + buffer->mNonContig = availToServer - part1; + mUnreleased = part1; + // optimization to avoid waking up the client too early + // FIXME need to test for recording + mDeferWake = part1 < ask && availToServer >= ask; + return part1 > 0 ? NO_ERROR : WOULD_BLOCK; +} - if (s >= frameCount) { - // common case, server didn't just wrap - if (s - frameCount >= serverBase ) { - serverBase += frameCount; - } - } else if (s >= serverBase + frameCount) { - // server just wrapped - serverBase += frameCount; +void ServerProxy::releaseBuffer(Buffer* buffer) +{ + if (mIsShutdown) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + return; + } + size_t stepCount = buffer->mFrameCount; + LOG_ALWAYS_FATAL_IF(stepCount > mUnreleased); + if (stepCount == 0) { + buffer->mRaw = NULL; + buffer->mNonContig = 0; + return; + } + mUnreleased -= stepCount; + audio_track_cblk_t* cblk = mCblk; + if (mIsOut) { + int32_t front = cblk->u.mStreaming.mFront; + android_atomic_release_store(stepCount + front, &cblk->u.mStreaming.mFront); + } else { + int32_t rear = cblk->u.mStreaming.mRear; + android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear); } - server = s; + mCblk->server += stepCount; - if (!(flags & CBLK_INVALID)) { - cv.signal(); + size_t half = mFrameCount / 2; + if (half == 0) { + half = 1; + } + size_t minimum = cblk->mMinimum; + if (minimum == 0) { + minimum = mIsOut ? half : 1; + } else if (minimum > half) { + minimum = half; + } + if (!mDeferWake && mAvailToClient + stepCount >= minimum) { + ALOGV("mAvailToClient=%u stepCount=%u minimum=%u", mAvailToClient, stepCount, minimum); + // could client be sleeping, or not need this increment and counter overflows? + int32_t old = android_atomic_inc(&cblk->mFutex); + if (old == -1) { + (void) __futex_syscall3(&cblk->mFutex, + mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1); + } } - lock.unlock(); - return true; + + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; } -void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const +// --------------------------------------------------------------------------- + +size_t AudioTrackServerProxy::framesReady() { - return (int8_t *)buffers + (offset - userBase) * frameSize; + LOG_ALWAYS_FATAL_IF(!mIsOut); + + if (mIsShutdown) { + return 0; + } + audio_track_cblk_t* cblk = mCblk; + // the acquire might not be necessary since not doing a subsequent read + int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear); + ssize_t filled = rear - cblk->u.mStreaming.mFront; + // pipe should not already be overfull + if (!(0 <= filled && (size_t) filled <= mFrameCount)) { + ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled); + mIsShutdown = true; + return 0; + } + // cache this value for later use by obtainBuffer(), with added barrier + // and racy if called by normal mixer thread + // ignores flush(), so framesReady() may report a larger mFrameCount than obtainBuffer() + return filled; } -uint32_t audio_track_cblk_t::framesAvailable(size_t frameCount, bool isOut) +// --------------------------------------------------------------------------- + +StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, + size_t frameCount, size_t frameSize) + : AudioTrackServerProxy(cblk, buffers, frameCount, frameSize), + mObserver(&cblk->u.mStatic.mSingleStateQueue), mPosition(0), + mEnd(frameCount), mFramesReadyIsCalledByMultipleThreads(false) { - Mutex::Autolock _l(lock); - return framesAvailable_l(frameCount, isOut); + mState.mLoopStart = 0; + mState.mLoopEnd = 0; + mState.mLoopCount = 0; } -uint32_t audio_track_cblk_t::framesAvailable_l(size_t frameCount, bool isOut) +void StaticAudioTrackServerProxy::framesReadyIsCalledByMultipleThreads() { - uint32_t u = user; - uint32_t s = server; + mFramesReadyIsCalledByMultipleThreads = true; +} - if (isOut) { - uint32_t limit = (s < loopStart) ? s : loopStart; - return limit + frameCount - u; - } else { - return frameCount + u - s; +size_t StaticAudioTrackServerProxy::framesReady() +{ + // FIXME + // This is racy if called by normal mixer thread, + // as we're reading 2 independent variables without a lock. + // Can't call mObserver.poll(), as we might be called from wrong thread. + // If looping is enabled, should return a higher number (since includes non-contiguous). + size_t position = mPosition; + if (!mFramesReadyIsCalledByMultipleThreads) { + ssize_t positionOrStatus = pollPosition(); + if (positionOrStatus >= 0) { + position = (size_t) positionOrStatus; + } } + size_t end = mEnd; + return position < end ? end - position : 0; } -uint32_t audio_track_cblk_t::framesReady(bool isOut) +ssize_t StaticAudioTrackServerProxy::pollPosition() { - uint32_t u = user; - uint32_t s = server; - - if (isOut) { - if (u < loopEnd) { - return u - s; - } else { - // do not block on mutex shared with client on AudioFlinger side - if (!tryLock()) { - ALOGW("framesReady() could not lock cblk"); - return 0; + size_t position = mPosition; + StaticAudioTrackState state; + if (mObserver.poll(state)) { + bool valid = false; + size_t loopStart = state.mLoopStart; + size_t loopEnd = state.mLoopEnd; + if (state.mLoopCount == 0) { + if (loopStart > mFrameCount) { + loopStart = mFrameCount; } - uint32_t frames = UINT_MAX; - if (loopCount >= 0) { - frames = (loopEnd - loopStart)*loopCount + u - s; + // ignore loopEnd + mPosition = position = loopStart; + mEnd = mFrameCount; + mState.mLoopCount = 0; + valid = true; + } else { + if (loopStart < loopEnd && loopEnd <= mFrameCount && + loopEnd - loopStart >= MIN_LOOP) { + if (!(loopStart <= position && position < loopEnd)) { + mPosition = position = loopStart; + } + mEnd = loopEnd; + mState = state; + valid = true; } - lock.unlock(); - return frames; } - } else { - return s - u; + if (!valid) { + ALOGE("%s client pushed an invalid state, shutting down", __func__); + mIsShutdown = true; + return (ssize_t) NO_INIT; + } + mCblk->u.mStatic.mBufferPosition = position; } + return (ssize_t) position; } -bool audio_track_cblk_t::tryLock() +status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer) { - // the code below simulates lock-with-timeout - // we MUST do this to protect the AudioFlinger server - // as this lock is shared with the client. - status_t err; + if (mIsShutdown) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + mUnreleased = 0; + return NO_INIT; + } + ssize_t positionOrStatus = pollPosition(); + if (positionOrStatus < 0) { + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; + mUnreleased = 0; + return (status_t) positionOrStatus; + } + size_t position = (size_t) positionOrStatus; + size_t avail; + if (position < mEnd) { + avail = mEnd - position; + size_t wanted = buffer->mFrameCount; + if (avail < wanted) { + buffer->mFrameCount = avail; + } else { + avail = wanted; + } + buffer->mRaw = &((char *) mBuffers)[position * mFrameSize]; + } else { + avail = 0; + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + } + buffer->mNonContig = 0; // FIXME should be > 0 for looping + mUnreleased = avail; + return NO_ERROR; +} - err = lock.tryLock(); - if (err == -EBUSY) { // just wait a bit - usleep(1000); - err = lock.tryLock(); +void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer) +{ + size_t stepCount = buffer->mFrameCount; + LOG_ALWAYS_FATAL_IF(stepCount > mUnreleased); + if (stepCount == 0) { + buffer->mRaw = NULL; + buffer->mNonContig = 0; + return; } - if (err != NO_ERROR) { - // probably, the client just died. - return false; + mUnreleased -= stepCount; + audio_track_cblk_t* cblk = mCblk; + size_t position = mPosition; + size_t newPosition = position + stepCount; + int32_t setFlags = 0; + if (!(position <= newPosition && newPosition <= mFrameCount)) { + ALOGW("%s newPosition %u outside [%u, %u]", __func__, newPosition, position, mFrameCount); + newPosition = mFrameCount; + } else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) { + if (mState.mLoopCount == -1 || --mState.mLoopCount != 0) { + newPosition = mState.mLoopStart; + setFlags = CBLK_LOOP_CYCLE; + } else { + mEnd = mFrameCount; // this is what allows playback to continue after the loop + setFlags = CBLK_LOOP_FINAL; + } } - return true; + if (newPosition == mFrameCount) { + setFlags |= CBLK_BUFFER_END; + } + mPosition = newPosition; + + cblk->server += stepCount; + cblk->u.mStatic.mBufferPosition = newPosition; + if (setFlags != 0) { + (void) android_atomic_or(setFlags, &cblk->flags); + // this would be a good place to wake a futex + } + + buffer->mFrameCount = 0; + buffer->mRaw = NULL; + buffer->mNonContig = 0; } +// --------------------------------------------------------------------------- + } // namespace android diff --git a/media/libmedia/IHDCP.cpp b/media/libmedia/IHDCP.cpp index f13addc..a46ff91 100644 --- a/media/libmedia/IHDCP.cpp +++ b/media/libmedia/IHDCP.cpp @@ -31,6 +31,7 @@ enum { HDCP_INIT_ASYNC, HDCP_SHUTDOWN_ASYNC, HDCP_ENCRYPT, + HDCP_ENCRYPT_NATIVE, HDCP_DECRYPT, }; @@ -108,6 +109,31 @@ struct BpHDCP : public BpInterface<IHDCP> { return err; } + virtual status_t encryptNative( + const sp<GraphicBuffer> &graphicBuffer, + size_t offset, size_t size, uint32_t streamCTR, + uint64_t *outInputCTR, void *outData) { + Parcel data, reply; + data.writeInterfaceToken(IHDCP::getInterfaceDescriptor()); + data.write(*graphicBuffer); + data.writeInt32(offset); + data.writeInt32(size); + data.writeInt32(streamCTR); + remote()->transact(HDCP_ENCRYPT_NATIVE, data, &reply); + + status_t err = reply.readInt32(); + + if (err != OK) { + *outInputCTR = 0; + return err; + } + + *outInputCTR = reply.readInt64(); + reply.read(outData, size); + + return err; + } + virtual status_t decrypt( const void *inData, size_t size, uint32_t streamCTR, uint64_t inputCTR, @@ -222,6 +248,34 @@ status_t BnHDCP::onTransact( return OK; } + case HDCP_ENCRYPT_NATIVE: + { + CHECK_INTERFACE(IHDCP, data, reply); + + sp<GraphicBuffer> graphicBuffer = new GraphicBuffer(); + data.read(*graphicBuffer); + size_t offset = data.readInt32(); + size_t size = data.readInt32(); + uint32_t streamCTR = data.readInt32(); + void *outData = malloc(size); + uint64_t inputCTR; + + status_t err = encryptNative(graphicBuffer, offset, size, + streamCTR, &inputCTR, outData); + + reply->writeInt32(err); + + if (err == OK) { + reply->writeInt64(inputCTR); + reply->write(outData, size); + } + + free(outData); + outData = NULL; + + return OK; + } + case HDCP_DECRYPT: { size_t size = data.readInt32(); diff --git a/media/libmedia/IMediaDeathNotifier.cpp b/media/libmedia/IMediaDeathNotifier.cpp index 9199db6..9db5b1b 100644 --- a/media/libmedia/IMediaDeathNotifier.cpp +++ b/media/libmedia/IMediaDeathNotifier.cpp @@ -49,10 +49,10 @@ IMediaDeathNotifier::getMediaPlayerService() } while (true); if (sDeathNotifier == NULL) { - sDeathNotifier = new DeathNotifier(); - } - binder->linkToDeath(sDeathNotifier); - sMediaPlayerService = interface_cast<IMediaPlayerService>(binder); + sDeathNotifier = new DeathNotifier(); + } + binder->linkToDeath(sDeathNotifier); + sMediaPlayerService = interface_cast<IMediaPlayerService>(binder); } ALOGE_IF(sMediaPlayerService == 0, "no media player service!?"); return sMediaPlayerService; diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp index 59e538f..8fe5bb3 100644 --- a/media/libmedia/JetPlayer.cpp +++ b/media/libmedia/JetPlayer.cpp @@ -39,7 +39,6 @@ JetPlayer::JetPlayer(void *javaJetPlayer, int maxTracks, int trackBufferSize) : mMaxTracks(maxTracks), mEasData(NULL), mEasJetFileLoc(NULL), - mAudioTrack(NULL), mTrackBufferSize(trackBufferSize) { ALOGV("JetPlayer constructor"); @@ -140,11 +139,10 @@ int JetPlayer::release() free(mEasJetFileLoc); mEasJetFileLoc = NULL; } - if (mAudioTrack) { + if (mAudioTrack != 0) { mAudioTrack->stop(); mAudioTrack->flush(); - delete mAudioTrack; - mAudioTrack = NULL; + mAudioTrack.clear(); } if (mAudioBuffer) { delete mAudioBuffer; diff --git a/media/libmedia/MediaScannerClient.cpp b/media/libmedia/MediaScannerClient.cpp index e1e3348..93a4a4c 100644 --- a/media/libmedia/MediaScannerClient.cpp +++ b/media/libmedia/MediaScannerClient.cpp @@ -16,7 +16,7 @@ #include <media/mediascanner.h> -#include <utils/StringArray.h> +#include "StringArray.h" #include "autodetect.h" #include "unicode/ucnv.h" diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp index ee70ef7..e1e88ec 100644 --- a/media/libmedia/SoundPool.cpp +++ b/media/libmedia/SoundPool.cpp @@ -547,8 +547,8 @@ void SoundChannel::init(SoundPool* soundPool) void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftVolume, float rightVolume, int priority, int loop, float rate) { - AudioTrack* oldTrack; - AudioTrack* newTrack; + sp<AudioTrack> oldTrack; + sp<AudioTrack> newTrack; status_t status; { // scope for the lock @@ -620,7 +620,7 @@ void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftV ALOGE("Error creating AudioTrack"); goto exit; } - ALOGV("setVolume %p", newTrack); + ALOGV("setVolume %p", newTrack.get()); newTrack->setVolume(leftVolume, rightVolume); newTrack->setLoop(0, frameCount, loop); @@ -643,11 +643,9 @@ void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftV } exit: - ALOGV("delete oldTrack %p", oldTrack); - delete oldTrack; + ALOGV("delete oldTrack %p", oldTrack.get()); if (status != NO_ERROR) { - delete newTrack; - mAudioTrack = NULL; + mAudioTrack.clear(); } } @@ -884,7 +882,7 @@ SoundChannel::~SoundChannel() } // do not call AudioTrack destructor with mLock held as it will wait for the AudioTrack // callback thread to exit which may need to execute process() and acquire the mLock. - delete mAudioTrack; + mAudioTrack.clear(); } void SoundChannel::dump() diff --git a/media/libmedia/StringArray.cpp b/media/libmedia/StringArray.cpp new file mode 100644 index 0000000..5f5b57a --- /dev/null +++ b/media/libmedia/StringArray.cpp @@ -0,0 +1,113 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +// +// Sortable array of strings. STL-ish, but STL-free. +// + +#include <stdlib.h> +#include <string.h> + +#include "StringArray.h" + +namespace android { + +// +// An expanding array of strings. Add, get, sort, delete. +// +StringArray::StringArray() + : mMax(0), mCurrent(0), mArray(NULL) +{ +} + +StringArray:: ~StringArray() { + for (int i = 0; i < mCurrent; i++) + delete[] mArray[i]; + delete[] mArray; +} + +// +// Add a string. A copy of the string is made. +// +bool StringArray::push_back(const char* str) { + if (mCurrent >= mMax) { + char** tmp; + + if (mMax == 0) + mMax = 16; // initial storage + else + mMax *= 2; + + tmp = new char*[mMax]; + if (tmp == NULL) + return false; + + memcpy(tmp, mArray, mCurrent * sizeof(char*)); + delete[] mArray; + mArray = tmp; + } + + int len = strlen(str); + mArray[mCurrent] = new char[len+1]; + memcpy(mArray[mCurrent], str, len+1); + mCurrent++; + + return true; +} + +// +// Delete an entry. +// +void StringArray::erase(int idx) { + if (idx < 0 || idx >= mCurrent) + return; + delete[] mArray[idx]; + if (idx < mCurrent-1) { + memmove(&mArray[idx], &mArray[idx+1], + (mCurrent-1 - idx) * sizeof(char*)); + } + mCurrent--; +} + +// +// Sort the array. +// +void StringArray::sort(int (*compare)(const void*, const void*)) { + qsort(mArray, mCurrent, sizeof(char*), compare); +} + +// +// Pass this to the sort routine to do an ascending alphabetical sort. +// +int StringArray::cmpAscendingAlpha(const void* pstr1, const void* pstr2) { + return strcmp(*(const char**)pstr1, *(const char**)pstr2); +} + +// +// Set entry N to specified string. +// [should use operator[] here] +// +void StringArray::setEntry(int idx, const char* str) { + if (idx < 0 || idx >= mCurrent) + return; + delete[] mArray[idx]; + int len = strlen(str); + mArray[idx] = new char[len+1]; + memcpy(mArray[idx], str, len+1); +} + + +}; // namespace android diff --git a/media/libmedia/StringArray.h b/media/libmedia/StringArray.h new file mode 100644 index 0000000..ae47085 --- /dev/null +++ b/media/libmedia/StringArray.h @@ -0,0 +1,83 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +// +// Sortable array of strings. STL-ish, but STL-free. +// +#ifndef _LIBS_MEDIA_STRING_ARRAY_H +#define _LIBS_MEDIA_STRING_ARRAY_H + +#include <stdlib.h> +#include <string.h> + +namespace android { + +// +// An expanding array of strings. Add, get, sort, delete. +// +class StringArray { +public: + StringArray(); + virtual ~StringArray(); + + // + // Add a string. A copy of the string is made. + // + bool push_back(const char* str); + + // + // Delete an entry. + // + void erase(int idx); + + // + // Sort the array. + // + void sort(int (*compare)(const void*, const void*)); + + // + // Pass this to the sort routine to do an ascending alphabetical sort. + // + static int cmpAscendingAlpha(const void* pstr1, const void* pstr2); + + // + // Get the #of items in the array. + // + inline int size(void) const { return mCurrent; } + + // + // Return entry N. + // [should use operator[] here] + // + const char* getEntry(int idx) const { + return (unsigned(idx) >= unsigned(mCurrent)) ? NULL : mArray[idx]; + } + + // + // Set entry N to specified string. + // [should use operator[] here] + // + void setEntry(int idx, const char* str); + +private: + int mMax; + int mCurrent; + char** mArray; +}; + +}; // namespace android + +#endif // _LIBS_MEDIA_STRING_ARRAY_H diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp index f55b697..f9ad31d 100644 --- a/media/libmedia/ToneGenerator.cpp +++ b/media/libmedia/ToneGenerator.cpp @@ -803,7 +803,6 @@ ToneGenerator::ToneGenerator(audio_stream_type_t streamType, float volume, bool ALOGV("ToneGenerator constructor: streamType=%d, volume=%f", streamType, volume); mState = TONE_IDLE; - mpAudioTrack = NULL; if (AudioSystem::getOutputSamplingRate(&mSamplingRate, streamType) != NO_ERROR) { ALOGE("Unable to marshal AudioFlinger"); @@ -855,10 +854,10 @@ ToneGenerator::ToneGenerator(audio_stream_type_t streamType, float volume, bool ToneGenerator::~ToneGenerator() { ALOGV("ToneGenerator destructor"); - if (mpAudioTrack != NULL) { + if (mpAudioTrack != 0) { stopTone(); - ALOGV("Delete Track: %p", mpAudioTrack); - delete mpAudioTrack; + ALOGV("Delete Track: %p", mpAudioTrack.get()); + mpAudioTrack.clear(); } } @@ -1047,14 +1046,9 @@ void ToneGenerator::stopTone() { //////////////////////////////////////////////////////////////////////////////// bool ToneGenerator::initAudioTrack() { - if (mpAudioTrack) { - delete mpAudioTrack; - mpAudioTrack = NULL; - } - // Open audio track in mono, PCM 16bit, default sampling rate, default buffer size mpAudioTrack = new AudioTrack(); - ALOGV("Create Track: %p", mpAudioTrack); + ALOGV("Create Track: %p", mpAudioTrack.get()); mpAudioTrack->set(mStreamType, 0, // sampleRate @@ -1066,7 +1060,9 @@ bool ToneGenerator::initAudioTrack() { this, // user 0, // notificationFrames 0, // sharedBuffer - mThreadCanCallJava); + mThreadCanCallJava, + 0, // sessionId + AudioTrack::TRANSFER_CALLBACK); if (mpAudioTrack->initCheck() != NO_ERROR) { ALOGE("AudioTrack->initCheck failed"); @@ -1081,12 +1077,10 @@ bool ToneGenerator::initAudioTrack() { initAudioTrack_exit: + ALOGV("Init failed: %p", mpAudioTrack.get()); + // Cleanup - if (mpAudioTrack != NULL) { - ALOGV("Delete Track I: %p", mpAudioTrack); - delete mpAudioTrack; - mpAudioTrack = NULL; - } + mpAudioTrack.clear(); return false; } diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk index d87bc7f..8f21632 100644 --- a/media/libmediaplayerservice/Android.mk +++ b/media/libmediaplayerservice/Android.mk @@ -34,6 +34,7 @@ LOCAL_SHARED_LIBRARIES := \ libsonivox \ libstagefright \ libstagefright_foundation \ + libstagefright_httplive \ libstagefright_omx \ libstagefright_wfd \ libutils \ diff --git a/media/libmediaplayerservice/HDCP.cpp b/media/libmediaplayerservice/HDCP.cpp index 469a02e..8a3188c 100644 --- a/media/libmediaplayerservice/HDCP.cpp +++ b/media/libmediaplayerservice/HDCP.cpp @@ -116,6 +116,24 @@ status_t HDCP::encrypt( return mHDCPModule->encrypt(inData, size, streamCTR, outInputCTR, outData); } +status_t HDCP::encryptNative( + const sp<GraphicBuffer> &graphicBuffer, + size_t offset, size_t size, uint32_t streamCTR, + uint64_t *outInputCTR, void *outData) { + Mutex::Autolock autoLock(mLock); + + CHECK(mIsEncryptionModule); + + if (mHDCPModule == NULL) { + *outInputCTR = 0; + + return NO_INIT; + } + + return mHDCPModule->encryptNative(graphicBuffer->handle, + offset, size, streamCTR, outInputCTR, outData); +} + status_t HDCP::decrypt( const void *inData, size_t size, uint32_t streamCTR, uint64_t outInputCTR, void *outData) { diff --git a/media/libmediaplayerservice/HDCP.h b/media/libmediaplayerservice/HDCP.h index 42e6467..c60c2e0 100644 --- a/media/libmediaplayerservice/HDCP.h +++ b/media/libmediaplayerservice/HDCP.h @@ -35,6 +35,11 @@ struct HDCP : public BnHDCP { const void *inData, size_t size, uint32_t streamCTR, uint64_t *outInputCTR, void *outData); + virtual status_t encryptNative( + const sp<GraphicBuffer> &graphicBuffer, + size_t offset, size_t size, uint32_t streamCTR, + uint64_t *outInputCTR, void *outData); + virtual status_t decrypt( const void *inData, size_t size, uint32_t streamCTR, uint64_t outInputCTR, void *outData); diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp index e600a3f..fa1ff36 100644 --- a/media/libmediaplayerservice/MediaPlayerService.cpp +++ b/media/libmediaplayerservice/MediaPlayerService.cpp @@ -1295,8 +1295,6 @@ MediaPlayerService::AudioOutput::AudioOutput(int sessionId) mSessionId(sessionId), mFlags(AUDIO_OUTPUT_FLAG_NONE) { ALOGV("AudioOutput(%d)", sessionId); - mTrack = 0; - mRecycledTrack = 0; mStreamType = AUDIO_STREAM_MUSIC; mLeftVolume = 1.0; mRightVolume = 1.0; @@ -1311,7 +1309,6 @@ MediaPlayerService::AudioOutput::AudioOutput(int sessionId) MediaPlayerService::AudioOutput::~AudioOutput() { close(); - delete mRecycledTrack; delete mCallbackData; } @@ -1422,7 +1419,7 @@ status_t MediaPlayerService::AudioOutput::open( } } - AudioTrack *t; + sp<AudioTrack> t; CallbackData *newcbd = NULL; if (mCallback != NULL) { newcbd = new CallbackData(this); @@ -1453,13 +1450,12 @@ status_t MediaPlayerService::AudioOutput::open( if ((t == 0) || (t->initCheck() != NO_ERROR)) { ALOGE("Unable to create audio track"); - delete t; delete newcbd; return NO_INIT; } - if (mRecycledTrack) { + if (mRecycledTrack != 0) { // check if the existing track can be reused as-is, or if a new track needs to be created. bool reuse = true; @@ -1484,11 +1480,10 @@ status_t MediaPlayerService::AudioOutput::open( ALOGV("chaining to next output"); close(); mTrack = mRecycledTrack; - mRecycledTrack = NULL; + mRecycledTrack.clear(); if (mCallbackData != NULL) { mCallbackData->setOutput(this); } - delete t; delete newcbd; return OK; } @@ -1499,8 +1494,7 @@ status_t MediaPlayerService::AudioOutput::open( mCallbackData->endTrackSwitch(); } mRecycledTrack->flush(); - delete mRecycledTrack; - mRecycledTrack = NULL; + mRecycledTrack.clear(); delete mCallbackData; mCallbackData = NULL; close(); @@ -1533,7 +1527,7 @@ void MediaPlayerService::AudioOutput::start() if (mCallbackData != NULL) { mCallbackData->endTrackSwitch(); } - if (mTrack) { + if (mTrack != 0) { mTrack->setVolume(mLeftVolume, mRightVolume); mTrack->setAuxEffectSendLevel(mSendLevel); mTrack->start(); @@ -1555,7 +1549,7 @@ void MediaPlayerService::AudioOutput::switchToNextOutput() { mNextOutput->mCallbackData = mCallbackData; mCallbackData = NULL; mNextOutput->mRecycledTrack = mTrack; - mTrack = NULL; + mTrack.clear(); mNextOutput->mSampleRateHz = mSampleRateHz; mNextOutput->mMsecsPerFrame = mMsecsPerFrame; mNextOutput->mBytesWritten = mBytesWritten; @@ -1568,7 +1562,7 @@ ssize_t MediaPlayerService::AudioOutput::write(const void* buffer, size_t size) LOG_FATAL_IF(mCallback != NULL, "Don't call write if supplying a callback."); //ALOGV("write(%p, %u)", buffer, size); - if (mTrack) { + if (mTrack != 0) { ssize_t ret = mTrack->write(buffer, size); mBytesWritten += ret; return ret; @@ -1579,26 +1573,25 @@ ssize_t MediaPlayerService::AudioOutput::write(const void* buffer, size_t size) void MediaPlayerService::AudioOutput::stop() { ALOGV("stop"); - if (mTrack) mTrack->stop(); + if (mTrack != 0) mTrack->stop(); } void MediaPlayerService::AudioOutput::flush() { ALOGV("flush"); - if (mTrack) mTrack->flush(); + if (mTrack != 0) mTrack->flush(); } void MediaPlayerService::AudioOutput::pause() { ALOGV("pause"); - if (mTrack) mTrack->pause(); + if (mTrack != 0) mTrack->pause(); } void MediaPlayerService::AudioOutput::close() { ALOGV("close"); - delete mTrack; - mTrack = 0; + mTrack.clear(); } void MediaPlayerService::AudioOutput::setVolume(float left, float right) @@ -1606,7 +1599,7 @@ void MediaPlayerService::AudioOutput::setVolume(float left, float right) ALOGV("setVolume(%f, %f)", left, right); mLeftVolume = left; mRightVolume = right; - if (mTrack) { + if (mTrack != 0) { mTrack->setVolume(left, right); } } @@ -1615,7 +1608,7 @@ status_t MediaPlayerService::AudioOutput::setPlaybackRatePermille(int32_t ratePe { ALOGV("setPlaybackRatePermille(%d)", ratePermille); status_t res = NO_ERROR; - if (mTrack) { + if (mTrack != 0) { res = mTrack->setSampleRate(ratePermille * mSampleRateHz / 1000); } else { res = NO_INIT; @@ -1631,7 +1624,7 @@ status_t MediaPlayerService::AudioOutput::setAuxEffectSendLevel(float level) { ALOGV("setAuxEffectSendLevel(%f)", level); mSendLevel = level; - if (mTrack) { + if (mTrack != 0) { return mTrack->setAuxEffectSendLevel(level); } return NO_ERROR; @@ -1641,7 +1634,7 @@ status_t MediaPlayerService::AudioOutput::attachAuxEffect(int effectId) { ALOGV("attachAuxEffect(%d)", effectId); mAuxEffectId = effectId; - if (mTrack) { + if (mTrack != 0) { return mTrack->attachAuxEffect(effectId); } return NO_ERROR; diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h index b33805d..e586156 100644 --- a/media/libmediaplayerservice/MediaPlayerService.h +++ b/media/libmediaplayerservice/MediaPlayerService.h @@ -78,7 +78,7 @@ class MediaPlayerService : public BnMediaPlayerService AudioOutput(int sessionId); virtual ~AudioOutput(); - virtual bool ready() const { return mTrack != NULL; } + virtual bool ready() const { return mTrack != 0; } virtual bool realtime() const { return true; } virtual ssize_t bufferSize() const; virtual ssize_t frameCount() const; @@ -120,8 +120,8 @@ class MediaPlayerService : public BnMediaPlayerService static void CallbackWrapper( int event, void *me, void *info); - AudioTrack* mTrack; - AudioTrack* mRecycledTrack; + sp<AudioTrack> mTrack; + sp<AudioTrack> mRecycledTrack; sp<AudioOutput> mNextOutput; AudioCallback mCallback; void * mCallbackCookie; diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp index 655ee55..c8901ce 100644 --- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp +++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp @@ -20,7 +20,6 @@ #include "HTTPLiveSource.h" -#include "ATSParser.h" #include "AnotherPacketSource.h" #include "LiveDataSource.h" #include "LiveSession.h" @@ -62,7 +61,10 @@ NuPlayer::HTTPLiveSource::HTTPLiveSource( NuPlayer::HTTPLiveSource::~HTTPLiveSource() { if (mLiveSession != NULL) { mLiveSession->disconnect(); + mLiveSession.clear(); + mLiveLooper->stop(); + mLiveLooper.clear(); } } @@ -76,112 +78,42 @@ void NuPlayer::HTTPLiveSource::prepareAsync() { mLiveSession = new LiveSession( notify, (mFlags & kFlagIncognito) ? LiveSession::kFlagIncognito : 0, - mUIDValid, mUID); + mUIDValid, + mUID); mLiveLooper->registerHandler(mLiveSession); - mLiveSession->connect( + mLiveSession->connectAsync( mURL.c_str(), mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders); - - mTSParser = new ATSParser; } void NuPlayer::HTTPLiveSource::start() { } -sp<MetaData> NuPlayer::HTTPLiveSource::getFormatMeta(bool audio) { - ATSParser::SourceType type = - audio ? ATSParser::AUDIO : ATSParser::VIDEO; - - sp<AnotherPacketSource> source = - static_cast<AnotherPacketSource *>(mTSParser->getSource(type).get()); +sp<AMessage> NuPlayer::HTTPLiveSource::getFormat(bool audio) { + sp<AMessage> format; + status_t err = mLiveSession->getStreamFormat( + audio ? LiveSession::STREAMTYPE_AUDIO + : LiveSession::STREAMTYPE_VIDEO, + &format); - if (source == NULL) { + if (err != OK) { return NULL; } - return source->getFormat(); + return format; } status_t NuPlayer::HTTPLiveSource::feedMoreTSData() { - if (mFinalResult != OK) { - return mFinalResult; - } - - sp<LiveDataSource> source = - static_cast<LiveDataSource *>(mLiveSession->getDataSource().get()); - - for (int32_t i = 0; i < 50; ++i) { - char buffer[188]; - ssize_t n = source->readAtNonBlocking(mOffset, buffer, sizeof(buffer)); - - if (n == -EWOULDBLOCK) { - break; - } else if (n < 0) { - if (n != ERROR_END_OF_STREAM) { - ALOGI("input data EOS reached, error %ld", n); - } else { - ALOGI("input data EOS reached."); - } - mTSParser->signalEOS(n); - mFinalResult = n; - break; - } else { - if (buffer[0] == 0x00) { - // XXX legacy - - uint8_t type = buffer[1]; - - sp<AMessage> extra = new AMessage; - - if (type & 2) { - int64_t mediaTimeUs; - memcpy(&mediaTimeUs, &buffer[2], sizeof(mediaTimeUs)); - - extra->setInt64(IStreamListener::kKeyMediaTimeUs, mediaTimeUs); - } - - mTSParser->signalDiscontinuity( - ((type & 1) == 0) - ? ATSParser::DISCONTINUITY_SEEK - : ATSParser::DISCONTINUITY_FORMATCHANGE, - extra); - } else { - status_t err = mTSParser->feedTSPacket(buffer, sizeof(buffer)); - - if (err != OK) { - ALOGE("TS Parser returned error %d", err); - mTSParser->signalEOS(err); - mFinalResult = err; - break; - } - } - - mOffset += n; - } - } - return OK; } status_t NuPlayer::HTTPLiveSource::dequeueAccessUnit( bool audio, sp<ABuffer> *accessUnit) { - ATSParser::SourceType type = - audio ? ATSParser::AUDIO : ATSParser::VIDEO; - - sp<AnotherPacketSource> source = - static_cast<AnotherPacketSource *>(mTSParser->getSource(type).get()); - - if (source == NULL) { - return -EWOULDBLOCK; - } - - status_t finalResult; - if (!source->hasBufferAvailable(&finalResult)) { - return finalResult == OK ? -EWOULDBLOCK : finalResult; - } - - return source->dequeueAccessUnit(accessUnit); + return mLiveSession->dequeueAccessUnit( + audio ? LiveSession::STREAMTYPE_AUDIO + : LiveSession::STREAMTYPE_VIDEO, + accessUnit); } status_t NuPlayer::HTTPLiveSource::getDuration(int64_t *durationUs) { @@ -189,15 +121,7 @@ status_t NuPlayer::HTTPLiveSource::getDuration(int64_t *durationUs) { } status_t NuPlayer::HTTPLiveSource::seekTo(int64_t seekTimeUs) { - // We need to make sure we're not seeking until we have seen the very first - // PTS timestamp in the whole stream (from the beginning of the stream). - while (!mTSParser->PTSTimeDeltaEstablished() && feedMoreTSData() == OK) { - usleep(100000); - } - - mLiveSession->seekTo(seekTimeUs); - - return OK; + return mLiveSession->seekTo(seekTimeUs); } void NuPlayer::HTTPLiveSource::onMessageReceived(const sp<AMessage> &msg) { @@ -249,6 +173,32 @@ void NuPlayer::HTTPLiveSource::onSessionNotify(const sp<AMessage> &msg) { break; } + case LiveSession::kWhatStreamsChanged: + { + uint32_t changedMask; + CHECK(msg->findInt32( + "changedMask", (int32_t *)&changedMask)); + + bool audio = changedMask & LiveSession::STREAMTYPE_AUDIO; + bool video = changedMask & LiveSession::STREAMTYPE_VIDEO; + + sp<AMessage> reply; + CHECK(msg->findMessage("reply", &reply)); + + sp<AMessage> notify = dupNotify(); + notify->setInt32("what", kWhatQueueDecoderShutdown); + notify->setInt32("audio", audio); + notify->setInt32("video", video); + notify->setMessage("reply", reply); + notify->post(); + break; + } + + case LiveSession::kWhatError: + { + break; + } + default: TRESPASS(); } diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h index 067d1da..aa9434b 100644 --- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h +++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.h @@ -23,7 +23,6 @@ namespace android { -struct ATSParser; struct LiveSession; struct NuPlayer::HTTPLiveSource : public NuPlayer::Source { @@ -37,18 +36,16 @@ struct NuPlayer::HTTPLiveSource : public NuPlayer::Source { virtual void prepareAsync(); virtual void start(); - virtual status_t feedMoreTSData(); - virtual status_t dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit); + virtual sp<AMessage> getFormat(bool audio); + virtual status_t feedMoreTSData(); virtual status_t getDuration(int64_t *durationUs); virtual status_t seekTo(int64_t seekTimeUs); protected: virtual ~HTTPLiveSource(); - virtual sp<MetaData> getFormatMeta(bool audio); - virtual void onMessageReceived(const sp<AMessage> &msg); private: @@ -70,7 +67,6 @@ private: off64_t mOffset; sp<ALooper> mLiveLooper; sp<LiveSession> mLiveSession; - sp<ATSParser> mTSParser; void onSessionNotify(const sp<AMessage> &msg); diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp index b89b1c8..7e81035 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp @@ -89,6 +89,38 @@ private: DISALLOW_EVIL_CONSTRUCTORS(SetSurfaceAction); }; +struct NuPlayer::ShutdownDecoderAction : public Action { + ShutdownDecoderAction(bool audio, bool video) + : mAudio(audio), + mVideo(video) { + } + + virtual void execute(NuPlayer *player) { + player->performDecoderShutdown(mAudio, mVideo); + } + +private: + bool mAudio; + bool mVideo; + + DISALLOW_EVIL_CONSTRUCTORS(ShutdownDecoderAction); +}; + +struct NuPlayer::PostMessageAction : public Action { + PostMessageAction(const sp<AMessage> &msg) + : mMessage(msg) { + } + + virtual void execute(NuPlayer *) { + mMessage->post(); + } + +private: + sp<AMessage> mMessage; + + DISALLOW_EVIL_CONSTRUCTORS(PostMessageAction); +}; + // Use this if there's no state necessary to save in order to execute // the action. struct NuPlayer::SimpleAction : public Action { @@ -335,7 +367,8 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { ALOGV("kWhatSetVideoNativeWindow"); mDeferredActions.push_back( - new SimpleAction(&NuPlayer::performDecoderShutdown)); + new ShutdownDecoderAction( + false /* audio */, true /* video */)); sp<RefBase> obj; CHECK(msg->findObject("native-window", &obj)); @@ -712,7 +745,8 @@ void NuPlayer::onMessageReceived(const sp<AMessage> &msg) { ALOGV("kWhatReset"); mDeferredActions.push_back( - new SimpleAction(&NuPlayer::performDecoderShutdown)); + new ShutdownDecoderAction( + true /* audio */, true /* video */)); mDeferredActions.push_back( new SimpleAction(&NuPlayer::performReset)); @@ -1023,6 +1057,9 @@ void NuPlayer::notifyListener(int msg, int ext1, int ext2) { } void NuPlayer::flushDecoder(bool audio, bool needShutdown) { + ALOGV("[%s] flushDecoder needShutdown=%d", + audio ? "audio" : "video", needShutdown); + if ((audio && mAudioDecoder == NULL) || (!audio && mVideoDecoder == NULL)) { ALOGI("flushDecoder %s without decoder present", audio ? "audio" : "video"); @@ -1173,20 +1210,29 @@ void NuPlayer::performDecoderFlush() { } } -void NuPlayer::performDecoderShutdown() { - ALOGV("performDecoderShutdown"); +void NuPlayer::performDecoderShutdown(bool audio, bool video) { + ALOGV("performDecoderShutdown audio=%d, video=%d", audio, video); - if (mAudioDecoder == NULL && mVideoDecoder == NULL) { + if ((!audio || mAudioDecoder == NULL) + && (!video || mVideoDecoder == NULL)) { return; } mTimeDiscontinuityPending = true; - if (mAudioDecoder != NULL) { + if (mFlushingAudio == NONE && (!audio || mAudioDecoder == NULL)) { + mFlushingAudio = FLUSHED; + } + + if (mFlushingVideo == NONE && (!video || mVideoDecoder == NULL)) { + mFlushingVideo = FLUSHED; + } + + if (audio && mAudioDecoder != NULL) { flushDecoder(true /* audio */, true /* needShutdown */); } - if (mVideoDecoder != NULL) { + if (video && mVideoDecoder != NULL) { flushDecoder(false /* audio */, true /* needShutdown */); } } @@ -1322,6 +1368,19 @@ void NuPlayer::onSourceNotify(const sp<AMessage> &msg) { break; } + case Source::kWhatQueueDecoderShutdown: + { + int32_t audio, video; + CHECK(msg->findInt32("audio", &audio)); + CHECK(msg->findInt32("video", &video)); + + sp<AMessage> reply; + CHECK(msg->findMessage("reply", &reply)); + + queueDecoderShutdown(audio, video, reply); + break; + } + default: TRESPASS(); } @@ -1355,4 +1414,19 @@ void NuPlayer::Source::onMessageReceived(const sp<AMessage> &msg) { TRESPASS(); } +void NuPlayer::queueDecoderShutdown( + bool audio, bool video, const sp<AMessage> &reply) { + ALOGI("queueDecoderShutdown audio=%d, video=%d", audio, video); + + mDeferredActions.push_back( + new ShutdownDecoderAction(audio, video)); + + mDeferredActions.push_back( + new SimpleAction(&NuPlayer::performScanSources)); + + mDeferredActions.push_back(new PostMessageAction(reply)); + + processDeferredActions(); +} + } // namespace android diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h index 50d0462..8b6c8c1 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayer.h +++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h @@ -80,6 +80,8 @@ private: struct Action; struct SeekAction; struct SetSurfaceAction; + struct ShutdownDecoderAction; + struct PostMessageAction; struct SimpleAction; enum { @@ -172,13 +174,16 @@ private: void performSeek(int64_t seekTimeUs); void performDecoderFlush(); - void performDecoderShutdown(); + void performDecoderShutdown(bool audio, bool video); void performReset(); void performScanSources(); void performSetSurface(const sp<NativeWindowWrapper> &wrapper); void onSourceNotify(const sp<AMessage> &msg); + void queueDecoderShutdown( + bool audio, bool video, const sp<AMessage> &reply); + DISALLOW_EVIL_CONSTRUCTORS(NuPlayer); }; diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp index 404b56f..b543d9d 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp @@ -95,11 +95,11 @@ void NuPlayer::Renderer::flush(bool audio) { } void NuPlayer::Renderer::signalTimeDiscontinuity() { - CHECK(mAudioQueue.empty()); - CHECK(mVideoQueue.empty()); + // CHECK(mAudioQueue.empty()); + // CHECK(mVideoQueue.empty()); mAnchorTimeMediaUs = -1; mAnchorTimeRealUs = -1; - mSyncQueues = mHasAudio && mHasVideo; + mSyncQueues = false; } void NuPlayer::Renderer::pause() { diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h index 1cbf575..81ffd21 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h +++ b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h @@ -42,6 +42,7 @@ struct NuPlayer::Source : public AHandler { kWhatVideoSizeChanged, kWhatBufferingStart, kWhatBufferingEnd, + kWhatQueueDecoderShutdown, }; // The provides message is used to notify the player about various diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp index cf41cf2..bf650b4 100644 --- a/media/libstagefright/ACodec.cpp +++ b/media/libstagefright/ACodec.cpp @@ -359,6 +359,7 @@ ACodec::ACodec() mNode(NULL), mSentFormat(false), mIsEncoder(false), + mUseMetadataOnEncoderOutput(false), mShutdownInProgress(false), mEncoderDelay(0), mEncoderPadding(0), @@ -483,7 +484,8 @@ status_t ACodec::allocateBuffersOnPort(OMX_U32 portIndex) { ? OMXCodec::kRequiresAllocateBufferOnInputPorts : OMXCodec::kRequiresAllocateBufferOnOutputPorts; - if (portIndex == kPortIndexInput && (mFlags & kFlagIsSecure)) { + if ((portIndex == kPortIndexInput && (mFlags & kFlagIsSecure)) + || mUseMetadataOnEncoderOutput) { mem.clear(); void *ptr; @@ -491,7 +493,10 @@ status_t ACodec::allocateBuffersOnPort(OMX_U32 portIndex) { mNode, portIndex, def.nBufferSize, &info.mBufferID, &ptr); - info.mData = new ABuffer(ptr, def.nBufferSize); + int32_t bufSize = mUseMetadataOnEncoderOutput ? + (4 + sizeof(buffer_handle_t)) : def.nBufferSize; + + info.mData = new ABuffer(ptr, bufSize); } else if (mQuirks & requiresAllocateBufferBit) { err = mOMX->allocateBufferWithBackup( mNode, portIndex, mem, &info.mBufferID); @@ -912,14 +917,14 @@ status_t ACodec::configureCodec( err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexInput, OMX_TRUE); if (err != OK) { - ALOGE("[%s] storeMetaDataInBuffers failed w/ err %d", - mComponentName.c_str(), err); + ALOGE("[%s] storeMetaDataInBuffers (input) failed w/ err %d", + mComponentName.c_str(), err); - return err; - } - } + return err; + } + } - int32_t prependSPSPPS; + int32_t prependSPSPPS = 0; if (encoder && msg->findInt32("prepend-sps-pps-to-idr-frames", &prependSPSPPS) && prependSPSPPS != 0) { @@ -946,7 +951,27 @@ status_t ACodec::configureCodec( } } - if (!strncasecmp(mime, "video/", 6)) { + // Only enable metadata mode on encoder output if encoder can prepend + // sps/pps to idr frames, since in metadata mode the bitstream is in an + // opaque handle, to which we don't have access. + int32_t video = !strncasecmp(mime, "video/", 6); + if (encoder && video) { + OMX_BOOL enable = (OMX_BOOL) (prependSPSPPS + && msg->findInt32("store-metadata-in-buffers-output", &storeMeta) + && storeMeta != 0); + + err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, enable); + + if (err != OK) { + ALOGE("[%s] storeMetaDataInBuffers (output) failed w/ err %d", + mComponentName.c_str(), err); + mUseMetadataOnEncoderOutput = 0; + } else { + mUseMetadataOnEncoderOutput = enable; + } + } + + if (video) { if (encoder) { err = setupVideoEncoder(mime, msg); } else { @@ -2321,10 +2346,15 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { ¶ms, sizeof(params)), (status_t)OK); + CHECK_GT(params.nChannels, 0); CHECK(params.nChannels == 1 || params.bInterleaved); CHECK_EQ(params.nBitPerSample, 16u); - CHECK_EQ((int)params.eNumData, (int)OMX_NumericalDataSigned); - CHECK_EQ((int)params.ePCMMode, (int)OMX_AUDIO_PCMModeLinear); + + CHECK_EQ((int)params.eNumData, + (int)OMX_NumericalDataSigned); + + CHECK_EQ((int)params.ePCMMode, + (int)OMX_AUDIO_PCMModeLinear); notify->setString("mime", MEDIA_MIMETYPE_AUDIO_RAW); notify->setInt32("channel-count", params.nChannels); @@ -2334,11 +2364,14 @@ void ACodec::sendFormatChange(const sp<AMessage> &reply) { if (mSkipCutBuffer != NULL) { size_t prevbufsize = mSkipCutBuffer->size(); if (prevbufsize != 0) { - ALOGW("Replacing SkipCutBuffer holding %d bytes", prevbufsize); + ALOGW("Replacing SkipCutBuffer holding %d " + "bytes", + prevbufsize); } } - mSkipCutBuffer = new SkipCutBuffer(mEncoderDelay * frameSize, - mEncoderPadding * frameSize); + mSkipCutBuffer = new SkipCutBuffer( + mEncoderDelay * frameSize, + mEncoderPadding * frameSize); } if (mChannelMaskPresent) { @@ -3062,7 +3095,15 @@ bool ACodec::BaseState::onOMXFillBufferDone( mCodec->sendFormatChange(reply); } - info->mData->setRange(rangeOffset, rangeLength); + if (mCodec->mUseMetadataOnEncoderOutput) { + native_handle_t* handle = + *(native_handle_t**)(info->mData->data() + 4); + info->mData->meta()->setPointer("handle", handle); + info->mData->meta()->setInt32("rangeOffset", rangeOffset); + info->mData->meta()->setInt32("rangeLength", rangeLength); + } else { + info->mData->setRange(rangeOffset, rangeLength); + } #if 0 if (mCodec->mNativeWindow == NULL) { if (IsIDR(info->mData)) { @@ -3220,6 +3261,7 @@ void ACodec::UninitializedState::stateEntered() { mCodec->mOMX.clear(); mCodec->mQuirks = 0; mCodec->mFlags = 0; + mCodec->mUseMetadataOnEncoderOutput = 0; mCodec->mComponentName.clear(); } diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk index acc3abf..9544dbc 100644 --- a/media/libstagefright/Android.mk +++ b/media/libstagefright/Android.mk @@ -69,7 +69,6 @@ LOCAL_C_INCLUDES:= \ LOCAL_SHARED_LIBRARIES := \ libbinder \ libcamera_client \ - libcrypto \ libcutils \ libdl \ libdrmframework \ @@ -97,7 +96,6 @@ LOCAL_STATIC_LIBRARIES := \ libvpx \ libwebm \ libstagefright_mpeg2ts \ - libstagefright_httplive \ libstagefright_id3 \ libFLAC \ diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp index 4208019..92efae8 100644 --- a/media/libstagefright/AudioPlayer.cpp +++ b/media/libstagefright/AudioPlayer.cpp @@ -36,8 +36,7 @@ AudioPlayer::AudioPlayer( const sp<MediaPlayerBase::AudioSink> &audioSink, bool allowDeepBuffering, AwesomePlayer *observer) - : mAudioTrack(NULL), - mInputBuffer(NULL), + : mInputBuffer(NULL), mSampleRate(0), mLatencyUs(0), mFrameSize(0), @@ -166,8 +165,7 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) { 0, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this, 0); if ((err = mAudioTrack->initCheck()) != OK) { - delete mAudioTrack; - mAudioTrack = NULL; + mAudioTrack.clear(); if (mFirstBuffer != NULL) { mFirstBuffer->release(); @@ -235,8 +233,7 @@ void AudioPlayer::reset() { } else { mAudioTrack->stop(); - delete mAudioTrack; - mAudioTrack = NULL; + mAudioTrack.clear(); } // Make sure to release any buffer we hold onto so that the @@ -297,7 +294,7 @@ bool AudioPlayer::reachedEOS(status_t *finalStatus) { status_t AudioPlayer::setPlaybackRatePermille(int32_t ratePermille) { if (mAudioSink.get() != NULL) { return mAudioSink->setPlaybackRatePermille(ratePermille); - } else if (mAudioTrack != NULL){ + } else if (mAudioTrack != 0){ return mAudioTrack->setSampleRate(ratePermille * mSampleRate / 1000); } else { return NO_INIT; diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp index 3cf4d5c..bdd842f 100644 --- a/media/libstagefright/AudioSource.cpp +++ b/media/libstagefright/AudioSource.cpp @@ -49,8 +49,7 @@ static void AudioRecordCallbackFunction(int event, void *user, void *info) { AudioSource::AudioSource( audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount) - : mRecord(NULL), - mStarted(false), + : mStarted(false), mSampleRate(sampleRate), mPrevSampleTimeUs(0), mNumFramesReceived(0), @@ -91,9 +90,6 @@ AudioSource::~AudioSource() { if (mStarted) { reset(); } - - delete mRecord; - mRecord = NULL; } status_t AudioSource::initCheck() const { @@ -122,8 +118,7 @@ status_t AudioSource::start(MetaData *params) { if (err == OK) { mStarted = true; } else { - delete mRecord; - mRecord = NULL; + mRecord.clear(); } diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp index bd28118..6c197e2 100644 --- a/media/libstagefright/AwesomePlayer.cpp +++ b/media/libstagefright/AwesomePlayer.cpp @@ -597,7 +597,7 @@ void AwesomePlayer::notifyListener_l(int msg, int ext1, int ext2) { bool AwesomePlayer::getBitrate(int64_t *bitrate) { off64_t size; - if (mDurationUs >= 0 && mCachedSource != NULL + if (mDurationUs > 0 && mCachedSource != NULL && mCachedSource->getSize(&size) == OK) { *bitrate = size * 8000000ll / mDurationUs; // in bits/sec return true; diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp index 145869e..42a9c7a 100644 --- a/media/libstagefright/MPEG4Extractor.cpp +++ b/media/libstagefright/MPEG4Extractor.cpp @@ -341,6 +341,7 @@ MPEG4Extractor::MPEG4Extractor(const sp<DataSource> &source) mDataSource(source), mInitCheck(NO_INIT), mHasVideo(false), + mHeaderTimescale(0), mFirstTrack(NULL), mLastTrack(NULL), mFileMetaData(new MetaData), @@ -817,6 +818,7 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('i', 'l', 's', 't'): case FOURCC('s', 'i', 'n', 'f'): case FOURCC('s', 'c', 'h', 'i'): + case FOURCC('e', 'd', 't', 's'): { if (chunk_type == FOURCC('s', 't', 'b', 'l')) { ALOGV("sampleTable chunk is %d bytes long.", (size_t)chunk_size); @@ -904,6 +906,68 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { break; } + case FOURCC('e', 'l', 's', 't'): + { + // See 14496-12 8.6.6 + uint8_t version; + if (mDataSource->readAt(data_offset, &version, 1) < 1) { + return ERROR_IO; + } + + uint32_t entry_count; + if (!mDataSource->getUInt32(data_offset + 4, &entry_count)) { + return ERROR_IO; + } + + if (entry_count != 1) { + // we only support a single entry at the moment, for gapless playback + ALOGW("ignoring edit list with %d entries", entry_count); + } else if (mHeaderTimescale == 0) { + ALOGW("ignoring edit list because timescale is 0"); + } else { + off64_t entriesoffset = data_offset + 8; + uint64_t segment_duration; + int64_t media_time; + + if (version == 1) { + if (!mDataSource->getUInt64(entriesoffset, &segment_duration) || + !mDataSource->getUInt64(entriesoffset + 8, (uint64_t*)&media_time)) { + return ERROR_IO; + } + } else if (version == 0) { + uint32_t sd; + int32_t mt; + if (!mDataSource->getUInt32(entriesoffset, &sd) || + !mDataSource->getUInt32(entriesoffset + 4, (uint32_t*)&mt)) { + return ERROR_IO; + } + segment_duration = sd; + media_time = mt; + } else { + return ERROR_IO; + } + + uint64_t halfscale = mHeaderTimescale / 2; + segment_duration = (segment_duration * 1000000 + halfscale)/ mHeaderTimescale; + media_time = (media_time * 1000000 + halfscale) / mHeaderTimescale; + + int64_t duration; + int32_t samplerate; + if (mLastTrack->meta->findInt64(kKeyDuration, &duration) && + mLastTrack->meta->findInt32(kKeySampleRate, &samplerate)) { + + int64_t delay = (media_time * samplerate + 500000) / 1000000; + mLastTrack->meta->setInt32(kKeyEncoderDelay, delay); + + int64_t paddingus = duration - (segment_duration + media_time); + int64_t paddingsamples = (paddingus * samplerate + 500000) / 1000000; + mLastTrack->meta->setInt32(kKeyEncoderPadding, paddingsamples); + } + } + *offset += chunk_size; + break; + } + case FOURCC('f', 'r', 'm', 'a'): { uint32_t original_fourcc; @@ -1564,24 +1628,26 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { case FOURCC('m', 'v', 'h', 'd'): { - if (chunk_data_size < 12) { + if (chunk_data_size < 24) { return ERROR_MALFORMED; } - uint8_t header[12]; + uint8_t header[24]; if (mDataSource->readAt( data_offset, header, sizeof(header)) < (ssize_t)sizeof(header)) { return ERROR_IO; } - int64_t creationTime; + uint64_t creationTime; if (header[0] == 1) { creationTime = U64_AT(&header[4]); + mHeaderTimescale = U32_AT(&header[20]); } else if (header[0] != 0) { return ERROR_MALFORMED; } else { creationTime = U32_AT(&header[4]); + mHeaderTimescale = U32_AT(&header[12]); } String8 s; diff --git a/media/libstagefright/SurfaceMediaSource.cpp b/media/libstagefright/SurfaceMediaSource.cpp index 409038a..71b6569 100644 --- a/media/libstagefright/SurfaceMediaSource.cpp +++ b/media/libstagefright/SurfaceMediaSource.cpp @@ -305,8 +305,9 @@ status_t SurfaceMediaSource::read( MediaBuffer **buffer, // First time seeing the buffer? Added it to the SMS slot if (item.mGraphicBuffer != NULL) { - mBufferSlot[item.mBuf] = item.mGraphicBuffer; + mSlots[item.mBuf].mGraphicBuffer = item.mGraphicBuffer; } + mSlots[item.mBuf].mFrameNumber = item.mFrameNumber; // check for the timing of this buffer if (mNumFramesReceived == 0 && !mUseAbsoluteTimestamps) { @@ -315,7 +316,8 @@ status_t SurfaceMediaSource::read( MediaBuffer **buffer, if (mStartTimeNs > 0) { if (item.mTimestamp < mStartTimeNs) { // This frame predates start of record, discard - mBufferQueue->releaseBuffer(item.mBuf, EGL_NO_DISPLAY, + mBufferQueue->releaseBuffer( + item.mBuf, item.mFrameNumber, EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE); continue; } @@ -345,17 +347,18 @@ status_t SurfaceMediaSource::read( MediaBuffer **buffer, // First time seeing the buffer? Added it to the SMS slot if (item.mGraphicBuffer != NULL) { - mBufferSlot[mCurrentSlot] = item.mGraphicBuffer; + mSlots[item.mBuf].mGraphicBuffer = item.mGraphicBuffer; } + mSlots[item.mBuf].mFrameNumber = item.mFrameNumber; - mCurrentBuffers.push_back(mBufferSlot[mCurrentSlot]); + mCurrentBuffers.push_back(mSlots[mCurrentSlot].mGraphicBuffer); int64_t prevTimeStamp = mCurrentTimestamp; mCurrentTimestamp = item.mTimestamp; mNumFramesEncoded++; // Pass the data to the MediaBuffer. Pass in only the metadata - passMetadataBuffer(buffer, mBufferSlot[mCurrentSlot]->handle); + passMetadataBuffer(buffer, mSlots[mCurrentSlot].mGraphicBuffer->handle); (*buffer)->setObserver(this); (*buffer)->add_ref(); @@ -405,15 +408,16 @@ void SurfaceMediaSource::signalBufferReturned(MediaBuffer *buffer) { } for (int id = 0; id < BufferQueue::NUM_BUFFER_SLOTS; id++) { - if (mBufferSlot[id] == NULL) { + if (mSlots[id].mGraphicBuffer == NULL) { continue; } - if (bufferHandle == mBufferSlot[id]->handle) { + if (bufferHandle == mSlots[id].mGraphicBuffer->handle) { ALOGV("Slot %d returned, matches handle = %p", id, - mBufferSlot[id]->handle); + mSlots[id].mGraphicBuffer->handle); - mBufferQueue->releaseBuffer(id, EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, + mBufferQueue->releaseBuffer(id, mSlots[id].mFrameNumber, + EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE); buffer->setObserver(0); @@ -469,7 +473,7 @@ void SurfaceMediaSource::onBuffersReleased() { mFrameAvailableCondition.signal(); for (int i = 0; i < BufferQueue::NUM_BUFFER_SLOTS; i++) { - mBufferSlot[i] = 0; + mSlots[i].mGraphicBuffer = 0; } } diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp index cf50dc9..1b20cbb 100644 --- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp +++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp @@ -604,6 +604,9 @@ void SoftAAC2::onReset() { // To make the codec behave the same before and after a reset, we need to invalidate the // streaminfo struct. This does that: mStreamInfo->sampleRate = 0; + + mSignalledError = false; + mOutputPortSettingsChange = NONE; } void SoftAAC2::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) { diff --git a/media/libstagefright/codecs/aacenc/SampleCode/Android.mk b/media/libstagefright/codecs/aacenc/SampleCode/Android.mk index 01016e7..d06dcf6 100644 --- a/media/libstagefright/codecs/aacenc/SampleCode/Android.mk +++ b/media/libstagefright/codecs/aacenc/SampleCode/Android.mk @@ -5,7 +5,7 @@ LOCAL_SRC_FILES := \ AAC_E_SAMPLES.c \ ../../common/cmnMemory.c -LOCAL_MODULE_TAGS := debug +LOCAL_MODULE_TAGS := optional LOCAL_MODULE := AACEncTest diff --git a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp index 4d4212f..3320688 100644 --- a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp +++ b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp @@ -457,6 +457,11 @@ void SoftAMR::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) { } } +void SoftAMR::onReset() { + mSignalledError = false; + mOutputPortSettingsChange = NONE; +} + } // namespace android android::SoftOMXComponent *createSoftOMXComponent( diff --git a/media/libstagefright/codecs/amrnb/dec/SoftAMR.h b/media/libstagefright/codecs/amrnb/dec/SoftAMR.h index 9a596e5..758d6ac 100644 --- a/media/libstagefright/codecs/amrnb/dec/SoftAMR.h +++ b/media/libstagefright/codecs/amrnb/dec/SoftAMR.h @@ -40,6 +40,7 @@ protected: virtual void onQueueFilled(OMX_U32 portIndex); virtual void onPortFlushCompleted(OMX_U32 portIndex); virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled); + virtual void onReset(); private: enum { diff --git a/media/libstagefright/codecs/amrwbenc/SampleCode/Android.mk b/media/libstagefright/codecs/amrwbenc/SampleCode/Android.mk index db34d08..c203f77 100644 --- a/media/libstagefright/codecs/amrwbenc/SampleCode/Android.mk +++ b/media/libstagefright/codecs/amrwbenc/SampleCode/Android.mk @@ -5,7 +5,7 @@ LOCAL_SRC_FILES := \ AMRWB_E_SAMPLE.c \ ../../common/cmnMemory.c -LOCAL_MODULE_TAGS := debug +LOCAL_MODULE_TAGS := optional LOCAL_MODULE := AMRWBEncTest LOCAL_ARM_MODE := arm diff --git a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp index 020cc0a..fb2a430 100644 --- a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp +++ b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp @@ -48,42 +48,32 @@ static const CodecProfileLevel kH263ProfileLevels[] = { { OMX_VIDEO_H263ProfileISWV2, OMX_VIDEO_H263Level45 }, }; -template<class T> -static void InitOMXParams(T *params) { - params->nSize = sizeof(T); - params->nVersion.s.nVersionMajor = 1; - params->nVersion.s.nVersionMinor = 0; - params->nVersion.s.nRevision = 0; - params->nVersion.s.nStep = 0; -} - SoftMPEG4::SoftMPEG4( const char *name, + const char *componentRole, + OMX_VIDEO_CODINGTYPE codingType, + const CodecProfileLevel *profileLevels, + size_t numProfileLevels, const OMX_CALLBACKTYPE *callbacks, OMX_PTR appData, OMX_COMPONENTTYPE **component) - : SimpleSoftOMXComponent(name, callbacks, appData, component), - mMode(MODE_MPEG4), + : SoftVideoDecoderOMXComponent( + name, componentRole, codingType, profileLevels, numProfileLevels, + 352 /* width */, 288 /* height */, callbacks, appData, component), + mMode(codingType == OMX_VIDEO_CodingH263 ? MODE_H263 : MODE_MPEG4), mHandle(new tagvideoDecControls), mInputBufferCount(0), - mWidth(352), - mHeight(288), - mCropLeft(0), - mCropTop(0), - mCropRight(mWidth - 1), - mCropBottom(mHeight - 1), mSignalledError(false), mInitialized(false), mFramesConfigured(false), mNumSamplesOutput(0), - mOutputPortSettingsChange(NONE) { - if (!strcmp(name, "OMX.google.h263.decoder")) { - mMode = MODE_H263; - } else { - CHECK(!strcmp(name, "OMX.google.mpeg4.decoder")); - } - - initPorts(); + mPvTime(0) { + initPorts( + kNumInputBuffers, + 8192 /* inputBufferSize */, + kNumOutputBuffers, + (mMode == MODE_MPEG4) + ? MEDIA_MIMETYPE_VIDEO_MPEG4 : MEDIA_MIMETYPE_VIDEO_H263); CHECK_EQ(initDecoder(), (status_t)OK); } @@ -96,219 +86,11 @@ SoftMPEG4::~SoftMPEG4() { mHandle = NULL; } -void SoftMPEG4::initPorts() { - OMX_PARAM_PORTDEFINITIONTYPE def; - InitOMXParams(&def); - - def.nPortIndex = 0; - def.eDir = OMX_DirInput; - def.nBufferCountMin = kNumInputBuffers; - def.nBufferCountActual = def.nBufferCountMin; - def.nBufferSize = 8192; - def.bEnabled = OMX_TRUE; - def.bPopulated = OMX_FALSE; - def.eDomain = OMX_PortDomainVideo; - def.bBuffersContiguous = OMX_FALSE; - def.nBufferAlignment = 1; - - def.format.video.cMIMEType = - (mMode == MODE_MPEG4) - ? const_cast<char *>(MEDIA_MIMETYPE_VIDEO_MPEG4) - : const_cast<char *>(MEDIA_MIMETYPE_VIDEO_H263); - - def.format.video.pNativeRender = NULL; - def.format.video.nFrameWidth = mWidth; - def.format.video.nFrameHeight = mHeight; - def.format.video.nStride = def.format.video.nFrameWidth; - def.format.video.nSliceHeight = def.format.video.nFrameHeight; - def.format.video.nBitrate = 0; - def.format.video.xFramerate = 0; - def.format.video.bFlagErrorConcealment = OMX_FALSE; - - def.format.video.eCompressionFormat = - mMode == MODE_MPEG4 ? OMX_VIDEO_CodingMPEG4 : OMX_VIDEO_CodingH263; - - def.format.video.eColorFormat = OMX_COLOR_FormatUnused; - def.format.video.pNativeWindow = NULL; - - addPort(def); - - def.nPortIndex = 1; - def.eDir = OMX_DirOutput; - def.nBufferCountMin = kNumOutputBuffers; - def.nBufferCountActual = def.nBufferCountMin; - def.bEnabled = OMX_TRUE; - def.bPopulated = OMX_FALSE; - def.eDomain = OMX_PortDomainVideo; - def.bBuffersContiguous = OMX_FALSE; - def.nBufferAlignment = 2; - - def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_RAW); - def.format.video.pNativeRender = NULL; - def.format.video.nFrameWidth = mWidth; - def.format.video.nFrameHeight = mHeight; - def.format.video.nStride = def.format.video.nFrameWidth; - def.format.video.nSliceHeight = def.format.video.nFrameHeight; - def.format.video.nBitrate = 0; - def.format.video.xFramerate = 0; - def.format.video.bFlagErrorConcealment = OMX_FALSE; - def.format.video.eCompressionFormat = OMX_VIDEO_CodingUnused; - def.format.video.eColorFormat = OMX_COLOR_FormatYUV420Planar; - def.format.video.pNativeWindow = NULL; - - def.nBufferSize = - (def.format.video.nFrameWidth * def.format.video.nFrameHeight * 3) / 2; - - addPort(def); -} - status_t SoftMPEG4::initDecoder() { memset(mHandle, 0, sizeof(tagvideoDecControls)); return OK; } -OMX_ERRORTYPE SoftMPEG4::internalGetParameter( - OMX_INDEXTYPE index, OMX_PTR params) { - switch (index) { - case OMX_IndexParamVideoPortFormat: - { - OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams = - (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params; - - if (formatParams->nPortIndex > 1) { - return OMX_ErrorUndefined; - } - - if (formatParams->nIndex != 0) { - return OMX_ErrorNoMore; - } - - if (formatParams->nPortIndex == 0) { - formatParams->eCompressionFormat = - (mMode == MODE_MPEG4) - ? OMX_VIDEO_CodingMPEG4 : OMX_VIDEO_CodingH263; - - formatParams->eColorFormat = OMX_COLOR_FormatUnused; - formatParams->xFramerate = 0; - } else { - CHECK_EQ(formatParams->nPortIndex, 1u); - - formatParams->eCompressionFormat = OMX_VIDEO_CodingUnused; - formatParams->eColorFormat = OMX_COLOR_FormatYUV420Planar; - formatParams->xFramerate = 0; - } - - return OMX_ErrorNone; - } - - case OMX_IndexParamVideoProfileLevelQuerySupported: - { - OMX_VIDEO_PARAM_PROFILELEVELTYPE *profileLevel = - (OMX_VIDEO_PARAM_PROFILELEVELTYPE *) params; - - if (profileLevel->nPortIndex != 0) { // Input port only - ALOGE("Invalid port index: %ld", profileLevel->nPortIndex); - return OMX_ErrorUnsupportedIndex; - } - - size_t index = profileLevel->nProfileIndex; - if (mMode == MODE_H263) { - size_t nProfileLevels = - sizeof(kH263ProfileLevels) / sizeof(kH263ProfileLevels[0]); - if (index >= nProfileLevels) { - return OMX_ErrorNoMore; - } - - profileLevel->eProfile = kH263ProfileLevels[index].mProfile; - profileLevel->eLevel = kH263ProfileLevels[index].mLevel; - } else { - size_t nProfileLevels = - sizeof(kM4VProfileLevels) / sizeof(kM4VProfileLevels[0]); - if (index >= nProfileLevels) { - return OMX_ErrorNoMore; - } - - profileLevel->eProfile = kM4VProfileLevels[index].mProfile; - profileLevel->eLevel = kM4VProfileLevels[index].mLevel; - } - return OMX_ErrorNone; - } - - default: - return SimpleSoftOMXComponent::internalGetParameter(index, params); - } -} - -OMX_ERRORTYPE SoftMPEG4::internalSetParameter( - OMX_INDEXTYPE index, const OMX_PTR params) { - switch (index) { - case OMX_IndexParamStandardComponentRole: - { - const OMX_PARAM_COMPONENTROLETYPE *roleParams = - (const OMX_PARAM_COMPONENTROLETYPE *)params; - - if (mMode == MODE_MPEG4) { - if (strncmp((const char *)roleParams->cRole, - "video_decoder.mpeg4", - OMX_MAX_STRINGNAME_SIZE - 1)) { - return OMX_ErrorUndefined; - } - } else { - if (strncmp((const char *)roleParams->cRole, - "video_decoder.h263", - OMX_MAX_STRINGNAME_SIZE - 1)) { - return OMX_ErrorUndefined; - } - } - - return OMX_ErrorNone; - } - - case OMX_IndexParamVideoPortFormat: - { - OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams = - (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params; - - if (formatParams->nPortIndex > 1) { - return OMX_ErrorUndefined; - } - - if (formatParams->nIndex != 0) { - return OMX_ErrorNoMore; - } - - return OMX_ErrorNone; - } - - default: - return SimpleSoftOMXComponent::internalSetParameter(index, params); - } -} - -OMX_ERRORTYPE SoftMPEG4::getConfig( - OMX_INDEXTYPE index, OMX_PTR params) { - switch (index) { - case OMX_IndexConfigCommonOutputCrop: - { - OMX_CONFIG_RECTTYPE *rectParams = (OMX_CONFIG_RECTTYPE *)params; - - if (rectParams->nPortIndex != 1) { - return OMX_ErrorUndefined; - } - - rectParams->nLeft = mCropLeft; - rectParams->nTop = mCropTop; - rectParams->nWidth = mCropRight - mCropLeft + 1; - rectParams->nHeight = mCropBottom - mCropTop + 1; - - return OMX_ErrorNone; - } - - default: - return OMX_ErrorUnsupportedIndex; - } -} - void SoftMPEG4::onQueueFilled(OMX_U32 portIndex) { if (mSignalledError || mOutputPortSettingsChange != NONE) { return; @@ -415,9 +197,14 @@ void SoftMPEG4::onQueueFilled(OMX_U32 portIndex) { uint32_t useExtTimestamp = (inHeader->nOffset == 0); - // decoder deals in ms, OMX in us. - uint32_t timestamp = - useExtTimestamp ? (inHeader->nTimeStamp + 500) / 1000 : 0xFFFFFFFF; + // decoder deals in ms (int32_t), OMX in us (int64_t) + // so use fake timestamp instead + uint32_t timestamp = 0xFFFFFFFF; + if (useExtTimestamp) { + mPvToOmxTimeMap.add(mPvTime, inHeader->nTimeStamp); + timestamp = mPvTime; + mPvTime++; + } int32_t bufferSize = inHeader->nFilledLen; int32_t tmp = bufferSize; @@ -441,7 +228,8 @@ void SoftMPEG4::onQueueFilled(OMX_U32 portIndex) { } // decoder deals in ms, OMX in us. - outHeader->nTimeStamp = timestamp * 1000; + outHeader->nTimeStamp = mPvToOmxTimeMap.valueFor(timestamp); + mPvToOmxTimeMap.removeItem(timestamp); inHeader->nOffset += bufferSize; inHeader->nFilledLen = 0; @@ -482,11 +270,11 @@ void SoftMPEG4::onQueueFilled(OMX_U32 portIndex) { } bool SoftMPEG4::portSettingsChanged() { - int32_t disp_width, disp_height; - PVGetVideoDimensions(mHandle, &disp_width, &disp_height); + uint32_t disp_width, disp_height; + PVGetVideoDimensions(mHandle, (int32 *)&disp_width, (int32 *)&disp_height); - int32_t buf_width, buf_height; - PVGetBufferDimensions(mHandle, &buf_width, &buf_height); + uint32_t buf_width, buf_height; + PVGetBufferDimensions(mHandle, (int32 *)&buf_width, (int32 *)&buf_height); CHECK_LE(disp_width, buf_width); CHECK_LE(disp_height, buf_height); @@ -494,12 +282,12 @@ bool SoftMPEG4::portSettingsChanged() { ALOGV("disp_width = %d, disp_height = %d, buf_width = %d, buf_height = %d", disp_width, disp_height, buf_width, buf_height); - if (mCropRight != disp_width - 1 - || mCropBottom != disp_height - 1) { + if (mCropWidth != disp_width + || mCropHeight != disp_height) { mCropLeft = 0; mCropTop = 0; - mCropRight = disp_width - 1; - mCropBottom = disp_height - 1; + mCropWidth = disp_width; + mCropHeight = disp_height; notify(OMX_EventPortSettingsChanged, 1, @@ -545,45 +333,22 @@ void SoftMPEG4::onPortFlushCompleted(OMX_U32 portIndex) { } } -void SoftMPEG4::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) { - if (portIndex != 1) { - return; - } - - switch (mOutputPortSettingsChange) { - case NONE: - break; - - case AWAITING_DISABLED: - { - CHECK(!enabled); - mOutputPortSettingsChange = AWAITING_ENABLED; - break; - } - - default: - { - CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED); - CHECK(enabled); - mOutputPortSettingsChange = NONE; - break; - } +void SoftMPEG4::onReset() { + SoftVideoDecoderOMXComponent::onReset(); + mPvToOmxTimeMap.clear(); + mSignalledError = false; + mFramesConfigured = false; + if (mInitialized) { + PVCleanUpVideoDecoder(mHandle); + mInitialized = false; } } void SoftMPEG4::updatePortDefinitions() { - OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(0)->mDef; - def->format.video.nFrameWidth = mWidth; - def->format.video.nFrameHeight = mHeight; - def->format.video.nStride = def->format.video.nFrameWidth; - def->format.video.nSliceHeight = def->format.video.nFrameHeight; - - def = &editPortInfo(1)->mDef; - def->format.video.nFrameWidth = mWidth; - def->format.video.nFrameHeight = mHeight; - def->format.video.nStride = def->format.video.nFrameWidth; - def->format.video.nSliceHeight = def->format.video.nFrameHeight; + SoftVideoDecoderOMXComponent::updatePortDefinitions(); + /* We have to align our width and height - this should affect stride! */ + OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(kOutputPortIndex)->mDef; def->nBufferSize = (((def->format.video.nFrameWidth + 15) & -16) * ((def->format.video.nFrameHeight + 15) & -16) * 3) / 2; @@ -594,6 +359,19 @@ void SoftMPEG4::updatePortDefinitions() { android::SoftOMXComponent *createSoftOMXComponent( const char *name, const OMX_CALLBACKTYPE *callbacks, OMX_PTR appData, OMX_COMPONENTTYPE **component) { - return new android::SoftMPEG4(name, callbacks, appData, component); + using namespace android; + if (!strcmp(name, "OMX.google.h263.decoder")) { + return new android::SoftMPEG4( + name, "video_decoder.h263", OMX_VIDEO_CodingH263, + kH263ProfileLevels, ARRAY_SIZE(kH263ProfileLevels), + callbacks, appData, component); + } else if (!strcmp(name, "OMX.google.mpeg4.decoder")) { + return new android::SoftMPEG4( + name, "video_decoder.mpeg4", OMX_VIDEO_CodingMPEG4, + kM4VProfileLevels, ARRAY_SIZE(kM4VProfileLevels), + callbacks, appData, component); + } else { + CHECK(!"Unknown component"); + } } diff --git a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.h b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.h index dff08a7..de14aaf 100644 --- a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.h +++ b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.h @@ -18,14 +18,18 @@ #define SOFT_MPEG4_H_ -#include "SimpleSoftOMXComponent.h" +#include "SoftVideoDecoderOMXComponent.h" struct tagvideoDecControls; namespace android { -struct SoftMPEG4 : public SimpleSoftOMXComponent { +struct SoftMPEG4 : public SoftVideoDecoderOMXComponent { SoftMPEG4(const char *name, + const char *componentRole, + OMX_VIDEO_CODINGTYPE codingType, + const CodecProfileLevel *profileLevels, + size_t numProfileLevels, const OMX_CALLBACKTYPE *callbacks, OMX_PTR appData, OMX_COMPONENTTYPE **component); @@ -33,17 +37,9 @@ struct SoftMPEG4 : public SimpleSoftOMXComponent { protected: virtual ~SoftMPEG4(); - virtual OMX_ERRORTYPE internalGetParameter( - OMX_INDEXTYPE index, OMX_PTR params); - - virtual OMX_ERRORTYPE internalSetParameter( - OMX_INDEXTYPE index, const OMX_PTR params); - - virtual OMX_ERRORTYPE getConfig(OMX_INDEXTYPE index, OMX_PTR params); - virtual void onQueueFilled(OMX_U32 portIndex); virtual void onPortFlushCompleted(OMX_U32 portIndex); - virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled); + virtual void onReset(); private: enum { @@ -54,32 +50,23 @@ private: enum { MODE_MPEG4, MODE_H263, - } mMode; tagvideoDecControls *mHandle; size_t mInputBufferCount; - int32_t mWidth, mHeight; - int32_t mCropLeft, mCropTop, mCropRight, mCropBottom; - bool mSignalledError; bool mInitialized; bool mFramesConfigured; int32_t mNumSamplesOutput; + int32_t mPvTime; + KeyedVector<int32_t, OMX_TICKS> mPvToOmxTimeMap; - enum { - NONE, - AWAITING_DISABLED, - AWAITING_ENABLED - } mOutputPortSettingsChange; - - void initPorts(); status_t initDecoder(); - void updatePortDefinitions(); + virtual void updatePortDefinitions(); bool portSettingsChanged(); DISALLOW_EVIL_CONSTRUCTORS(SoftMPEG4); diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp index 9f25536..7c382fb 100644 --- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp +++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp @@ -361,6 +361,8 @@ void SoftMP3::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) { void SoftMP3::onReset() { pvmp3_InitDecoder(mConfig, mDecoderBuf); mIsFirst = true; + mSignalledError = false; + mOutputPortSettingsChange = NONE; } } // namespace android diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp index a400b4c..43d0263 100644 --- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp +++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp @@ -29,26 +29,19 @@ namespace android { -template<class T> -static void InitOMXParams(T *params) { - params->nSize = sizeof(T); - params->nVersion.s.nVersionMajor = 1; - params->nVersion.s.nVersionMinor = 0; - params->nVersion.s.nRevision = 0; - params->nVersion.s.nStep = 0; -} - SoftVPX::SoftVPX( const char *name, const OMX_CALLBACKTYPE *callbacks, OMX_PTR appData, OMX_COMPONENTTYPE **component) - : SimpleSoftOMXComponent(name, callbacks, appData, component), - mCtx(NULL), - mWidth(320), - mHeight(240), - mOutputPortSettingsChange(NONE) { - initPorts(); + : SoftVideoDecoderOMXComponent( + name, "video_decoder.vpx", OMX_VIDEO_CodingVPX, + NULL /* profileLevels */, 0 /* numProfileLevels */, + 320 /* width */, 240 /* height */, callbacks, appData, component), + mCtx(NULL) { + initPorts(kNumBuffers, 768 * 1024 /* inputBufferSize */, + kNumBuffers, MEDIA_MIMETYPE_VIDEO_VPX); + CHECK_EQ(initDecoder(), (status_t)OK); } @@ -58,65 +51,6 @@ SoftVPX::~SoftVPX() { mCtx = NULL; } -void SoftVPX::initPorts() { - OMX_PARAM_PORTDEFINITIONTYPE def; - InitOMXParams(&def); - - def.nPortIndex = 0; - def.eDir = OMX_DirInput; - def.nBufferCountMin = kNumBuffers; - def.nBufferCountActual = def.nBufferCountMin; - def.nBufferSize = 768 * 1024; - def.bEnabled = OMX_TRUE; - def.bPopulated = OMX_FALSE; - def.eDomain = OMX_PortDomainVideo; - def.bBuffersContiguous = OMX_FALSE; - def.nBufferAlignment = 1; - - def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_VPX); - def.format.video.pNativeRender = NULL; - def.format.video.nFrameWidth = mWidth; - def.format.video.nFrameHeight = mHeight; - def.format.video.nStride = def.format.video.nFrameWidth; - def.format.video.nSliceHeight = def.format.video.nFrameHeight; - def.format.video.nBitrate = 0; - def.format.video.xFramerate = 0; - def.format.video.bFlagErrorConcealment = OMX_FALSE; - def.format.video.eCompressionFormat = OMX_VIDEO_CodingVPX; - def.format.video.eColorFormat = OMX_COLOR_FormatUnused; - def.format.video.pNativeWindow = NULL; - - addPort(def); - - def.nPortIndex = 1; - def.eDir = OMX_DirOutput; - def.nBufferCountMin = kNumBuffers; - def.nBufferCountActual = def.nBufferCountMin; - def.bEnabled = OMX_TRUE; - def.bPopulated = OMX_FALSE; - def.eDomain = OMX_PortDomainVideo; - def.bBuffersContiguous = OMX_FALSE; - def.nBufferAlignment = 2; - - def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_RAW); - def.format.video.pNativeRender = NULL; - def.format.video.nFrameWidth = mWidth; - def.format.video.nFrameHeight = mHeight; - def.format.video.nStride = def.format.video.nFrameWidth; - def.format.video.nSliceHeight = def.format.video.nFrameHeight; - def.format.video.nBitrate = 0; - def.format.video.xFramerate = 0; - def.format.video.bFlagErrorConcealment = OMX_FALSE; - def.format.video.eCompressionFormat = OMX_VIDEO_CodingUnused; - def.format.video.eColorFormat = OMX_COLOR_FormatYUV420Planar; - def.format.video.pNativeWindow = NULL; - - def.nBufferSize = - (def.format.video.nFrameWidth * def.format.video.nFrameHeight * 3) / 2; - - addPort(def); -} - static int GetCPUCoreCount() { int cpuCoreCount = 1; #if defined(_SC_NPROCESSORS_ONLN) @@ -145,80 +79,6 @@ status_t SoftVPX::initDecoder() { return OK; } -OMX_ERRORTYPE SoftVPX::internalGetParameter( - OMX_INDEXTYPE index, OMX_PTR params) { - switch (index) { - case OMX_IndexParamVideoPortFormat: - { - OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams = - (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params; - - if (formatParams->nPortIndex > 1) { - return OMX_ErrorUndefined; - } - - if (formatParams->nIndex != 0) { - return OMX_ErrorNoMore; - } - - if (formatParams->nPortIndex == 0) { - formatParams->eCompressionFormat = OMX_VIDEO_CodingVPX; - formatParams->eColorFormat = OMX_COLOR_FormatUnused; - formatParams->xFramerate = 0; - } else { - CHECK_EQ(formatParams->nPortIndex, 1u); - - formatParams->eCompressionFormat = OMX_VIDEO_CodingUnused; - formatParams->eColorFormat = OMX_COLOR_FormatYUV420Planar; - formatParams->xFramerate = 0; - } - - return OMX_ErrorNone; - } - - default: - return SimpleSoftOMXComponent::internalGetParameter(index, params); - } -} - -OMX_ERRORTYPE SoftVPX::internalSetParameter( - OMX_INDEXTYPE index, const OMX_PTR params) { - switch (index) { - case OMX_IndexParamStandardComponentRole: - { - const OMX_PARAM_COMPONENTROLETYPE *roleParams = - (const OMX_PARAM_COMPONENTROLETYPE *)params; - - if (strncmp((const char *)roleParams->cRole, - "video_decoder.vpx", - OMX_MAX_STRINGNAME_SIZE - 1)) { - return OMX_ErrorUndefined; - } - - return OMX_ErrorNone; - } - - case OMX_IndexParamVideoPortFormat: - { - OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams = - (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params; - - if (formatParams->nPortIndex > 1) { - return OMX_ErrorUndefined; - } - - if (formatParams->nIndex != 0) { - return OMX_ErrorNoMore; - } - - return OMX_ErrorNone; - } - - default: - return SimpleSoftOMXComponent::internalSetParameter(index, params); - } -} - void SoftVPX::onQueueFilled(OMX_U32 portIndex) { if (mOutputPortSettingsChange != NONE) { return; @@ -226,6 +86,7 @@ void SoftVPX::onQueueFilled(OMX_U32 portIndex) { List<BufferInfo *> &inQueue = getPortQueue(0); List<BufferInfo *> &outQueue = getPortQueue(1); + bool EOSseen = false; while (!inQueue.empty() && !outQueue.empty()) { BufferInfo *inInfo = *inQueue.begin(); @@ -235,17 +96,20 @@ void SoftVPX::onQueueFilled(OMX_U32 portIndex) { OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { - inQueue.erase(inQueue.begin()); - inInfo->mOwnedByUs = false; - notifyEmptyBufferDone(inHeader); - - outHeader->nFilledLen = 0; - outHeader->nFlags = OMX_BUFFERFLAG_EOS; - - outQueue.erase(outQueue.begin()); - outInfo->mOwnedByUs = false; - notifyFillBufferDone(outHeader); - return; + EOSseen = true; + if (inHeader->nFilledLen == 0) { + inQueue.erase(inQueue.begin()); + inInfo->mOwnedByUs = false; + notifyEmptyBufferDone(inHeader); + + outHeader->nFilledLen = 0; + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + + outQueue.erase(outQueue.begin()); + outInfo->mOwnedByUs = false; + notifyFillBufferDone(outHeader); + return; + } } if (vpx_codec_decode( @@ -266,8 +130,8 @@ void SoftVPX::onQueueFilled(OMX_U32 portIndex) { if (img != NULL) { CHECK_EQ(img->fmt, IMG_FMT_I420); - int32_t width = img->d_w; - int32_t height = img->d_h; + uint32_t width = img->d_w; + uint32_t height = img->d_h; if (width != mWidth || height != mHeight) { mWidth = width; @@ -282,7 +146,7 @@ void SoftVPX::onQueueFilled(OMX_U32 portIndex) { outHeader->nOffset = 0; outHeader->nFilledLen = (width * height * 3) / 2; - outHeader->nFlags = 0; + outHeader->nFlags = EOSseen ? OMX_BUFFERFLAG_EOS : 0; outHeader->nTimeStamp = inHeader->nTimeStamp; const uint8_t *srcLine = (const uint8_t *)img->planes[PLANE_Y]; @@ -325,53 +189,6 @@ void SoftVPX::onQueueFilled(OMX_U32 portIndex) { } } -void SoftVPX::onPortFlushCompleted(OMX_U32 portIndex) { -} - -void SoftVPX::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) { - if (portIndex != 1) { - return; - } - - switch (mOutputPortSettingsChange) { - case NONE: - break; - - case AWAITING_DISABLED: - { - CHECK(!enabled); - mOutputPortSettingsChange = AWAITING_ENABLED; - break; - } - - default: - { - CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED); - CHECK(enabled); - mOutputPortSettingsChange = NONE; - break; - } - } -} - -void SoftVPX::updatePortDefinitions() { - OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(0)->mDef; - def->format.video.nFrameWidth = mWidth; - def->format.video.nFrameHeight = mHeight; - def->format.video.nStride = def->format.video.nFrameWidth; - def->format.video.nSliceHeight = def->format.video.nFrameHeight; - - def = &editPortInfo(1)->mDef; - def->format.video.nFrameWidth = mWidth; - def->format.video.nFrameHeight = mHeight; - def->format.video.nStride = def->format.video.nFrameWidth; - def->format.video.nSliceHeight = def->format.video.nFrameHeight; - - def->nBufferSize = - (def->format.video.nFrameWidth - * def->format.video.nFrameHeight * 3) / 2; -} - } // namespace android android::SoftOMXComponent *createSoftOMXComponent( diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.h b/media/libstagefright/codecs/on2/dec/SoftVPX.h index 3e814a2..626307b 100644 --- a/media/libstagefright/codecs/on2/dec/SoftVPX.h +++ b/media/libstagefright/codecs/on2/dec/SoftVPX.h @@ -18,11 +18,11 @@ #define SOFT_VPX_H_ -#include "SimpleSoftOMXComponent.h" +#include "SoftVideoDecoderOMXComponent.h" namespace android { -struct SoftVPX : public SimpleSoftOMXComponent { +struct SoftVPX : public SoftVideoDecoderOMXComponent { SoftVPX(const char *name, const OMX_CALLBACKTYPE *callbacks, OMX_PTR appData, @@ -31,15 +31,7 @@ struct SoftVPX : public SimpleSoftOMXComponent { protected: virtual ~SoftVPX(); - virtual OMX_ERRORTYPE internalGetParameter( - OMX_INDEXTYPE index, OMX_PTR params); - - virtual OMX_ERRORTYPE internalSetParameter( - OMX_INDEXTYPE index, const OMX_PTR params); - virtual void onQueueFilled(OMX_U32 portIndex); - virtual void onPortFlushCompleted(OMX_U32 portIndex); - virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled); private: enum { @@ -48,20 +40,8 @@ private: void *mCtx; - int32_t mWidth; - int32_t mHeight; - - enum { - NONE, - AWAITING_DISABLED, - AWAITING_ENABLED - } mOutputPortSettingsChange; - - void initPorts(); status_t initDecoder(); - void updatePortDefinitions(); - DISALLOW_EVIL_CONSTRUCTORS(SoftVPX); }; diff --git a/media/libstagefright/codecs/on2/h264dec/Android.mk b/media/libstagefright/codecs/on2/h264dec/Android.mk index 2539f98..655b2ab 100644 --- a/media/libstagefright/codecs/on2/h264dec/Android.mk +++ b/media/libstagefright/codecs/on2/h264dec/Android.mk @@ -119,7 +119,7 @@ LOCAL_C_INCLUDES := $(LOCAL_PATH)/inc LOCAL_SHARED_LIBRARIES := libstagefright_soft_h264dec -LOCAL_MODULE_TAGS := debug +LOCAL_MODULE_TAGS := optional LOCAL_MODULE := decoder diff --git a/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp b/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp index 6e36651..7ddb13c 100644 --- a/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp +++ b/media/libstagefright/codecs/on2/h264dec/SoftAVC.cpp @@ -47,38 +47,28 @@ static const CodecProfileLevel kProfileLevels[] = { { OMX_VIDEO_AVCProfileBaseline, OMX_VIDEO_AVCLevel51 }, }; -template<class T> -static void InitOMXParams(T *params) { - params->nSize = sizeof(T); - params->nVersion.s.nVersionMajor = 1; - params->nVersion.s.nVersionMinor = 0; - params->nVersion.s.nRevision = 0; - params->nVersion.s.nStep = 0; -} - SoftAVC::SoftAVC( const char *name, const OMX_CALLBACKTYPE *callbacks, OMX_PTR appData, OMX_COMPONENTTYPE **component) - : SimpleSoftOMXComponent(name, callbacks, appData, component), + : SoftVideoDecoderOMXComponent( + name, "video_decoder.avc", OMX_VIDEO_CodingAVC, + kProfileLevels, ARRAY_SIZE(kProfileLevels), + 320 /* width */, 240 /* height */, callbacks, appData, component), mHandle(NULL), mInputBufferCount(0), - mWidth(320), - mHeight(240), mPictureSize(mWidth * mHeight * 3 / 2), - mCropLeft(0), - mCropTop(0), - mCropWidth(mWidth), - mCropHeight(mHeight), mFirstPicture(NULL), mFirstPictureId(-1), mPicId(0), mHeadersDecoded(false), mEOSStatus(INPUT_DATA_AVAILABLE), - mOutputPortSettingsChange(NONE), mSignalledError(false) { - initPorts(); + initPorts( + kNumInputBuffers, 8192 /* inputBufferSize */, + kNumOutputBuffers, MEDIA_MIMETYPE_VIDEO_AVC); + CHECK_EQ(initDecoder(), (status_t)OK); } @@ -100,65 +90,6 @@ SoftAVC::~SoftAVC() { delete[] mFirstPicture; } -void SoftAVC::initPorts() { - OMX_PARAM_PORTDEFINITIONTYPE def; - InitOMXParams(&def); - - def.nPortIndex = kInputPortIndex; - def.eDir = OMX_DirInput; - def.nBufferCountMin = kNumInputBuffers; - def.nBufferCountActual = def.nBufferCountMin; - def.nBufferSize = 8192; - def.bEnabled = OMX_TRUE; - def.bPopulated = OMX_FALSE; - def.eDomain = OMX_PortDomainVideo; - def.bBuffersContiguous = OMX_FALSE; - def.nBufferAlignment = 1; - - def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_AVC); - def.format.video.pNativeRender = NULL; - def.format.video.nFrameWidth = mWidth; - def.format.video.nFrameHeight = mHeight; - def.format.video.nStride = def.format.video.nFrameWidth; - def.format.video.nSliceHeight = def.format.video.nFrameHeight; - def.format.video.nBitrate = 0; - def.format.video.xFramerate = 0; - def.format.video.bFlagErrorConcealment = OMX_FALSE; - def.format.video.eCompressionFormat = OMX_VIDEO_CodingAVC; - def.format.video.eColorFormat = OMX_COLOR_FormatUnused; - def.format.video.pNativeWindow = NULL; - - addPort(def); - - def.nPortIndex = kOutputPortIndex; - def.eDir = OMX_DirOutput; - def.nBufferCountMin = kNumOutputBuffers; - def.nBufferCountActual = def.nBufferCountMin; - def.bEnabled = OMX_TRUE; - def.bPopulated = OMX_FALSE; - def.eDomain = OMX_PortDomainVideo; - def.bBuffersContiguous = OMX_FALSE; - def.nBufferAlignment = 2; - - def.format.video.cMIMEType = const_cast<char *>(MEDIA_MIMETYPE_VIDEO_RAW); - def.format.video.pNativeRender = NULL; - def.format.video.nFrameWidth = mWidth; - def.format.video.nFrameHeight = mHeight; - def.format.video.nStride = def.format.video.nFrameWidth; - def.format.video.nSliceHeight = def.format.video.nFrameHeight; - def.format.video.nBitrate = 0; - def.format.video.xFramerate = 0; - def.format.video.bFlagErrorConcealment = OMX_FALSE; - def.format.video.eCompressionFormat = OMX_VIDEO_CodingUnused; - def.format.video.eColorFormat = OMX_COLOR_FormatYUV420Planar; - def.format.video.pNativeWindow = NULL; - - def.nBufferSize = - (def.format.video.nFrameWidth * def.format.video.nFrameHeight * 3) / 2; - - addPort(def); -} - status_t SoftAVC::initDecoder() { // Force decoder to output buffers in display order. if (H264SwDecInit(&mHandle, 0) == H264SWDEC_OK) { @@ -167,126 +98,6 @@ status_t SoftAVC::initDecoder() { return UNKNOWN_ERROR; } -OMX_ERRORTYPE SoftAVC::internalGetParameter( - OMX_INDEXTYPE index, OMX_PTR params) { - switch (index) { - case OMX_IndexParamVideoPortFormat: - { - OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams = - (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params; - - if (formatParams->nPortIndex > kOutputPortIndex) { - return OMX_ErrorUndefined; - } - - if (formatParams->nIndex != 0) { - return OMX_ErrorNoMore; - } - - if (formatParams->nPortIndex == kInputPortIndex) { - formatParams->eCompressionFormat = OMX_VIDEO_CodingAVC; - formatParams->eColorFormat = OMX_COLOR_FormatUnused; - formatParams->xFramerate = 0; - } else { - CHECK(formatParams->nPortIndex == kOutputPortIndex); - - formatParams->eCompressionFormat = OMX_VIDEO_CodingUnused; - formatParams->eColorFormat = OMX_COLOR_FormatYUV420Planar; - formatParams->xFramerate = 0; - } - - return OMX_ErrorNone; - } - - case OMX_IndexParamVideoProfileLevelQuerySupported: - { - OMX_VIDEO_PARAM_PROFILELEVELTYPE *profileLevel = - (OMX_VIDEO_PARAM_PROFILELEVELTYPE *) params; - - if (profileLevel->nPortIndex != kInputPortIndex) { - ALOGE("Invalid port index: %ld", profileLevel->nPortIndex); - return OMX_ErrorUnsupportedIndex; - } - - size_t index = profileLevel->nProfileIndex; - size_t nProfileLevels = - sizeof(kProfileLevels) / sizeof(kProfileLevels[0]); - if (index >= nProfileLevels) { - return OMX_ErrorNoMore; - } - - profileLevel->eProfile = kProfileLevels[index].mProfile; - profileLevel->eLevel = kProfileLevels[index].mLevel; - return OMX_ErrorNone; - } - - default: - return SimpleSoftOMXComponent::internalGetParameter(index, params); - } -} - -OMX_ERRORTYPE SoftAVC::internalSetParameter( - OMX_INDEXTYPE index, const OMX_PTR params) { - switch (index) { - case OMX_IndexParamStandardComponentRole: - { - const OMX_PARAM_COMPONENTROLETYPE *roleParams = - (const OMX_PARAM_COMPONENTROLETYPE *)params; - - if (strncmp((const char *)roleParams->cRole, - "video_decoder.avc", - OMX_MAX_STRINGNAME_SIZE - 1)) { - return OMX_ErrorUndefined; - } - - return OMX_ErrorNone; - } - - case OMX_IndexParamVideoPortFormat: - { - OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams = - (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params; - - if (formatParams->nPortIndex > kOutputPortIndex) { - return OMX_ErrorUndefined; - } - - if (formatParams->nIndex != 0) { - return OMX_ErrorNoMore; - } - - return OMX_ErrorNone; - } - - default: - return SimpleSoftOMXComponent::internalSetParameter(index, params); - } -} - -OMX_ERRORTYPE SoftAVC::getConfig( - OMX_INDEXTYPE index, OMX_PTR params) { - switch (index) { - case OMX_IndexConfigCommonOutputCrop: - { - OMX_CONFIG_RECTTYPE *rectParams = (OMX_CONFIG_RECTTYPE *)params; - - if (rectParams->nPortIndex != 1) { - return OMX_ErrorUndefined; - } - - rectParams->nLeft = mCropLeft; - rectParams->nTop = mCropTop; - rectParams->nWidth = mCropWidth; - rectParams->nHeight = mCropHeight; - - return OMX_ErrorNone; - } - - default: - return OMX_ErrorUnsupportedIndex; - } -} - void SoftAVC::onQueueFilled(OMX_U32 portIndex) { if (mSignalledError || mOutputPortSettingsChange != NONE) { return; @@ -298,13 +109,21 @@ void SoftAVC::onQueueFilled(OMX_U32 portIndex) { List<BufferInfo *> &inQueue = getPortQueue(kInputPortIndex); List<BufferInfo *> &outQueue = getPortQueue(kOutputPortIndex); + + if (mHeadersDecoded) { + // Dequeue any already decoded output frames to free up space + // in the output queue. + + drainAllOutputBuffers(false /* eos */); + } + H264SwDecRet ret = H264SWDEC_PIC_RDY; bool portSettingsChanged = false; while ((mEOSStatus != INPUT_DATA_AVAILABLE || !inQueue.empty()) && outQueue.size() == kNumOutputBuffers) { if (mEOSStatus == INPUT_EOS_SEEN) { - drainAllOutputBuffers(); + drainAllOutputBuffers(true /* eos */); return; } @@ -392,15 +211,7 @@ void SoftAVC::onQueueFilled(OMX_U32 portIndex) { mFirstPictureId = -1; } - while (!outQueue.empty() && - mHeadersDecoded && - H264SwDecNextPicture(mHandle, &decodedPicture, 0) - == H264SWDEC_PIC_RDY) { - - int32_t picId = decodedPicture.picId; - uint8_t *data = (uint8_t *) decodedPicture.pOutputPicture; - drainOneOutputBuffer(picId, data); - } + drainAllOutputBuffers(false /* eos */); } } @@ -409,8 +220,6 @@ bool SoftAVC::handlePortSettingChangeEvent(const H264SwDecInfo *info) { mWidth = info->picWidth; mHeight = info->picHeight; mPictureSize = mWidth * mHeight * 3 / 2; - mCropWidth = mWidth; - mCropHeight = mHeight; updatePortDefinitions(); notify(OMX_EventPortSettingsChanged, 1, 0, NULL); mOutputPortSettingsChange = AWAITING_DISABLED; @@ -463,43 +272,38 @@ void SoftAVC::drainOneOutputBuffer(int32_t picId, uint8_t* data) { notifyFillBufferDone(outHeader); } -bool SoftAVC::drainAllOutputBuffers() { +void SoftAVC::drainAllOutputBuffers(bool eos) { List<BufferInfo *> &outQueue = getPortQueue(kOutputPortIndex); H264SwDecPicture decodedPicture; + if (mHeadersDecoded) { + while (!outQueue.empty() + && H264SWDEC_PIC_RDY == H264SwDecNextPicture( + mHandle, &decodedPicture, eos /* flush */)) { + int32_t picId = decodedPicture.picId; + uint8_t *data = (uint8_t *) decodedPicture.pOutputPicture; + drainOneOutputBuffer(picId, data); + } + } + + if (!eos) { + return; + } + while (!outQueue.empty()) { BufferInfo *outInfo = *outQueue.begin(); outQueue.erase(outQueue.begin()); OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; - if (mHeadersDecoded && - H264SWDEC_PIC_RDY == - H264SwDecNextPicture(mHandle, &decodedPicture, 1 /* flush */)) { - int32_t picId = decodedPicture.picId; - CHECK(mPicToHeaderMap.indexOfKey(picId) >= 0); - - memcpy(outHeader->pBuffer + outHeader->nOffset, - decodedPicture.pOutputPicture, - mPictureSize); - - OMX_BUFFERHEADERTYPE *header = mPicToHeaderMap.valueFor(picId); - outHeader->nTimeStamp = header->nTimeStamp; - outHeader->nFlags = header->nFlags; - outHeader->nFilledLen = mPictureSize; - mPicToHeaderMap.removeItem(picId); - delete header; - } else { - outHeader->nTimeStamp = 0; - outHeader->nFilledLen = 0; - outHeader->nFlags = OMX_BUFFERFLAG_EOS; - mEOSStatus = OUTPUT_FRAMES_FLUSHED; - } + outHeader->nTimeStamp = 0; + outHeader->nFilledLen = 0; + outHeader->nFlags = OMX_BUFFERFLAG_EOS; outInfo->mOwnedByUs = false; notifyFillBufferDone(outHeader); - } - return true; + mEOSStatus = OUTPUT_FRAMES_FLUSHED; + } } void SoftAVC::onPortFlushCompleted(OMX_U32 portIndex) { @@ -508,44 +312,9 @@ void SoftAVC::onPortFlushCompleted(OMX_U32 portIndex) { } } -void SoftAVC::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) { - switch (mOutputPortSettingsChange) { - case NONE: - break; - - case AWAITING_DISABLED: - { - CHECK(!enabled); - mOutputPortSettingsChange = AWAITING_ENABLED; - break; - } - - default: - { - CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED); - CHECK(enabled); - mOutputPortSettingsChange = NONE; - break; - } - } -} - -void SoftAVC::updatePortDefinitions() { - OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(0)->mDef; - def->format.video.nFrameWidth = mWidth; - def->format.video.nFrameHeight = mHeight; - def->format.video.nStride = def->format.video.nFrameWidth; - def->format.video.nSliceHeight = def->format.video.nFrameHeight; - - def = &editPortInfo(1)->mDef; - def->format.video.nFrameWidth = mWidth; - def->format.video.nFrameHeight = mHeight; - def->format.video.nStride = def->format.video.nFrameWidth; - def->format.video.nSliceHeight = def->format.video.nFrameHeight; - - def->nBufferSize = - (def->format.video.nFrameWidth - * def->format.video.nFrameHeight * 3) / 2; +void SoftAVC::onReset() { + SoftVideoDecoderOMXComponent::onReset(); + mSignalledError = false; } } // namespace android diff --git a/media/libstagefright/codecs/on2/h264dec/SoftAVC.h b/media/libstagefright/codecs/on2/h264dec/SoftAVC.h index 879b014..ee69926 100644 --- a/media/libstagefright/codecs/on2/h264dec/SoftAVC.h +++ b/media/libstagefright/codecs/on2/h264dec/SoftAVC.h @@ -18,7 +18,7 @@ #define SOFT_AVC_H_ -#include "SimpleSoftOMXComponent.h" +#include "SoftVideoDecoderOMXComponent.h" #include <utils/KeyedVector.h> #include "H264SwDecApi.h" @@ -26,7 +26,7 @@ namespace android { -struct SoftAVC : public SimpleSoftOMXComponent { +struct SoftAVC : public SoftVideoDecoderOMXComponent { SoftAVC(const char *name, const OMX_CALLBACKTYPE *callbacks, OMX_PTR appData, @@ -35,22 +35,12 @@ struct SoftAVC : public SimpleSoftOMXComponent { protected: virtual ~SoftAVC(); - virtual OMX_ERRORTYPE internalGetParameter( - OMX_INDEXTYPE index, OMX_PTR params); - - virtual OMX_ERRORTYPE internalSetParameter( - OMX_INDEXTYPE index, const OMX_PTR params); - - virtual OMX_ERRORTYPE getConfig(OMX_INDEXTYPE index, OMX_PTR params); - virtual void onQueueFilled(OMX_U32 portIndex); virtual void onPortFlushCompleted(OMX_U32 portIndex); - virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled); + virtual void onReset(); private: enum { - kInputPortIndex = 0, - kOutputPortIndex = 1, kNumInputBuffers = 8, kNumOutputBuffers = 2, }; @@ -65,9 +55,7 @@ private: size_t mInputBufferCount; - uint32_t mWidth, mHeight, mPictureSize; - uint32_t mCropLeft, mCropTop; - uint32_t mCropWidth, mCropHeight; + uint32_t mPictureSize; uint8_t *mFirstPicture; int32_t mFirstPictureId; @@ -81,19 +69,10 @@ private: EOSStatus mEOSStatus; - enum OutputPortSettingChange { - NONE, - AWAITING_DISABLED, - AWAITING_ENABLED - }; - OutputPortSettingChange mOutputPortSettingsChange; - bool mSignalledError; - void initPorts(); status_t initDecoder(); - void updatePortDefinitions(); - bool drainAllOutputBuffers(); + void drainAllOutputBuffers(bool eos); void drainOneOutputBuffer(int32_t picId, uint8_t *data); void saveFirstOutputBuffer(int32_t pidId, uint8_t *data); bool handleCropRectEvent(const CropParams* crop); diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp index 4115324..51bb958 100644 --- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp +++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp @@ -424,6 +424,8 @@ void SoftVorbis::onReset() { delete mVi; mVi = NULL; } + + mOutputPortSettingsChange = NONE; } void SoftVorbis::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) { diff --git a/media/libstagefright/foundation/AHierarchicalStateMachine.cpp b/media/libstagefright/foundation/AHierarchicalStateMachine.cpp index 40c5a3c..f7a00d8 100644 --- a/media/libstagefright/foundation/AHierarchicalStateMachine.cpp +++ b/media/libstagefright/foundation/AHierarchicalStateMachine.cpp @@ -14,6 +14,10 @@ * limitations under the License. */ +//#define LOG_NDEBUG 0 +#define LOG_TAG "AHierarchicalStateMachine" +#include <utils/Log.h> + #include <media/stagefright/foundation/AHierarchicalStateMachine.h> #include <media/stagefright/foundation/ADebug.h> diff --git a/media/libstagefright/httplive/Android.mk b/media/libstagefright/httplive/Android.mk index a3fa7a3..85bd492 100644 --- a/media/libstagefright/httplive/Android.mk +++ b/media/libstagefright/httplive/Android.mk @@ -6,16 +6,25 @@ LOCAL_SRC_FILES:= \ LiveDataSource.cpp \ LiveSession.cpp \ M3UParser.cpp \ + PlaylistFetcher.cpp \ LOCAL_C_INCLUDES:= \ $(TOP)/frameworks/av/media/libstagefright \ $(TOP)/frameworks/native/include/media/openmax \ $(TOP)/external/openssl/include +LOCAL_SHARED_LIBRARIES := \ + libcrypto \ + libcutils \ + libmedia \ + libstagefright \ + libstagefright_foundation \ + libutils \ + LOCAL_MODULE:= libstagefright_httplive ifeq ($(TARGET_ARCH),arm) LOCAL_CFLAGS += -Wno-psabi endif -include $(BUILD_STATIC_LIBRARY) +include $(BUILD_SHARED_LIBRARY) diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index 505bdb3..e91c60b 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -18,12 +18,13 @@ #define LOG_TAG "LiveSession" #include <utils/Log.h> -#include "include/LiveSession.h" +#include "LiveSession.h" -#include "LiveDataSource.h" +#include "M3UParser.h" +#include "PlaylistFetcher.h" -#include "include/M3UParser.h" #include "include/HTTPBase.h" +#include "mpeg2ts/AnotherPacketSource.h" #include <cutils/properties.h> #include <media/stagefright/foundation/hexdump.h> @@ -33,6 +34,8 @@ #include <media/stagefright/DataSource.h> #include <media/stagefright/FileSource.h> #include <media/stagefright/MediaErrors.h> +#include <media/stagefright/MetaData.h> +#include <media/stagefright/Utils.h> #include <ctype.h> #include <openssl/aes.h> @@ -47,37 +50,107 @@ LiveSession::LiveSession( mUIDValid(uidValid), mUID(uid), mInPreparationPhase(true), - mDataSource(new LiveDataSource), mHTTPDataSource( HTTPBase::Create( (mFlags & kFlagIncognito) ? HTTPBase::kFlagIncognito : 0)), mPrevBandwidthIndex(-1), - mLastPlaylistFetchTimeUs(-1), - mSeqNumber(-1), - mSeekTimeUs(-1), - mNumRetries(0), - mStartOfPlayback(true), - mDurationUs(-1), - mDurationFixed(false), - mSeekDone(false), - mDisconnectPending(false), - mMonitorQueueGeneration(0), - mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY) { + mStreamMask(0), + mCheckBandwidthGeneration(0), + mLastDequeuedTimeUs(0ll), + mReconfigurationInProgress(false), + mDisconnectReplyID(0) { if (mUIDValid) { mHTTPDataSource->setUID(mUID); } + + mPacketSources.add( + STREAMTYPE_AUDIO, new AnotherPacketSource(NULL /* meta */)); + + mPacketSources.add( + STREAMTYPE_VIDEO, new AnotherPacketSource(NULL /* meta */)); + + mPacketSources.add( + STREAMTYPE_SUBTITLES, new AnotherPacketSource(NULL /* meta */)); } LiveSession::~LiveSession() { } -sp<DataSource> LiveSession::getDataSource() { - return mDataSource; +status_t LiveSession::dequeueAccessUnit( + StreamType stream, sp<ABuffer> *accessUnit) { + if (!(mStreamMask & stream)) { + return UNKNOWN_ERROR; + } + + sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream); + + status_t finalResult; + if (!packetSource->hasBufferAvailable(&finalResult)) { + return finalResult == OK ? -EAGAIN : finalResult; + } + + status_t err = packetSource->dequeueAccessUnit(accessUnit); + + const char *streamStr; + switch (stream) { + case STREAMTYPE_AUDIO: + streamStr = "audio"; + break; + case STREAMTYPE_VIDEO: + streamStr = "video"; + break; + case STREAMTYPE_SUBTITLES: + streamStr = "subs"; + break; + default: + TRESPASS(); + } + + if (err == INFO_DISCONTINUITY) { + int32_t type; + CHECK((*accessUnit)->meta()->findInt32("discontinuity", &type)); + + sp<AMessage> extra; + if (!(*accessUnit)->meta()->findMessage("extra", &extra)) { + extra.clear(); + } + + ALOGI("[%s] read discontinuity of type %d, extra = %s", + streamStr, + type, + extra == NULL ? "NULL" : extra->debugString().c_str()); + } else if (err == OK) { + int64_t timeUs; + CHECK((*accessUnit)->meta()->findInt64("timeUs", &timeUs)); + ALOGV("[%s] read buffer at time %lld us", streamStr, timeUs); + + mLastDequeuedTimeUs = timeUs; + } else { + ALOGI("[%s] encountered error %d", streamStr, err); + } + + return err; +} + +status_t LiveSession::getStreamFormat(StreamType stream, sp<AMessage> *format) { + if (!(mStreamMask & stream)) { + return UNKNOWN_ERROR; + } + + sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream); + + sp<MetaData> meta = packetSource->getFormat(); + + if (meta == NULL) { + return -EAGAIN; + } + + return convertMetaDataToMessage(meta, format); } -void LiveSession::connect( +void LiveSession::connectAsync( const char *url, const KeyedVector<String8, String8> *headers) { sp<AMessage> msg = new AMessage(kWhatConnect, id()); msg->setString("url", url); @@ -91,55 +164,184 @@ void LiveSession::connect( msg->post(); } -void LiveSession::disconnect() { - Mutex::Autolock autoLock(mLock); - mDisconnectPending = true; +status_t LiveSession::disconnect() { + sp<AMessage> msg = new AMessage(kWhatDisconnect, id()); - mHTTPDataSource->disconnect(); + sp<AMessage> response; + status_t err = msg->postAndAwaitResponse(&response); - (new AMessage(kWhatDisconnect, id()))->post(); + return err; } -void LiveSession::seekTo(int64_t timeUs) { - Mutex::Autolock autoLock(mLock); - mSeekDone = false; - +status_t LiveSession::seekTo(int64_t timeUs) { sp<AMessage> msg = new AMessage(kWhatSeek, id()); msg->setInt64("timeUs", timeUs); - msg->post(); - while (!mSeekDone) { - mCondition.wait(mLock); - } + sp<AMessage> response; + status_t err = msg->postAndAwaitResponse(&response); + + return err; } void LiveSession::onMessageReceived(const sp<AMessage> &msg) { switch (msg->what()) { case kWhatConnect: + { onConnect(msg); break; + } case kWhatDisconnect: - onDisconnect(); + { + CHECK(msg->senderAwaitsResponse(&mDisconnectReplyID)); + + if (mReconfigurationInProgress) { + break; + } + + finishDisconnect(); break; + } - case kWhatMonitorQueue: + case kWhatSeek: + { + uint32_t replyID; + CHECK(msg->senderAwaitsResponse(&replyID)); + + status_t err = onSeek(msg); + + sp<AMessage> response = new AMessage; + response->setInt32("err", err); + + response->postReply(replyID); + break; + } + + case kWhatFetcherNotify: + { + int32_t what; + CHECK(msg->findInt32("what", &what)); + + switch (what) { + case PlaylistFetcher::kWhatStarted: + break; + case PlaylistFetcher::kWhatPaused: + case PlaylistFetcher::kWhatStopped: + { + if (what == PlaylistFetcher::kWhatStopped) { + AString uri; + CHECK(msg->findString("uri", &uri)); + mFetcherInfos.removeItem(uri); + } + + if (mContinuation != NULL) { + CHECK_GT(mContinuationCounter, 0); + if (--mContinuationCounter == 0) { + mContinuation->post(); + } + } + break; + } + + case PlaylistFetcher::kWhatDurationUpdate: + { + AString uri; + CHECK(msg->findString("uri", &uri)); + + int64_t durationUs; + CHECK(msg->findInt64("durationUs", &durationUs)); + + FetcherInfo *info = &mFetcherInfos.editValueFor(uri); + info->mDurationUs = durationUs; + break; + } + + case PlaylistFetcher::kWhatError: + { + status_t err; + CHECK(msg->findInt32("err", &err)); + + ALOGE("XXX Received error %d from PlaylistFetcher.", err); + + if (mInPreparationPhase) { + postPrepared(err); + } + + mPacketSources.valueFor(STREAMTYPE_AUDIO)->signalEOS(err); + + mPacketSources.valueFor(STREAMTYPE_VIDEO)->signalEOS(err); + + mPacketSources.valueFor( + STREAMTYPE_SUBTITLES)->signalEOS(err); + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatError); + notify->setInt32("err", err); + notify->post(); + break; + } + + case PlaylistFetcher::kWhatTemporarilyDoneFetching: + { + AString uri; + CHECK(msg->findString("uri", &uri)); + + FetcherInfo *info = &mFetcherInfos.editValueFor(uri); + info->mIsPrepared = true; + + if (mInPreparationPhase) { + bool allFetchersPrepared = true; + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { + if (!mFetcherInfos.valueAt(i).mIsPrepared) { + allFetchersPrepared = false; + break; + } + } + + if (allFetchersPrepared) { + postPrepared(OK); + } + } + break; + } + + default: + TRESPASS(); + } + + break; + } + + case kWhatCheckBandwidth: { int32_t generation; CHECK(msg->findInt32("generation", &generation)); - if (generation != mMonitorQueueGeneration) { - // Stale event + if (generation != mCheckBandwidthGeneration) { break; } - onMonitorQueue(); + onCheckBandwidth(); break; } - case kWhatSeek: - onSeek(msg); + case kWhatChangeConfiguration2: + { + onChangeConfiguration2(msg); + break; + } + + case kWhatChangeConfiguration3: + { + onChangeConfiguration3(msg); + break; + } + + case kWhatFinishDisconnect2: + { + onFinishDisconnect2(); break; + } default: TRESPASS(); @@ -172,48 +374,127 @@ void LiveSession::onConnect(const sp<AMessage> &msg) { headers = NULL; } +#if 1 ALOGI("onConnect <URL suppressed>"); +#else + ALOGI("onConnect %s", url.c_str()); +#endif mMasterURL = url; bool dummy; - sp<M3UParser> playlist = fetchPlaylist(url.c_str(), &dummy); + mPlaylist = fetchPlaylist(url.c_str(), NULL /* curPlaylistHash */, &dummy); - if (playlist == NULL) { + if (mPlaylist == NULL) { ALOGE("unable to fetch master playlist '%s'.", url.c_str()); - signalEOS(ERROR_IO); + postPrepared(ERROR_IO); return; } - if (playlist->isVariantPlaylist()) { - for (size_t i = 0; i < playlist->size(); ++i) { + // We trust the content provider to make a reasonable choice of preferred + // initial bandwidth by listing it first in the variant playlist. + // At startup we really don't have a good estimate on the available + // network bandwidth since we haven't tranferred any data yet. Once + // we have we can make a better informed choice. + size_t initialBandwidth = 0; + size_t initialBandwidthIndex = 0; + + if (mPlaylist->isVariantPlaylist()) { + for (size_t i = 0; i < mPlaylist->size(); ++i) { BandwidthItem item; + item.mPlaylistIndex = i; + sp<AMessage> meta; - playlist->itemAt(i, &item.mURI, &meta); + AString uri; + mPlaylist->itemAt(i, &uri, &meta); unsigned long bandwidth; CHECK(meta->findInt32("bandwidth", (int32_t *)&item.mBandwidth)); + if (initialBandwidth == 0) { + initialBandwidth = item.mBandwidth; + } + mBandwidthItems.push(item); } CHECK_GT(mBandwidthItems.size(), 0u); mBandwidthItems.sort(SortByBandwidth); + + for (size_t i = 0; i < mBandwidthItems.size(); ++i) { + if (mBandwidthItems.itemAt(i).mBandwidth == initialBandwidth) { + initialBandwidthIndex = i; + break; + } + } + } else { + // dummy item. + BandwidthItem item; + item.mPlaylistIndex = 0; + item.mBandwidth = 0; + mBandwidthItems.push(item); } - postMonitorQueue(); + changeConfiguration(0ll /* timeUs */, initialBandwidthIndex); } -void LiveSession::onDisconnect() { - ALOGI("onDisconnect"); +void LiveSession::finishDisconnect() { + // No reconfiguration is currently pending, make sure none will trigger + // during disconnection either. + cancelCheckBandwidthEvent(); + + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { + mFetcherInfos.valueAt(i).mFetcher->stopAsync(); + } + + sp<AMessage> msg = new AMessage(kWhatFinishDisconnect2, id()); - signalEOS(ERROR_END_OF_STREAM); + mContinuationCounter = mFetcherInfos.size(); + mContinuation = msg; - Mutex::Autolock autoLock(mLock); - mDisconnectPending = false; + if (mContinuationCounter == 0) { + msg->post(); + } +} + +void LiveSession::onFinishDisconnect2() { + mContinuation.clear(); + + mPacketSources.valueFor(STREAMTYPE_AUDIO)->signalEOS(ERROR_END_OF_STREAM); + mPacketSources.valueFor(STREAMTYPE_VIDEO)->signalEOS(ERROR_END_OF_STREAM); + + mPacketSources.valueFor( + STREAMTYPE_SUBTITLES)->signalEOS(ERROR_END_OF_STREAM); + + sp<AMessage> response = new AMessage; + response->setInt32("err", OK); + + response->postReply(mDisconnectReplyID); + mDisconnectReplyID = 0; +} + +sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) { + ssize_t index = mFetcherInfos.indexOfKey(uri); + + if (index >= 0) { + return NULL; + } + + sp<AMessage> notify = new AMessage(kWhatFetcherNotify, id()); + notify->setString("uri", uri); + + FetcherInfo info; + info.mFetcher = new PlaylistFetcher(notify, this, uri); + info.mDurationUs = -1ll; + info.mIsPrepared = false; + looper()->registerHandler(info.mFetcher); + + mFetcherInfos.add(uri, info); + + return info.mFetcher; } status_t LiveSession::fetchFile( @@ -229,14 +510,6 @@ status_t LiveSession::fetchFile( && strncasecmp(url, "https://", 8)) { return ERROR_UNSUPPORTED; } else { - { - Mutex::Autolock autoLock(mLock); - - if (mDisconnectPending) { - return ERROR_IO; - } - } - KeyedVector<String8, String8> headers = mExtraHeaders; if (range_offset > 0 || range_length >= 0) { headers.add( @@ -315,7 +588,8 @@ status_t LiveSession::fetchFile( return OK; } -sp<M3UParser> LiveSession::fetchPlaylist(const char *url, bool *unchanged) { +sp<M3UParser> LiveSession::fetchPlaylist( + const char *url, uint8_t *curPlaylistHash, bool *unchanged) { ALOGV("fetchPlaylist '%s'", url); *unchanged = false; @@ -339,13 +613,8 @@ sp<M3UParser> LiveSession::fetchPlaylist(const char *url, bool *unchanged) { MD5_Final(hash, &m); - if (mPlaylist != NULL && !memcmp(hash, mPlaylistHash, 16)) { + if (curPlaylistHash != NULL && !memcmp(hash, curPlaylistHash, 16)) { // playlist unchanged - - if (mRefreshState != THIRD_UNCHANGED_RELOAD_ATTEMPT) { - mRefreshState = (RefreshState)(mRefreshState + 1); - } - *unchanged = true; ALOGV("Playlist unchanged, refresh state is now %d", @@ -354,9 +623,9 @@ sp<M3UParser> LiveSession::fetchPlaylist(const char *url, bool *unchanged) { return NULL; } - memcpy(mPlaylistHash, hash, sizeof(hash)); - - mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY; + if (curPlaylistHash != NULL) { + memcpy(curPlaylistHash, hash, sizeof(hash)); + } #endif sp<M3UParser> playlist = @@ -371,37 +640,6 @@ sp<M3UParser> LiveSession::fetchPlaylist(const char *url, bool *unchanged) { return playlist; } -int64_t LiveSession::getSegmentStartTimeUs(int32_t seqNumber) const { - CHECK(mPlaylist != NULL); - - int32_t firstSeqNumberInPlaylist; - if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( - "media-sequence", &firstSeqNumberInPlaylist)) { - firstSeqNumberInPlaylist = 0; - } - - int32_t lastSeqNumberInPlaylist = - firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; - - CHECK_GE(seqNumber, firstSeqNumberInPlaylist); - CHECK_LE(seqNumber, lastSeqNumberInPlaylist); - - int64_t segmentStartUs = 0ll; - for (int32_t index = 0; - index < seqNumber - firstSeqNumberInPlaylist; ++index) { - sp<AMessage> itemMeta; - CHECK(mPlaylist->itemAt( - index, NULL /* uri */, &itemMeta)); - - int64_t itemDurationUs; - CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); - - segmentStartUs += itemDurationUs; - } - - return segmentStartUs; -} - static double uniformRand() { return (double)rand() / RAND_MAX; } @@ -412,36 +650,50 @@ size_t LiveSession::getBandwidthIndex() { } #if 1 - int32_t bandwidthBps; - if (mHTTPDataSource != NULL - && mHTTPDataSource->estimateBandwidth(&bandwidthBps)) { - ALOGV("bandwidth estimated at %.2f kbps", bandwidthBps / 1024.0f); - } else { - ALOGV("no bandwidth estimate."); - return 0; // Pick the lowest bandwidth stream by default. - } - char value[PROPERTY_VALUE_MAX]; - if (property_get("media.httplive.max-bw", value, NULL)) { + ssize_t index = -1; + if (property_get("media.httplive.bw-index", value, NULL)) { char *end; - long maxBw = strtoul(value, &end, 10); - if (end > value && *end == '\0') { - if (maxBw > 0 && bandwidthBps > maxBw) { - ALOGV("bandwidth capped to %ld bps", maxBw); - bandwidthBps = maxBw; - } + index = strtol(value, &end, 10); + CHECK(end > value && *end == '\0'); + + if (index >= 0 && (size_t)index >= mBandwidthItems.size()) { + index = mBandwidthItems.size() - 1; } } - // Consider only 80% of the available bandwidth usable. - bandwidthBps = (bandwidthBps * 8) / 10; + if (index < 0) { + int32_t bandwidthBps; + if (mHTTPDataSource != NULL + && mHTTPDataSource->estimateBandwidth(&bandwidthBps)) { + ALOGV("bandwidth estimated at %.2f kbps", bandwidthBps / 1024.0f); + } else { + ALOGV("no bandwidth estimate."); + return 0; // Pick the lowest bandwidth stream by default. + } - // Pick the highest bandwidth stream below or equal to estimated bandwidth. + char value[PROPERTY_VALUE_MAX]; + if (property_get("media.httplive.max-bw", value, NULL)) { + char *end; + long maxBw = strtoul(value, &end, 10); + if (end > value && *end == '\0') { + if (maxBw > 0 && bandwidthBps > maxBw) { + ALOGV("bandwidth capped to %ld bps", maxBw); + bandwidthBps = maxBw; + } + } + } - size_t index = mBandwidthItems.size() - 1; - while (index > 0 && mBandwidthItems.itemAt(index).mBandwidth - > (size_t)bandwidthBps) { - --index; + // Consider only 80% of the available bandwidth usable. + bandwidthBps = (bandwidthBps * 8) / 10; + + // Pick the highest bandwidth stream below or equal to estimated bandwidth. + + index = mBandwidthItems.size() - 1; + while (index > 0 && mBandwidthItems.itemAt(index).mBandwidth + > (size_t)bandwidthBps) { + --index; + } } #elif 0 // Change bandwidth at random() @@ -452,6 +704,8 @@ size_t LiveSession::getBandwidthIndex() { // to lowest) const size_t kMinIndex = 0; + static ssize_t mPrevBandwidthIndex = -1; + size_t index; if (mPrevBandwidthIndex < 0) { index = kMinIndex; @@ -463,6 +717,7 @@ size_t LiveSession::getBandwidthIndex() { index = kMinIndex; } } + mPrevBandwidthIndex = index; #elif 0 // Pick the highest bandwidth stream below or equal to 1.2 Mbit/sec @@ -470,570 +725,381 @@ size_t LiveSession::getBandwidthIndex() { while (index > 0 && mBandwidthItems.itemAt(index).mBandwidth > 1200000) { --index; } +#elif 1 + char value[PROPERTY_VALUE_MAX]; + size_t index; + if (property_get("media.httplive.bw-index", value, NULL)) { + char *end; + index = strtoul(value, &end, 10); + CHECK(end > value && *end == '\0'); + + if (index >= mBandwidthItems.size()) { + index = mBandwidthItems.size() - 1; + } + } else { + index = 0; + } #else size_t index = mBandwidthItems.size() - 1; // Highest bandwidth stream #endif + CHECK_GE(index, 0); + return index; } -bool LiveSession::timeToRefreshPlaylist(int64_t nowUs) const { - if (mPlaylist == NULL) { - CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY); - return true; - } - - int32_t targetDurationSecs; - CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); - - int64_t targetDurationUs = targetDurationSecs * 1000000ll; - - int64_t minPlaylistAgeUs; - - switch (mRefreshState) { - case INITIAL_MINIMUM_RELOAD_DELAY: - { - size_t n = mPlaylist->size(); - if (n > 0) { - sp<AMessage> itemMeta; - CHECK(mPlaylist->itemAt(n - 1, NULL /* uri */, &itemMeta)); - - int64_t itemDurationUs; - CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); - - minPlaylistAgeUs = itemDurationUs; - break; - } - - // fall through - } - - case FIRST_UNCHANGED_RELOAD_ATTEMPT: - { - minPlaylistAgeUs = targetDurationUs / 2; - break; - } - - case SECOND_UNCHANGED_RELOAD_ATTEMPT: - { - minPlaylistAgeUs = (targetDurationUs * 3) / 2; - break; - } - - case THIRD_UNCHANGED_RELOAD_ATTEMPT: - { - minPlaylistAgeUs = targetDurationUs * 3; - break; - } +status_t LiveSession::onSeek(const sp<AMessage> &msg) { + int64_t timeUs; + CHECK(msg->findInt64("timeUs", &timeUs)); - default: - TRESPASS(); - break; + if (!mReconfigurationInProgress) { + changeConfiguration(timeUs, getBandwidthIndex()); } - return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs; + return OK; } -void LiveSession::onDownloadNext() { - size_t bandwidthIndex = getBandwidthIndex(); - -rinse_repeat: - int64_t nowUs = ALooper::GetNowUs(); - - if (mLastPlaylistFetchTimeUs < 0 - || (ssize_t)bandwidthIndex != mPrevBandwidthIndex - || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) { - AString url; - if (mBandwidthItems.size() > 0) { - url = mBandwidthItems.editItemAt(bandwidthIndex).mURI; - } else { - url = mMasterURL; - } - - if ((ssize_t)bandwidthIndex != mPrevBandwidthIndex) { - // If we switch bandwidths, do not pay any heed to whether - // playlists changed since the last time... - mPlaylist.clear(); - } - - bool unchanged; - sp<M3UParser> playlist = fetchPlaylist(url.c_str(), &unchanged); - if (playlist == NULL) { - if (unchanged) { - // We succeeded in fetching the playlist, but it was - // unchanged from the last time we tried. - } else { - ALOGE("failed to load playlist at url '%s'", url.c_str()); - signalEOS(ERROR_IO); - - return; - } - } else { - mPlaylist = playlist; - } - - if (!mDurationFixed) { - Mutex::Autolock autoLock(mLock); - - if (!mPlaylist->isComplete() && !mPlaylist->isEvent()) { - mDurationUs = -1; - mDurationFixed = true; - } else { - mDurationUs = 0; - for (size_t i = 0; i < mPlaylist->size(); ++i) { - sp<AMessage> itemMeta; - CHECK(mPlaylist->itemAt( - i, NULL /* uri */, &itemMeta)); - - int64_t itemDurationUs; - CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); - - mDurationUs += itemDurationUs; - } +status_t LiveSession::getDuration(int64_t *durationUs) const { + int64_t maxDurationUs = 0ll; + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { + int64_t fetcherDurationUs = mFetcherInfos.valueAt(i).mDurationUs; - mDurationFixed = mPlaylist->isComplete(); - } + if (fetcherDurationUs >= 0ll && fetcherDurationUs > maxDurationUs) { + maxDurationUs = fetcherDurationUs; } - - mLastPlaylistFetchTimeUs = ALooper::GetNowUs(); } - int32_t firstSeqNumberInPlaylist; - if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( - "media-sequence", &firstSeqNumberInPlaylist)) { - firstSeqNumberInPlaylist = 0; - } + *durationUs = maxDurationUs; - bool seekDiscontinuity = false; - bool explicitDiscontinuity = false; - bool bandwidthChanged = false; - - if (mSeekTimeUs >= 0) { - if (mPlaylist->isComplete() || mPlaylist->isEvent()) { - size_t index = 0; - int64_t segmentStartUs = 0; - while (index < mPlaylist->size()) { - sp<AMessage> itemMeta; - CHECK(mPlaylist->itemAt( - index, NULL /* uri */, &itemMeta)); - - int64_t itemDurationUs; - CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); + return OK; +} - if (mSeekTimeUs < segmentStartUs + itemDurationUs) { - break; - } +bool LiveSession::isSeekable() const { + int64_t durationUs; + return getDuration(&durationUs) == OK && durationUs >= 0; +} - segmentStartUs += itemDurationUs; - ++index; - } +bool LiveSession::hasDynamicDuration() const { + return false; +} - if (index < mPlaylist->size()) { - int32_t newSeqNumber = firstSeqNumberInPlaylist + index; +void LiveSession::changeConfiguration(int64_t timeUs, size_t bandwidthIndex) { + CHECK(!mReconfigurationInProgress); + mReconfigurationInProgress = true; - ALOGI("seeking to seq no %d", newSeqNumber); + mPrevBandwidthIndex = bandwidthIndex; - mSeqNumber = newSeqNumber; + ALOGV("changeConfiguration => timeUs:%lld us, bwIndex:%d", + timeUs, bandwidthIndex); - mDataSource->reset(); + mPlaylist->pickRandomMediaItems(); - // reseting the data source will have had the - // side effect of discarding any previously queued - // bandwidth change discontinuity. - // Therefore we'll need to treat these seek - // discontinuities as involving a bandwidth change - // even if they aren't directly. - seekDiscontinuity = true; - bandwidthChanged = true; - } - } + CHECK_LT(bandwidthIndex, mBandwidthItems.size()); + const BandwidthItem &item = mBandwidthItems.itemAt(bandwidthIndex); - mSeekTimeUs = -1; + uint32_t streamMask = 0; - Mutex::Autolock autoLock(mLock); - mSeekDone = true; - mCondition.broadcast(); + AString audioURI; + if (mPlaylist->getAudioURI(item.mPlaylistIndex, &audioURI)) { + streamMask |= STREAMTYPE_AUDIO; } - const int32_t lastSeqNumberInPlaylist = - firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; - - if (mSeqNumber < 0) { - if (mPlaylist->isComplete()) { - mSeqNumber = firstSeqNumberInPlaylist; - } else { - // If this is a live session, start 3 segments from the end. - mSeqNumber = lastSeqNumberInPlaylist - 3; - if (mSeqNumber < firstSeqNumberInPlaylist) { - mSeqNumber = firstSeqNumberInPlaylist; - } - } + AString videoURI; + if (mPlaylist->getVideoURI(item.mPlaylistIndex, &videoURI)) { + streamMask |= STREAMTYPE_VIDEO; } - if (mSeqNumber < firstSeqNumberInPlaylist - || mSeqNumber > lastSeqNumberInPlaylist) { - if (mPrevBandwidthIndex != (ssize_t)bandwidthIndex) { - // Go back to the previous bandwidth. + AString subtitleURI; + if (mPlaylist->getSubtitleURI(item.mPlaylistIndex, &subtitleURI)) { + streamMask |= STREAMTYPE_SUBTITLES; + } - ALOGI("new bandwidth does not have the sequence number " - "we're looking for, switching back to previous bandwidth"); + // Step 1, stop and discard fetchers that are no longer needed. + // Pause those that we'll reuse. + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { + const AString &uri = mFetcherInfos.keyAt(i); - mLastPlaylistFetchTimeUs = -1; - bandwidthIndex = mPrevBandwidthIndex; - goto rinse_repeat; - } + bool discardFetcher = true; - if (!mPlaylist->isComplete() && mNumRetries < kMaxNumRetries) { - ++mNumRetries; - - if (mSeqNumber > lastSeqNumberInPlaylist) { - mLastPlaylistFetchTimeUs = -1; - postMonitorQueue(3000000ll); - return; + // If we're seeking all current fetchers are discarded. + if (timeUs < 0ll) { + if (((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) + || ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) + || ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI)) { + discardFetcher = false; } + } - // we've missed the boat, let's start from the lowest sequence - // number available and signal a discontinuity. - - ALOGI("We've missed the boat, restarting playback."); - mSeqNumber = lastSeqNumberInPlaylist; - explicitDiscontinuity = true; - - // fall through + if (discardFetcher) { + mFetcherInfos.valueAt(i).mFetcher->stopAsync(); } else { - ALOGE("Cannot find sequence number %d in playlist " - "(contains %d - %d)", - mSeqNumber, firstSeqNumberInPlaylist, - firstSeqNumberInPlaylist + mPlaylist->size() - 1); - - signalEOS(ERROR_END_OF_STREAM); - return; + mFetcherInfos.valueAt(i).mFetcher->pauseAsync(); } } - mNumRetries = 0; - - AString uri; - sp<AMessage> itemMeta; - CHECK(mPlaylist->itemAt( - mSeqNumber - firstSeqNumberInPlaylist, - &uri, - &itemMeta)); - - int32_t val; - if (itemMeta->findInt32("discontinuity", &val) && val != 0) { - explicitDiscontinuity = true; + sp<AMessage> msg = new AMessage(kWhatChangeConfiguration2, id()); + msg->setInt32("streamMask", streamMask); + msg->setInt64("timeUs", timeUs); + if (streamMask & STREAMTYPE_AUDIO) { + msg->setString("audioURI", audioURI.c_str()); } - - int64_t range_offset, range_length; - if (!itemMeta->findInt64("range-offset", &range_offset) - || !itemMeta->findInt64("range-length", &range_length)) { - range_offset = 0; - range_length = -1; + if (streamMask & STREAMTYPE_VIDEO) { + msg->setString("videoURI", videoURI.c_str()); } - - ALOGV("fetching segment %d from (%d .. %d)", - mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist); - - sp<ABuffer> buffer; - status_t err = fetchFile(uri.c_str(), &buffer, range_offset, range_length); - if (err != OK) { - ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str()); - signalEOS(err); - return; + if (streamMask & STREAMTYPE_SUBTITLES) { + msg->setString("subtitleURI", subtitleURI.c_str()); } - CHECK(buffer != NULL); - - err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer); + // Every time a fetcher acknowledges the stopAsync or pauseAsync request + // we'll decrement mContinuationCounter, once it reaches zero, i.e. all + // fetchers have completed their asynchronous operation, we'll post + // mContinuation, which then is handled below in onChangeConfiguration2. + mContinuationCounter = mFetcherInfos.size(); + mContinuation = msg; - if (err != OK) { - ALOGE("decryptBuffer failed w/ error %d", err); - - signalEOS(err); - return; + if (mContinuationCounter == 0) { + msg->post(); } +} - if (buffer->size() == 0 || buffer->data()[0] != 0x47) { - // Not a transport stream??? - - ALOGE("This doesn't look like a transport stream..."); - - mBandwidthItems.removeAt(bandwidthIndex); - - if (mBandwidthItems.isEmpty()) { - signalEOS(ERROR_UNSUPPORTED); - return; - } +void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) { + mContinuation.clear(); - ALOGI("Retrying with a different bandwidth stream."); + // All fetchers are either suspended or have been removed now. - mLastPlaylistFetchTimeUs = -1; - bandwidthIndex = getBandwidthIndex(); - mPrevBandwidthIndex = bandwidthIndex; - mSeqNumber = -1; + uint32_t streamMask; + CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask)); - goto rinse_repeat; + AString audioURI, videoURI, subtitleURI; + if (streamMask & STREAMTYPE_AUDIO) { + CHECK(msg->findString("audioURI", &audioURI)); + ALOGV("audioURI = '%s'", audioURI.c_str()); } - - if ((size_t)mPrevBandwidthIndex != bandwidthIndex) { - bandwidthChanged = true; + if (streamMask & STREAMTYPE_VIDEO) { + CHECK(msg->findString("videoURI", &videoURI)); + ALOGV("videoURI = '%s'", videoURI.c_str()); } - - if (mPrevBandwidthIndex < 0) { - // Don't signal a bandwidth change at the very beginning of - // playback. - bandwidthChanged = false; + if (streamMask & STREAMTYPE_SUBTITLES) { + CHECK(msg->findString("subtitleURI", &subtitleURI)); + ALOGV("subtitleURI = '%s'", subtitleURI.c_str()); } - if (mStartOfPlayback) { - seekDiscontinuity = true; - mStartOfPlayback = false; + // Determine which decoders to shutdown on the player side, + // a decoder has to be shutdown if either + // 1) its streamtype was active before but now longer isn't. + // or + // 2) its streamtype was already active and still is but the URI + // has changed. + uint32_t changedMask = 0; + if (((mStreamMask & streamMask & STREAMTYPE_AUDIO) + && !(audioURI == mAudioURI)) + || (mStreamMask & ~streamMask & STREAMTYPE_AUDIO)) { + changedMask |= STREAMTYPE_AUDIO; + } + if (((mStreamMask & streamMask & STREAMTYPE_VIDEO) + && !(videoURI == mVideoURI)) + || (mStreamMask & ~streamMask & STREAMTYPE_VIDEO)) { + changedMask |= STREAMTYPE_VIDEO; } - if (seekDiscontinuity || explicitDiscontinuity || bandwidthChanged) { - // Signal discontinuity. - - ALOGI("queueing discontinuity (seek=%d, explicit=%d, bandwidthChanged=%d)", - seekDiscontinuity, explicitDiscontinuity, bandwidthChanged); + if (changedMask == 0) { + // If nothing changed as far as the audio/video decoders + // are concerned we can proceed. + onChangeConfiguration3(msg); + return; + } - sp<ABuffer> tmp = new ABuffer(188); - memset(tmp->data(), 0, tmp->size()); + // Something changed, inform the player which will shutdown the + // corresponding decoders and will post the reply once that's done. + // Handling the reply will continue executing below in + // onChangeConfiguration3. + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatStreamsChanged); + notify->setInt32("changedMask", changedMask); - // signal a 'hard' discontinuity for explicit or bandwidthChanged. - uint8_t type = (explicitDiscontinuity || bandwidthChanged) ? 1 : 0; + msg->setWhat(kWhatChangeConfiguration3); + msg->setTarget(id()); - if (mPlaylist->isComplete() || mPlaylist->isEvent()) { - // If this was a live event this made no sense since - // we don't have access to all the segment before the current - // one. - int64_t segmentStartTimeUs = getSegmentStartTimeUs(mSeqNumber); - memcpy(tmp->data() + 2, &segmentStartTimeUs, sizeof(segmentStartTimeUs)); + notify->setMessage("reply", msg); + notify->post(); +} - type |= 2; - } +void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) { + // All remaining fetchers are still suspended, the player has shutdown + // any decoders that needed it. - tmp->data()[1] = type; + uint32_t streamMask; + CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask)); - mDataSource->queueBuffer(tmp); + AString audioURI, videoURI, subtitleURI; + if (streamMask & STREAMTYPE_AUDIO) { + CHECK(msg->findString("audioURI", &audioURI)); + } + if (streamMask & STREAMTYPE_VIDEO) { + CHECK(msg->findString("videoURI", &videoURI)); + } + if (streamMask & STREAMTYPE_SUBTITLES) { + CHECK(msg->findString("subtitleURI", &subtitleURI)); } - mDataSource->queueBuffer(buffer); + int64_t timeUs; + CHECK(msg->findInt64("timeUs", &timeUs)); - mPrevBandwidthIndex = bandwidthIndex; - ++mSeqNumber; + if (timeUs < 0ll) { + timeUs = mLastDequeuedTimeUs; + } - postMonitorQueue(); -} + mStreamMask = streamMask; + mAudioURI = audioURI; + mVideoURI = videoURI; + mSubtitleURI = subtitleURI; -void LiveSession::signalEOS(status_t err) { - if (mInPreparationPhase && mNotify != NULL) { - sp<AMessage> notify = mNotify->dup(); + // Resume all existing fetchers and assign them packet sources. + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { + const AString &uri = mFetcherInfos.keyAt(i); - notify->setInt32( - "what", - err == ERROR_END_OF_STREAM - ? kWhatPrepared : kWhatPreparationFailed); + uint32_t resumeMask = 0; - if (err != ERROR_END_OF_STREAM) { - notify->setInt32("err", err); + sp<AnotherPacketSource> audioSource; + if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) { + audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO); + resumeMask |= STREAMTYPE_AUDIO; } - notify->post(); - - mInPreparationPhase = false; - } - - mDataSource->queueEOS(err); -} - -void LiveSession::onMonitorQueue() { - if (mSeekTimeUs >= 0 - || mDataSource->countQueuedBuffers() < kMaxNumQueuedFragments) { - onDownloadNext(); - } else { - if (mInPreparationPhase) { - if (mNotify != NULL) { - sp<AMessage> notify = mNotify->dup(); - notify->setInt32("what", kWhatPrepared); - notify->post(); - } - - mInPreparationPhase = false; + sp<AnotherPacketSource> videoSource; + if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) { + videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO); + resumeMask |= STREAMTYPE_VIDEO; } - postMonitorQueue(1000000ll); - } -} + sp<AnotherPacketSource> subtitleSource; + if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) { + subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES); + resumeMask |= STREAMTYPE_SUBTITLES; + } -status_t LiveSession::decryptBuffer( - size_t playlistIndex, const sp<ABuffer> &buffer) { - sp<AMessage> itemMeta; - bool found = false; - AString method; + CHECK_NE(resumeMask, 0u); - for (ssize_t i = playlistIndex; i >= 0; --i) { - AString uri; - CHECK(mPlaylist->itemAt(i, &uri, &itemMeta)); + ALOGV("resuming fetchers for mask 0x%08x", resumeMask); - if (itemMeta->findString("cipher-method", &method)) { - found = true; - break; - } - } + streamMask &= ~resumeMask; - if (!found) { - method = "NONE"; + mFetcherInfos.valueAt(i).mFetcher->startAsync( + audioSource, videoSource, subtitleSource); } - if (method == "NONE") { - return OK; - } else if (!(method == "AES-128")) { - ALOGE("Unsupported cipher method '%s'", method.c_str()); - return ERROR_UNSUPPORTED; - } + // streamMask now only contains the types that need a new fetcher created. - AString keyURI; - if (!itemMeta->findString("cipher-uri", &keyURI)) { - ALOGE("Missing key uri"); - return ERROR_MALFORMED; + if (streamMask != 0) { + ALOGV("creating new fetchers for mask 0x%08x", streamMask); } - ssize_t index = mAESKeyForURI.indexOfKey(keyURI); - - sp<ABuffer> key; - if (index >= 0) { - key = mAESKeyForURI.valueAt(index); - } else { - key = new ABuffer(16); - - sp<HTTPBase> keySource = - HTTPBase::Create( - (mFlags & kFlagIncognito) - ? HTTPBase::kFlagIncognito - : 0); + while (streamMask != 0) { + StreamType streamType = (StreamType)(streamMask & ~(streamMask - 1)); - if (mUIDValid) { - keySource->setUID(mUID); + AString uri; + switch (streamType) { + case STREAMTYPE_AUDIO: + uri = audioURI; + break; + case STREAMTYPE_VIDEO: + uri = videoURI; + break; + case STREAMTYPE_SUBTITLES: + uri = subtitleURI; + break; + default: + TRESPASS(); } - status_t err = - keySource->connect( - keyURI.c_str(), - mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders); - - if (err == OK) { - size_t offset = 0; - while (offset < 16) { - ssize_t n = keySource->readAt( - offset, key->data() + offset, 16 - offset); - if (n <= 0) { - err = ERROR_IO; - break; - } + sp<PlaylistFetcher> fetcher = addFetcher(uri.c_str()); + CHECK(fetcher != NULL); - offset += n; - } - } + sp<AnotherPacketSource> audioSource; + if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) { + audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO); + audioSource->clear(); - if (err != OK) { - ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str()); - return ERROR_IO; + streamMask &= ~STREAMTYPE_AUDIO; } - mAESKeyForURI.add(keyURI, key); - } + sp<AnotherPacketSource> videoSource; + if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) { + videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO); + videoSource->clear(); - AES_KEY aes_key; - if (AES_set_decrypt_key(key->data(), 128, &aes_key) != 0) { - ALOGE("failed to set AES decryption key."); - return UNKNOWN_ERROR; - } - - unsigned char aes_ivec[16]; - - AString iv; - if (itemMeta->findString("cipher-iv", &iv)) { - if ((!iv.startsWith("0x") && !iv.startsWith("0X")) - || iv.size() != 16 * 2 + 2) { - ALOGE("malformed cipher IV '%s'.", iv.c_str()); - return ERROR_MALFORMED; + streamMask &= ~STREAMTYPE_VIDEO; } - memset(aes_ivec, 0, sizeof(aes_ivec)); - for (size_t i = 0; i < 16; ++i) { - char c1 = tolower(iv.c_str()[2 + 2 * i]); - char c2 = tolower(iv.c_str()[3 + 2 * i]); - if (!isxdigit(c1) || !isxdigit(c2)) { - ALOGE("malformed cipher IV '%s'.", iv.c_str()); - return ERROR_MALFORMED; - } - uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10; - uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10; + sp<AnotherPacketSource> subtitleSource; + if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) { + subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES); + subtitleSource->clear(); - aes_ivec[i] = nibble1 << 4 | nibble2; + streamMask &= ~STREAMTYPE_SUBTITLES; } - } else { - memset(aes_ivec, 0, sizeof(aes_ivec)); - aes_ivec[15] = mSeqNumber & 0xff; - aes_ivec[14] = (mSeqNumber >> 8) & 0xff; - aes_ivec[13] = (mSeqNumber >> 16) & 0xff; - aes_ivec[12] = (mSeqNumber >> 24) & 0xff; + + fetcher->startAsync(audioSource, videoSource, subtitleSource, timeUs); } - AES_cbc_encrypt( - buffer->data(), buffer->data(), buffer->size(), - &aes_key, aes_ivec, AES_DECRYPT); + // All fetchers have now been started, the configuration change + // has completed. - // hexdump(buffer->data(), buffer->size()); + scheduleCheckBandwidthEvent(); - size_t n = buffer->size(); - CHECK_GT(n, 0u); + ALOGV("XXX configuration change completed."); - size_t pad = buffer->data()[n - 1]; + mReconfigurationInProgress = false; - CHECK_GT(pad, 0u); - CHECK_LE(pad, 16u); - CHECK_GE((size_t)n, pad); - for (size_t i = 0; i < pad; ++i) { - CHECK_EQ((unsigned)buffer->data()[n - 1 - i], pad); + if (mDisconnectReplyID != 0) { + finishDisconnect(); } +} - n -= pad; - - buffer->setRange(buffer->offset(), n); - - return OK; +void LiveSession::scheduleCheckBandwidthEvent() { + sp<AMessage> msg = new AMessage(kWhatCheckBandwidth, id()); + msg->setInt32("generation", mCheckBandwidthGeneration); + msg->post(10000000ll); } -void LiveSession::postMonitorQueue(int64_t delayUs) { - sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id()); - msg->setInt32("generation", ++mMonitorQueueGeneration); - msg->post(delayUs); +void LiveSession::cancelCheckBandwidthEvent() { + ++mCheckBandwidthGeneration; } -void LiveSession::onSeek(const sp<AMessage> &msg) { - int64_t timeUs; - CHECK(msg->findInt64("timeUs", &timeUs)); +void LiveSession::onCheckBandwidth() { + if (mReconfigurationInProgress) { + scheduleCheckBandwidthEvent(); + return; + } + + size_t bandwidthIndex = getBandwidthIndex(); + if (mPrevBandwidthIndex < 0 + || bandwidthIndex != (size_t)mPrevBandwidthIndex) { + changeConfiguration(-1ll /* timeUs */, bandwidthIndex); + } - mSeekTimeUs = timeUs; - postMonitorQueue(); + // Handling the kWhatCheckBandwidth even here does _not_ automatically + // schedule another one on return, only an explicit call to + // scheduleCheckBandwidthEvent will do that. + // This ensures that only one configuration change is ongoing at any + // one time, once that completes it'll schedule another check bandwidth + // event. } -status_t LiveSession::getDuration(int64_t *durationUs) const { - Mutex::Autolock autoLock(mLock); - *durationUs = mDurationUs; +void LiveSession::postPrepared(status_t err) { + CHECK(mInPreparationPhase); - return OK; -} + sp<AMessage> notify = mNotify->dup(); + if (err == OK || err == ERROR_END_OF_STREAM) { + notify->setInt32("what", kWhatPrepared); + } else { + notify->setInt32("what", kWhatPreparationFailed); + notify->setInt32("err", err); + } -bool LiveSession::isSeekable() const { - int64_t durationUs; - return getDuration(&durationUs) == OK && durationUs >= 0; -} + notify->post(); -bool LiveSession::hasDynamicDuration() const { - return !mDurationFixed; + mInPreparationPhase = false; } } // namespace android diff --git a/media/libstagefright/include/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index db44a33..b134725 100644 --- a/media/libstagefright/include/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -25,10 +25,12 @@ namespace android { struct ABuffer; +struct AnotherPacketSource; struct DataSource; +struct HTTPBase; struct LiveDataSource; struct M3UParser; -struct HTTPBase; +struct PlaylistFetcher; struct LiveSession : public AHandler { enum Flags { @@ -39,24 +41,32 @@ struct LiveSession : public AHandler { const sp<AMessage> ¬ify, uint32_t flags = 0, bool uidValid = false, uid_t uid = 0); - sp<DataSource> getDataSource(); + enum StreamType { + STREAMTYPE_AUDIO = 1, + STREAMTYPE_VIDEO = 2, + STREAMTYPE_SUBTITLES = 4, + }; + status_t dequeueAccessUnit(StreamType stream, sp<ABuffer> *accessUnit); + + status_t getStreamFormat(StreamType stream, sp<AMessage> *format); - void connect( + void connectAsync( const char *url, const KeyedVector<String8, String8> *headers = NULL); - void disconnect(); + status_t disconnect(); // Blocks until seek is complete. - void seekTo(int64_t timeUs); + status_t seekTo(int64_t timeUs); status_t getDuration(int64_t *durationUs) const; bool isSeekable() const; bool hasDynamicDuration() const; - // Posted notification's "what" field will carry one of the following: enum { + kWhatStreamsChanged, + kWhatError, kWhatPrepared, kWhatPreparationFailed, }; @@ -67,23 +77,30 @@ protected: virtual void onMessageReceived(const sp<AMessage> &msg); private: - enum { - kMaxNumQueuedFragments = 3, - kMaxNumRetries = 5, - }; + friend struct PlaylistFetcher; enum { - kWhatConnect = 'conn', - kWhatDisconnect = 'disc', - kWhatMonitorQueue = 'moni', - kWhatSeek = 'seek', + kWhatConnect = 'conn', + kWhatDisconnect = 'disc', + kWhatSeek = 'seek', + kWhatFetcherNotify = 'notf', + kWhatCheckBandwidth = 'bndw', + kWhatChangeConfiguration2 = 'chC2', + kWhatChangeConfiguration3 = 'chC3', + kWhatFinishDisconnect2 = 'fin2', }; struct BandwidthItem { - AString mURI; + size_t mPlaylistIndex; unsigned long mBandwidth; }; + struct FetcherInfo { + sp<PlaylistFetcher> mFetcher; + int64_t mDurationUs; + bool mIsPrepared; + }; + sp<AMessage> mNotify; uint32_t mFlags; bool mUIDValid; @@ -91,71 +108,61 @@ private: bool mInPreparationPhase; - sp<LiveDataSource> mDataSource; - sp<HTTPBase> mHTTPDataSource; + KeyedVector<String8, String8> mExtraHeaders; AString mMasterURL; - KeyedVector<String8, String8> mExtraHeaders; Vector<BandwidthItem> mBandwidthItems; - - KeyedVector<AString, sp<ABuffer> > mAESKeyForURI; - ssize_t mPrevBandwidthIndex; - int64_t mLastPlaylistFetchTimeUs; + sp<M3UParser> mPlaylist; - int32_t mSeqNumber; - int64_t mSeekTimeUs; - int32_t mNumRetries; - bool mStartOfPlayback; - - mutable Mutex mLock; - Condition mCondition; - int64_t mDurationUs; - bool mDurationFixed; // Duration has been determined once and for all. - bool mSeekDone; - bool mDisconnectPending; - - int32_t mMonitorQueueGeneration; - - enum RefreshState { - INITIAL_MINIMUM_RELOAD_DELAY, - FIRST_UNCHANGED_RELOAD_ATTEMPT, - SECOND_UNCHANGED_RELOAD_ATTEMPT, - THIRD_UNCHANGED_RELOAD_ATTEMPT - }; - RefreshState mRefreshState; - uint8_t mPlaylistHash[16]; + KeyedVector<AString, FetcherInfo> mFetcherInfos; + AString mAudioURI, mVideoURI, mSubtitleURI; + uint32_t mStreamMask; + + KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources; + + int32_t mCheckBandwidthGeneration; + + size_t mContinuationCounter; + sp<AMessage> mContinuation; + + int64_t mLastDequeuedTimeUs; + + bool mReconfigurationInProgress; + uint32_t mDisconnectReplyID; + + sp<PlaylistFetcher> addFetcher(const char *uri); void onConnect(const sp<AMessage> &msg); - void onDisconnect(); - void onDownloadNext(); - void onMonitorQueue(); - void onSeek(const sp<AMessage> &msg); + status_t onSeek(const sp<AMessage> &msg); + void onFinishDisconnect2(); status_t fetchFile( const char *url, sp<ABuffer> *out, int64_t range_offset = 0, int64_t range_length = -1); - sp<M3UParser> fetchPlaylist(const char *url, bool *unchanged); + sp<M3UParser> fetchPlaylist( + const char *url, uint8_t *curPlaylistHash, bool *unchanged); + size_t getBandwidthIndex(); - status_t decryptBuffer( - size_t playlistIndex, const sp<ABuffer> &buffer); + static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *); - void postMonitorQueue(int64_t delayUs = 0); + void changeConfiguration(int64_t timeUs, size_t bandwidthIndex); + void onChangeConfiguration2(const sp<AMessage> &msg); + void onChangeConfiguration3(const sp<AMessage> &msg); - bool timeToRefreshPlaylist(int64_t nowUs) const; + void scheduleCheckBandwidthEvent(); + void cancelCheckBandwidthEvent(); - static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *); + void onCheckBandwidth(); - // Returns the media time in us of the segment specified by seqNumber. - // This is computed by summing the durations of all segments before it. - int64_t getSegmentStartTimeUs(int32_t seqNumber) const; + void finishDisconnect(); - void signalEOS(status_t err); + void postPrepared(status_t err); DISALLOW_EVIL_CONSTRUCTORS(LiveSession); }; diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp index 68bbca2..be66252 100644 --- a/media/libstagefright/httplive/M3UParser.cpp +++ b/media/libstagefright/httplive/M3UParser.cpp @@ -18,14 +18,153 @@ #define LOG_TAG "M3UParser" #include <utils/Log.h> -#include "include/M3UParser.h" +#include "M3UParser.h" +#include <cutils/properties.h> #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/foundation/AMessage.h> #include <media/stagefright/MediaErrors.h> namespace android { +struct M3UParser::MediaGroup : public RefBase { + enum Type { + TYPE_AUDIO, + TYPE_VIDEO, + TYPE_SUBS, + }; + + enum FlagBits { + FLAG_AUTOSELECT = 1, + FLAG_DEFAULT = 2, + FLAG_FORCED = 4, + FLAG_HAS_LANGUAGE = 8, + FLAG_HAS_URI = 16, + }; + + MediaGroup(Type type); + + Type type() const; + + status_t addMedia( + const char *name, + const char *uri, + const char *language, + uint32_t flags); + + bool getActiveURI(AString *uri) const; + + void pickRandomMediaItems(); + +protected: + virtual ~MediaGroup(); + +private: + struct Media { + AString mName; + AString mURI; + AString mLanguage; + uint32_t mFlags; + }; + + Type mType; + Vector<Media> mMediaItems; + + ssize_t mSelectedIndex; + + DISALLOW_EVIL_CONSTRUCTORS(MediaGroup); +}; + +M3UParser::MediaGroup::MediaGroup(Type type) + : mType(type), + mSelectedIndex(-1) { +} + +M3UParser::MediaGroup::~MediaGroup() { +} + +M3UParser::MediaGroup::Type M3UParser::MediaGroup::type() const { + return mType; +} + +status_t M3UParser::MediaGroup::addMedia( + const char *name, + const char *uri, + const char *language, + uint32_t flags) { + mMediaItems.push(); + Media &item = mMediaItems.editItemAt(mMediaItems.size() - 1); + + item.mName = name; + + if (uri) { + item.mURI = uri; + } + + if (language) { + item.mLanguage = language; + } + + item.mFlags = flags; + + return OK; +} + +void M3UParser::MediaGroup::pickRandomMediaItems() { +#if 1 + switch (mType) { + case TYPE_AUDIO: + { + char value[PROPERTY_VALUE_MAX]; + if (property_get("media.httplive.audio-index", value, NULL)) { + char *end; + mSelectedIndex = strtoul(value, &end, 10); + CHECK(end > value && *end == '\0'); + + if (mSelectedIndex >= mMediaItems.size()) { + mSelectedIndex = mMediaItems.size() - 1; + } + } else { + mSelectedIndex = 0; + } + break; + } + + case TYPE_VIDEO: + { + mSelectedIndex = 0; + break; + } + + case TYPE_SUBS: + { + mSelectedIndex = -1; + break; + } + + default: + TRESPASS(); + } +#else + mSelectedIndex = (rand() * mMediaItems.size()) / RAND_MAX; +#endif +} + +bool M3UParser::MediaGroup::getActiveURI(AString *uri) const { + for (size_t i = 0; i < mMediaItems.size(); ++i) { + if (mSelectedIndex >= 0 && i == (size_t)mSelectedIndex) { + const Media &item = mMediaItems.itemAt(i); + + *uri = item.mURI; + return true; + } + } + + return false; +} + +//////////////////////////////////////////////////////////////////////////////// + M3UParser::M3UParser( const char *baseURI, const void *data, size_t size) : mInitCheck(NO_INIT), @@ -92,6 +231,58 @@ bool M3UParser::itemAt(size_t index, AString *uri, sp<AMessage> *meta) { return true; } +void M3UParser::pickRandomMediaItems() { + for (size_t i = 0; i < mMediaGroups.size(); ++i) { + mMediaGroups.valueAt(i)->pickRandomMediaItems(); + } +} + +bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const { + if (!mIsVariantPlaylist) { + *uri = mBaseURI; + + // Assume media without any more specific attribute contains + // audio and video, but no subtitles. + return !strcmp("audio", key) || !strcmp("video", key); + } + + CHECK_LT(index, mItems.size()); + + sp<AMessage> meta = mItems.itemAt(index).mMeta; + + AString groupID; + if (!meta->findString(key, &groupID)) { + *uri = mItems.itemAt(index).mURI; + + // Assume media without any more specific attribute contains + // audio and video, but no subtitles. + return !strcmp("audio", key) || !strcmp("video", key); + } + + sp<MediaGroup> group = mMediaGroups.valueFor(groupID); + if (!group->getActiveURI(uri)) { + return false; + } + + if ((*uri).empty()) { + *uri = mItems.itemAt(index).mURI; + } + + return true; +} + +bool M3UParser::getAudioURI(size_t index, AString *uri) const { + return getTypeURI(index, "audio", uri); +} + +bool M3UParser::getVideoURI(size_t index, AString *uri) const { + return getTypeURI(index, "video", uri); +} + +bool M3UParser::getSubtitleURI(size_t index, AString *uri) const { + return getTypeURI(index, "subtitles", uri); +} + static bool MakeURL(const char *baseURL, const char *url, AString *out) { out->clear(); @@ -241,6 +432,8 @@ status_t M3UParser::parse(const void *_data, size_t size) { segmentRangeOffset = offset + length; } + } else if (line.startsWith("#EXT-X-MEDIA")) { + err = parseMedia(line); } if (err != OK) { @@ -322,9 +515,31 @@ status_t M3UParser::parseMetaDataDuration( return OK; } -// static +// Find the next occurence of the character "what" at or after "offset", +// but ignore occurences between quotation marks. +// Return the index of the occurrence or -1 if not found. +static ssize_t FindNextUnquoted( + const AString &line, char what, size_t offset) { + CHECK_NE((int)what, (int)'"'); + + bool quoted = false; + while (offset < line.size()) { + char c = line.c_str()[offset]; + + if (c == '"') { + quoted = !quoted; + } else if (c == what && !quoted) { + return offset; + } + + ++offset; + } + + return -1; +} + status_t M3UParser::parseStreamInf( - const AString &line, sp<AMessage> *meta) { + const AString &line, sp<AMessage> *meta) const { ssize_t colonPos = line.find(":"); if (colonPos < 0) { @@ -334,7 +549,7 @@ status_t M3UParser::parseStreamInf( size_t offset = colonPos + 1; while (offset < line.size()) { - ssize_t end = line.find(",", offset); + ssize_t end = FindNextUnquoted(line, ',', offset); if (end < 0) { end = line.size(); } @@ -371,33 +586,35 @@ status_t M3UParser::parseStreamInf( *meta = new AMessage; } (*meta)->setInt32("bandwidth", x); - } - } + } else if (!strcasecmp("audio", key.c_str()) + || !strcasecmp("video", key.c_str()) + || !strcasecmp("subtitles", key.c_str())) { + if (val.size() < 2 + || val.c_str()[0] != '"' + || val.c_str()[val.size() - 1] != '"') { + ALOGE("Expected quoted string for %s attribute, " + "got '%s' instead.", + key.c_str(), val.c_str()); + + return ERROR_MALFORMED; + } - return OK; -} + AString groupID(val, 1, val.size() - 2); + ssize_t groupIndex = mMediaGroups.indexOfKey(groupID); -// Find the next occurence of the character "what" at or after "offset", -// but ignore occurences between quotation marks. -// Return the index of the occurrence or -1 if not found. -static ssize_t FindNextUnquoted( - const AString &line, char what, size_t offset) { - CHECK_NE((int)what, (int)'"'); + if (groupIndex < 0) { + ALOGE("Undefined media group '%s' referenced in stream info.", + groupID.c_str()); - bool quoted = false; - while (offset < line.size()) { - char c = line.c_str()[offset]; + return ERROR_MALFORMED; + } - if (c == '"') { - quoted = !quoted; - } else if (c == what && !quoted) { - return offset; + key.tolower(); + (*meta)->setString(key.c_str(), groupID.c_str()); } - - ++offset; } - return -1; + return OK; } // static @@ -515,6 +732,234 @@ status_t M3UParser::parseByteRange( return OK; } +status_t M3UParser::parseMedia(const AString &line) { + ssize_t colonPos = line.find(":"); + + if (colonPos < 0) { + return ERROR_MALFORMED; + } + + bool haveGroupType = false; + MediaGroup::Type groupType = MediaGroup::TYPE_AUDIO; + + bool haveGroupID = false; + AString groupID; + + bool haveGroupLanguage = false; + AString groupLanguage; + + bool haveGroupName = false; + AString groupName; + + bool haveGroupAutoselect = false; + bool groupAutoselect = false; + + bool haveGroupDefault = false; + bool groupDefault = false; + + bool haveGroupForced = false; + bool groupForced = false; + + bool haveGroupURI = false; + AString groupURI; + + size_t offset = colonPos + 1; + + while (offset < line.size()) { + ssize_t end = FindNextUnquoted(line, ',', offset); + if (end < 0) { + end = line.size(); + } + + AString attr(line, offset, end - offset); + attr.trim(); + + offset = end + 1; + + ssize_t equalPos = attr.find("="); + if (equalPos < 0) { + continue; + } + + AString key(attr, 0, equalPos); + key.trim(); + + AString val(attr, equalPos + 1, attr.size() - equalPos - 1); + val.trim(); + + ALOGV("key=%s value=%s", key.c_str(), val.c_str()); + + if (!strcasecmp("type", key.c_str())) { + if (!strcasecmp("subtitles", val.c_str())) { + groupType = MediaGroup::TYPE_SUBS; + } else if (!strcasecmp("audio", val.c_str())) { + groupType = MediaGroup::TYPE_AUDIO; + } else if (!strcasecmp("video", val.c_str())) { + groupType = MediaGroup::TYPE_VIDEO; + } else { + ALOGE("Invalid media group type '%s'", val.c_str()); + return ERROR_MALFORMED; + } + + haveGroupType = true; + } else if (!strcasecmp("group-id", key.c_str())) { + if (val.size() < 2 + || val.c_str()[0] != '"' + || val.c_str()[val.size() - 1] != '"') { + ALOGE("Expected quoted string for GROUP-ID, got '%s' instead.", + val.c_str()); + + return ERROR_MALFORMED; + } + + groupID.setTo(val, 1, val.size() - 2); + haveGroupID = true; + } else if (!strcasecmp("language", key.c_str())) { + if (val.size() < 2 + || val.c_str()[0] != '"' + || val.c_str()[val.size() - 1] != '"') { + ALOGE("Expected quoted string for LANGUAGE, got '%s' instead.", + val.c_str()); + + return ERROR_MALFORMED; + } + + groupLanguage.setTo(val, 1, val.size() - 2); + haveGroupLanguage = true; + } else if (!strcasecmp("name", key.c_str())) { + if (val.size() < 2 + || val.c_str()[0] != '"' + || val.c_str()[val.size() - 1] != '"') { + ALOGE("Expected quoted string for NAME, got '%s' instead.", + val.c_str()); + + return ERROR_MALFORMED; + } + + groupName.setTo(val, 1, val.size() - 2); + haveGroupName = true; + } else if (!strcasecmp("autoselect", key.c_str())) { + groupAutoselect = false; + if (!strcasecmp("YES", val.c_str())) { + groupAutoselect = true; + } else if (!strcasecmp("NO", val.c_str())) { + groupAutoselect = false; + } else { + ALOGE("Expected YES or NO for AUTOSELECT attribute, " + "got '%s' instead.", + val.c_str()); + + return ERROR_MALFORMED; + } + + haveGroupAutoselect = true; + } else if (!strcasecmp("default", key.c_str())) { + groupDefault = false; + if (!strcasecmp("YES", val.c_str())) { + groupDefault = true; + } else if (!strcasecmp("NO", val.c_str())) { + groupDefault = false; + } else { + ALOGE("Expected YES or NO for DEFAULT attribute, " + "got '%s' instead.", + val.c_str()); + + return ERROR_MALFORMED; + } + + haveGroupDefault = true; + } else if (!strcasecmp("forced", key.c_str())) { + groupForced = false; + if (!strcasecmp("YES", val.c_str())) { + groupForced = true; + } else if (!strcasecmp("NO", val.c_str())) { + groupForced = false; + } else { + ALOGE("Expected YES or NO for FORCED attribute, " + "got '%s' instead.", + val.c_str()); + + return ERROR_MALFORMED; + } + + haveGroupForced = true; + } else if (!strcasecmp("uri", key.c_str())) { + if (val.size() < 2 + || val.c_str()[0] != '"' + || val.c_str()[val.size() - 1] != '"') { + ALOGE("Expected quoted string for URI, got '%s' instead.", + val.c_str()); + + return ERROR_MALFORMED; + } + + AString tmp(val, 1, val.size() - 2); + + if (!MakeURL(mBaseURI.c_str(), tmp.c_str(), &groupURI)) { + ALOGI("Failed to make absolute URI from '%s'.", tmp.c_str()); + } + + haveGroupURI = true; + } + } + + if (!haveGroupType || !haveGroupID || !haveGroupName) { + ALOGE("Incomplete EXT-X-MEDIA element."); + return ERROR_MALFORMED; + } + + uint32_t flags = 0; + if (haveGroupAutoselect && groupAutoselect) { + flags |= MediaGroup::FLAG_AUTOSELECT; + } + if (haveGroupDefault && groupDefault) { + flags |= MediaGroup::FLAG_DEFAULT; + } + if (haveGroupForced) { + if (groupType != MediaGroup::TYPE_SUBS) { + ALOGE("The FORCED attribute MUST not be present on anything " + "but SUBS media."); + + return ERROR_MALFORMED; + } + + if (groupForced) { + flags |= MediaGroup::FLAG_FORCED; + } + } + if (haveGroupLanguage) { + flags |= MediaGroup::FLAG_HAS_LANGUAGE; + } + if (haveGroupURI) { + flags |= MediaGroup::FLAG_HAS_URI; + } + + ssize_t groupIndex = mMediaGroups.indexOfKey(groupID); + sp<MediaGroup> group; + + if (groupIndex < 0) { + group = new MediaGroup(groupType); + mMediaGroups.add(groupID, group); + } else { + group = mMediaGroups.valueAt(groupIndex); + + if (group->type() != groupType) { + ALOGE("Attempt to put media item under group of different type " + "(groupType = %d, item type = %d", + group->type(), + groupType); + + return ERROR_MALFORMED; + } + } + + return group->addMedia( + groupName.c_str(), + haveGroupURI ? groupURI.c_str() : NULL, + haveGroupLanguage ? groupLanguage.c_str() : NULL, + flags); +} + // static status_t M3UParser::ParseInt32(const char *s, int32_t *x) { char *end; diff --git a/media/libstagefright/include/M3UParser.h b/media/libstagefright/httplive/M3UParser.h index 2d2f50f..abea286 100644 --- a/media/libstagefright/include/M3UParser.h +++ b/media/libstagefright/httplive/M3UParser.h @@ -40,10 +40,18 @@ struct M3UParser : public RefBase { size_t size(); bool itemAt(size_t index, AString *uri, sp<AMessage> *meta = NULL); + void pickRandomMediaItems(); + + bool getAudioURI(size_t index, AString *uri) const; + bool getVideoURI(size_t index, AString *uri) const; + bool getSubtitleURI(size_t index, AString *uri) const; + protected: virtual ~M3UParser(); private: + struct MediaGroup; + struct Item { AString mURI; sp<AMessage> mMeta; @@ -60,6 +68,9 @@ private: sp<AMessage> mMeta; Vector<Item> mItems; + // Media groups keyed by group ID. + KeyedVector<AString, sp<MediaGroup> > mMediaGroups; + status_t parse(const void *data, size_t size); static status_t parseMetaData( @@ -68,8 +79,8 @@ private: static status_t parseMetaDataDuration( const AString &line, sp<AMessage> *meta, const char *key); - static status_t parseStreamInf( - const AString &line, sp<AMessage> *meta); + status_t parseStreamInf( + const AString &line, sp<AMessage> *meta) const; static status_t parseCipherInfo( const AString &line, sp<AMessage> *meta, const AString &baseURI); @@ -78,6 +89,10 @@ private: const AString &line, uint64_t curOffset, uint64_t *length, uint64_t *offset); + status_t parseMedia(const AString &line); + + bool getTypeURI(size_t index, const char *key, AString *uri) const; + static status_t ParseInt32(const char *s, int32_t *x); static status_t ParseDouble(const char *s, double *x); diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp new file mode 100644 index 0000000..8ae70b7 --- /dev/null +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -0,0 +1,969 @@ +/* + * Copyright (C) 2012 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "PlaylistFetcher" +#include <utils/Log.h> + +#include "PlaylistFetcher.h" + +#include "LiveDataSource.h" +#include "LiveSession.h" +#include "M3UParser.h" + +#include "include/avc_utils.h" +#include "include/HTTPBase.h" +#include "include/ID3.h" +#include "mpeg2ts/AnotherPacketSource.h" + +#include <media/IStreamSource.h> +#include <media/stagefright/foundation/ABitReader.h> +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/hexdump.h> +#include <media/stagefright/FileSource.h> +#include <media/stagefright/MediaDefs.h> +#include <media/stagefright/MetaData.h> +#include <media/stagefright/Utils.h> + +#include <ctype.h> +#include <openssl/aes.h> +#include <openssl/md5.h> + +namespace android { + +// static +const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll; + +PlaylistFetcher::PlaylistFetcher( + const sp<AMessage> ¬ify, + const sp<LiveSession> &session, + const char *uri) + : mNotify(notify), + mSession(session), + mURI(uri), + mStreamTypeMask(0), + mStartTimeUs(-1ll), + mLastPlaylistFetchTimeUs(-1ll), + mSeqNumber(-1), + mNumRetries(0), + mStartup(true), + mNextPTSTimeUs(-1ll), + mMonitorQueueGeneration(0), + mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY), + mFirstPTSValid(false), + mAbsoluteTimeAnchorUs(0ll) { + memset(mPlaylistHash, 0, sizeof(mPlaylistHash)); +} + +PlaylistFetcher::~PlaylistFetcher() { +} + +int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const { + CHECK(mPlaylist != NULL); + + int32_t firstSeqNumberInPlaylist; + if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( + "media-sequence", &firstSeqNumberInPlaylist)) { + firstSeqNumberInPlaylist = 0; + } + + int32_t lastSeqNumberInPlaylist = + firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; + + CHECK_GE(seqNumber, firstSeqNumberInPlaylist); + CHECK_LE(seqNumber, lastSeqNumberInPlaylist); + + int64_t segmentStartUs = 0ll; + for (int32_t index = 0; + index < seqNumber - firstSeqNumberInPlaylist; ++index) { + sp<AMessage> itemMeta; + CHECK(mPlaylist->itemAt( + index, NULL /* uri */, &itemMeta)); + + int64_t itemDurationUs; + CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); + + segmentStartUs += itemDurationUs; + } + + return segmentStartUs; +} + +bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const { + if (mPlaylist == NULL) { + CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY); + return true; + } + + int32_t targetDurationSecs; + CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); + + int64_t targetDurationUs = targetDurationSecs * 1000000ll; + + int64_t minPlaylistAgeUs; + + switch (mRefreshState) { + case INITIAL_MINIMUM_RELOAD_DELAY: + { + size_t n = mPlaylist->size(); + if (n > 0) { + sp<AMessage> itemMeta; + CHECK(mPlaylist->itemAt(n - 1, NULL /* uri */, &itemMeta)); + + int64_t itemDurationUs; + CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); + + minPlaylistAgeUs = itemDurationUs; + break; + } + + // fall through + } + + case FIRST_UNCHANGED_RELOAD_ATTEMPT: + { + minPlaylistAgeUs = targetDurationUs / 2; + break; + } + + case SECOND_UNCHANGED_RELOAD_ATTEMPT: + { + minPlaylistAgeUs = (targetDurationUs * 3) / 2; + break; + } + + case THIRD_UNCHANGED_RELOAD_ATTEMPT: + { + minPlaylistAgeUs = targetDurationUs * 3; + break; + } + + default: + TRESPASS(); + break; + } + + return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs; +} + +status_t PlaylistFetcher::decryptBuffer( + size_t playlistIndex, const sp<ABuffer> &buffer) { + sp<AMessage> itemMeta; + bool found = false; + AString method; + + for (ssize_t i = playlistIndex; i >= 0; --i) { + AString uri; + CHECK(mPlaylist->itemAt(i, &uri, &itemMeta)); + + if (itemMeta->findString("cipher-method", &method)) { + found = true; + break; + } + } + + if (!found) { + method = "NONE"; + } + + if (method == "NONE") { + return OK; + } else if (!(method == "AES-128")) { + ALOGE("Unsupported cipher method '%s'", method.c_str()); + return ERROR_UNSUPPORTED; + } + + AString keyURI; + if (!itemMeta->findString("cipher-uri", &keyURI)) { + ALOGE("Missing key uri"); + return ERROR_MALFORMED; + } + + ssize_t index = mAESKeyForURI.indexOfKey(keyURI); + + sp<ABuffer> key; + if (index >= 0) { + key = mAESKeyForURI.valueAt(index); + } else { + status_t err = mSession->fetchFile(keyURI.c_str(), &key); + + if (err != OK) { + ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str()); + return ERROR_IO; + } else if (key->size() != 16) { + ALOGE("key file '%s' wasn't 16 bytes in size.", keyURI.c_str()); + return ERROR_MALFORMED; + } + + mAESKeyForURI.add(keyURI, key); + } + + AES_KEY aes_key; + if (AES_set_decrypt_key(key->data(), 128, &aes_key) != 0) { + ALOGE("failed to set AES decryption key."); + return UNKNOWN_ERROR; + } + + unsigned char aes_ivec[16]; + + AString iv; + if (itemMeta->findString("cipher-iv", &iv)) { + if ((!iv.startsWith("0x") && !iv.startsWith("0X")) + || iv.size() != 16 * 2 + 2) { + ALOGE("malformed cipher IV '%s'.", iv.c_str()); + return ERROR_MALFORMED; + } + + memset(aes_ivec, 0, sizeof(aes_ivec)); + for (size_t i = 0; i < 16; ++i) { + char c1 = tolower(iv.c_str()[2 + 2 * i]); + char c2 = tolower(iv.c_str()[3 + 2 * i]); + if (!isxdigit(c1) || !isxdigit(c2)) { + ALOGE("malformed cipher IV '%s'.", iv.c_str()); + return ERROR_MALFORMED; + } + uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10; + uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10; + + aes_ivec[i] = nibble1 << 4 | nibble2; + } + } else { + memset(aes_ivec, 0, sizeof(aes_ivec)); + aes_ivec[15] = mSeqNumber & 0xff; + aes_ivec[14] = (mSeqNumber >> 8) & 0xff; + aes_ivec[13] = (mSeqNumber >> 16) & 0xff; + aes_ivec[12] = (mSeqNumber >> 24) & 0xff; + } + + AES_cbc_encrypt( + buffer->data(), buffer->data(), buffer->size(), + &aes_key, aes_ivec, AES_DECRYPT); + + // hexdump(buffer->data(), buffer->size()); + + size_t n = buffer->size(); + CHECK_GT(n, 0u); + + size_t pad = buffer->data()[n - 1]; + + CHECK_GT(pad, 0u); + CHECK_LE(pad, 16u); + CHECK_GE((size_t)n, pad); + for (size_t i = 0; i < pad; ++i) { + CHECK_EQ((unsigned)buffer->data()[n - 1 - i], pad); + } + + n -= pad; + + buffer->setRange(buffer->offset(), n); + + return OK; +} + +void PlaylistFetcher::postMonitorQueue(int64_t delayUs) { + sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id()); + msg->setInt32("generation", mMonitorQueueGeneration); + msg->post(delayUs); +} + +void PlaylistFetcher::cancelMonitorQueue() { + ++mMonitorQueueGeneration; +} + +void PlaylistFetcher::startAsync( + const sp<AnotherPacketSource> &audioSource, + const sp<AnotherPacketSource> &videoSource, + const sp<AnotherPacketSource> &subtitleSource, + int64_t startTimeUs) { + sp<AMessage> msg = new AMessage(kWhatStart, id()); + + uint32_t streamTypeMask = 0ul; + + if (audioSource != NULL) { + msg->setPointer("audioSource", audioSource.get()); + streamTypeMask |= LiveSession::STREAMTYPE_AUDIO; + } + + if (videoSource != NULL) { + msg->setPointer("videoSource", videoSource.get()); + streamTypeMask |= LiveSession::STREAMTYPE_VIDEO; + } + + if (subtitleSource != NULL) { + msg->setPointer("subtitleSource", subtitleSource.get()); + streamTypeMask |= LiveSession::STREAMTYPE_SUBTITLES; + } + + msg->setInt32("streamTypeMask", streamTypeMask); + msg->setInt64("startTimeUs", startTimeUs); + msg->post(); +} + +void PlaylistFetcher::pauseAsync() { + (new AMessage(kWhatPause, id()))->post(); +} + +void PlaylistFetcher::stopAsync() { + (new AMessage(kWhatStop, id()))->post(); +} + +void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatStart: + { + status_t err = onStart(msg); + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatStarted); + notify->setInt32("err", err); + notify->post(); + break; + } + + case kWhatPause: + { + onPause(); + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatPaused); + notify->post(); + break; + } + + case kWhatStop: + { + onStop(); + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatStopped); + notify->post(); + break; + } + + case kWhatMonitorQueue: + { + int32_t generation; + CHECK(msg->findInt32("generation", &generation)); + + if (generation != mMonitorQueueGeneration) { + // Stale event + break; + } + + onMonitorQueue(); + break; + } + + default: + TRESPASS(); + } +} + +status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { + mPacketSources.clear(); + + uint32_t streamTypeMask; + CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask)); + + int64_t startTimeUs; + CHECK(msg->findInt64("startTimeUs", &startTimeUs)); + + if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) { + void *ptr; + CHECK(msg->findPointer("audioSource", &ptr)); + + mPacketSources.add( + LiveSession::STREAMTYPE_AUDIO, + static_cast<AnotherPacketSource *>(ptr)); + } + + if (streamTypeMask & LiveSession::STREAMTYPE_VIDEO) { + void *ptr; + CHECK(msg->findPointer("videoSource", &ptr)); + + mPacketSources.add( + LiveSession::STREAMTYPE_VIDEO, + static_cast<AnotherPacketSource *>(ptr)); + } + + if (streamTypeMask & LiveSession::STREAMTYPE_SUBTITLES) { + void *ptr; + CHECK(msg->findPointer("subtitleSource", &ptr)); + + mPacketSources.add( + LiveSession::STREAMTYPE_SUBTITLES, + static_cast<AnotherPacketSource *>(ptr)); + } + + mStreamTypeMask = streamTypeMask; + mStartTimeUs = startTimeUs; + + if (mStartTimeUs >= 0ll) { + mSeqNumber = -1; + mStartup = true; + } + + postMonitorQueue(); + + return OK; +} + +void PlaylistFetcher::onPause() { + cancelMonitorQueue(); + + mPacketSources.clear(); + mStreamTypeMask = 0; +} + +void PlaylistFetcher::onStop() { + cancelMonitorQueue(); + + for (size_t i = 0; i < mPacketSources.size(); ++i) { + mPacketSources.valueAt(i)->clear(); + } + + mPacketSources.clear(); + mStreamTypeMask = 0; +} + +void PlaylistFetcher::notifyError(status_t err) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatError); + notify->setInt32("err", err); + notify->post(); +} + +void PlaylistFetcher::queueDiscontinuity( + ATSParser::DiscontinuityType type, const sp<AMessage> &extra) { + for (size_t i = 0; i < mPacketSources.size(); ++i) { + mPacketSources.valueAt(i)->queueDiscontinuity(type, extra); + } +} + +void PlaylistFetcher::onMonitorQueue() { + bool downloadMore = false; + + status_t finalResult; + if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) { + sp<AnotherPacketSource> packetSource = + mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES); + + downloadMore = packetSource->hasBufferAvailable(&finalResult); + } else { + bool first = true; + int64_t minBufferedDurationUs = 0ll; + + for (size_t i = 0; i < mPacketSources.size(); ++i) { + if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) { + continue; + } + + int64_t bufferedDurationUs = + mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult); + + if (first || bufferedDurationUs < minBufferedDurationUs) { + minBufferedDurationUs = bufferedDurationUs; + first = false; + } + } + + downloadMore = + !first && (minBufferedDurationUs < kMinBufferedDurationUs); + } + + if (finalResult == OK && downloadMore) { + onDownloadNext(); + } else { + // Nothing to do yet, try again in a second. + + sp<AMessage> msg = mNotify->dup(); + msg->setInt32("what", kWhatTemporarilyDoneFetching); + msg->post(); + + postMonitorQueue(1000000ll); + } +} + +void PlaylistFetcher::onDownloadNext() { + int64_t nowUs = ALooper::GetNowUs(); + + if (mLastPlaylistFetchTimeUs < 0ll + || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) { + bool unchanged; + sp<M3UParser> playlist = mSession->fetchPlaylist( + mURI.c_str(), mPlaylistHash, &unchanged); + + if (playlist == NULL) { + if (unchanged) { + // We succeeded in fetching the playlist, but it was + // unchanged from the last time we tried. + + if (mRefreshState != THIRD_UNCHANGED_RELOAD_ATTEMPT) { + mRefreshState = (RefreshState)(mRefreshState + 1); + } + } else { + ALOGE("failed to load playlist at url '%s'", mURI.c_str()); + notifyError(ERROR_IO); + return; + } + } else { + mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY; + mPlaylist = playlist; + + if (mPlaylist->isComplete() || mPlaylist->isEvent()) { + updateDuration(); + } + } + + mLastPlaylistFetchTimeUs = ALooper::GetNowUs(); + } + + int32_t firstSeqNumberInPlaylist; + if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( + "media-sequence", &firstSeqNumberInPlaylist)) { + firstSeqNumberInPlaylist = 0; + } + + bool seekDiscontinuity = false; + bool explicitDiscontinuity = false; + + const int32_t lastSeqNumberInPlaylist = + firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; + + if (mSeqNumber < 0) { + CHECK_GE(mStartTimeUs, 0ll); + + if (mPlaylist->isComplete() || mPlaylist->isEvent()) { + mSeqNumber = getSeqNumberForTime(mStartTimeUs); + } else { + // If this is a live session, start 3 segments from the end. + mSeqNumber = lastSeqNumberInPlaylist - 3; + if (mSeqNumber < firstSeqNumberInPlaylist) { + mSeqNumber = firstSeqNumberInPlaylist; + } + } + + mStartTimeUs = -1ll; + } + + if (mSeqNumber < firstSeqNumberInPlaylist + || mSeqNumber > lastSeqNumberInPlaylist) { + if (!mPlaylist->isComplete() && mNumRetries < kMaxNumRetries) { + ++mNumRetries; + + if (mSeqNumber > lastSeqNumberInPlaylist) { + mLastPlaylistFetchTimeUs = -1; + postMonitorQueue(3000000ll); + return; + } + + // we've missed the boat, let's start from the lowest sequence + // number available and signal a discontinuity. + + ALOGI("We've missed the boat, restarting playback."); + mSeqNumber = lastSeqNumberInPlaylist; + explicitDiscontinuity = true; + + // fall through + } else { + ALOGE("Cannot find sequence number %d in playlist " + "(contains %d - %d)", + mSeqNumber, firstSeqNumberInPlaylist, + firstSeqNumberInPlaylist + mPlaylist->size() - 1); + + notifyError(ERROR_END_OF_STREAM); + return; + } + } + + mNumRetries = 0; + + AString uri; + sp<AMessage> itemMeta; + CHECK(mPlaylist->itemAt( + mSeqNumber - firstSeqNumberInPlaylist, + &uri, + &itemMeta)); + + int32_t val; + if (itemMeta->findInt32("discontinuity", &val) && val != 0) { + explicitDiscontinuity = true; + } + + int64_t range_offset, range_length; + if (!itemMeta->findInt64("range-offset", &range_offset) + || !itemMeta->findInt64("range-length", &range_length)) { + range_offset = 0; + range_length = -1; + } + + ALOGV("fetching segment %d from (%d .. %d)", + mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist); + + ALOGV("fetching '%s'", uri.c_str()); + + sp<ABuffer> buffer; + status_t err = mSession->fetchFile( + uri.c_str(), &buffer, range_offset, range_length); + + if (err != OK) { + ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str()); + notifyError(err); + return; + } + + CHECK(buffer != NULL); + + err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer); + + if (err != OK) { + ALOGE("decryptBuffer failed w/ error %d", err); + + notifyError(err); + return; + } + + if (mStartup || seekDiscontinuity || explicitDiscontinuity) { + // Signal discontinuity. + + if (mPlaylist->isComplete() || mPlaylist->isEvent()) { + // If this was a live event this made no sense since + // we don't have access to all the segment before the current + // one. + mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber); + } + + if (seekDiscontinuity || explicitDiscontinuity) { + ALOGI("queueing discontinuity (seek=%d, explicit=%d)", + seekDiscontinuity, explicitDiscontinuity); + + queueDiscontinuity( + explicitDiscontinuity + ? ATSParser::DISCONTINUITY_FORMATCHANGE + : ATSParser::DISCONTINUITY_SEEK, + NULL /* extra */); + } + } + + err = extractAndQueueAccessUnits(buffer); + + if (err != OK) { + notifyError(err); + return; + } + + ++mSeqNumber; + + postMonitorQueue(); + + mStartup = false; +} + +int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const { + int32_t firstSeqNumberInPlaylist; + if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( + "media-sequence", &firstSeqNumberInPlaylist)) { + firstSeqNumberInPlaylist = 0; + } + + size_t index = 0; + int64_t segmentStartUs = 0; + while (index < mPlaylist->size()) { + sp<AMessage> itemMeta; + CHECK(mPlaylist->itemAt( + index, NULL /* uri */, &itemMeta)); + + int64_t itemDurationUs; + CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); + + if (timeUs < segmentStartUs + itemDurationUs) { + break; + } + + segmentStartUs += itemDurationUs; + ++index; + } + + if (index >= mPlaylist->size()) { + index = mPlaylist->size() - 1; + } + + return firstSeqNumberInPlaylist + index; +} + +status_t PlaylistFetcher::extractAndQueueAccessUnits( + const sp<ABuffer> &buffer) { + if (buffer->size() > 0 && buffer->data()[0] == 0x47) { + // Let's assume this is an MPEG2 transport stream. + + if ((buffer->size() % 188) != 0) { + ALOGE("MPEG2 transport stream is not an even multiple of 188 " + "bytes in length."); + return ERROR_MALFORMED; + } + + if (mTSParser == NULL) { + mTSParser = new ATSParser; + } + + if (mNextPTSTimeUs >= 0ll) { + sp<AMessage> extra = new AMessage; + extra->setInt64(IStreamListener::kKeyMediaTimeUs, mNextPTSTimeUs); + + mTSParser->signalDiscontinuity( + ATSParser::DISCONTINUITY_SEEK, extra); + + mNextPTSTimeUs = -1ll; + } + + size_t offset = 0; + while (offset < buffer->size()) { + status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188); + + if (err != OK) { + return err; + } + + offset += 188; + } + + for (size_t i = mPacketSources.size(); i-- > 0;) { + sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); + + ATSParser::SourceType type; + switch (mPacketSources.keyAt(i)) { + case LiveSession::STREAMTYPE_VIDEO: + type = ATSParser::VIDEO; + break; + + case LiveSession::STREAMTYPE_AUDIO: + type = ATSParser::AUDIO; + break; + + case LiveSession::STREAMTYPE_SUBTITLES: + { + ALOGE("MPEG2 Transport streams do not contain subtitles."); + return ERROR_MALFORMED; + break; + } + + default: + TRESPASS(); + } + + sp<AnotherPacketSource> source = + static_cast<AnotherPacketSource *>( + mTSParser->getSource(type).get()); + + if (source == NULL) { + ALOGW("MPEG2 Transport stream does not contain %s data.", + type == ATSParser::VIDEO ? "video" : "audio"); + + mStreamTypeMask &= ~mPacketSources.keyAt(i); + mPacketSources.removeItemsAt(i); + continue; + } + + sp<ABuffer> accessUnit; + status_t finalResult; + while (source->hasBufferAvailable(&finalResult) + && source->dequeueAccessUnit(&accessUnit) == OK) { + // Note that we do NOT dequeue any discontinuities. + + packetSource->queueAccessUnit(accessUnit); + } + + if (packetSource->getFormat() == NULL) { + packetSource->setFormat(source->getFormat()); + } + } + + return OK; + } else if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) { + if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) { + ALOGE("This stream only contains subtitles."); + return ERROR_MALFORMED; + } + + const sp<AnotherPacketSource> packetSource = + mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES); + + buffer->meta()->setInt64("timeUs", 0ll); + + packetSource->queueAccessUnit(buffer); + return OK; + } + + if (mNextPTSTimeUs >= 0ll) { + mFirstPTSValid = false; + mAbsoluteTimeAnchorUs = mNextPTSTimeUs; + mNextPTSTimeUs = -1ll; + } + + // This better be an ISO 13818-7 (AAC) or ISO 13818-1 (MPEG) audio + // stream prefixed by an ID3 tag. + + bool firstID3Tag = true; + uint64_t PTS = 0; + + for (;;) { + // Make sure to skip all ID3 tags preceding the audio data. + // At least one must be present to provide the PTS timestamp. + + ID3 id3(buffer->data(), buffer->size(), true /* ignoreV1 */); + if (!id3.isValid()) { + if (firstID3Tag) { + ALOGE("Unable to parse ID3 tag."); + return ERROR_MALFORMED; + } else { + break; + } + } + + if (firstID3Tag) { + bool found = false; + + ID3::Iterator it(id3, "PRIV"); + while (!it.done()) { + size_t length; + const uint8_t *data = it.getData(&length); + + static const char *kMatchName = + "com.apple.streaming.transportStreamTimestamp"; + static const size_t kMatchNameLen = strlen(kMatchName); + + if (length == kMatchNameLen + 1 + 8 + && !strncmp((const char *)data, kMatchName, kMatchNameLen)) { + found = true; + PTS = U64_AT(&data[kMatchNameLen + 1]); + } + + it.next(); + } + + if (!found) { + ALOGE("Unable to extract transportStreamTimestamp from ID3 tag."); + return ERROR_MALFORMED; + } + } + + // skip the ID3 tag + buffer->setRange( + buffer->offset() + id3.rawSize(), buffer->size() - id3.rawSize()); + + firstID3Tag = false; + } + + if (!mFirstPTSValid) { + mFirstPTSValid = true; + mFirstPTS = PTS; + } + PTS -= mFirstPTS; + + int64_t timeUs = (PTS * 100ll) / 9ll + mAbsoluteTimeAnchorUs; + + if (mStreamTypeMask != LiveSession::STREAMTYPE_AUDIO) { + ALOGW("This stream only contains audio data!"); + + mStreamTypeMask &= LiveSession::STREAMTYPE_AUDIO; + + if (mStreamTypeMask == 0) { + return OK; + } + } + + sp<AnotherPacketSource> packetSource = + mPacketSources.valueFor(LiveSession::STREAMTYPE_AUDIO); + + if (packetSource->getFormat() == NULL && buffer->size() >= 7) { + ABitReader bits(buffer->data(), buffer->size()); + + // adts_fixed_header + + CHECK_EQ(bits.getBits(12), 0xfffu); + bits.skipBits(3); // ID, layer + bool protection_absent = bits.getBits(1) != 0; + + unsigned profile = bits.getBits(2); + CHECK_NE(profile, 3u); + unsigned sampling_freq_index = bits.getBits(4); + bits.getBits(1); // private_bit + unsigned channel_configuration = bits.getBits(3); + CHECK_NE(channel_configuration, 0u); + bits.skipBits(2); // original_copy, home + + sp<MetaData> meta = MakeAACCodecSpecificData( + profile, sampling_freq_index, channel_configuration); + + meta->setInt32(kKeyIsADTS, true); + + packetSource->setFormat(meta); + } + + int64_t numSamples = 0ll; + int32_t sampleRate; + CHECK(packetSource->getFormat()->findInt32(kKeySampleRate, &sampleRate)); + + size_t offset = 0; + while (offset < buffer->size()) { + const uint8_t *adtsHeader = buffer->data() + offset; + CHECK_LT(offset + 5, buffer->size()); + + unsigned aac_frame_length = + ((adtsHeader[3] & 3) << 11) + | (adtsHeader[4] << 3) + | (adtsHeader[5] >> 5); + + CHECK_LE(offset + aac_frame_length, buffer->size()); + + sp<ABuffer> unit = new ABuffer(aac_frame_length); + memcpy(unit->data(), adtsHeader, aac_frame_length); + + int64_t unitTimeUs = timeUs + numSamples * 1000000ll / sampleRate; + unit->meta()->setInt64("timeUs", unitTimeUs); + + // Each AAC frame encodes 1024 samples. + numSamples += 1024; + + packetSource->queueAccessUnit(unit); + + offset += aac_frame_length; + } + + return OK; +} + +void PlaylistFetcher::updateDuration() { + int64_t durationUs = 0ll; + for (size_t index = 0; index < mPlaylist->size(); ++index) { + sp<AMessage> itemMeta; + CHECK(mPlaylist->itemAt( + index, NULL /* uri */, &itemMeta)); + + int64_t itemDurationUs; + CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); + + durationUs += itemDurationUs; + } + + sp<AMessage> msg = mNotify->dup(); + msg->setInt32("what", kWhatDurationUpdate); + msg->setInt64("durationUs", durationUs); + msg->post(); +} + +} // namespace android diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h new file mode 100644 index 0000000..5a2b901 --- /dev/null +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -0,0 +1,155 @@ +/* + * Copyright (C) 2012 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef PLAYLIST_FETCHER_H_ + +#define PLAYLIST_FETCHER_H_ + +#include <media/stagefright/foundation/AHandler.h> + +#include "mpeg2ts/ATSParser.h" +#include "LiveSession.h" + +namespace android { + +struct ABuffer; +struct AnotherPacketSource; +struct DataSource; +struct HTTPBase; +struct LiveDataSource; +struct M3UParser; +struct String8; + +struct PlaylistFetcher : public AHandler { + enum { + kWhatStarted, + kWhatPaused, + kWhatStopped, + kWhatError, + kWhatDurationUpdate, + kWhatTemporarilyDoneFetching, + kWhatPrepared, + kWhatPreparationFailed, + }; + + PlaylistFetcher( + const sp<AMessage> ¬ify, + const sp<LiveSession> &session, + const char *uri); + + sp<DataSource> getDataSource(); + + void startAsync( + const sp<AnotherPacketSource> &audioSource, + const sp<AnotherPacketSource> &videoSource, + const sp<AnotherPacketSource> &subtitleSource, + int64_t startTimeUs = -1ll); + + void pauseAsync(); + + void stopAsync(); + +protected: + virtual ~PlaylistFetcher(); + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kMaxNumRetries = 5, + }; + + enum { + kWhatStart = 'strt', + kWhatPause = 'paus', + kWhatStop = 'stop', + kWhatMonitorQueue = 'moni', + }; + + static const int64_t kMinBufferedDurationUs; + + sp<AMessage> mNotify; + sp<LiveSession> mSession; + AString mURI; + + uint32_t mStreamTypeMask; + int64_t mStartTimeUs; + + KeyedVector<LiveSession::StreamType, sp<AnotherPacketSource> > + mPacketSources; + + KeyedVector<AString, sp<ABuffer> > mAESKeyForURI; + + int64_t mLastPlaylistFetchTimeUs; + sp<M3UParser> mPlaylist; + int32_t mSeqNumber; + int32_t mNumRetries; + bool mStartup; + int64_t mNextPTSTimeUs; + + int32_t mMonitorQueueGeneration; + + enum RefreshState { + INITIAL_MINIMUM_RELOAD_DELAY, + FIRST_UNCHANGED_RELOAD_ATTEMPT, + SECOND_UNCHANGED_RELOAD_ATTEMPT, + THIRD_UNCHANGED_RELOAD_ATTEMPT + }; + RefreshState mRefreshState; + + uint8_t mPlaylistHash[16]; + + sp<ATSParser> mTSParser; + + bool mFirstPTSValid; + uint64_t mFirstPTS; + int64_t mAbsoluteTimeAnchorUs; + + status_t decryptBuffer( + size_t playlistIndex, const sp<ABuffer> &buffer); + + void postMonitorQueue(int64_t delayUs = 0); + void cancelMonitorQueue(); + + bool timeToRefreshPlaylist(int64_t nowUs) const; + + // Returns the media time in us of the segment specified by seqNumber. + // This is computed by summing the durations of all segments before it. + int64_t getSegmentStartTimeUs(int32_t seqNumber) const; + + status_t onStart(const sp<AMessage> &msg); + void onPause(); + void onStop(); + void onMonitorQueue(); + void onDownloadNext(); + + status_t extractAndQueueAccessUnits(const sp<ABuffer> &buffer); + + void notifyError(status_t err); + + void queueDiscontinuity( + ATSParser::DiscontinuityType type, const sp<AMessage> &extra); + + int32_t getSeqNumberForTime(int64_t timeUs) const; + + void updateDuration(); + + DISALLOW_EVIL_CONSTRUCTORS(PlaylistFetcher); +}; + +} // namespace android + +#endif // PLAYLIST_FETCHER_H_ + diff --git a/media/libstagefright/id3/Android.mk b/media/libstagefright/id3/Android.mk index 80a1a3a..bf6f7bb 100644 --- a/media/libstagefright/id3/Android.mk +++ b/media/libstagefright/id3/Android.mk @@ -21,7 +21,7 @@ LOCAL_SHARED_LIBRARIES := \ LOCAL_STATIC_LIBRARIES := \ libstagefright_id3 -LOCAL_MODULE_TAGS := debug +LOCAL_MODULE_TAGS := optional LOCAL_MODULE := testid3 diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp index 22c2f5a..34d671a 100644 --- a/media/libstagefright/id3/ID3.cpp +++ b/media/libstagefright/id3/ID3.cpp @@ -30,12 +30,55 @@ namespace android { static const size_t kMaxMetadataSize = 3 * 1024 * 1024; +struct MemorySource : public DataSource { + MemorySource(const uint8_t *data, size_t size) + : mData(data), + mSize(size) { + } + + virtual status_t initCheck() const { + return OK; + } + + virtual ssize_t readAt(off64_t offset, void *data, size_t size) { + off64_t available = (offset >= mSize) ? 0ll : mSize - offset; + + size_t copy = (available > size) ? size : available; + memcpy(data, mData + offset, copy); + + return copy; + } + +private: + const uint8_t *mData; + size_t mSize; + + DISALLOW_EVIL_CONSTRUCTORS(MemorySource); +}; + ID3::ID3(const sp<DataSource> &source, bool ignoreV1) : mIsValid(false), mData(NULL), mSize(0), mFirstFrameOffset(0), - mVersion(ID3_UNKNOWN) { + mVersion(ID3_UNKNOWN), + mRawSize(0) { + mIsValid = parseV2(source); + + if (!mIsValid && !ignoreV1) { + mIsValid = parseV1(source); + } +} + +ID3::ID3(const uint8_t *data, size_t size, bool ignoreV1) + : mIsValid(false), + mData(NULL), + mSize(0), + mFirstFrameOffset(0), + mVersion(ID3_UNKNOWN), + mRawSize(0) { + sp<MemorySource> source = new MemorySource(data, size); + mIsValid = parseV2(source); if (!mIsValid && !ignoreV1) { @@ -140,6 +183,7 @@ struct id3_header { } mSize = size; + mRawSize = mSize + sizeof(header); if (source->readAt(sizeof(header), mData, mSize) != (ssize_t)mSize) { free(mData); @@ -313,17 +357,22 @@ bool ID3::removeUnsynchronizationV2_4(bool iTunesHack) { } if (flags & 2) { - // Unsynchronization added. + // This file has "unsynchronization", so we have to replace occurrences + // of 0xff 0x00 with just 0xff in order to get the real data. + size_t readOffset = offset + 11; + size_t writeOffset = offset + 11; for (size_t i = 0; i + 1 < dataSize; ++i) { - if (mData[offset + 10 + i] == 0xff - && mData[offset + 11 + i] == 0x00) { - memmove(&mData[offset + 11 + i], &mData[offset + 12 + i], - mSize - offset - 12 - i); + if (mData[readOffset - 1] == 0xff + && mData[readOffset] == 0x00) { + ++readOffset; --mSize; --dataSize; } + mData[writeOffset++] = mData[readOffset++]; } + // move the remaining data following this frame + memmove(&mData[writeOffset], &mData[readOffset], oldSize - readOffset); flags &= ~2; } @@ -505,7 +554,7 @@ void ID3::Iterator::getstring(String8 *id, bool otherdata) const { int32_t i = n - 4; while(--i >= 0 && *++frameData != 0) ; int skipped = (frameData - mFrameData); - if (skipped >= n) { + if (skipped >= (int)n) { return; } n -= skipped; diff --git a/media/libstagefright/include/ID3.h b/media/libstagefright/include/ID3.h index 3028f56..cca83ab 100644 --- a/media/libstagefright/include/ID3.h +++ b/media/libstagefright/include/ID3.h @@ -36,6 +36,7 @@ struct ID3 { }; ID3(const sp<DataSource> &source, bool ignoreV1 = false); + ID3(const uint8_t *data, size_t size, bool ignoreV1 = false); ~ID3(); bool isValid() const; @@ -71,6 +72,8 @@ struct ID3 { Iterator &operator=(const Iterator &); }; + size_t rawSize() const { return mRawSize; } + private: bool mIsValid; uint8_t *mData; @@ -78,6 +81,10 @@ private: size_t mFirstFrameOffset; Version mVersion; + // size of the ID3 tag including header before any unsynchronization. + // only valid for IDV2+ + size_t mRawSize; + bool parseV1(const sp<DataSource> &source); bool parseV2(const sp<DataSource> &source); void removeUnsynchronization(); diff --git a/media/libstagefright/include/MPEG2TSExtractor.h b/media/libstagefright/include/MPEG2TSExtractor.h index fe74a42..c5e86a6 100644 --- a/media/libstagefright/include/MPEG2TSExtractor.h +++ b/media/libstagefright/include/MPEG2TSExtractor.h @@ -31,7 +31,6 @@ struct ATSParser; struct DataSource; struct MPEG2TSSource; struct String8; -struct LiveSession; struct MPEG2TSExtractor : public MediaExtractor { MPEG2TSExtractor(const sp<DataSource> &source); @@ -44,16 +43,12 @@ struct MPEG2TSExtractor : public MediaExtractor { virtual uint32_t flags() const; - void setLiveSession(const sp<LiveSession> &liveSession); - void seekTo(int64_t seekTimeUs); - private: friend struct MPEG2TSSource; mutable Mutex mLock; sp<DataSource> mDataSource; - sp<LiveSession> mLiveSession; sp<ATSParser> mParser; diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h index 35eff96..bbec1c4 100644 --- a/media/libstagefright/include/MPEG4Extractor.h +++ b/media/libstagefright/include/MPEG4Extractor.h @@ -82,6 +82,7 @@ private: sp<DataSource> mDataSource; status_t mInitCheck; bool mHasVideo; + uint32_t mHeaderTimescale; Track *mFirstTrack, *mLastTrack; diff --git a/media/libstagefright/include/SoftVideoDecoderOMXComponent.h b/media/libstagefright/include/SoftVideoDecoderOMXComponent.h new file mode 100644 index 0000000..d050fa6 --- /dev/null +++ b/media/libstagefright/include/SoftVideoDecoderOMXComponent.h @@ -0,0 +1,93 @@ +/* + * Copyright (C) 2013 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef SOFT_VIDEO_DECODER_OMX_COMPONENT_H_ + +#define SOFT_VIDEO_DECODER_OMX_COMPONENT_H_ + +#include "SimpleSoftOMXComponent.h" + +#include <media/stagefright/foundation/AHandlerReflector.h> +#include <media/IOMX.h> + +#include <utils/RefBase.h> +#include <utils/threads.h> +#include <utils/Vector.h> + +#define ARRAY_SIZE(a) (sizeof(a) / sizeof(*(a))) + +namespace android { + +struct SoftVideoDecoderOMXComponent : public SimpleSoftOMXComponent { + SoftVideoDecoderOMXComponent( + const char *name, + const char *componentRole, + OMX_VIDEO_CODINGTYPE codingType, + const CodecProfileLevel *profileLevels, + size_t numProfileLevels, + int32_t width, + int32_t height, + const OMX_CALLBACKTYPE *callbacks, + OMX_PTR appData, + OMX_COMPONENTTYPE **component); + +protected: + virtual void onPortEnableCompleted(OMX_U32 portIndex, bool enabled); + virtual void onReset(); + + virtual OMX_ERRORTYPE internalGetParameter( + OMX_INDEXTYPE index, OMX_PTR params); + + virtual OMX_ERRORTYPE internalSetParameter( + OMX_INDEXTYPE index, const OMX_PTR params); + + virtual OMX_ERRORTYPE getConfig( + OMX_INDEXTYPE index, OMX_PTR params); + + void initPorts(OMX_U32 numInputBuffers, + OMX_U32 inputBufferSize, + OMX_U32 numOutputBuffers, + const char *mimeType); + + virtual void updatePortDefinitions(); + + enum { + kInputPortIndex = 0, + kOutputPortIndex = 1, + kMaxPortIndex = 1, + }; + + uint32_t mWidth, mHeight; + uint32_t mCropLeft, mCropTop, mCropWidth, mCropHeight; + + enum { + NONE, + AWAITING_DISABLED, + AWAITING_ENABLED + } mOutputPortSettingsChange; + +private: + const char *mComponentRole; + OMX_VIDEO_CODINGTYPE mCodingType; + const CodecProfileLevel *mProfileLevels; + size_t mNumProfileLevels; + + DISALLOW_EVIL_CONSTRUCTORS(SoftVideoDecoderOMXComponent); +}; + +} // namespace android + +#endif // SOFT_VIDEO_DECODER_OMX_COMPONENT_H_ diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp index 3de3a61..3153c8b 100644 --- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp +++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp @@ -32,9 +32,22 @@ const int64_t kNearEOSMarkUs = 2000000ll; // 2 secs AnotherPacketSource::AnotherPacketSource(const sp<MetaData> &meta) : mIsAudio(false), - mFormat(meta), + mFormat(NULL), mLastQueuedTimeUs(0), mEOSResult(OK) { + setFormat(meta); +} + +void AnotherPacketSource::setFormat(const sp<MetaData> &meta) { + CHECK(mFormat == NULL); + + mIsAudio = false; + + if (meta == NULL) { + return; + } + + mFormat = meta; const char *mime; CHECK(meta->findCString(kKeyMIMEType, &mime)); @@ -45,11 +58,6 @@ AnotherPacketSource::AnotherPacketSource(const sp<MetaData> &meta) } } -void AnotherPacketSource::setFormat(const sp<MetaData> &meta) { - CHECK(mFormat == NULL); - mFormat = meta; -} - AnotherPacketSource::~AnotherPacketSource() { } @@ -152,6 +160,15 @@ void AnotherPacketSource::queueAccessUnit(const sp<ABuffer> &buffer) { mCondition.signal(); } +void AnotherPacketSource::clear() { + Mutex::Autolock autoLock(mLock); + + mBuffers.clear(); + mEOSResult = OK; + + mFormat = NULL; +} + void AnotherPacketSource::queueDiscontinuity( ATSParser::DiscontinuityType type, const sp<AMessage> &extra) { diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.h b/media/libstagefright/mpeg2ts/AnotherPacketSource.h index 1db4068..e16cf78 100644 --- a/media/libstagefright/mpeg2ts/AnotherPacketSource.h +++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.h @@ -41,6 +41,8 @@ struct AnotherPacketSource : public MediaSource { virtual status_t read( MediaBuffer **buffer, const ReadOptions *options = NULL); + void clear(); + bool hasBufferAvailable(status_t *finalResult); // Returns the difference between the last and the first queued diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp index e1589b4..d449c34 100644 --- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp +++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp @@ -19,7 +19,6 @@ #include <utils/Log.h> #include "include/MPEG2TSExtractor.h" -#include "include/LiveSession.h" #include "include/NuCachedSource2.h" #include <media/stagefright/foundation/ADebug.h> @@ -79,15 +78,7 @@ status_t MPEG2TSSource::stop() { } sp<MetaData> MPEG2TSSource::getFormat() { - sp<MetaData> meta = mImpl->getFormat(); - - int64_t durationUs; - if (mExtractor->mLiveSession != NULL - && mExtractor->mLiveSession->getDuration(&durationUs) == OK) { - meta->setInt64(kKeyDuration, durationUs); - } - - return meta; + return mImpl->getFormat(); } status_t MPEG2TSSource::read( @@ -97,7 +88,7 @@ status_t MPEG2TSSource::read( int64_t seekTimeUs; ReadOptions::SeekMode seekMode; if (mSeekable && options && options->getSeekTo(&seekTimeUs, &seekMode)) { - mExtractor->seekTo(seekTimeUs); + return ERROR_UNSUPPORTED; } status_t finalResult; @@ -216,32 +207,8 @@ status_t MPEG2TSExtractor::feedMore() { return mParser->feedTSPacket(packet, kTSPacketSize); } -void MPEG2TSExtractor::setLiveSession(const sp<LiveSession> &liveSession) { - Mutex::Autolock autoLock(mLock); - - mLiveSession = liveSession; -} - -void MPEG2TSExtractor::seekTo(int64_t seekTimeUs) { - Mutex::Autolock autoLock(mLock); - - if (mLiveSession == NULL) { - return; - } - - mLiveSession->seekTo(seekTimeUs); -} - uint32_t MPEG2TSExtractor::flags() const { - Mutex::Autolock autoLock(mLock); - - uint32_t flags = CAN_PAUSE; - - if (mLiveSession != NULL && mLiveSession->isSeekable()) { - flags |= CAN_SEEK_FORWARD | CAN_SEEK_BACKWARD | CAN_SEEK; - } - - return flags; + return CAN_PAUSE; } //////////////////////////////////////////////////////////////////////////////// diff --git a/media/libstagefright/omx/Android.mk b/media/libstagefright/omx/Android.mk index a8b4939..cd912e7 100644 --- a/media/libstagefright/omx/Android.mk +++ b/media/libstagefright/omx/Android.mk @@ -9,6 +9,7 @@ LOCAL_SRC_FILES:= \ SimpleSoftOMXComponent.cpp \ SoftOMXComponent.cpp \ SoftOMXPlugin.cpp \ + SoftVideoDecoderOMXComponent.cpp \ LOCAL_C_INCLUDES += \ $(TOP)/frameworks/av/media/libstagefright \ diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp index ef27879..b3a8463 100644 --- a/media/libstagefright/omx/GraphicBufferSource.cpp +++ b/media/libstagefright/omx/GraphicBufferSource.cpp @@ -206,24 +206,15 @@ void GraphicBufferSource::codecBufferEmptied(OMX_BUFFERHEADERTYPE* header) { // Find matching entry in our cached copy of the BufferQueue slots. // If we find a match, release that slot. If we don't, the BufferQueue // has dropped that GraphicBuffer, and there's nothing for us to release. - // - // (We could store "id" in CodecBuffer and avoid the slot search.) - int id; - for (id = 0; id < BufferQueue::NUM_BUFFER_SLOTS; id++) { - if (mBufferSlot[id] == NULL) { - continue; - } - - if (mBufferSlot[id]->handle == codecBuffer.mGraphicBuffer->handle) { - ALOGV("cbi %d matches bq slot %d, handle=%p", - cbi, id, mBufferSlot[id]->handle); - - mBufferQueue->releaseBuffer(id, EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, - Fence::NO_FENCE); - break; - } - } - if (id == BufferQueue::NUM_BUFFER_SLOTS) { + int id = codecBuffer.mBuf; + if (mBufferSlot[id] != NULL && + mBufferSlot[id]->handle == codecBuffer.mGraphicBuffer->handle) { + ALOGV("cbi %d matches bq slot %d, handle=%p", + cbi, id, mBufferSlot[id]->handle); + + mBufferQueue->releaseBuffer(id, codecBuffer.mFrameNumber, + EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE); + } else { ALOGV("codecBufferEmptied: no match for emptied buffer in cbi %d", cbi); } @@ -287,11 +278,11 @@ bool GraphicBufferSource::fillCodecBuffer_l() { mBufferSlot[item.mBuf] = item.mGraphicBuffer; } - err = submitBuffer_l(mBufferSlot[item.mBuf], item.mTimestamp / 1000, cbi); + err = submitBuffer_l(item, cbi); if (err != OK) { ALOGV("submitBuffer_l failed, releasing bq buf %d", item.mBuf); - mBufferQueue->releaseBuffer(item.mBuf, EGL_NO_DISPLAY, - EGL_NO_SYNC_KHR, Fence::NO_FENCE); + mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber, + EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE); } else { ALOGV("buffer submitted (bq %d, cbi %d)", item.mBuf, cbi); } @@ -326,11 +317,13 @@ status_t GraphicBufferSource::signalEndOfInputStream() { return OK; } -status_t GraphicBufferSource::submitBuffer_l(sp<GraphicBuffer>& graphicBuffer, - int64_t timestampUsec, int cbi) { +status_t GraphicBufferSource::submitBuffer_l( + const BufferQueue::BufferItem &item, int cbi) { ALOGV("submitBuffer_l cbi=%d", cbi); CodecBuffer& codecBuffer(mCodecBuffers.editItemAt(cbi)); - codecBuffer.mGraphicBuffer = graphicBuffer; + codecBuffer.mGraphicBuffer = mBufferSlot[item.mBuf]; + codecBuffer.mBuf = item.mBuf; + codecBuffer.mFrameNumber = item.mFrameNumber; OMX_BUFFERHEADERTYPE* header = codecBuffer.mHeader; CHECK(header->nAllocLen >= 4 + sizeof(buffer_handle_t)); @@ -342,7 +335,7 @@ status_t GraphicBufferSource::submitBuffer_l(sp<GraphicBuffer>& graphicBuffer, status_t err = mNodeInstance->emptyDirectBuffer(header, 0, 4 + sizeof(buffer_handle_t), OMX_BUFFERFLAG_ENDOFFRAME, - timestampUsec); + item.mTimestamp / 1000); if (err != OK) { ALOGW("WARNING: emptyDirectBuffer failed: 0x%x", err); codecBuffer.mGraphicBuffer = NULL; @@ -431,8 +424,8 @@ void GraphicBufferSource::onFrameAvailable() { BufferQueue::BufferItem item; status_t err = mBufferQueue->acquireBuffer(&item); if (err == OK) { - mBufferQueue->releaseBuffer(item.mBuf, EGL_NO_DISPLAY, - EGL_NO_SYNC_KHR, item.mFence); + mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber, + EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, item.mFence); } return; } diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h index 562d342..8c6b470 100644 --- a/media/libstagefright/omx/GraphicBufferSource.h +++ b/media/libstagefright/omx/GraphicBufferSource.h @@ -104,6 +104,13 @@ private: // (mGraphicBuffer == NULL) or in use by the codec. struct CodecBuffer { OMX_BUFFERHEADERTYPE* mHeader; + + // buffer producer's frame-number for buffer + uint64_t mFrameNumber; + + // buffer producer's buffer slot for buffer + int mBuf; + sp<GraphicBuffer> mGraphicBuffer; }; @@ -130,8 +137,7 @@ private: // Marks the mCodecBuffers entry as in-use, copies the GraphicBuffer // reference into the codec buffer, and submits the data to the codec. - status_t submitBuffer_l(sp<GraphicBuffer>& graphicBuffer, - int64_t timestampUsec, int cbi); + status_t submitBuffer_l(const BufferQueue::BufferItem &item, int cbi); // Submits an empty buffer, with the EOS flag set. Returns without // doing anything if we don't have a codec buffer available. diff --git a/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp new file mode 100644 index 0000000..08a3d42 --- /dev/null +++ b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp @@ -0,0 +1,290 @@ +/* + * Copyright (C) 2013 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "SoftVideoDecoderOMXComponent" +#include <utils/Log.h> + +#include "include/SoftVideoDecoderOMXComponent.h" + +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/ALooper.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/MediaDefs.h> + +namespace android { + +template<class T> +static void InitOMXParams(T *params) { + params->nSize = sizeof(T); + params->nVersion.s.nVersionMajor = 1; + params->nVersion.s.nVersionMinor = 0; + params->nVersion.s.nRevision = 0; + params->nVersion.s.nStep = 0; +} + +SoftVideoDecoderOMXComponent::SoftVideoDecoderOMXComponent( + const char *name, + const char *componentRole, + OMX_VIDEO_CODINGTYPE codingType, + const CodecProfileLevel *profileLevels, + size_t numProfileLevels, + int32_t width, + int32_t height, + const OMX_CALLBACKTYPE *callbacks, + OMX_PTR appData, + OMX_COMPONENTTYPE **component) + : SimpleSoftOMXComponent(name, callbacks, appData, component), + mWidth(width), + mHeight(height), + mCropLeft(0), + mCropTop(0), + mCropWidth(width), + mCropHeight(height), + mOutputPortSettingsChange(NONE), + mComponentRole(componentRole), + mCodingType(codingType), + mProfileLevels(profileLevels), + mNumProfileLevels(numProfileLevels) { +} + +void SoftVideoDecoderOMXComponent::initPorts( + OMX_U32 numInputBuffers, + OMX_U32 inputBufferSize, + OMX_U32 numOutputBuffers, + const char *mimeType) { + OMX_PARAM_PORTDEFINITIONTYPE def; + InitOMXParams(&def); + + def.nPortIndex = kInputPortIndex; + def.eDir = OMX_DirInput; + def.nBufferCountMin = numInputBuffers; + def.nBufferCountActual = def.nBufferCountMin; + def.nBufferSize = inputBufferSize; + def.bEnabled = OMX_TRUE; + def.bPopulated = OMX_FALSE; + def.eDomain = OMX_PortDomainVideo; + def.bBuffersContiguous = OMX_FALSE; + def.nBufferAlignment = 1; + + def.format.video.cMIMEType = const_cast<char *>(mimeType); + def.format.video.pNativeRender = NULL; + /* size is initialized in updatePortDefinitions() */ + def.format.video.nBitrate = 0; + def.format.video.xFramerate = 0; + def.format.video.bFlagErrorConcealment = OMX_FALSE; + def.format.video.eCompressionFormat = mCodingType; + def.format.video.eColorFormat = OMX_COLOR_FormatUnused; + def.format.video.pNativeWindow = NULL; + + addPort(def); + + def.nPortIndex = kOutputPortIndex; + def.eDir = OMX_DirOutput; + def.nBufferCountMin = numOutputBuffers; + def.nBufferCountActual = def.nBufferCountMin; + def.bEnabled = OMX_TRUE; + def.bPopulated = OMX_FALSE; + def.eDomain = OMX_PortDomainVideo; + def.bBuffersContiguous = OMX_FALSE; + def.nBufferAlignment = 2; + + def.format.video.cMIMEType = const_cast<char *>("video/raw"); + def.format.video.pNativeRender = NULL; + /* size is initialized in updatePortDefinitions() */ + def.format.video.nBitrate = 0; + def.format.video.xFramerate = 0; + def.format.video.bFlagErrorConcealment = OMX_FALSE; + def.format.video.eCompressionFormat = OMX_VIDEO_CodingUnused; + def.format.video.eColorFormat = OMX_COLOR_FormatYUV420Planar; + def.format.video.pNativeWindow = NULL; + + addPort(def); + + updatePortDefinitions(); +} + +void SoftVideoDecoderOMXComponent::updatePortDefinitions() { + OMX_PARAM_PORTDEFINITIONTYPE *def = &editPortInfo(kInputPortIndex)->mDef; + def->format.video.nFrameWidth = mWidth; + def->format.video.nFrameHeight = mHeight; + def->format.video.nStride = def->format.video.nFrameWidth; + def->format.video.nSliceHeight = def->format.video.nFrameHeight; + + def = &editPortInfo(kOutputPortIndex)->mDef; + def->format.video.nFrameWidth = mWidth; + def->format.video.nFrameHeight = mHeight; + def->format.video.nStride = def->format.video.nFrameWidth; + def->format.video.nSliceHeight = def->format.video.nFrameHeight; + + def->nBufferSize = + (def->format.video.nFrameWidth * + def->format.video.nFrameHeight * 3) / 2; + + mCropLeft = 0; + mCropTop = 0; + mCropWidth = mWidth; + mCropHeight = mHeight; +} + +OMX_ERRORTYPE SoftVideoDecoderOMXComponent::internalGetParameter( + OMX_INDEXTYPE index, OMX_PTR params) { + switch (index) { + case OMX_IndexParamVideoPortFormat: + { + OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams = + (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params; + + if (formatParams->nPortIndex > kMaxPortIndex) { + return OMX_ErrorUndefined; + } + + if (formatParams->nIndex != 0) { + return OMX_ErrorNoMore; + } + + if (formatParams->nPortIndex == kInputPortIndex) { + formatParams->eCompressionFormat = mCodingType; + formatParams->eColorFormat = OMX_COLOR_FormatUnused; + formatParams->xFramerate = 0; + } else { + CHECK_EQ(formatParams->nPortIndex, 1u); + + formatParams->eCompressionFormat = OMX_VIDEO_CodingUnused; + formatParams->eColorFormat = OMX_COLOR_FormatYUV420Planar; + formatParams->xFramerate = 0; + } + + return OMX_ErrorNone; + } + + case OMX_IndexParamVideoProfileLevelQuerySupported: + { + OMX_VIDEO_PARAM_PROFILELEVELTYPE *profileLevel = + (OMX_VIDEO_PARAM_PROFILELEVELTYPE *) params; + + if (profileLevel->nPortIndex != kInputPortIndex) { + ALOGE("Invalid port index: %ld", profileLevel->nPortIndex); + return OMX_ErrorUnsupportedIndex; + } + + if (index >= mNumProfileLevels) { + return OMX_ErrorNoMore; + } + + profileLevel->eProfile = mProfileLevels[index].mProfile; + profileLevel->eLevel = mProfileLevels[index].mLevel; + return OMX_ErrorNone; + } + + default: + return SimpleSoftOMXComponent::internalGetParameter(index, params); + } +} + +OMX_ERRORTYPE SoftVideoDecoderOMXComponent::internalSetParameter( + OMX_INDEXTYPE index, const OMX_PTR params) { + switch (index) { + case OMX_IndexParamStandardComponentRole: + { + const OMX_PARAM_COMPONENTROLETYPE *roleParams = + (const OMX_PARAM_COMPONENTROLETYPE *)params; + + if (strncmp((const char *)roleParams->cRole, + mComponentRole, + OMX_MAX_STRINGNAME_SIZE - 1)) { + return OMX_ErrorUndefined; + } + + return OMX_ErrorNone; + } + + case OMX_IndexParamVideoPortFormat: + { + OMX_VIDEO_PARAM_PORTFORMATTYPE *formatParams = + (OMX_VIDEO_PARAM_PORTFORMATTYPE *)params; + + if (formatParams->nPortIndex > kMaxPortIndex) { + return OMX_ErrorUndefined; + } + + if (formatParams->nIndex != 0) { + return OMX_ErrorNoMore; + } + + return OMX_ErrorNone; + } + + default: + return SimpleSoftOMXComponent::internalSetParameter(index, params); + } +} + +OMX_ERRORTYPE SoftVideoDecoderOMXComponent::getConfig( + OMX_INDEXTYPE index, OMX_PTR params) { + switch (index) { + case OMX_IndexConfigCommonOutputCrop: + { + OMX_CONFIG_RECTTYPE *rectParams = (OMX_CONFIG_RECTTYPE *)params; + + if (rectParams->nPortIndex != kOutputPortIndex) { + return OMX_ErrorUndefined; + } + + rectParams->nLeft = mCropLeft; + rectParams->nTop = mCropTop; + rectParams->nWidth = mCropWidth; + rectParams->nHeight = mCropHeight; + + return OMX_ErrorNone; + } + + default: + return OMX_ErrorUnsupportedIndex; + } +} + +void SoftVideoDecoderOMXComponent::onReset() { + mOutputPortSettingsChange = NONE; +} + +void SoftVideoDecoderOMXComponent::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) { + if (portIndex != kOutputPortIndex) { + return; + } + + switch (mOutputPortSettingsChange) { + case NONE: + break; + + case AWAITING_DISABLED: + { + CHECK(!enabled); + mOutputPortSettingsChange = AWAITING_ENABLED; + break; + } + + default: + { + CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED); + CHECK(enabled); + mOutputPortSettingsChange = NONE; + break; + } + } +} + +} // namespace android diff --git a/media/libstagefright/rtsp/Android.mk b/media/libstagefright/rtsp/Android.mk index 9e2724d..e77c69c 100644 --- a/media/libstagefright/rtsp/Android.mk +++ b/media/libstagefright/rtsp/Android.mk @@ -51,7 +51,7 @@ LOCAL_C_INCLUDES:= \ LOCAL_CFLAGS += -Wno-multichar -LOCAL_MODULE_TAGS := debug +LOCAL_MODULE_TAGS := optional LOCAL_MODULE:= rtp_test diff --git a/media/libstagefright/wifi-display/Android.mk b/media/libstagefright/wifi-display/Android.mk index 061ae89..404b41e 100644 --- a/media/libstagefright/wifi-display/Android.mk +++ b/media/libstagefright/wifi-display/Android.mk @@ -4,10 +4,17 @@ include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ ANetworkSession.cpp \ + MediaReceiver.cpp \ MediaSender.cpp \ Parameters.cpp \ ParsedMessage.cpp \ + rtp/RTPAssembler.cpp \ + rtp/RTPReceiver.cpp \ rtp/RTPSender.cpp \ + sink/DirectRenderer.cpp \ + sink/WifiDisplaySink.cpp \ + SNTPClient.cpp \ + TimeSyncer.cpp \ source/Converter.cpp \ source/MediaPuller.cpp \ source/PlaybackSession.cpp \ @@ -57,6 +64,67 @@ LOCAL_SHARED_LIBRARIES:= \ LOCAL_MODULE:= wfd -LOCAL_MODULE_TAGS := debug +include $(BUILD_EXECUTABLE) + +################################################################################ + +include $(CLEAR_VARS) + +LOCAL_SRC_FILES:= \ + udptest.cpp \ + +LOCAL_SHARED_LIBRARIES:= \ + libbinder \ + libgui \ + libmedia \ + libstagefright \ + libstagefright_foundation \ + libstagefright_wfd \ + libutils \ + liblog \ + +LOCAL_MODULE:= udptest + +include $(BUILD_EXECUTABLE) + +################################################################################ + +include $(CLEAR_VARS) + +LOCAL_SRC_FILES:= \ + rtptest.cpp \ + +LOCAL_SHARED_LIBRARIES:= \ + libbinder \ + libgui \ + libmedia \ + libstagefright \ + libstagefright_foundation \ + libstagefright_wfd \ + libutils \ + liblog \ + +LOCAL_MODULE:= rtptest + +include $(BUILD_EXECUTABLE) + +################################################################################ + +include $(CLEAR_VARS) + +LOCAL_SRC_FILES:= \ + nettest.cpp \ + +LOCAL_SHARED_LIBRARIES:= \ + libbinder \ + libgui \ + libmedia \ + libstagefright \ + libstagefright_foundation \ + libstagefright_wfd \ + libutils \ + liblog \ + +LOCAL_MODULE:= nettest include $(BUILD_EXECUTABLE) diff --git a/media/libstagefright/wifi-display/MediaReceiver.cpp b/media/libstagefright/wifi-display/MediaReceiver.cpp new file mode 100644 index 0000000..364acb9 --- /dev/null +++ b/media/libstagefright/wifi-display/MediaReceiver.cpp @@ -0,0 +1,328 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "MediaReceiver" +#include <utils/Log.h> + +#include "MediaReceiver.h" + +#include "ANetworkSession.h" +#include "AnotherPacketSource.h" +#include "rtp/RTPReceiver.h" + +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/MetaData.h> +#include <media/stagefright/Utils.h> + +namespace android { + +MediaReceiver::MediaReceiver( + const sp<ANetworkSession> &netSession, + const sp<AMessage> ¬ify) + : mNetSession(netSession), + mNotify(notify), + mMode(MODE_UNDEFINED), + mGeneration(0), + mInitStatus(OK), + mInitDoneCount(0) { +} + +MediaReceiver::~MediaReceiver() { +} + +ssize_t MediaReceiver::addTrack( + RTPReceiver::TransportMode rtpMode, + RTPReceiver::TransportMode rtcpMode, + int32_t *localRTPPort) { + if (mMode != MODE_UNDEFINED) { + return INVALID_OPERATION; + } + + size_t trackIndex = mTrackInfos.size(); + + TrackInfo info; + + sp<AMessage> notify = new AMessage(kWhatReceiverNotify, id()); + notify->setInt32("generation", mGeneration); + notify->setSize("trackIndex", trackIndex); + + info.mReceiver = new RTPReceiver(mNetSession, notify); + looper()->registerHandler(info.mReceiver); + + info.mReceiver->registerPacketType( + 33, RTPReceiver::PACKETIZATION_TRANSPORT_STREAM); + + info.mReceiver->registerPacketType( + 96, RTPReceiver::PACKETIZATION_AAC); + + info.mReceiver->registerPacketType( + 97, RTPReceiver::PACKETIZATION_H264); + + status_t err = info.mReceiver->initAsync( + rtpMode, + rtcpMode, + localRTPPort); + + if (err != OK) { + looper()->unregisterHandler(info.mReceiver->id()); + info.mReceiver.clear(); + + return err; + } + + mTrackInfos.push_back(info); + + return trackIndex; +} + +status_t MediaReceiver::connectTrack( + size_t trackIndex, + const char *remoteHost, + int32_t remoteRTPPort, + int32_t remoteRTCPPort) { + if (trackIndex >= mTrackInfos.size()) { + return -ERANGE; + } + + TrackInfo *info = &mTrackInfos.editItemAt(trackIndex); + return info->mReceiver->connect(remoteHost, remoteRTPPort, remoteRTCPPort); +} + +status_t MediaReceiver::initAsync(Mode mode) { + if ((mode == MODE_TRANSPORT_STREAM || mode == MODE_TRANSPORT_STREAM_RAW) + && mTrackInfos.size() > 1) { + return INVALID_OPERATION; + } + + sp<AMessage> msg = new AMessage(kWhatInit, id()); + msg->setInt32("mode", mode); + msg->post(); + + return OK; +} + +void MediaReceiver::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatInit: + { + int32_t mode; + CHECK(msg->findInt32("mode", &mode)); + + CHECK_EQ(mMode, MODE_UNDEFINED); + mMode = (Mode)mode; + + if (mInitStatus != OK || mInitDoneCount == mTrackInfos.size()) { + notifyInitDone(mInitStatus); + } + + mTSParser = new ATSParser( + ATSParser::ALIGNED_VIDEO_DATA + | ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE); + + mFormatKnownMask = 0; + break; + } + + case kWhatReceiverNotify: + { + int32_t generation; + CHECK(msg->findInt32("generation", &generation)); + if (generation != mGeneration) { + break; + } + + onReceiverNotify(msg); + break; + } + + default: + TRESPASS(); + } +} + +void MediaReceiver::onReceiverNotify(const sp<AMessage> &msg) { + int32_t what; + CHECK(msg->findInt32("what", &what)); + + switch (what) { + case RTPReceiver::kWhatInitDone: + { + ++mInitDoneCount; + + int32_t err; + CHECK(msg->findInt32("err", &err)); + + if (err != OK) { + mInitStatus = err; + ++mGeneration; + } + + if (mMode != MODE_UNDEFINED) { + if (mInitStatus != OK || mInitDoneCount == mTrackInfos.size()) { + notifyInitDone(mInitStatus); + } + } + break; + } + + case RTPReceiver::kWhatError: + { + int32_t err; + CHECK(msg->findInt32("err", &err)); + + notifyError(err); + break; + } + + case RTPReceiver::kWhatAccessUnit: + { + size_t trackIndex; + CHECK(msg->findSize("trackIndex", &trackIndex)); + + sp<ABuffer> accessUnit; + CHECK(msg->findBuffer("accessUnit", &accessUnit)); + + int32_t followsDiscontinuity; + if (!msg->findInt32( + "followsDiscontinuity", &followsDiscontinuity)) { + followsDiscontinuity = 0; + } + + if (mMode == MODE_TRANSPORT_STREAM) { + if (followsDiscontinuity) { + mTSParser->signalDiscontinuity( + ATSParser::DISCONTINUITY_TIME, NULL /* extra */); + } + + for (size_t offset = 0; + offset < accessUnit->size(); offset += 188) { + status_t err = mTSParser->feedTSPacket( + accessUnit->data() + offset, 188); + + if (err != OK) { + notifyError(err); + break; + } + } + + drainPackets(0 /* trackIndex */, ATSParser::VIDEO); + drainPackets(1 /* trackIndex */, ATSParser::AUDIO); + } else { + postAccessUnit(trackIndex, accessUnit, NULL); + } + break; + } + + case RTPReceiver::kWhatPacketLost: + { + notifyPacketLost(); + break; + } + + default: + TRESPASS(); + } +} + +void MediaReceiver::drainPackets( + size_t trackIndex, ATSParser::SourceType type) { + sp<AnotherPacketSource> source = + static_cast<AnotherPacketSource *>( + mTSParser->getSource(type).get()); + + if (source == NULL) { + return; + } + + sp<AMessage> format; + if (!(mFormatKnownMask & (1ul << trackIndex))) { + sp<MetaData> meta = source->getFormat(); + CHECK(meta != NULL); + + CHECK_EQ((status_t)OK, convertMetaDataToMessage(meta, &format)); + + mFormatKnownMask |= 1ul << trackIndex; + } + + status_t finalResult; + while (source->hasBufferAvailable(&finalResult)) { + sp<ABuffer> accessUnit; + status_t err = source->dequeueAccessUnit(&accessUnit); + if (err == OK) { + postAccessUnit(trackIndex, accessUnit, format); + format.clear(); + } else if (err != INFO_DISCONTINUITY) { + notifyError(err); + } + } + + if (finalResult != OK) { + notifyError(finalResult); + } +} + +void MediaReceiver::notifyInitDone(status_t err) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatInitDone); + notify->setInt32("err", err); + notify->post(); +} + +void MediaReceiver::notifyError(status_t err) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatError); + notify->setInt32("err", err); + notify->post(); +} + +void MediaReceiver::notifyPacketLost() { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatPacketLost); + notify->post(); +} + +void MediaReceiver::postAccessUnit( + size_t trackIndex, + const sp<ABuffer> &accessUnit, + const sp<AMessage> &format) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatAccessUnit); + notify->setSize("trackIndex", trackIndex); + notify->setBuffer("accessUnit", accessUnit); + + if (format != NULL) { + notify->setMessage("format", format); + } + + notify->post(); +} + +status_t MediaReceiver::informSender( + size_t trackIndex, const sp<AMessage> ¶ms) { + if (trackIndex >= mTrackInfos.size()) { + return -ERANGE; + } + + TrackInfo *info = &mTrackInfos.editItemAt(trackIndex); + return info->mReceiver->informSender(params); +} + +} // namespace android + + diff --git a/media/libstagefright/wifi-display/MediaReceiver.h b/media/libstagefright/wifi-display/MediaReceiver.h new file mode 100644 index 0000000..afbb407 --- /dev/null +++ b/media/libstagefright/wifi-display/MediaReceiver.h @@ -0,0 +1,111 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include <media/stagefright/foundation/AHandler.h> + +#include "ATSParser.h" +#include "rtp/RTPReceiver.h" + +namespace android { + +struct ABuffer; +struct ANetworkSession; +struct AMessage; +struct ATSParser; + +// This class facilitates receiving of media data for one or more tracks +// over RTP. Either a 1:1 track to RTP channel mapping is used or a single +// RTP channel provides the data for a transport stream that is consequently +// demuxed and its track's data provided to the observer. +struct MediaReceiver : public AHandler { + enum { + kWhatInitDone, + kWhatError, + kWhatAccessUnit, + kWhatPacketLost, + }; + + MediaReceiver( + const sp<ANetworkSession> &netSession, + const sp<AMessage> ¬ify); + + ssize_t addTrack( + RTPReceiver::TransportMode rtpMode, + RTPReceiver::TransportMode rtcpMode, + int32_t *localRTPPort); + + status_t connectTrack( + size_t trackIndex, + const char *remoteHost, + int32_t remoteRTPPort, + int32_t remoteRTCPPort); + + enum Mode { + MODE_UNDEFINED, + MODE_TRANSPORT_STREAM, + MODE_TRANSPORT_STREAM_RAW, + MODE_ELEMENTARY_STREAMS, + }; + status_t initAsync(Mode mode); + + status_t informSender(size_t trackIndex, const sp<AMessage> ¶ms); + +protected: + virtual void onMessageReceived(const sp<AMessage> &msg); + virtual ~MediaReceiver(); + +private: + enum { + kWhatInit, + kWhatReceiverNotify, + }; + + struct TrackInfo { + sp<RTPReceiver> mReceiver; + }; + + sp<ANetworkSession> mNetSession; + sp<AMessage> mNotify; + + Mode mMode; + int32_t mGeneration; + + Vector<TrackInfo> mTrackInfos; + + status_t mInitStatus; + size_t mInitDoneCount; + + sp<ATSParser> mTSParser; + uint32_t mFormatKnownMask; + + void onReceiverNotify(const sp<AMessage> &msg); + + void drainPackets(size_t trackIndex, ATSParser::SourceType type); + + void notifyInitDone(status_t err); + void notifyError(status_t err); + void notifyPacketLost(); + + void postAccessUnit( + size_t trackIndex, + const sp<ABuffer> &accessUnit, + const sp<AMessage> &format); + + DISALLOW_EVIL_CONSTRUCTORS(MediaReceiver); +}; + +} // namespace android + diff --git a/media/libstagefright/wifi-display/MediaSender.cpp b/media/libstagefright/wifi-display/MediaSender.cpp index 8a3566f..a230cd8 100644 --- a/media/libstagefright/wifi-display/MediaSender.cpp +++ b/media/libstagefright/wifi-display/MediaSender.cpp @@ -27,9 +27,11 @@ #include "include/avc_utils.h" #include <media/IHDCP.h> +#include <media/stagefright/MediaBuffer.h> #include <media/stagefright/foundation/ABuffer.h> #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/foundation/AMessage.h> +#include <ui/GraphicBuffer.h> namespace android { @@ -341,6 +343,22 @@ void MediaSender::onSenderNotify(const sp<AMessage> &msg) { break; } + case kWhatInformSender: + { + int64_t avgLatencyUs; + CHECK(msg->findInt64("avgLatencyUs", &avgLatencyUs)); + + int64_t maxLatencyUs; + CHECK(msg->findInt64("maxLatencyUs", &maxLatencyUs)); + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatInformSender); + notify->setInt64("avgLatencyUs", avgLatencyUs); + notify->setInt64("maxLatencyUs", maxLatencyUs); + notify->post(); + break; + } + default: TRESPASS(); } @@ -392,11 +410,36 @@ status_t MediaSender::packetizeAccessUnit( info.mPacketizerTrackIndex, accessUnit); } - status_t err = mHDCP->encrypt( - accessUnit->data(), accessUnit->size(), - trackIndex /* streamCTR */, - &inputCTR, - accessUnit->data()); + status_t err; + native_handle_t* handle; + if (accessUnit->meta()->findPointer("handle", (void**)&handle) + && handle != NULL) { + int32_t rangeLength, rangeOffset; + sp<AMessage> notify; + CHECK(accessUnit->meta()->findInt32("rangeOffset", &rangeOffset)); + CHECK(accessUnit->meta()->findInt32("rangeLength", &rangeLength)); + CHECK(accessUnit->meta()->findMessage("notify", ¬ify) + && notify != NULL); + CHECK_GE(accessUnit->size(), rangeLength); + + sp<GraphicBuffer> grbuf(new GraphicBuffer( + rangeOffset + rangeLength, 1, HAL_PIXEL_FORMAT_Y8, + GRALLOC_USAGE_HW_VIDEO_ENCODER, rangeOffset + rangeLength, + handle, false)); + + err = mHDCP->encryptNative( + grbuf, rangeOffset, rangeLength, + trackIndex /* streamCTR */, + &inputCTR, + accessUnit->data()); + notify->post(); + } else { + err = mHDCP->encrypt( + accessUnit->data(), accessUnit->size(), + trackIndex /* streamCTR */, + &inputCTR, + accessUnit->data()); + } if (err != OK) { ALOGE("Failed to HDCP-encrypt media data (err %d)", diff --git a/media/libstagefright/wifi-display/MediaSender.h b/media/libstagefright/wifi-display/MediaSender.h index 64722c5..04538ea 100644 --- a/media/libstagefright/wifi-display/MediaSender.h +++ b/media/libstagefright/wifi-display/MediaSender.h @@ -43,6 +43,7 @@ struct MediaSender : public AHandler { kWhatInitDone, kWhatError, kWhatNetworkStall, + kWhatInformSender, }; MediaSender( diff --git a/media/libstagefright/wifi-display/SNTPClient.cpp b/media/libstagefright/wifi-display/SNTPClient.cpp new file mode 100644 index 0000000..5c0af6a --- /dev/null +++ b/media/libstagefright/wifi-display/SNTPClient.cpp @@ -0,0 +1,174 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "SNTPClient.h" + +#include <media/stagefright/foundation/ALooper.h> +#include <media/stagefright/Utils.h> + +#include <arpa/inet.h> +#include <netdb.h> +#include <netinet/in.h> +#include <sys/socket.h> +#include <unistd.h> + +namespace android { + +SNTPClient::SNTPClient() { +} + +status_t SNTPClient::requestTime(const char *host) { + struct hostent *ent; + int64_t requestTimeNTP, requestTimeUs; + ssize_t n; + int64_t responseTimeUs, responseTimeNTP; + int64_t originateTimeNTP, receiveTimeNTP, transmitTimeNTP; + int64_t roundTripTimeNTP, clockOffsetNTP; + + status_t err = UNKNOWN_ERROR; + + int s = socket(AF_INET, SOCK_DGRAM, 0); + + if (s < 0) { + err = -errno; + + goto bail; + } + + ent = gethostbyname(host); + + if (ent == NULL) { + err = -ENOENT; + goto bail2; + } + + struct sockaddr_in hostAddr; + memset(hostAddr.sin_zero, 0, sizeof(hostAddr.sin_zero)); + hostAddr.sin_family = AF_INET; + hostAddr.sin_port = htons(kNTPPort); + hostAddr.sin_addr.s_addr = *(in_addr_t *)ent->h_addr; + + uint8_t packet[kNTPPacketSize]; + memset(packet, 0, sizeof(packet)); + + packet[0] = kNTPModeClient | (kNTPVersion << 3); + + requestTimeNTP = getNowNTP(); + requestTimeUs = ALooper::GetNowUs(); + writeTimeStamp(&packet[kNTPTransmitTimeOffset], requestTimeNTP); + + n = sendto( + s, packet, sizeof(packet), 0, + (const struct sockaddr *)&hostAddr, sizeof(hostAddr)); + + if (n < 0) { + err = -errno; + goto bail2; + } + + memset(packet, 0, sizeof(packet)); + + do { + n = recv(s, packet, sizeof(packet), 0); + } while (n < 0 && errno == EINTR); + + if (n < 0) { + err = -errno; + goto bail2; + } + + responseTimeUs = ALooper::GetNowUs(); + + responseTimeNTP = requestTimeNTP + makeNTP(responseTimeUs - requestTimeUs); + + originateTimeNTP = readTimeStamp(&packet[kNTPOriginateTimeOffset]); + receiveTimeNTP = readTimeStamp(&packet[kNTPReceiveTimeOffset]); + transmitTimeNTP = readTimeStamp(&packet[kNTPTransmitTimeOffset]); + + roundTripTimeNTP = + makeNTP(responseTimeUs - requestTimeUs) + - (transmitTimeNTP - receiveTimeNTP); + + clockOffsetNTP = + ((receiveTimeNTP - originateTimeNTP) + + (transmitTimeNTP - responseTimeNTP)) / 2; + + mTimeReferenceNTP = responseTimeNTP + clockOffsetNTP; + mTimeReferenceUs = responseTimeUs; + mRoundTripTimeNTP = roundTripTimeNTP; + + err = OK; + +bail2: + close(s); + s = -1; + +bail: + return err; +} + +int64_t SNTPClient::adjustTimeUs(int64_t timeUs) const { + uint64_t nowNTP = + mTimeReferenceNTP + makeNTP(timeUs - mTimeReferenceUs); + + int64_t nowUs = + (nowNTP >> 32) * 1000000ll + + ((nowNTP & 0xffffffff) * 1000000ll) / (1ll << 32); + + nowUs -= ((70ll * 365 + 17) * 24) * 60 * 60 * 1000000ll; + + return nowUs; +} + +// static +void SNTPClient::writeTimeStamp(uint8_t *dst, uint64_t ntpTime) { + *dst++ = (ntpTime >> 56) & 0xff; + *dst++ = (ntpTime >> 48) & 0xff; + *dst++ = (ntpTime >> 40) & 0xff; + *dst++ = (ntpTime >> 32) & 0xff; + *dst++ = (ntpTime >> 24) & 0xff; + *dst++ = (ntpTime >> 16) & 0xff; + *dst++ = (ntpTime >> 8) & 0xff; + *dst++ = ntpTime & 0xff; +} + +// static +uint64_t SNTPClient::readTimeStamp(const uint8_t *dst) { + return U64_AT(dst); +} + +// static +uint64_t SNTPClient::getNowNTP() { + struct timeval tv; + gettimeofday(&tv, NULL /* time zone */); + + uint64_t nowUs = tv.tv_sec * 1000000ll + tv.tv_usec; + + nowUs += ((70ll * 365 + 17) * 24) * 60 * 60 * 1000000ll; + + return makeNTP(nowUs); +} + +// static +uint64_t SNTPClient::makeNTP(uint64_t deltaUs) { + uint64_t hi = deltaUs / 1000000ll; + uint64_t lo = ((1ll << 32) * (deltaUs % 1000000ll)) / 1000000ll; + + return (hi << 32) | lo; +} + +} // namespace android + diff --git a/media/libstagefright/wifi-display/SNTPClient.h b/media/libstagefright/wifi-display/SNTPClient.h new file mode 100644 index 0000000..967d1fc --- /dev/null +++ b/media/libstagefright/wifi-display/SNTPClient.h @@ -0,0 +1,62 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef SNTP_CLIENT_H_ + +#define SNTP_CLIENT_H_ + +#include <media/stagefright/foundation/ABase.h> +#include <utils/Errors.h> + +namespace android { + +// Implementation of the SNTP (Simple Network Time Protocol) +struct SNTPClient { + SNTPClient(); + + status_t requestTime(const char *host); + + // given a time obtained from ALooper::GetNowUs() + // return the number of us elapsed since Jan 1 1970 00:00:00 (UTC). + int64_t adjustTimeUs(int64_t timeUs) const; + +private: + enum { + kNTPPort = 123, + kNTPPacketSize = 48, + kNTPModeClient = 3, + kNTPVersion = 3, + kNTPTransmitTimeOffset = 40, + kNTPOriginateTimeOffset = 24, + kNTPReceiveTimeOffset = 32, + }; + + uint64_t mTimeReferenceNTP; + int64_t mTimeReferenceUs; + int64_t mRoundTripTimeNTP; + + static void writeTimeStamp(uint8_t *dst, uint64_t ntpTime); + static uint64_t readTimeStamp(const uint8_t *dst); + + static uint64_t getNowNTP(); + static uint64_t makeNTP(uint64_t deltaUs); + + DISALLOW_EVIL_CONSTRUCTORS(SNTPClient); +}; + +} // namespace android + +#endif // SNTP_CLIENT_H_ diff --git a/media/libstagefright/wifi-display/TimeSyncer.cpp b/media/libstagefright/wifi-display/TimeSyncer.cpp new file mode 100644 index 0000000..cb429bc --- /dev/null +++ b/media/libstagefright/wifi-display/TimeSyncer.cpp @@ -0,0 +1,338 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NEBUG 0 +#define LOG_TAG "TimeSyncer" +#include <utils/Log.h> + +#include "TimeSyncer.h" + +#include "ANetworkSession.h" + +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AHandler.h> +#include <media/stagefright/foundation/ALooper.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/Utils.h> + +namespace android { + +TimeSyncer::TimeSyncer( + const sp<ANetworkSession> &netSession, const sp<AMessage> ¬ify) + : mNetSession(netSession), + mNotify(notify), + mIsServer(false), + mConnected(false), + mUDPSession(0), + mSeqNo(0), + mTotalTimeUs(0.0), + mPendingT1(0ll), + mTimeoutGeneration(0) { +} + +TimeSyncer::~TimeSyncer() { +} + +void TimeSyncer::startServer(unsigned localPort) { + sp<AMessage> msg = new AMessage(kWhatStartServer, id()); + msg->setInt32("localPort", localPort); + msg->post(); +} + +void TimeSyncer::startClient(const char *remoteHost, unsigned remotePort) { + sp<AMessage> msg = new AMessage(kWhatStartClient, id()); + msg->setString("remoteHost", remoteHost); + msg->setInt32("remotePort", remotePort); + msg->post(); +} + +void TimeSyncer::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatStartClient: + { + AString remoteHost; + CHECK(msg->findString("remoteHost", &remoteHost)); + + int32_t remotePort; + CHECK(msg->findInt32("remotePort", &remotePort)); + + sp<AMessage> notify = new AMessage(kWhatUDPNotify, id()); + + CHECK_EQ((status_t)OK, + mNetSession->createUDPSession( + 0 /* localPort */, + remoteHost.c_str(), + remotePort, + notify, + &mUDPSession)); + + postSendPacket(); + break; + } + + case kWhatStartServer: + { + mIsServer = true; + + int32_t localPort; + CHECK(msg->findInt32("localPort", &localPort)); + + sp<AMessage> notify = new AMessage(kWhatUDPNotify, id()); + + CHECK_EQ((status_t)OK, + mNetSession->createUDPSession( + localPort, notify, &mUDPSession)); + + break; + } + + case kWhatSendPacket: + { + if (mHistory.size() == 0) { + ALOGI("starting batch"); + } + + TimeInfo ti; + memset(&ti, 0, sizeof(ti)); + + ti.mT1 = ALooper::GetNowUs(); + + CHECK_EQ((status_t)OK, + mNetSession->sendRequest( + mUDPSession, &ti, sizeof(ti))); + + mPendingT1 = ti.mT1; + postTimeout(); + break; + } + + case kWhatTimedOut: + { + int32_t generation; + CHECK(msg->findInt32("generation", &generation)); + + if (generation != mTimeoutGeneration) { + break; + } + + ALOGI("timed out, sending another request"); + postSendPacket(); + break; + } + + case kWhatUDPNotify: + { + int32_t reason; + CHECK(msg->findInt32("reason", &reason)); + + switch (reason) { + case ANetworkSession::kWhatError: + { + int32_t sessionID; + CHECK(msg->findInt32("sessionID", &sessionID)); + + int32_t err; + CHECK(msg->findInt32("err", &err)); + + AString detail; + CHECK(msg->findString("detail", &detail)); + + ALOGE("An error occurred in session %d (%d, '%s/%s').", + sessionID, + err, + detail.c_str(), + strerror(-err)); + + mNetSession->destroySession(sessionID); + + cancelTimeout(); + + notifyError(err); + break; + } + + case ANetworkSession::kWhatDatagram: + { + int32_t sessionID; + CHECK(msg->findInt32("sessionID", &sessionID)); + + sp<ABuffer> packet; + CHECK(msg->findBuffer("data", &packet)); + + int64_t arrivalTimeUs; + CHECK(packet->meta()->findInt64( + "arrivalTimeUs", &arrivalTimeUs)); + + CHECK_EQ(packet->size(), sizeof(TimeInfo)); + + TimeInfo *ti = (TimeInfo *)packet->data(); + + if (mIsServer) { + if (!mConnected) { + AString fromAddr; + CHECK(msg->findString("fromAddr", &fromAddr)); + + int32_t fromPort; + CHECK(msg->findInt32("fromPort", &fromPort)); + + CHECK_EQ((status_t)OK, + mNetSession->connectUDPSession( + mUDPSession, fromAddr.c_str(), fromPort)); + + mConnected = true; + } + + ti->mT2 = arrivalTimeUs; + ti->mT3 = ALooper::GetNowUs(); + + CHECK_EQ((status_t)OK, + mNetSession->sendRequest( + mUDPSession, ti, sizeof(*ti))); + } else { + if (ti->mT1 != mPendingT1) { + break; + } + + cancelTimeout(); + mPendingT1 = 0; + + ti->mT4 = arrivalTimeUs; + + // One way delay for a packet to travel from client + // to server or back (assumed to be the same either way). + int64_t delay = + (ti->mT2 - ti->mT1 + ti->mT4 - ti->mT3) / 2; + + // Offset between the client clock (T1, T4) and the + // server clock (T2, T3) timestamps. + int64_t offset = + (ti->mT2 - ti->mT1 - ti->mT4 + ti->mT3) / 2; + + mHistory.push_back(*ti); + + ALOGV("delay = %lld us,\toffset %lld us", + delay, + offset); + + if (mHistory.size() < kNumPacketsPerBatch) { + postSendPacket(1000000ll / 30); + } else { + notifyOffset(); + + ALOGI("batch done"); + + mHistory.clear(); + postSendPacket(kBatchDelayUs); + } + } + break; + } + + default: + TRESPASS(); + } + + break; + } + + default: + TRESPASS(); + } +} + +void TimeSyncer::postSendPacket(int64_t delayUs) { + (new AMessage(kWhatSendPacket, id()))->post(delayUs); +} + +void TimeSyncer::postTimeout() { + sp<AMessage> msg = new AMessage(kWhatTimedOut, id()); + msg->setInt32("generation", mTimeoutGeneration); + msg->post(kTimeoutDelayUs); +} + +void TimeSyncer::cancelTimeout() { + ++mTimeoutGeneration; +} + +void TimeSyncer::notifyError(status_t err) { + if (mNotify == NULL) { + looper()->stop(); + return; + } + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatError); + notify->setInt32("err", err); + notify->post(); +} + +// static +int TimeSyncer::CompareRountripTime(const TimeInfo *ti1, const TimeInfo *ti2) { + int64_t rt1 = ti1->mT4 - ti1->mT1; + int64_t rt2 = ti2->mT4 - ti2->mT1; + + if (rt1 < rt2) { + return -1; + } else if (rt1 > rt2) { + return 1; + } + + return 0; +} + +void TimeSyncer::notifyOffset() { + mHistory.sort(CompareRountripTime); + + int64_t sum = 0ll; + size_t count = 0; + + // Only consider the third of the information associated with the best + // (smallest) roundtrip times. + for (size_t i = 0; i < mHistory.size() / 3; ++i) { + const TimeInfo *ti = &mHistory[i]; + +#if 0 + // One way delay for a packet to travel from client + // to server or back (assumed to be the same either way). + int64_t delay = + (ti->mT2 - ti->mT1 + ti->mT4 - ti->mT3) / 2; +#endif + + // Offset between the client clock (T1, T4) and the + // server clock (T2, T3) timestamps. + int64_t offset = + (ti->mT2 - ti->mT1 - ti->mT4 + ti->mT3) / 2; + + ALOGV("(%d) RT: %lld us, offset: %lld us", + i, ti->mT4 - ti->mT1, offset); + + sum += offset; + ++count; + } + + if (mNotify == NULL) { + ALOGI("avg. offset is %lld", sum / count); + return; + } + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatTimeOffset); + notify->setInt64("offset", sum / count); + notify->post(); +} + +} // namespace android diff --git a/media/libstagefright/wifi-display/TimeSyncer.h b/media/libstagefright/wifi-display/TimeSyncer.h new file mode 100644 index 0000000..4e7571f --- /dev/null +++ b/media/libstagefright/wifi-display/TimeSyncer.h @@ -0,0 +1,109 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef TIME_SYNCER_H_ + +#define TIME_SYNCER_H_ + +#include <media/stagefright/foundation/AHandler.h> + +namespace android { + +struct ANetworkSession; + +/* + TimeSyncer allows us to synchronize time between a client and a server. + The client sends a UDP packet containing its send-time to the server, + the server sends that packet back to the client amended with information + about when it was received as well as the time the reply was sent back. + Finally the client receives the reply and has now enough information to + compute the clock offset between client and server assuming that packet + exchange is symmetric, i.e. time for a packet client->server and + server->client is roughly equal. + This exchange is repeated a number of times and the average offset computed + over the 30% of packets that had the lowest roundtrip times. + The offset is determined every 10 secs to account for slight differences in + clock frequency. +*/ +struct TimeSyncer : public AHandler { + enum { + kWhatError, + kWhatTimeOffset, + }; + TimeSyncer( + const sp<ANetworkSession> &netSession, + const sp<AMessage> ¬ify); + + void startServer(unsigned localPort); + void startClient(const char *remoteHost, unsigned remotePort); + +protected: + virtual ~TimeSyncer(); + + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kWhatStartServer, + kWhatStartClient, + kWhatUDPNotify, + kWhatSendPacket, + kWhatTimedOut, + }; + + struct TimeInfo { + int64_t mT1; // client timestamp at send + int64_t mT2; // server timestamp at receive + int64_t mT3; // server timestamp at send + int64_t mT4; // client timestamp at receive + }; + + enum { + kNumPacketsPerBatch = 30, + }; + static const int64_t kTimeoutDelayUs = 500000ll; + static const int64_t kBatchDelayUs = 60000000ll; // every minute + + sp<ANetworkSession> mNetSession; + sp<AMessage> mNotify; + + bool mIsServer; + bool mConnected; + int32_t mUDPSession; + uint32_t mSeqNo; + double mTotalTimeUs; + + Vector<TimeInfo> mHistory; + + int64_t mPendingT1; + int32_t mTimeoutGeneration; + + void postSendPacket(int64_t delayUs = 0ll); + + void postTimeout(); + void cancelTimeout(); + + void notifyError(status_t err); + void notifyOffset(); + + static int CompareRountripTime(const TimeInfo *ti1, const TimeInfo *ti2); + + DISALLOW_EVIL_CONSTRUCTORS(TimeSyncer); +}; + +} // namespace android + +#endif // TIME_SYNCER_H_ diff --git a/media/libstagefright/wifi-display/VideoFormats.cpp b/media/libstagefright/wifi-display/VideoFormats.cpp index 458b163..04e02c1 100644 --- a/media/libstagefright/wifi-display/VideoFormats.cpp +++ b/media/libstagefright/wifi-display/VideoFormats.cpp @@ -24,7 +24,8 @@ namespace android { -VideoFormats::config_t VideoFormats::mConfigs[][32] = { +// static +const VideoFormats::config_t VideoFormats::mResolutionTable[][32] = { { // CEA Resolutions { 640, 480, 60, false, 0, 0}, @@ -133,6 +134,8 @@ VideoFormats::config_t VideoFormats::mConfigs[][32] = { }; VideoFormats::VideoFormats() { + memcpy(mConfigs, mResolutionTable, sizeof(mConfigs)); + for (size_t i = 0; i < kNumResolutionTypes; ++i) { mResolutionEnabled[i] = 0; } @@ -175,6 +178,29 @@ void VideoFormats::enableAll() { } } +void VideoFormats::enableResolutionUpto( + ResolutionType type, size_t index, + ProfileType profile, LevelType level) { + size_t width, height, fps, score; + bool interlaced; + if (!GetConfiguration(type, index, &width, &height, + &fps, &interlaced)) { + ALOGE("Maximum resolution not found!"); + return; + } + score = width * height * fps * (!interlaced + 1); + for (size_t i = 0; i < kNumResolutionTypes; ++i) { + for (size_t j = 0; j < 32; j++) { + if (GetConfiguration((ResolutionType)i, j, + &width, &height, &fps, &interlaced) + && score >= width * height * fps * (!interlaced + 1)) { + setResolutionEnabled((ResolutionType)i, j); + setProfileLevel((ResolutionType)i, j, profile, level); + } + } + } +} + void VideoFormats::setResolutionEnabled( ResolutionType type, size_t index, bool enabled) { CHECK_LT(type, kNumResolutionTypes); @@ -182,11 +208,56 @@ void VideoFormats::setResolutionEnabled( if (enabled) { mResolutionEnabled[type] |= (1ul << index); + mConfigs[type][index].profile = (1ul << PROFILE_CBP); + mConfigs[type][index].level = (1ul << LEVEL_31); } else { mResolutionEnabled[type] &= ~(1ul << index); + mConfigs[type][index].profile = 0; + mConfigs[type][index].level = 0; } } +void VideoFormats::setProfileLevel( + ResolutionType type, size_t index, + ProfileType profile, LevelType level) { + CHECK_LT(type, kNumResolutionTypes); + CHECK(GetConfiguration(type, index, NULL, NULL, NULL, NULL)); + + mConfigs[type][index].profile = (1ul << profile); + mConfigs[type][index].level = (1ul << level); +} + +void VideoFormats::getProfileLevel( + ResolutionType type, size_t index, + ProfileType *profile, LevelType *level) const{ + CHECK_LT(type, kNumResolutionTypes); + CHECK(GetConfiguration(type, index, NULL, NULL, NULL, NULL)); + + int i, bestProfile = -1, bestLevel = -1; + + for (i = 0; i < kNumProfileTypes; ++i) { + if (mConfigs[type][index].profile & (1ul << i)) { + bestProfile = i; + } + } + + for (i = 0; i < kNumLevelTypes; ++i) { + if (mConfigs[type][index].level & (1ul << i)) { + bestLevel = i; + } + } + + if (bestProfile == -1 || bestLevel == -1) { + ALOGE("Profile or level not set for resolution type %d, index %d", + type, index); + bestProfile = PROFILE_CBP; + bestLevel = LEVEL_31; + } + + *profile = (ProfileType) bestProfile; + *level = (LevelType) bestLevel; +} + bool VideoFormats::isResolutionEnabled( ResolutionType type, size_t index) const { CHECK_LT(type, kNumResolutionTypes); @@ -207,7 +278,7 @@ bool VideoFormats::GetConfiguration( return false; } - const config_t *config = &mConfigs[type][index]; + const config_t *config = &mResolutionTable[type][index]; if (config->width == 0) { return false; @@ -251,9 +322,12 @@ bool VideoFormats::parseH264Codec(const char *spec) { if (res[i] & (1ul << j)){ mResolutionEnabled[i] |= (1ul << j); if (profile > mConfigs[i][j].profile) { + // prefer higher profile (even if level is lower) mConfigs[i][j].profile = profile; - if (level > mConfigs[i][j].level) - mConfigs[i][j].level = level; + mConfigs[i][j].level = level; + } else if (profile == mConfigs[i][j].profile && + level > mConfigs[i][j].level) { + mConfigs[i][j].level = level; } } } @@ -262,9 +336,51 @@ bool VideoFormats::parseH264Codec(const char *spec) { return true; } +// static +bool VideoFormats::GetProfileLevel( + ProfileType profile, LevelType level, unsigned *profileIdc, + unsigned *levelIdc, unsigned *constraintSet) { + CHECK_LT(profile, kNumProfileTypes); + CHECK_LT(level, kNumLevelTypes); + + static const unsigned kProfileIDC[kNumProfileTypes] = { + 66, // PROFILE_CBP + 100, // PROFILE_CHP + }; + + static const unsigned kLevelIDC[kNumLevelTypes] = { + 31, // LEVEL_31 + 32, // LEVEL_32 + 40, // LEVEL_40 + 41, // LEVEL_41 + 42, // LEVEL_42 + }; + + static const unsigned kConstraintSet[kNumProfileTypes] = { + 0xc0, // PROFILE_CBP + 0x0c, // PROFILE_CHP + }; + + if (profileIdc) { + *profileIdc = kProfileIDC[profile]; + } + + if (levelIdc) { + *levelIdc = kLevelIDC[level]; + } + + if (constraintSet) { + *constraintSet = kConstraintSet[profile]; + } + + return true; +} + bool VideoFormats::parseFormatSpec(const char *spec) { CHECK_EQ(kNumResolutionTypes, 3); + disableAll(); + unsigned native, dummy; unsigned res[3]; size_t size = strlen(spec); @@ -320,8 +436,10 @@ AString VideoFormats::getFormatSpec(bool forM4Message) const { // max-vres (none or 2 byte) return StringPrintf( - "%02x 00 02 02 %08x %08x %08x 00 0000 0000 00 none none", + "%02x 00 %02x %02x %08x %08x %08x 00 0000 0000 00 none none", forM4Message ? 0x00 : ((mNativeIndex << 3) | mNativeType), + mConfigs[mNativeType][mNativeIndex].profile, + mConfigs[mNativeType][mNativeIndex].level, mResolutionEnabled[0], mResolutionEnabled[1], mResolutionEnabled[2]); @@ -332,7 +450,9 @@ bool VideoFormats::PickBestFormat( const VideoFormats &sinkSupported, const VideoFormats &sourceSupported, ResolutionType *chosenType, - size_t *chosenIndex) { + size_t *chosenIndex, + ProfileType *chosenProfile, + LevelType *chosenLevel) { #if 0 // Support for the native format is a great idea, the spec includes // these features, but nobody supports it and the tests don't validate it. @@ -412,6 +532,18 @@ bool VideoFormats::PickBestFormat( *chosenType = (ResolutionType)bestType; *chosenIndex = bestIndex; + // Pick the best profile/level supported by both sink and source. + ProfileType srcProfile, sinkProfile; + LevelType srcLevel, sinkLevel; + sourceSupported.getProfileLevel( + (ResolutionType)bestType, bestIndex, + &srcProfile, &srcLevel); + sinkSupported.getProfileLevel( + (ResolutionType)bestType, bestIndex, + &sinkProfile, &sinkLevel); + *chosenProfile = srcProfile < sinkProfile ? srcProfile : sinkProfile; + *chosenLevel = srcLevel < sinkLevel ? srcLevel : sinkLevel; + return true; } diff --git a/media/libstagefright/wifi-display/VideoFormats.h b/media/libstagefright/wifi-display/VideoFormats.h index 01de246..fd38fd1 100644 --- a/media/libstagefright/wifi-display/VideoFormats.h +++ b/media/libstagefright/wifi-display/VideoFormats.h @@ -69,17 +69,33 @@ struct VideoFormats { void disableAll(); void enableAll(); + void enableResolutionUpto( + ResolutionType type, size_t index, + ProfileType profile, LevelType level); void setResolutionEnabled( ResolutionType type, size_t index, bool enabled = true); bool isResolutionEnabled(ResolutionType type, size_t index) const; + void setProfileLevel( + ResolutionType type, size_t index, + ProfileType profile, LevelType level); + + void getProfileLevel( + ResolutionType type, size_t index, + ProfileType *profile, LevelType *level) const; + static bool GetConfiguration( ResolutionType type, size_t index, size_t *width, size_t *height, size_t *framesPerSecond, bool *interlaced); + static bool GetProfileLevel( + ProfileType profile, LevelType level, + unsigned *profileIdc, unsigned *levelIdc, + unsigned *constraintSet); + bool parseFormatSpec(const char *spec); AString getFormatSpec(bool forM4Message = false) const; @@ -87,7 +103,9 @@ struct VideoFormats { const VideoFormats &sinkSupported, const VideoFormats &sourceSupported, ResolutionType *chosenType, - size_t *chosenIndex); + size_t *chosenIndex, + ProfileType *chosenProfile, + LevelType *chosenLevel); private: bool parseH264Codec(const char *spec); @@ -95,7 +113,8 @@ private: size_t mNativeIndex; uint32_t mResolutionEnabled[kNumResolutionTypes]; - static config_t mConfigs[kNumResolutionTypes][32]; + static const config_t mResolutionTable[kNumResolutionTypes][32]; + config_t mConfigs[kNumResolutionTypes][32]; DISALLOW_EVIL_CONSTRUCTORS(VideoFormats); }; diff --git a/media/libstagefright/wifi-display/nettest.cpp b/media/libstagefright/wifi-display/nettest.cpp new file mode 100644 index 0000000..0779bf5 --- /dev/null +++ b/media/libstagefright/wifi-display/nettest.cpp @@ -0,0 +1,400 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NEBUG 0 +#define LOG_TAG "nettest" +#include <utils/Log.h> + +#include "ANetworkSession.h" +#include "TimeSyncer.h" + +#include <binder/ProcessState.h> +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AHandler.h> +#include <media/stagefright/foundation/ALooper.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/foundation/hexdump.h> +#include <media/stagefright/DataSource.h> +#include <media/stagefright/MediaDefs.h> +#include <media/stagefright/NuMediaExtractor.h> +#include <media/stagefright/Utils.h> + +namespace android { + +struct TestHandler : public AHandler { + TestHandler(const sp<ANetworkSession> &netSession); + + void listen(int32_t port); + void connect(const char *host, int32_t port); + +protected: + virtual ~TestHandler(); + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kTimeSyncerPort = 8123, + }; + + enum { + kWhatListen, + kWhatConnect, + kWhatTimeSyncerNotify, + kWhatNetNotify, + kWhatSendMore, + kWhatStop, + }; + + sp<ANetworkSession> mNetSession; + sp<TimeSyncer> mTimeSyncer; + + int32_t mServerSessionID; + int32_t mSessionID; + + int64_t mTimeOffsetUs; + bool mTimeOffsetValid; + + int32_t mCounter; + + int64_t mMaxDelayMs; + + void dumpDelay(int32_t counter, int64_t delayMs); + + DISALLOW_EVIL_CONSTRUCTORS(TestHandler); +}; + +TestHandler::TestHandler(const sp<ANetworkSession> &netSession) + : mNetSession(netSession), + mServerSessionID(0), + mSessionID(0), + mTimeOffsetUs(-1ll), + mTimeOffsetValid(false), + mCounter(0), + mMaxDelayMs(-1ll) { +} + +TestHandler::~TestHandler() { +} + +void TestHandler::listen(int32_t port) { + sp<AMessage> msg = new AMessage(kWhatListen, id()); + msg->setInt32("port", port); + msg->post(); +} + +void TestHandler::connect(const char *host, int32_t port) { + sp<AMessage> msg = new AMessage(kWhatConnect, id()); + msg->setString("host", host); + msg->setInt32("port", port); + msg->post(); +} + +void TestHandler::dumpDelay(int32_t counter, int64_t delayMs) { + static const int64_t kMinDelayMs = 0; + static const int64_t kMaxDelayMs = 300; + + const char *kPattern = "########################################"; + size_t kPatternSize = strlen(kPattern); + + int n = (kPatternSize * (delayMs - kMinDelayMs)) + / (kMaxDelayMs - kMinDelayMs); + + if (n < 0) { + n = 0; + } else if ((size_t)n > kPatternSize) { + n = kPatternSize; + } + + if (delayMs > mMaxDelayMs) { + mMaxDelayMs = delayMs; + } + + ALOGI("[%d] (%4lld ms / %4lld ms) %s", + counter, + delayMs, + mMaxDelayMs, + kPattern + kPatternSize - n); +} + +void TestHandler::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatListen: + { + sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id()); + mTimeSyncer = new TimeSyncer(mNetSession, notify); + looper()->registerHandler(mTimeSyncer); + + notify = new AMessage(kWhatNetNotify, id()); + + int32_t port; + CHECK(msg->findInt32("port", &port)); + + struct in_addr ifaceAddr; + ifaceAddr.s_addr = INADDR_ANY; + + CHECK_EQ((status_t)OK, + mNetSession->createTCPDatagramSession( + ifaceAddr, + port, + notify, + &mServerSessionID)); + break; + } + + case kWhatConnect: + { + sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id()); + mTimeSyncer = new TimeSyncer(mNetSession, notify); + looper()->registerHandler(mTimeSyncer); + mTimeSyncer->startServer(kTimeSyncerPort); + + AString host; + CHECK(msg->findString("host", &host)); + + int32_t port; + CHECK(msg->findInt32("port", &port)); + + notify = new AMessage(kWhatNetNotify, id()); + + CHECK_EQ((status_t)OK, + mNetSession->createTCPDatagramSession( + 0 /* localPort */, + host.c_str(), + port, + notify, + &mSessionID)); + break; + } + + case kWhatNetNotify: + { + int32_t reason; + CHECK(msg->findInt32("reason", &reason)); + + switch (reason) { + case ANetworkSession::kWhatConnected: + { + ALOGI("kWhatConnected"); + + (new AMessage(kWhatSendMore, id()))->post(); + break; + } + + case ANetworkSession::kWhatClientConnected: + { + ALOGI("kWhatClientConnected"); + + CHECK_EQ(mSessionID, 0); + CHECK(msg->findInt32("sessionID", &mSessionID)); + + AString clientIP; + CHECK(msg->findString("client-ip", &clientIP)); + + mTimeSyncer->startClient(clientIP.c_str(), kTimeSyncerPort); + break; + } + + case ANetworkSession::kWhatDatagram: + { + sp<ABuffer> packet; + CHECK(msg->findBuffer("data", &packet)); + + CHECK_EQ(packet->size(), 12u); + + int32_t counter = U32_AT(packet->data()); + int64_t timeUs = U64_AT(packet->data() + 4); + + if (mTimeOffsetValid) { + timeUs -= mTimeOffsetUs; + int64_t nowUs = ALooper::GetNowUs(); + int64_t delayMs = (nowUs - timeUs) / 1000ll; + + dumpDelay(counter, delayMs); + } else { + ALOGI("received %d", counter); + } + break; + } + + case ANetworkSession::kWhatError: + { + ALOGE("kWhatError"); + break; + } + + default: + TRESPASS(); + } + break; + } + + case kWhatTimeSyncerNotify: + { + CHECK(msg->findInt64("offset", &mTimeOffsetUs)); + mTimeOffsetValid = true; + break; + } + + case kWhatSendMore: + { + uint8_t buffer[4 + 8]; + buffer[0] = mCounter >> 24; + buffer[1] = (mCounter >> 16) & 0xff; + buffer[2] = (mCounter >> 8) & 0xff; + buffer[3] = mCounter & 0xff; + + int64_t nowUs = ALooper::GetNowUs(); + + buffer[4] = nowUs >> 56; + buffer[5] = (nowUs >> 48) & 0xff; + buffer[6] = (nowUs >> 40) & 0xff; + buffer[7] = (nowUs >> 32) & 0xff; + buffer[8] = (nowUs >> 24) & 0xff; + buffer[9] = (nowUs >> 16) & 0xff; + buffer[10] = (nowUs >> 8) & 0xff; + buffer[11] = nowUs & 0xff; + + ++mCounter; + + CHECK_EQ((status_t)OK, + mNetSession->sendRequest( + mSessionID, + buffer, + sizeof(buffer), + true /* timeValid */, + nowUs)); + + msg->post(100000ll); + break; + } + + case kWhatStop: + { + if (mSessionID != 0) { + mNetSession->destroySession(mSessionID); + mSessionID = 0; + } + + if (mServerSessionID != 0) { + mNetSession->destroySession(mServerSessionID); + mServerSessionID = 0; + } + + looper()->stop(); + break; + } + + default: + TRESPASS(); + } +} + +} // namespace android + +static void usage(const char *me) { + fprintf(stderr, + "usage: %s -c host:port\tconnect to remote host\n" + " -l port \tlisten\n", + me); +} + +int main(int argc, char **argv) { + using namespace android; + + // srand(time(NULL)); + + ProcessState::self()->startThreadPool(); + + DataSource::RegisterDefaultSniffers(); + + int32_t connectToPort = -1; + AString connectToHost; + + int32_t listenOnPort = -1; + + int res; + while ((res = getopt(argc, argv, "hc:l:")) >= 0) { + switch (res) { + case 'c': + { + const char *colonPos = strrchr(optarg, ':'); + + if (colonPos == NULL) { + usage(argv[0]); + exit(1); + } + + connectToHost.setTo(optarg, colonPos - optarg); + + char *end; + connectToPort = strtol(colonPos + 1, &end, 10); + + if (*end != '\0' || end == colonPos + 1 + || connectToPort < 0 || connectToPort > 65535) { + fprintf(stderr, "Illegal port specified.\n"); + exit(1); + } + break; + } + + case 'l': + { + char *end; + listenOnPort = strtol(optarg, &end, 10); + + if (*end != '\0' || end == optarg + || listenOnPort < 0 || listenOnPort > 65535) { + fprintf(stderr, "Illegal port specified.\n"); + exit(1); + } + break; + } + + case '?': + case 'h': + usage(argv[0]); + exit(1); + } + } + + if ((listenOnPort < 0 && connectToPort < 0) + || (listenOnPort >= 0 && connectToPort >= 0)) { + fprintf(stderr, + "You need to select either client or server mode.\n"); + exit(1); + } + + sp<ANetworkSession> netSession = new ANetworkSession; + netSession->start(); + + sp<ALooper> looper = new ALooper; + + sp<TestHandler> handler = new TestHandler(netSession); + looper->registerHandler(handler); + + if (listenOnPort) { + handler->listen(listenOnPort); + } + + if (connectToPort >= 0) { + handler->connect(connectToHost.c_str(), connectToPort); + } + + looper->start(true /* runOnCallingThread */); + + return 0; +} diff --git a/media/libstagefright/wifi-display/rtp/RTPAssembler.cpp b/media/libstagefright/wifi-display/rtp/RTPAssembler.cpp new file mode 100644 index 0000000..7a96081 --- /dev/null +++ b/media/libstagefright/wifi-display/rtp/RTPAssembler.cpp @@ -0,0 +1,328 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "RTPAssembler" +#include <utils/Log.h> + +#include "RTPAssembler.h" + +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/foundation/hexdump.h> +#include <media/stagefright/MediaErrors.h> + +namespace android { + +RTPReceiver::Assembler::Assembler(const sp<AMessage> ¬ify) + : mNotify(notify) { +} + +void RTPReceiver::Assembler::postAccessUnit( + const sp<ABuffer> &accessUnit, bool followsDiscontinuity) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", RTPReceiver::kWhatAccessUnit); + notify->setBuffer("accessUnit", accessUnit); + notify->setInt32("followsDiscontinuity", followsDiscontinuity); + notify->post(); +} +//////////////////////////////////////////////////////////////////////////////// + +RTPReceiver::TSAssembler::TSAssembler(const sp<AMessage> ¬ify) + : Assembler(notify), + mSawDiscontinuity(false) { +} + +void RTPReceiver::TSAssembler::signalDiscontinuity() { + mSawDiscontinuity = true; +} + +status_t RTPReceiver::TSAssembler::processPacket(const sp<ABuffer> &packet) { + int32_t rtpTime; + CHECK(packet->meta()->findInt32("rtp-time", &rtpTime)); + + packet->meta()->setInt64("timeUs", (rtpTime * 100ll) / 9); + + postAccessUnit(packet, mSawDiscontinuity); + + if (mSawDiscontinuity) { + mSawDiscontinuity = false; + } + + return OK; +} + +//////////////////////////////////////////////////////////////////////////////// + +RTPReceiver::H264Assembler::H264Assembler(const sp<AMessage> ¬ify) + : Assembler(notify), + mState(0), + mIndicator(0), + mNALType(0), + mAccessUnitRTPTime(0) { +} + +void RTPReceiver::H264Assembler::signalDiscontinuity() { + reset(); +} + +status_t RTPReceiver::H264Assembler::processPacket(const sp<ABuffer> &packet) { + status_t err = internalProcessPacket(packet); + + if (err != OK) { + reset(); + } + + return err; +} + +status_t RTPReceiver::H264Assembler::internalProcessPacket( + const sp<ABuffer> &packet) { + const uint8_t *data = packet->data(); + size_t size = packet->size(); + + switch (mState) { + case 0: + { + if (size < 1 || (data[0] & 0x80)) { + ALOGV("Malformed H264 RTP packet (empty or F-bit set)"); + return ERROR_MALFORMED; + } + + unsigned nalType = data[0] & 0x1f; + if (nalType >= 1 && nalType <= 23) { + addSingleNALUnit(packet); + ALOGV("added single NAL packet"); + } else if (nalType == 28) { + // FU-A + unsigned indicator = data[0]; + CHECK((indicator & 0x1f) == 28); + + if (size < 2) { + ALOGV("Malformed H264 FU-A packet (single byte)"); + return ERROR_MALFORMED; + } + + if (!(data[1] & 0x80)) { + ALOGV("Malformed H264 FU-A packet (no start bit)"); + return ERROR_MALFORMED; + } + + mIndicator = data[0]; + mNALType = data[1] & 0x1f; + uint32_t nri = (data[0] >> 5) & 3; + + clearAccumulator(); + + uint8_t byte = mNALType | (nri << 5); + appendToAccumulator(&byte, 1); + appendToAccumulator(data + 2, size - 2); + + int32_t rtpTime; + CHECK(packet->meta()->findInt32("rtp-time", &rtpTime)); + mAccumulator->meta()->setInt32("rtp-time", rtpTime); + + if (data[1] & 0x40) { + // Huh? End bit also set on the first buffer. + addSingleNALUnit(mAccumulator); + clearAccumulator(); + + ALOGV("added FU-A"); + break; + } + + mState = 1; + } else if (nalType == 24) { + // STAP-A + + status_t err = addSingleTimeAggregationPacket(packet); + if (err != OK) { + return err; + } + } else { + ALOGV("Malformed H264 packet (unknown type %d)", nalType); + return ERROR_UNSUPPORTED; + } + break; + } + + case 1: + { + if (size < 2 + || data[0] != mIndicator + || (data[1] & 0x1f) != mNALType + || (data[1] & 0x80)) { + ALOGV("Malformed H264 FU-A packet (indicator, " + "type or start bit mismatch)"); + + return ERROR_MALFORMED; + } + + appendToAccumulator(data + 2, size - 2); + + if (data[1] & 0x40) { + addSingleNALUnit(mAccumulator); + + clearAccumulator(); + mState = 0; + + ALOGV("added FU-A"); + } + break; + } + + default: + TRESPASS(); + } + + int32_t marker; + CHECK(packet->meta()->findInt32("M", &marker)); + + if (marker) { + flushAccessUnit(); + } + + return OK; +} + +void RTPReceiver::H264Assembler::reset() { + mNALUnits.clear(); + + clearAccumulator(); + mState = 0; +} + +void RTPReceiver::H264Assembler::clearAccumulator() { + if (mAccumulator != NULL) { + // XXX Too expensive. + mAccumulator.clear(); + } +} + +void RTPReceiver::H264Assembler::appendToAccumulator( + const void *data, size_t size) { + if (mAccumulator == NULL) { + mAccumulator = new ABuffer(size); + memcpy(mAccumulator->data(), data, size); + return; + } + + if (mAccumulator->size() + size > mAccumulator->capacity()) { + sp<ABuffer> buf = new ABuffer(mAccumulator->size() + size); + memcpy(buf->data(), mAccumulator->data(), mAccumulator->size()); + buf->setRange(0, mAccumulator->size()); + + int32_t rtpTime; + if (mAccumulator->meta()->findInt32("rtp-time", &rtpTime)) { + buf->meta()->setInt32("rtp-time", rtpTime); + } + + mAccumulator = buf; + } + + memcpy(mAccumulator->data() + mAccumulator->size(), data, size); + mAccumulator->setRange(0, mAccumulator->size() + size); +} + +void RTPReceiver::H264Assembler::addSingleNALUnit(const sp<ABuffer> &packet) { + if (mNALUnits.empty()) { + int32_t rtpTime; + CHECK(packet->meta()->findInt32("rtp-time", &rtpTime)); + + mAccessUnitRTPTime = rtpTime; + } + + mNALUnits.push_back(packet); +} + +void RTPReceiver::H264Assembler::flushAccessUnit() { + if (mNALUnits.empty()) { + return; + } + + size_t totalSize = 0; + for (List<sp<ABuffer> >::iterator it = mNALUnits.begin(); + it != mNALUnits.end(); ++it) { + totalSize += 4 + (*it)->size(); + } + + sp<ABuffer> accessUnit = new ABuffer(totalSize); + size_t offset = 0; + for (List<sp<ABuffer> >::iterator it = mNALUnits.begin(); + it != mNALUnits.end(); ++it) { + const sp<ABuffer> nalUnit = *it; + + memcpy(accessUnit->data() + offset, "\x00\x00\x00\x01", 4); + + memcpy(accessUnit->data() + offset + 4, + nalUnit->data(), + nalUnit->size()); + + offset += 4 + nalUnit->size(); + } + + mNALUnits.clear(); + + accessUnit->meta()->setInt64("timeUs", mAccessUnitRTPTime * 100ll / 9ll); + postAccessUnit(accessUnit, false /* followsDiscontinuity */); +} + +status_t RTPReceiver::H264Assembler::addSingleTimeAggregationPacket( + const sp<ABuffer> &packet) { + const uint8_t *data = packet->data(); + size_t size = packet->size(); + + if (size < 3) { + ALOGV("Malformed H264 STAP-A packet (too small)"); + return ERROR_MALFORMED; + } + + int32_t rtpTime; + CHECK(packet->meta()->findInt32("rtp-time", &rtpTime)); + + ++data; + --size; + while (size >= 2) { + size_t nalSize = (data[0] << 8) | data[1]; + + if (size < nalSize + 2) { + ALOGV("Malformed H264 STAP-A packet (incomplete NAL unit)"); + return ERROR_MALFORMED; + } + + sp<ABuffer> unit = new ABuffer(nalSize); + memcpy(unit->data(), &data[2], nalSize); + + unit->meta()->setInt32("rtp-time", rtpTime); + + addSingleNALUnit(unit); + + data += 2 + nalSize; + size -= 2 + nalSize; + } + + if (size != 0) { + ALOGV("Unexpected padding at end of STAP-A packet."); + } + + ALOGV("added STAP-A"); + + return OK; +} + +} // namespace android + diff --git a/media/libstagefright/wifi-display/rtp/RTPAssembler.h b/media/libstagefright/wifi-display/rtp/RTPAssembler.h new file mode 100644 index 0000000..e456d32 --- /dev/null +++ b/media/libstagefright/wifi-display/rtp/RTPAssembler.h @@ -0,0 +1,92 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef RTP_ASSEMBLER_H_ + +#define RTP_ASSEMBLER_H_ + +#include "RTPReceiver.h" + +namespace android { + +// A helper class to reassemble the payload of RTP packets into access +// units depending on the packetization scheme. +struct RTPReceiver::Assembler : public RefBase { + Assembler(const sp<AMessage> ¬ify); + + virtual void signalDiscontinuity() = 0; + virtual status_t processPacket(const sp<ABuffer> &packet) = 0; + +protected: + virtual ~Assembler() {} + + void postAccessUnit( + const sp<ABuffer> &accessUnit, bool followsDiscontinuity); + +private: + sp<AMessage> mNotify; + + DISALLOW_EVIL_CONSTRUCTORS(Assembler); +}; + +struct RTPReceiver::TSAssembler : public RTPReceiver::Assembler { + TSAssembler(const sp<AMessage> ¬ify); + + virtual void signalDiscontinuity(); + virtual status_t processPacket(const sp<ABuffer> &packet); + +private: + bool mSawDiscontinuity; + + DISALLOW_EVIL_CONSTRUCTORS(TSAssembler); +}; + +struct RTPReceiver::H264Assembler : public RTPReceiver::Assembler { + H264Assembler(const sp<AMessage> ¬ify); + + virtual void signalDiscontinuity(); + virtual status_t processPacket(const sp<ABuffer> &packet); + +private: + int32_t mState; + + uint8_t mIndicator; + uint8_t mNALType; + + sp<ABuffer> mAccumulator; + + List<sp<ABuffer> > mNALUnits; + int32_t mAccessUnitRTPTime; + + status_t internalProcessPacket(const sp<ABuffer> &packet); + + void addSingleNALUnit(const sp<ABuffer> &packet); + status_t addSingleTimeAggregationPacket(const sp<ABuffer> &packet); + + void flushAccessUnit(); + + void clearAccumulator(); + void appendToAccumulator(const void *data, size_t size); + + void reset(); + + DISALLOW_EVIL_CONSTRUCTORS(H264Assembler); +}; + +} // namespace android + +#endif // RTP_ASSEMBLER_H_ + diff --git a/media/libstagefright/wifi-display/rtp/RTPReceiver.cpp b/media/libstagefright/wifi-display/rtp/RTPReceiver.cpp new file mode 100644 index 0000000..2d22e79 --- /dev/null +++ b/media/libstagefright/wifi-display/rtp/RTPReceiver.cpp @@ -0,0 +1,1153 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "RTPReceiver" +#include <utils/Log.h> + +#include "RTPAssembler.h" +#include "RTPReceiver.h" + +#include "ANetworkSession.h" + +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/foundation/hexdump.h> +#include <media/stagefright/MediaErrors.h> +#include <media/stagefright/Utils.h> + +#define TRACK_PACKET_LOSS 0 + +namespace android { + +//////////////////////////////////////////////////////////////////////////////// + +struct RTPReceiver::Source : public AHandler { + Source(RTPReceiver *receiver, uint32_t ssrc); + + void onPacketReceived(uint16_t seq, const sp<ABuffer> &buffer); + + void addReportBlock(uint32_t ssrc, const sp<ABuffer> &buf); + +protected: + virtual ~Source(); + + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kWhatRetransmit, + kWhatDeclareLost, + }; + + static const uint32_t kMinSequential = 2; + static const uint32_t kMaxDropout = 3000; + static const uint32_t kMaxMisorder = 100; + static const uint32_t kRTPSeqMod = 1u << 16; + static const int64_t kReportIntervalUs = 10000000ll; + + RTPReceiver *mReceiver; + uint32_t mSSRC; + bool mFirst; + uint16_t mMaxSeq; + uint32_t mCycles; + uint32_t mBaseSeq; + uint32_t mReceived; + uint32_t mExpectedPrior; + uint32_t mReceivedPrior; + + int64_t mFirstArrivalTimeUs; + int64_t mFirstRTPTimeUs; + + // Ordered by extended seq number. + List<sp<ABuffer> > mPackets; + + enum StatusBits { + STATUS_DECLARED_LOST = 1, + STATUS_REQUESTED_RETRANSMISSION = 2, + STATUS_ARRIVED_LATE = 4, + }; +#if TRACK_PACKET_LOSS + KeyedVector<int32_t, uint32_t> mLostPackets; +#endif + + void modifyPacketStatus(int32_t extSeqNo, uint32_t mask); + + int32_t mAwaitingExtSeqNo; + bool mRequestedRetransmission; + + int32_t mActivePacketType; + sp<Assembler> mActiveAssembler; + + int64_t mNextReportTimeUs; + + int32_t mNumDeclaredLost; + int32_t mNumDeclaredLostPrior; + + int32_t mRetransmitGeneration; + int32_t mDeclareLostGeneration; + bool mDeclareLostTimerPending; + + void queuePacket(const sp<ABuffer> &packet); + void dequeueMore(); + + sp<ABuffer> getNextPacket(); + void resync(); + + void postRetransmitTimer(int64_t delayUs); + void postDeclareLostTimer(int64_t delayUs); + void cancelTimers(); + + DISALLOW_EVIL_CONSTRUCTORS(Source); +}; + +//////////////////////////////////////////////////////////////////////////////// + +RTPReceiver::Source::Source(RTPReceiver *receiver, uint32_t ssrc) + : mReceiver(receiver), + mSSRC(ssrc), + mFirst(true), + mMaxSeq(0), + mCycles(0), + mBaseSeq(0), + mReceived(0), + mExpectedPrior(0), + mReceivedPrior(0), + mFirstArrivalTimeUs(-1ll), + mFirstRTPTimeUs(-1ll), + mAwaitingExtSeqNo(-1), + mRequestedRetransmission(false), + mActivePacketType(-1), + mNextReportTimeUs(-1ll), + mNumDeclaredLost(0), + mNumDeclaredLostPrior(0), + mRetransmitGeneration(0), + mDeclareLostGeneration(0), + mDeclareLostTimerPending(false) { +} + +RTPReceiver::Source::~Source() { +} + +void RTPReceiver::Source::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatRetransmit: + { + int32_t generation; + CHECK(msg->findInt32("generation", &generation)); + + if (generation != mRetransmitGeneration) { + break; + } + + mRequestedRetransmission = true; + mReceiver->requestRetransmission(mSSRC, mAwaitingExtSeqNo); + + modifyPacketStatus( + mAwaitingExtSeqNo, STATUS_REQUESTED_RETRANSMISSION); + break; + } + + case kWhatDeclareLost: + { + int32_t generation; + CHECK(msg->findInt32("generation", &generation)); + + if (generation != mDeclareLostGeneration) { + break; + } + + cancelTimers(); + + ALOGV("Lost packet extSeqNo %d %s", + mAwaitingExtSeqNo, + mRequestedRetransmission ? "*" : ""); + + mRequestedRetransmission = false; + if (mActiveAssembler != NULL) { + mActiveAssembler->signalDiscontinuity(); + } + + modifyPacketStatus(mAwaitingExtSeqNo, STATUS_DECLARED_LOST); + + // resync(); + ++mAwaitingExtSeqNo; + ++mNumDeclaredLost; + + mReceiver->notifyPacketLost(); + + dequeueMore(); + break; + } + + default: + TRESPASS(); + } +} + +void RTPReceiver::Source::onPacketReceived( + uint16_t seq, const sp<ABuffer> &buffer) { + if (mFirst) { + buffer->setInt32Data(mCycles | seq); + queuePacket(buffer); + + mFirst = false; + mBaseSeq = seq; + mMaxSeq = seq; + ++mReceived; + return; + } + + uint16_t udelta = seq - mMaxSeq; + + if (udelta < kMaxDropout) { + // In order, with permissible gap. + + if (seq < mMaxSeq) { + // Sequence number wrapped - count another 64K cycle + mCycles += kRTPSeqMod; + } + + mMaxSeq = seq; + + ++mReceived; + } else if (udelta <= kRTPSeqMod - kMaxMisorder) { + // The sequence number made a very large jump + return; + } else { + // Duplicate or reordered packet. + } + + buffer->setInt32Data(mCycles | seq); + queuePacket(buffer); +} + +void RTPReceiver::Source::queuePacket(const sp<ABuffer> &packet) { + int32_t newExtendedSeqNo = packet->int32Data(); + + if (mFirstArrivalTimeUs < 0ll) { + mFirstArrivalTimeUs = ALooper::GetNowUs(); + + uint32_t rtpTime; + CHECK(packet->meta()->findInt32("rtp-time", (int32_t *)&rtpTime)); + + mFirstRTPTimeUs = (rtpTime * 100ll) / 9ll; + } + + if (mAwaitingExtSeqNo >= 0 && newExtendedSeqNo < mAwaitingExtSeqNo) { + // We're no longer interested in these. They're old. + ALOGV("dropping stale extSeqNo %d", newExtendedSeqNo); + + modifyPacketStatus(newExtendedSeqNo, STATUS_ARRIVED_LATE); + return; + } + + if (mPackets.empty()) { + mPackets.push_back(packet); + dequeueMore(); + return; + } + + List<sp<ABuffer> >::iterator firstIt = mPackets.begin(); + List<sp<ABuffer> >::iterator it = --mPackets.end(); + for (;;) { + int32_t extendedSeqNo = (*it)->int32Data(); + + if (extendedSeqNo == newExtendedSeqNo) { + // Duplicate packet. + return; + } + + if (extendedSeqNo < newExtendedSeqNo) { + // Insert new packet after the one at "it". + mPackets.insert(++it, packet); + break; + } + + if (it == firstIt) { + // Insert new packet before the first existing one. + mPackets.insert(it, packet); + break; + } + + --it; + } + + dequeueMore(); +} + +void RTPReceiver::Source::dequeueMore() { + int64_t nowUs = ALooper::GetNowUs(); + if (mNextReportTimeUs < 0ll || nowUs >= mNextReportTimeUs) { + if (mNextReportTimeUs >= 0ll) { + uint32_t expected = (mMaxSeq | mCycles) - mBaseSeq + 1; + + uint32_t expectedInterval = expected - mExpectedPrior; + mExpectedPrior = expected; + + uint32_t receivedInterval = mReceived - mReceivedPrior; + mReceivedPrior = mReceived; + + int64_t lostInterval = + (int64_t)expectedInterval - (int64_t)receivedInterval; + + int32_t declaredLostInterval = + mNumDeclaredLost - mNumDeclaredLostPrior; + + mNumDeclaredLostPrior = mNumDeclaredLost; + + if (declaredLostInterval > 0) { + ALOGI("lost %lld packets (%.2f %%), declared %d lost\n", + lostInterval, + 100.0f * lostInterval / expectedInterval, + declaredLostInterval); + } + } + + mNextReportTimeUs = nowUs + kReportIntervalUs; + +#if TRACK_PACKET_LOSS + for (size_t i = 0; i < mLostPackets.size(); ++i) { + int32_t key = mLostPackets.keyAt(i); + uint32_t value = mLostPackets.valueAt(i); + + AString status; + if (value & STATUS_REQUESTED_RETRANSMISSION) { + status.append("retrans "); + } + if (value & STATUS_ARRIVED_LATE) { + status.append("arrived-late "); + } + ALOGI("Packet %d declared lost %s", key, status.c_str()); + } +#endif + } + + sp<ABuffer> packet; + while ((packet = getNextPacket()) != NULL) { + if (mDeclareLostTimerPending) { + cancelTimers(); + } + + CHECK_GE(mAwaitingExtSeqNo, 0); +#if TRACK_PACKET_LOSS + mLostPackets.removeItem(mAwaitingExtSeqNo); +#endif + + int32_t packetType; + CHECK(packet->meta()->findInt32("PT", &packetType)); + + if (packetType != mActivePacketType) { + mActiveAssembler = mReceiver->makeAssembler(packetType); + mActivePacketType = packetType; + } + + if (mActiveAssembler != NULL) { + status_t err = mActiveAssembler->processPacket(packet); + if (err != OK) { + ALOGV("assembler returned error %d", err); + } + } + + ++mAwaitingExtSeqNo; + } + + if (mDeclareLostTimerPending) { + return; + } + + if (mPackets.empty()) { + return; + } + + CHECK_GE(mAwaitingExtSeqNo, 0); + + const sp<ABuffer> &firstPacket = *mPackets.begin(); + + uint32_t rtpTime; + CHECK(firstPacket->meta()->findInt32( + "rtp-time", (int32_t *)&rtpTime)); + + + int64_t rtpUs = (rtpTime * 100ll) / 9ll; + + int64_t maxArrivalTimeUs = + mFirstArrivalTimeUs + rtpUs - mFirstRTPTimeUs; + + nowUs = ALooper::GetNowUs(); + + CHECK_LT(mAwaitingExtSeqNo, firstPacket->int32Data()); + + ALOGV("waiting for %d, comparing against %d, %lld us left", + mAwaitingExtSeqNo, + firstPacket->int32Data(), + maxArrivalTimeUs - nowUs); + + postDeclareLostTimer(maxArrivalTimeUs + kPacketLostAfterUs); + + if (kRequestRetransmissionAfterUs > 0ll) { + postRetransmitTimer( + maxArrivalTimeUs + kRequestRetransmissionAfterUs); + } +} + +sp<ABuffer> RTPReceiver::Source::getNextPacket() { + if (mPackets.empty()) { + return NULL; + } + + int32_t extSeqNo = (*mPackets.begin())->int32Data(); + + if (mAwaitingExtSeqNo < 0) { + mAwaitingExtSeqNo = extSeqNo; + } else if (extSeqNo != mAwaitingExtSeqNo) { + return NULL; + } + + sp<ABuffer> packet = *mPackets.begin(); + mPackets.erase(mPackets.begin()); + + return packet; +} + +void RTPReceiver::Source::resync() { + mAwaitingExtSeqNo = -1; +} + +void RTPReceiver::Source::addReportBlock( + uint32_t ssrc, const sp<ABuffer> &buf) { + uint32_t extMaxSeq = mMaxSeq | mCycles; + uint32_t expected = extMaxSeq - mBaseSeq + 1; + + int64_t lost = (int64_t)expected - (int64_t)mReceived; + if (lost > 0x7fffff) { + lost = 0x7fffff; + } else if (lost < -0x800000) { + lost = -0x800000; + } + + uint32_t expectedInterval = expected - mExpectedPrior; + mExpectedPrior = expected; + + uint32_t receivedInterval = mReceived - mReceivedPrior; + mReceivedPrior = mReceived; + + int64_t lostInterval = expectedInterval - receivedInterval; + + uint8_t fractionLost; + if (expectedInterval == 0 || lostInterval <=0) { + fractionLost = 0; + } else { + fractionLost = (lostInterval << 8) / expectedInterval; + } + + uint8_t *ptr = buf->data() + buf->size(); + + ptr[0] = ssrc >> 24; + ptr[1] = (ssrc >> 16) & 0xff; + ptr[2] = (ssrc >> 8) & 0xff; + ptr[3] = ssrc & 0xff; + + ptr[4] = fractionLost; + + ptr[5] = (lost >> 16) & 0xff; + ptr[6] = (lost >> 8) & 0xff; + ptr[7] = lost & 0xff; + + ptr[8] = extMaxSeq >> 24; + ptr[9] = (extMaxSeq >> 16) & 0xff; + ptr[10] = (extMaxSeq >> 8) & 0xff; + ptr[11] = extMaxSeq & 0xff; + + // XXX TODO: + + ptr[12] = 0x00; // interarrival jitter + ptr[13] = 0x00; + ptr[14] = 0x00; + ptr[15] = 0x00; + + ptr[16] = 0x00; // last SR + ptr[17] = 0x00; + ptr[18] = 0x00; + ptr[19] = 0x00; + + ptr[20] = 0x00; // delay since last SR + ptr[21] = 0x00; + ptr[22] = 0x00; + ptr[23] = 0x00; + + buf->setRange(buf->offset(), buf->size() + 24); +} + +//////////////////////////////////////////////////////////////////////////////// + +RTPReceiver::RTPReceiver( + const sp<ANetworkSession> &netSession, + const sp<AMessage> ¬ify, + uint32_t flags) + : mNetSession(netSession), + mNotify(notify), + mFlags(flags), + mRTPMode(TRANSPORT_UNDEFINED), + mRTCPMode(TRANSPORT_UNDEFINED), + mRTPSessionID(0), + mRTCPSessionID(0), + mRTPConnected(false), + mRTCPConnected(false), + mRTPClientSessionID(0), + mRTCPClientSessionID(0) { +} + +RTPReceiver::~RTPReceiver() { + if (mRTCPClientSessionID != 0) { + mNetSession->destroySession(mRTCPClientSessionID); + mRTCPClientSessionID = 0; + } + + if (mRTPClientSessionID != 0) { + mNetSession->destroySession(mRTPClientSessionID); + mRTPClientSessionID = 0; + } + + if (mRTCPSessionID != 0) { + mNetSession->destroySession(mRTCPSessionID); + mRTCPSessionID = 0; + } + + if (mRTPSessionID != 0) { + mNetSession->destroySession(mRTPSessionID); + mRTPSessionID = 0; + } +} + +status_t RTPReceiver::initAsync( + TransportMode rtpMode, + TransportMode rtcpMode, + int32_t *outLocalRTPPort) { + if (mRTPMode != TRANSPORT_UNDEFINED + || rtpMode == TRANSPORT_UNDEFINED + || rtpMode == TRANSPORT_NONE + || rtcpMode == TRANSPORT_UNDEFINED) { + return INVALID_OPERATION; + } + + CHECK_NE(rtpMode, TRANSPORT_TCP_INTERLEAVED); + CHECK_NE(rtcpMode, TRANSPORT_TCP_INTERLEAVED); + + sp<AMessage> rtpNotify = new AMessage(kWhatRTPNotify, id()); + + sp<AMessage> rtcpNotify; + if (rtcpMode != TRANSPORT_NONE) { + rtcpNotify = new AMessage(kWhatRTCPNotify, id()); + } + + CHECK_EQ(mRTPSessionID, 0); + CHECK_EQ(mRTCPSessionID, 0); + + int32_t localRTPPort; + + struct in_addr ifaceAddr; + ifaceAddr.s_addr = INADDR_ANY; + + for (;;) { + localRTPPort = PickRandomRTPPort(); + + status_t err; + if (rtpMode == TRANSPORT_UDP) { + err = mNetSession->createUDPSession( + localRTPPort, + rtpNotify, + &mRTPSessionID); + } else { + CHECK_EQ(rtpMode, TRANSPORT_TCP); + err = mNetSession->createTCPDatagramSession( + ifaceAddr, + localRTPPort, + rtpNotify, + &mRTPSessionID); + } + + if (err != OK) { + continue; + } + + if (rtcpMode == TRANSPORT_NONE) { + break; + } else if (rtcpMode == TRANSPORT_UDP) { + err = mNetSession->createUDPSession( + localRTPPort + 1, + rtcpNotify, + &mRTCPSessionID); + } else { + CHECK_EQ(rtpMode, TRANSPORT_TCP); + err = mNetSession->createTCPDatagramSession( + ifaceAddr, + localRTPPort + 1, + rtcpNotify, + &mRTCPSessionID); + } + + if (err == OK) { + break; + } + + mNetSession->destroySession(mRTPSessionID); + mRTPSessionID = 0; + } + + mRTPMode = rtpMode; + mRTCPMode = rtcpMode; + *outLocalRTPPort = localRTPPort; + + return OK; +} + +status_t RTPReceiver::connect( + const char *remoteHost, int32_t remoteRTPPort, int32_t remoteRTCPPort) { + status_t err; + + if (mRTPMode == TRANSPORT_UDP) { + CHECK(!mRTPConnected); + + err = mNetSession->connectUDPSession( + mRTPSessionID, remoteHost, remoteRTPPort); + + if (err != OK) { + notifyInitDone(err); + return err; + } + + ALOGI("connectUDPSession RTP successful."); + + mRTPConnected = true; + } + + if (mRTCPMode == TRANSPORT_UDP) { + CHECK(!mRTCPConnected); + + err = mNetSession->connectUDPSession( + mRTCPSessionID, remoteHost, remoteRTCPPort); + + if (err != OK) { + notifyInitDone(err); + return err; + } + + scheduleSendRR(); + + ALOGI("connectUDPSession RTCP successful."); + + mRTCPConnected = true; + } + + if (mRTPConnected + && (mRTCPConnected || mRTCPMode == TRANSPORT_NONE)) { + notifyInitDone(OK); + } + + return OK; +} + +status_t RTPReceiver::informSender(const sp<AMessage> ¶ms) { + if (!mRTCPConnected) { + return INVALID_OPERATION; + } + + int64_t avgLatencyUs; + CHECK(params->findInt64("avgLatencyUs", &avgLatencyUs)); + + int64_t maxLatencyUs; + CHECK(params->findInt64("maxLatencyUs", &maxLatencyUs)); + + sp<ABuffer> buf = new ABuffer(28); + + uint8_t *ptr = buf->data(); + ptr[0] = 0x80 | 0; + ptr[1] = 204; // APP + ptr[2] = 0; + + CHECK((buf->size() % 4) == 0u); + ptr[3] = (buf->size() / 4) - 1; + + ptr[4] = kSourceID >> 24; // SSRC + ptr[5] = (kSourceID >> 16) & 0xff; + ptr[6] = (kSourceID >> 8) & 0xff; + ptr[7] = kSourceID & 0xff; + ptr[8] = 'l'; + ptr[9] = 'a'; + ptr[10] = 't'; + ptr[11] = 'e'; + + ptr[12] = avgLatencyUs >> 56; + ptr[13] = (avgLatencyUs >> 48) & 0xff; + ptr[14] = (avgLatencyUs >> 40) & 0xff; + ptr[15] = (avgLatencyUs >> 32) & 0xff; + ptr[16] = (avgLatencyUs >> 24) & 0xff; + ptr[17] = (avgLatencyUs >> 16) & 0xff; + ptr[18] = (avgLatencyUs >> 8) & 0xff; + ptr[19] = avgLatencyUs & 0xff; + + ptr[20] = maxLatencyUs >> 56; + ptr[21] = (maxLatencyUs >> 48) & 0xff; + ptr[22] = (maxLatencyUs >> 40) & 0xff; + ptr[23] = (maxLatencyUs >> 32) & 0xff; + ptr[24] = (maxLatencyUs >> 24) & 0xff; + ptr[25] = (maxLatencyUs >> 16) & 0xff; + ptr[26] = (maxLatencyUs >> 8) & 0xff; + ptr[27] = maxLatencyUs & 0xff; + + mNetSession->sendRequest(mRTCPSessionID, buf->data(), buf->size()); + + return OK; +} + +void RTPReceiver::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatRTPNotify: + case kWhatRTCPNotify: + onNetNotify(msg->what() == kWhatRTPNotify, msg); + break; + + case kWhatSendRR: + { + onSendRR(); + break; + } + + default: + TRESPASS(); + } +} + +void RTPReceiver::onNetNotify(bool isRTP, const sp<AMessage> &msg) { + int32_t reason; + CHECK(msg->findInt32("reason", &reason)); + + switch (reason) { + case ANetworkSession::kWhatError: + { + int32_t sessionID; + CHECK(msg->findInt32("sessionID", &sessionID)); + + int32_t err; + CHECK(msg->findInt32("err", &err)); + + int32_t errorOccuredDuringSend; + CHECK(msg->findInt32("send", &errorOccuredDuringSend)); + + AString detail; + CHECK(msg->findString("detail", &detail)); + + ALOGE("An error occurred during %s in session %d " + "(%d, '%s' (%s)).", + errorOccuredDuringSend ? "send" : "receive", + sessionID, + err, + detail.c_str(), + strerror(-err)); + + mNetSession->destroySession(sessionID); + + if (sessionID == mRTPSessionID) { + mRTPSessionID = 0; + } else if (sessionID == mRTCPSessionID) { + mRTCPSessionID = 0; + } else if (sessionID == mRTPClientSessionID) { + mRTPClientSessionID = 0; + } else if (sessionID == mRTCPClientSessionID) { + mRTCPClientSessionID = 0; + } + + if (!mRTPConnected + || (mRTCPMode != TRANSPORT_NONE && !mRTCPConnected)) { + notifyInitDone(err); + break; + } + + notifyError(err); + break; + } + + case ANetworkSession::kWhatDatagram: + { + sp<ABuffer> data; + CHECK(msg->findBuffer("data", &data)); + + if (isRTP) { + if (mFlags & FLAG_AUTO_CONNECT) { + AString fromAddr; + CHECK(msg->findString("fromAddr", &fromAddr)); + + int32_t fromPort; + CHECK(msg->findInt32("fromPort", &fromPort)); + + CHECK_EQ((status_t)OK, + connect( + fromAddr.c_str(), fromPort, fromPort + 1)); + + mFlags &= ~FLAG_AUTO_CONNECT; + } + + onRTPData(data); + } else { + onRTCPData(data); + } + break; + } + + case ANetworkSession::kWhatClientConnected: + { + int32_t sessionID; + CHECK(msg->findInt32("sessionID", &sessionID)); + + if (isRTP) { + CHECK_EQ(mRTPMode, TRANSPORT_TCP); + + if (mRTPClientSessionID != 0) { + // We only allow a single client connection. + mNetSession->destroySession(sessionID); + sessionID = 0; + break; + } + + mRTPClientSessionID = sessionID; + mRTPConnected = true; + } else { + CHECK_EQ(mRTCPMode, TRANSPORT_TCP); + + if (mRTCPClientSessionID != 0) { + // We only allow a single client connection. + mNetSession->destroySession(sessionID); + sessionID = 0; + break; + } + + mRTCPClientSessionID = sessionID; + mRTCPConnected = true; + } + + if (mRTPConnected + && (mRTCPConnected || mRTCPMode == TRANSPORT_NONE)) { + notifyInitDone(OK); + } + break; + } + } +} + +void RTPReceiver::notifyInitDone(status_t err) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatInitDone); + notify->setInt32("err", err); + notify->post(); +} + +void RTPReceiver::notifyError(status_t err) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatError); + notify->setInt32("err", err); + notify->post(); +} + +void RTPReceiver::notifyPacketLost() { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatPacketLost); + notify->post(); +} + +status_t RTPReceiver::onRTPData(const sp<ABuffer> &buffer) { + size_t size = buffer->size(); + if (size < 12) { + // Too short to be a valid RTP header. + return ERROR_MALFORMED; + } + + const uint8_t *data = buffer->data(); + + if ((data[0] >> 6) != 2) { + // Unsupported version. + return ERROR_UNSUPPORTED; + } + + if (data[0] & 0x20) { + // Padding present. + + size_t paddingLength = data[size - 1]; + + if (paddingLength + 12 > size) { + // If we removed this much padding we'd end up with something + // that's too short to be a valid RTP header. + return ERROR_MALFORMED; + } + + size -= paddingLength; + } + + int numCSRCs = data[0] & 0x0f; + + size_t payloadOffset = 12 + 4 * numCSRCs; + + if (size < payloadOffset) { + // Not enough data to fit the basic header and all the CSRC entries. + return ERROR_MALFORMED; + } + + if (data[0] & 0x10) { + // Header eXtension present. + + if (size < payloadOffset + 4) { + // Not enough data to fit the basic header, all CSRC entries + // and the first 4 bytes of the extension header. + + return ERROR_MALFORMED; + } + + const uint8_t *extensionData = &data[payloadOffset]; + + size_t extensionLength = + 4 * (extensionData[2] << 8 | extensionData[3]); + + if (size < payloadOffset + 4 + extensionLength) { + return ERROR_MALFORMED; + } + + payloadOffset += 4 + extensionLength; + } + + uint32_t srcId = U32_AT(&data[8]); + uint32_t rtpTime = U32_AT(&data[4]); + uint16_t seqNo = U16_AT(&data[2]); + + sp<AMessage> meta = buffer->meta(); + meta->setInt32("ssrc", srcId); + meta->setInt32("rtp-time", rtpTime); + meta->setInt32("PT", data[1] & 0x7f); + meta->setInt32("M", data[1] >> 7); + + buffer->setRange(payloadOffset, size - payloadOffset); + + ssize_t index = mSources.indexOfKey(srcId); + sp<Source> source; + if (index < 0) { + source = new Source(this, srcId); + looper()->registerHandler(source); + + mSources.add(srcId, source); + } else { + source = mSources.valueAt(index); + } + + source->onPacketReceived(seqNo, buffer); + + return OK; +} + +status_t RTPReceiver::onRTCPData(const sp<ABuffer> &data) { + ALOGI("onRTCPData"); + return OK; +} + +void RTPReceiver::addSDES(const sp<ABuffer> &buffer) { + uint8_t *data = buffer->data() + buffer->size(); + data[0] = 0x80 | 1; + data[1] = 202; // SDES + data[4] = kSourceID >> 24; // SSRC + data[5] = (kSourceID >> 16) & 0xff; + data[6] = (kSourceID >> 8) & 0xff; + data[7] = kSourceID & 0xff; + + size_t offset = 8; + + data[offset++] = 1; // CNAME + + AString cname = "stagefright@somewhere"; + data[offset++] = cname.size(); + + memcpy(&data[offset], cname.c_str(), cname.size()); + offset += cname.size(); + + data[offset++] = 6; // TOOL + + AString tool = "stagefright/1.0"; + data[offset++] = tool.size(); + + memcpy(&data[offset], tool.c_str(), tool.size()); + offset += tool.size(); + + data[offset++] = 0; + + if ((offset % 4) > 0) { + size_t count = 4 - (offset % 4); + switch (count) { + case 3: + data[offset++] = 0; + case 2: + data[offset++] = 0; + case 1: + data[offset++] = 0; + } + } + + size_t numWords = (offset / 4) - 1; + data[2] = numWords >> 8; + data[3] = numWords & 0xff; + + buffer->setRange(buffer->offset(), buffer->size() + offset); +} + +void RTPReceiver::scheduleSendRR() { + (new AMessage(kWhatSendRR, id()))->post(5000000ll); +} + +void RTPReceiver::onSendRR() { + sp<ABuffer> buf = new ABuffer(kMaxUDPPacketSize); + buf->setRange(0, 0); + + uint8_t *ptr = buf->data(); + ptr[0] = 0x80 | 0; + ptr[1] = 201; // RR + ptr[2] = 0; + ptr[3] = 1; + ptr[4] = kSourceID >> 24; // SSRC + ptr[5] = (kSourceID >> 16) & 0xff; + ptr[6] = (kSourceID >> 8) & 0xff; + ptr[7] = kSourceID & 0xff; + + buf->setRange(0, 8); + + size_t numReportBlocks = 0; + for (size_t i = 0; i < mSources.size(); ++i) { + uint32_t ssrc = mSources.keyAt(i); + sp<Source> source = mSources.valueAt(i); + + if (numReportBlocks > 31 || buf->size() + 24 > buf->capacity()) { + // Cannot fit another report block. + break; + } + + source->addReportBlock(ssrc, buf); + ++numReportBlocks; + } + + ptr[0] |= numReportBlocks; // 5 bit + + size_t sizeInWordsMinus1 = 1 + 6 * numReportBlocks; + ptr[2] = sizeInWordsMinus1 >> 8; + ptr[3] = sizeInWordsMinus1 & 0xff; + + buf->setRange(0, (sizeInWordsMinus1 + 1) * 4); + + addSDES(buf); + + mNetSession->sendRequest(mRTCPSessionID, buf->data(), buf->size()); + + scheduleSendRR(); +} + +status_t RTPReceiver::registerPacketType( + uint8_t packetType, PacketizationMode mode) { + mPacketTypes.add(packetType, mode); + + return OK; +} + +sp<RTPReceiver::Assembler> RTPReceiver::makeAssembler(uint8_t packetType) { + ssize_t index = mPacketTypes.indexOfKey(packetType); + if (index < 0) { + return NULL; + } + + PacketizationMode mode = mPacketTypes.valueAt(index); + + switch (mode) { + case PACKETIZATION_NONE: + case PACKETIZATION_TRANSPORT_STREAM: + return new TSAssembler(mNotify); + + case PACKETIZATION_H264: + return new H264Assembler(mNotify); + + default: + return NULL; + } +} + +void RTPReceiver::requestRetransmission(uint32_t senderSSRC, int32_t extSeqNo) { + int32_t blp = 0; + + sp<ABuffer> buf = new ABuffer(16); + buf->setRange(0, 0); + + uint8_t *ptr = buf->data(); + ptr[0] = 0x80 | 1; // generic NACK + ptr[1] = 205; // TSFB + ptr[2] = 0; + ptr[3] = 3; + ptr[8] = (senderSSRC >> 24) & 0xff; + ptr[9] = (senderSSRC >> 16) & 0xff; + ptr[10] = (senderSSRC >> 8) & 0xff; + ptr[11] = (senderSSRC & 0xff); + ptr[8] = (kSourceID >> 24) & 0xff; + ptr[9] = (kSourceID >> 16) & 0xff; + ptr[10] = (kSourceID >> 8) & 0xff; + ptr[11] = (kSourceID & 0xff); + ptr[12] = (extSeqNo >> 8) & 0xff; + ptr[13] = (extSeqNo & 0xff); + ptr[14] = (blp >> 8) & 0xff; + ptr[15] = (blp & 0xff); + + buf->setRange(0, 16); + + mNetSession->sendRequest(mRTCPSessionID, buf->data(), buf->size()); +} + +void RTPReceiver::Source::modifyPacketStatus(int32_t extSeqNo, uint32_t mask) { +#if TRACK_PACKET_LOSS + ssize_t index = mLostPackets.indexOfKey(extSeqNo); + if (index < 0) { + mLostPackets.add(extSeqNo, mask); + } else { + mLostPackets.editValueAt(index) |= mask; + } +#endif +} + +void RTPReceiver::Source::postRetransmitTimer(int64_t timeUs) { + int64_t delayUs = timeUs - ALooper::GetNowUs(); + sp<AMessage> msg = new AMessage(kWhatRetransmit, id()); + msg->setInt32("generation", mRetransmitGeneration); + msg->post(delayUs); +} + +void RTPReceiver::Source::postDeclareLostTimer(int64_t timeUs) { + CHECK(!mDeclareLostTimerPending); + mDeclareLostTimerPending = true; + + int64_t delayUs = timeUs - ALooper::GetNowUs(); + sp<AMessage> msg = new AMessage(kWhatDeclareLost, id()); + msg->setInt32("generation", mDeclareLostGeneration); + msg->post(delayUs); +} + +void RTPReceiver::Source::cancelTimers() { + ++mRetransmitGeneration; + ++mDeclareLostGeneration; + mDeclareLostTimerPending = false; +} + +} // namespace android + diff --git a/media/libstagefright/wifi-display/rtp/RTPReceiver.h b/media/libstagefright/wifi-display/rtp/RTPReceiver.h new file mode 100644 index 0000000..240ab2e --- /dev/null +++ b/media/libstagefright/wifi-display/rtp/RTPReceiver.h @@ -0,0 +1,125 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef RTP_RECEIVER_H_ + +#define RTP_RECEIVER_H_ + +#include "RTPBase.h" + +#include <media/stagefright/foundation/AHandler.h> + +namespace android { + +struct ABuffer; +struct ANetworkSession; + +// An object of this class facilitates receiving of media data on an RTP +// channel. The channel is established over a UDP or TCP connection depending +// on which "TransportMode" was chosen. In addition different RTP packetization +// schemes are supported such as "Transport Stream Packets over RTP", +// or "AVC/H.264 encapsulation as specified in RFC 3984 (non-interleaved mode)" +struct RTPReceiver : public RTPBase, public AHandler { + enum { + kWhatInitDone, + kWhatError, + kWhatAccessUnit, + kWhatPacketLost, + }; + + enum Flags { + FLAG_AUTO_CONNECT = 1, + }; + RTPReceiver( + const sp<ANetworkSession> &netSession, + const sp<AMessage> ¬ify, + uint32_t flags = 0); + + status_t registerPacketType( + uint8_t packetType, PacketizationMode mode); + + status_t initAsync( + TransportMode rtpMode, + TransportMode rtcpMode, + int32_t *outLocalRTPPort); + + status_t connect( + const char *remoteHost, + int32_t remoteRTPPort, + int32_t remoteRTCPPort); + + status_t informSender(const sp<AMessage> ¶ms); + +protected: + virtual ~RTPReceiver(); + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kWhatRTPNotify, + kWhatRTCPNotify, + kWhatSendRR, + }; + + enum { + kSourceID = 0xdeadbeef, + kPacketLostAfterUs = 100000, + kRequestRetransmissionAfterUs = -1, + }; + + struct Assembler; + struct H264Assembler; + struct Source; + struct TSAssembler; + + sp<ANetworkSession> mNetSession; + sp<AMessage> mNotify; + uint32_t mFlags; + TransportMode mRTPMode; + TransportMode mRTCPMode; + int32_t mRTPSessionID; + int32_t mRTCPSessionID; + bool mRTPConnected; + bool mRTCPConnected; + + int32_t mRTPClientSessionID; // in TRANSPORT_TCP mode. + int32_t mRTCPClientSessionID; // in TRANSPORT_TCP mode. + + KeyedVector<uint8_t, PacketizationMode> mPacketTypes; + KeyedVector<uint32_t, sp<Source> > mSources; + + void onNetNotify(bool isRTP, const sp<AMessage> &msg); + status_t onRTPData(const sp<ABuffer> &data); + status_t onRTCPData(const sp<ABuffer> &data); + void onSendRR(); + + void scheduleSendRR(); + void addSDES(const sp<ABuffer> &buffer); + + void notifyInitDone(status_t err); + void notifyError(status_t err); + void notifyPacketLost(); + + sp<Assembler> makeAssembler(uint8_t packetType); + + void requestRetransmission(uint32_t senderSSRC, int32_t extSeqNo); + + DISALLOW_EVIL_CONSTRUCTORS(RTPReceiver); +}; + +} // namespace android + +#endif // RTP_RECEIVER_H_ diff --git a/media/libstagefright/wifi-display/rtp/RTPSender.cpp b/media/libstagefright/wifi-display/rtp/RTPSender.cpp index 095fd97..6bbe650 100644 --- a/media/libstagefright/wifi-display/rtp/RTPSender.cpp +++ b/media/libstagefright/wifi-display/rtp/RTPSender.cpp @@ -767,6 +767,17 @@ status_t RTPSender::parseTSFB(const uint8_t *data, size_t size) { } status_t RTPSender::parseAPP(const uint8_t *data, size_t size) { + if (!memcmp("late", &data[8], 4)) { + int64_t avgLatencyUs = (int64_t)U64_AT(&data[12]); + int64_t maxLatencyUs = (int64_t)U64_AT(&data[20]); + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatInformSender); + notify->setInt64("avgLatencyUs", avgLatencyUs); + notify->setInt64("maxLatencyUs", maxLatencyUs); + notify->post(); + } + return OK; } diff --git a/media/libstagefright/wifi-display/rtp/RTPSender.h b/media/libstagefright/wifi-display/rtp/RTPSender.h index 7dc138a..fefcab7 100644 --- a/media/libstagefright/wifi-display/rtp/RTPSender.h +++ b/media/libstagefright/wifi-display/rtp/RTPSender.h @@ -37,6 +37,7 @@ struct RTPSender : public RTPBase, public AHandler { kWhatInitDone, kWhatError, kWhatNetworkStall, + kWhatInformSender, }; RTPSender( const sp<ANetworkSession> &netSession, diff --git a/media/libstagefright/wifi-display/rtptest.cpp b/media/libstagefright/wifi-display/rtptest.cpp new file mode 100644 index 0000000..764a38b --- /dev/null +++ b/media/libstagefright/wifi-display/rtptest.cpp @@ -0,0 +1,565 @@ +/* + * Copyright 2013, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NEBUG 0 +#define LOG_TAG "rtptest" +#include <utils/Log.h> + +#include "ANetworkSession.h" +#include "rtp/RTPSender.h" +#include "rtp/RTPReceiver.h" +#include "TimeSyncer.h" + +#include <binder/ProcessState.h> +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AHandler.h> +#include <media/stagefright/foundation/ALooper.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/foundation/hexdump.h> +#include <media/stagefright/DataSource.h> +#include <media/stagefright/MediaDefs.h> +#include <media/stagefright/NuMediaExtractor.h> +#include <media/stagefright/Utils.h> + +#define MEDIA_FILENAME "/sdcard/Frame Counter HD 30FPS_1080p.mp4" + +namespace android { + +struct PacketSource : public RefBase { + PacketSource() {} + + virtual sp<ABuffer> getNextAccessUnit() = 0; + +protected: + virtual ~PacketSource() {} + +private: + DISALLOW_EVIL_CONSTRUCTORS(PacketSource); +}; + +struct MediaPacketSource : public PacketSource { + MediaPacketSource() + : mMaxSampleSize(1024 * 1024) { + mExtractor = new NuMediaExtractor; + CHECK_EQ((status_t)OK, + mExtractor->setDataSource(MEDIA_FILENAME)); + + bool haveVideo = false; + for (size_t i = 0; i < mExtractor->countTracks(); ++i) { + sp<AMessage> format; + CHECK_EQ((status_t)OK, mExtractor->getTrackFormat(i, &format)); + + AString mime; + CHECK(format->findString("mime", &mime)); + + if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mime.c_str())) { + mExtractor->selectTrack(i); + haveVideo = true; + break; + } + } + + CHECK(haveVideo); + } + + virtual sp<ABuffer> getNextAccessUnit() { + int64_t timeUs; + status_t err = mExtractor->getSampleTime(&timeUs); + + if (err != OK) { + return NULL; + } + + sp<ABuffer> accessUnit = new ABuffer(mMaxSampleSize); + CHECK_EQ((status_t)OK, mExtractor->readSampleData(accessUnit)); + + accessUnit->meta()->setInt64("timeUs", timeUs); + + CHECK_EQ((status_t)OK, mExtractor->advance()); + + return accessUnit; + } + +protected: + virtual ~MediaPacketSource() { + } + +private: + sp<NuMediaExtractor> mExtractor; + size_t mMaxSampleSize; + + DISALLOW_EVIL_CONSTRUCTORS(MediaPacketSource); +}; + +struct SimplePacketSource : public PacketSource { + SimplePacketSource() + : mCounter(0) { + } + + virtual sp<ABuffer> getNextAccessUnit() { + sp<ABuffer> buffer = new ABuffer(4); + uint8_t *dst = buffer->data(); + dst[0] = mCounter >> 24; + dst[1] = (mCounter >> 16) & 0xff; + dst[2] = (mCounter >> 8) & 0xff; + dst[3] = mCounter & 0xff; + + buffer->meta()->setInt64("timeUs", mCounter * 1000000ll / kFrameRate); + + ++mCounter; + + return buffer; + } + +protected: + virtual ~SimplePacketSource() { + } + +private: + enum { + kFrameRate = 30 + }; + + uint32_t mCounter; + + DISALLOW_EVIL_CONSTRUCTORS(SimplePacketSource); +}; + +struct TestHandler : public AHandler { + TestHandler(const sp<ANetworkSession> &netSession); + + void listen(); + void connect(const char *host, int32_t port); + +protected: + virtual ~TestHandler(); + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kWhatListen, + kWhatConnect, + kWhatReceiverNotify, + kWhatSenderNotify, + kWhatSendMore, + kWhatStop, + kWhatTimeSyncerNotify, + }; + +#if 1 + static const RTPBase::TransportMode kRTPMode = RTPBase::TRANSPORT_UDP; + static const RTPBase::TransportMode kRTCPMode = RTPBase::TRANSPORT_UDP; +#else + static const RTPBase::TransportMode kRTPMode = RTPBase::TRANSPORT_TCP; + static const RTPBase::TransportMode kRTCPMode = RTPBase::TRANSPORT_NONE; +#endif + +#if 1 + static const RTPBase::PacketizationMode kPacketizationMode + = RTPBase::PACKETIZATION_H264; +#else + static const RTPBase::PacketizationMode kPacketizationMode + = RTPBase::PACKETIZATION_NONE; +#endif + + sp<ANetworkSession> mNetSession; + sp<PacketSource> mSource; + sp<RTPSender> mSender; + sp<RTPReceiver> mReceiver; + + sp<TimeSyncer> mTimeSyncer; + bool mTimeSyncerStarted; + + int64_t mFirstTimeRealUs; + int64_t mFirstTimeMediaUs; + + int64_t mTimeOffsetUs; + bool mTimeOffsetValid; + + status_t readMore(); + + DISALLOW_EVIL_CONSTRUCTORS(TestHandler); +}; + +TestHandler::TestHandler(const sp<ANetworkSession> &netSession) + : mNetSession(netSession), + mTimeSyncerStarted(false), + mFirstTimeRealUs(-1ll), + mFirstTimeMediaUs(-1ll), + mTimeOffsetUs(-1ll), + mTimeOffsetValid(false) { +} + +TestHandler::~TestHandler() { +} + +void TestHandler::listen() { + sp<AMessage> msg = new AMessage(kWhatListen, id()); + msg->post(); +} + +void TestHandler::connect(const char *host, int32_t port) { + sp<AMessage> msg = new AMessage(kWhatConnect, id()); + msg->setString("host", host); + msg->setInt32("port", port); + msg->post(); +} + +static void dumpDelay(int64_t delayMs) { + static const int64_t kMinDelayMs = 0; + static const int64_t kMaxDelayMs = 300; + + const char *kPattern = "########################################"; + size_t kPatternSize = strlen(kPattern); + + int n = (kPatternSize * (delayMs - kMinDelayMs)) + / (kMaxDelayMs - kMinDelayMs); + + if (n < 0) { + n = 0; + } else if ((size_t)n > kPatternSize) { + n = kPatternSize; + } + + ALOGI("(%4lld ms) %s\n", + delayMs, + kPattern + kPatternSize - n); +} + +void TestHandler::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatListen: + { + sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id()); + mTimeSyncer = new TimeSyncer(mNetSession, notify); + looper()->registerHandler(mTimeSyncer); + + notify = new AMessage(kWhatReceiverNotify, id()); + mReceiver = new RTPReceiver( + mNetSession, notify, RTPReceiver::FLAG_AUTO_CONNECT); + looper()->registerHandler(mReceiver); + + CHECK_EQ((status_t)OK, + mReceiver->registerPacketType(33, kPacketizationMode)); + + int32_t receiverRTPPort; + CHECK_EQ((status_t)OK, + mReceiver->initAsync( + kRTPMode, + kRTCPMode, + &receiverRTPPort)); + + printf("picked receiverRTPPort %d\n", receiverRTPPort); + +#if 0 + CHECK_EQ((status_t)OK, + mReceiver->connect( + "127.0.0.1", senderRTPPort, senderRTPPort + 1)); +#endif + break; + } + + case kWhatConnect: + { + AString host; + CHECK(msg->findString("host", &host)); + + sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id()); + mTimeSyncer = new TimeSyncer(mNetSession, notify); + looper()->registerHandler(mTimeSyncer); + mTimeSyncer->startServer(8123); + + int32_t receiverRTPPort; + CHECK(msg->findInt32("port", &receiverRTPPort)); + +#if 1 + mSource = new MediaPacketSource; +#else + mSource = new SimplePacketSource; +#endif + + notify = new AMessage(kWhatSenderNotify, id()); + mSender = new RTPSender(mNetSession, notify); + + looper()->registerHandler(mSender); + + int32_t senderRTPPort; + CHECK_EQ((status_t)OK, + mSender->initAsync( + host.c_str(), + receiverRTPPort, + kRTPMode, + kRTCPMode == RTPBase::TRANSPORT_NONE + ? -1 : receiverRTPPort + 1, + kRTCPMode, + &senderRTPPort)); + + printf("picked senderRTPPort %d\n", senderRTPPort); + break; + } + + case kWhatSenderNotify: + { + ALOGI("kWhatSenderNotify"); + + int32_t what; + CHECK(msg->findInt32("what", &what)); + + switch (what) { + case RTPSender::kWhatInitDone: + { + int32_t err; + CHECK(msg->findInt32("err", &err)); + + ALOGI("RTPSender::initAsync completed w/ err %d", err); + + if (err == OK) { + err = readMore(); + + if (err != OK) { + (new AMessage(kWhatStop, id()))->post(); + } + } + break; + } + + case RTPSender::kWhatError: + break; + } + break; + } + + case kWhatReceiverNotify: + { + ALOGV("kWhatReceiverNotify"); + + int32_t what; + CHECK(msg->findInt32("what", &what)); + + switch (what) { + case RTPReceiver::kWhatInitDone: + { + int32_t err; + CHECK(msg->findInt32("err", &err)); + + ALOGI("RTPReceiver::initAsync completed w/ err %d", err); + break; + } + + case RTPReceiver::kWhatError: + break; + + case RTPReceiver::kWhatAccessUnit: + { +#if 0 + if (!mTimeSyncerStarted) { + mTimeSyncer->startClient("172.18.41.216", 8123); + mTimeSyncerStarted = true; + } + + sp<ABuffer> accessUnit; + CHECK(msg->findBuffer("accessUnit", &accessUnit)); + + int64_t timeUs; + CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); + + if (mTimeOffsetValid) { + timeUs -= mTimeOffsetUs; + int64_t nowUs = ALooper::GetNowUs(); + int64_t delayMs = (nowUs - timeUs) / 1000ll; + + dumpDelay(delayMs); + } +#endif + break; + } + + case RTPReceiver::kWhatPacketLost: + ALOGV("kWhatPacketLost"); + break; + + default: + TRESPASS(); + } + break; + } + + case kWhatSendMore: + { + sp<ABuffer> accessUnit; + CHECK(msg->findBuffer("accessUnit", &accessUnit)); + + CHECK_EQ((status_t)OK, + mSender->queueBuffer( + accessUnit, + 33, + kPacketizationMode)); + + status_t err = readMore(); + + if (err != OK) { + (new AMessage(kWhatStop, id()))->post(); + } + break; + } + + case kWhatStop: + { + if (mReceiver != NULL) { + looper()->unregisterHandler(mReceiver->id()); + mReceiver.clear(); + } + + if (mSender != NULL) { + looper()->unregisterHandler(mSender->id()); + mSender.clear(); + } + + mSource.clear(); + + looper()->stop(); + break; + } + + case kWhatTimeSyncerNotify: + { + CHECK(msg->findInt64("offset", &mTimeOffsetUs)); + mTimeOffsetValid = true; + break; + } + + default: + TRESPASS(); + } +} + +status_t TestHandler::readMore() { + sp<ABuffer> accessUnit = mSource->getNextAccessUnit(); + + if (accessUnit == NULL) { + return ERROR_END_OF_STREAM; + } + + int64_t timeUs; + CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); + + int64_t nowUs = ALooper::GetNowUs(); + int64_t whenUs; + + if (mFirstTimeRealUs < 0ll) { + mFirstTimeRealUs = whenUs = nowUs; + mFirstTimeMediaUs = timeUs; + } else { + whenUs = mFirstTimeRealUs + timeUs - mFirstTimeMediaUs; + } + + accessUnit->meta()->setInt64("timeUs", whenUs); + + sp<AMessage> msg = new AMessage(kWhatSendMore, id()); + msg->setBuffer("accessUnit", accessUnit); + msg->post(whenUs - nowUs); + + return OK; +} + +} // namespace android + +static void usage(const char *me) { + fprintf(stderr, + "usage: %s -c host:port\tconnect to remote host\n" + " -l \tlisten\n", + me); +} + +int main(int argc, char **argv) { + using namespace android; + + // srand(time(NULL)); + + ProcessState::self()->startThreadPool(); + + DataSource::RegisterDefaultSniffers(); + + bool listen = false; + int32_t connectToPort = -1; + AString connectToHost; + + int res; + while ((res = getopt(argc, argv, "hc:l")) >= 0) { + switch (res) { + case 'c': + { + const char *colonPos = strrchr(optarg, ':'); + + if (colonPos == NULL) { + usage(argv[0]); + exit(1); + } + + connectToHost.setTo(optarg, colonPos - optarg); + + char *end; + connectToPort = strtol(colonPos + 1, &end, 10); + + if (*end != '\0' || end == colonPos + 1 + || connectToPort < 1 || connectToPort > 65535) { + fprintf(stderr, "Illegal port specified.\n"); + exit(1); + } + break; + } + + case 'l': + { + listen = true; + break; + } + + case '?': + case 'h': + usage(argv[0]); + exit(1); + } + } + + if (!listen && connectToPort < 0) { + fprintf(stderr, + "You need to select either client or server mode.\n"); + exit(1); + } + + sp<ANetworkSession> netSession = new ANetworkSession; + netSession->start(); + + sp<ALooper> looper = new ALooper; + + sp<TestHandler> handler = new TestHandler(netSession); + looper->registerHandler(handler); + + if (listen) { + handler->listen(); + } + + if (connectToPort >= 0) { + handler->connect(connectToHost.c_str(), connectToPort); + } + + looper->start(true /* runOnCallingThread */); + + return 0; +} + diff --git a/media/libstagefright/wifi-display/sink/DirectRenderer.cpp b/media/libstagefright/wifi-display/sink/DirectRenderer.cpp new file mode 100644 index 0000000..15f9c88 --- /dev/null +++ b/media/libstagefright/wifi-display/sink/DirectRenderer.cpp @@ -0,0 +1,625 @@ +/* + * Copyright 2012, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "DirectRenderer" +#include <utils/Log.h> + +#include "DirectRenderer.h" + +#include <gui/SurfaceComposerClient.h> +#include <gui/Surface.h> +#include <media/AudioTrack.h> +#include <media/ICrypto.h> +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/foundation/hexdump.h> +#include <media/stagefright/MediaCodec.h> +#include <media/stagefright/MediaErrors.h> +#include <media/stagefright/MetaData.h> +#include <media/stagefright/Utils.h> + +namespace android { + +/* + Drives the decoding process using a MediaCodec instance. Input buffers + queued by calls to "queueInputBuffer" are fed to the decoder as soon + as the decoder is ready for them, the client is notified about output + buffers as the decoder spits them out. +*/ +struct DirectRenderer::DecoderContext : public AHandler { + enum { + kWhatOutputBufferReady, + }; + DecoderContext(const sp<AMessage> ¬ify); + + status_t init( + const sp<AMessage> &format, + const sp<IGraphicBufferProducer> &surfaceTex); + + void queueInputBuffer(const sp<ABuffer> &accessUnit); + + status_t renderOutputBufferAndRelease(size_t index); + status_t releaseOutputBuffer(size_t index); + +protected: + virtual ~DecoderContext(); + + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kWhatDecoderNotify, + }; + + sp<AMessage> mNotify; + sp<ALooper> mDecoderLooper; + sp<MediaCodec> mDecoder; + Vector<sp<ABuffer> > mDecoderInputBuffers; + Vector<sp<ABuffer> > mDecoderOutputBuffers; + List<size_t> mDecoderInputBuffersAvailable; + bool mDecoderNotificationPending; + + List<sp<ABuffer> > mAccessUnits; + + void onDecoderNotify(); + void scheduleDecoderNotification(); + void queueDecoderInputBuffers(); + + void queueOutputBuffer( + size_t index, int64_t timeUs, const sp<ABuffer> &buffer); + + DISALLOW_EVIL_CONSTRUCTORS(DecoderContext); +}; + +//////////////////////////////////////////////////////////////////////////////// + +/* + A "push" audio renderer. The primary function of this renderer is to use + an AudioTrack in push mode and making sure not to block the event loop + be ensuring that calls to AudioTrack::write never block. This is done by + estimating an upper bound of data that can be written to the AudioTrack + buffer without delay. +*/ +struct DirectRenderer::AudioRenderer : public AHandler { + AudioRenderer(const sp<DecoderContext> &decoderContext); + + void queueInputBuffer( + size_t index, int64_t timeUs, const sp<ABuffer> &buffer); + +protected: + virtual ~AudioRenderer(); + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kWhatPushAudio, + }; + + struct BufferInfo { + size_t mIndex; + int64_t mTimeUs; + sp<ABuffer> mBuffer; + }; + + sp<DecoderContext> mDecoderContext; + sp<AudioTrack> mAudioTrack; + + List<BufferInfo> mInputBuffers; + bool mPushPending; + + size_t mNumFramesWritten; + + void schedulePushIfNecessary(); + void onPushAudio(); + + ssize_t writeNonBlocking(const uint8_t *data, size_t size); + + DISALLOW_EVIL_CONSTRUCTORS(AudioRenderer); +}; + +//////////////////////////////////////////////////////////////////////////////// + +DirectRenderer::DecoderContext::DecoderContext(const sp<AMessage> ¬ify) + : mNotify(notify), + mDecoderNotificationPending(false) { +} + +DirectRenderer::DecoderContext::~DecoderContext() { + if (mDecoder != NULL) { + mDecoder->release(); + mDecoder.clear(); + + mDecoderLooper->stop(); + mDecoderLooper.clear(); + } +} + +status_t DirectRenderer::DecoderContext::init( + const sp<AMessage> &format, + const sp<IGraphicBufferProducer> &surfaceTex) { + CHECK(mDecoder == NULL); + + AString mime; + CHECK(format->findString("mime", &mime)); + + mDecoderLooper = new ALooper; + mDecoderLooper->setName("video codec looper"); + + mDecoderLooper->start( + false /* runOnCallingThread */, + false /* canCallJava */, + PRIORITY_DEFAULT); + + mDecoder = MediaCodec::CreateByType( + mDecoderLooper, mime.c_str(), false /* encoder */); + + CHECK(mDecoder != NULL); + + status_t err = mDecoder->configure( + format, + surfaceTex == NULL + ? NULL : new Surface(surfaceTex), + NULL /* crypto */, + 0 /* flags */); + CHECK_EQ(err, (status_t)OK); + + err = mDecoder->start(); + CHECK_EQ(err, (status_t)OK); + + err = mDecoder->getInputBuffers( + &mDecoderInputBuffers); + CHECK_EQ(err, (status_t)OK); + + err = mDecoder->getOutputBuffers( + &mDecoderOutputBuffers); + CHECK_EQ(err, (status_t)OK); + + scheduleDecoderNotification(); + + return OK; +} + +void DirectRenderer::DecoderContext::queueInputBuffer( + const sp<ABuffer> &accessUnit) { + CHECK(mDecoder != NULL); + + mAccessUnits.push_back(accessUnit); + queueDecoderInputBuffers(); +} + +status_t DirectRenderer::DecoderContext::renderOutputBufferAndRelease( + size_t index) { + return mDecoder->renderOutputBufferAndRelease(index); +} + +status_t DirectRenderer::DecoderContext::releaseOutputBuffer(size_t index) { + return mDecoder->releaseOutputBuffer(index); +} + +void DirectRenderer::DecoderContext::queueDecoderInputBuffers() { + if (mDecoder == NULL) { + return; + } + + bool submittedMore = false; + + while (!mAccessUnits.empty() + && !mDecoderInputBuffersAvailable.empty()) { + size_t index = *mDecoderInputBuffersAvailable.begin(); + + mDecoderInputBuffersAvailable.erase( + mDecoderInputBuffersAvailable.begin()); + + sp<ABuffer> srcBuffer = *mAccessUnits.begin(); + mAccessUnits.erase(mAccessUnits.begin()); + + const sp<ABuffer> &dstBuffer = + mDecoderInputBuffers.itemAt(index); + + memcpy(dstBuffer->data(), srcBuffer->data(), srcBuffer->size()); + + int64_t timeUs; + CHECK(srcBuffer->meta()->findInt64("timeUs", &timeUs)); + + status_t err = mDecoder->queueInputBuffer( + index, + 0 /* offset */, + srcBuffer->size(), + timeUs, + 0 /* flags */); + CHECK_EQ(err, (status_t)OK); + + submittedMore = true; + } + + if (submittedMore) { + scheduleDecoderNotification(); + } +} + +void DirectRenderer::DecoderContext::onMessageReceived( + const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatDecoderNotify: + { + onDecoderNotify(); + break; + } + + default: + TRESPASS(); + } +} + +void DirectRenderer::DecoderContext::onDecoderNotify() { + mDecoderNotificationPending = false; + + for (;;) { + size_t index; + status_t err = mDecoder->dequeueInputBuffer(&index); + + if (err == OK) { + mDecoderInputBuffersAvailable.push_back(index); + } else if (err == -EAGAIN) { + break; + } else { + TRESPASS(); + } + } + + queueDecoderInputBuffers(); + + for (;;) { + size_t index; + size_t offset; + size_t size; + int64_t timeUs; + uint32_t flags; + status_t err = mDecoder->dequeueOutputBuffer( + &index, + &offset, + &size, + &timeUs, + &flags); + + if (err == OK) { + queueOutputBuffer( + index, timeUs, mDecoderOutputBuffers.itemAt(index)); + } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) { + err = mDecoder->getOutputBuffers( + &mDecoderOutputBuffers); + CHECK_EQ(err, (status_t)OK); + } else if (err == INFO_FORMAT_CHANGED) { + // We don't care. + } else if (err == -EAGAIN) { + break; + } else { + TRESPASS(); + } + } + + scheduleDecoderNotification(); +} + +void DirectRenderer::DecoderContext::scheduleDecoderNotification() { + if (mDecoderNotificationPending) { + return; + } + + sp<AMessage> notify = + new AMessage(kWhatDecoderNotify, id()); + + mDecoder->requestActivityNotification(notify); + mDecoderNotificationPending = true; +} + +void DirectRenderer::DecoderContext::queueOutputBuffer( + size_t index, int64_t timeUs, const sp<ABuffer> &buffer) { + sp<AMessage> msg = mNotify->dup(); + msg->setInt32("what", kWhatOutputBufferReady); + msg->setSize("index", index); + msg->setInt64("timeUs", timeUs); + msg->setBuffer("buffer", buffer); + msg->post(); +} + +//////////////////////////////////////////////////////////////////////////////// + +DirectRenderer::AudioRenderer::AudioRenderer( + const sp<DecoderContext> &decoderContext) + : mDecoderContext(decoderContext), + mPushPending(false), + mNumFramesWritten(0) { + mAudioTrack = new AudioTrack( + AUDIO_STREAM_DEFAULT, + 48000.0f, + AUDIO_FORMAT_PCM, + AUDIO_CHANNEL_OUT_STEREO, + (int)0 /* frameCount */); + + CHECK_EQ((status_t)OK, mAudioTrack->initCheck()); + + mAudioTrack->start(); +} + +DirectRenderer::AudioRenderer::~AudioRenderer() { +} + +void DirectRenderer::AudioRenderer::queueInputBuffer( + size_t index, int64_t timeUs, const sp<ABuffer> &buffer) { + BufferInfo info; + info.mIndex = index; + info.mTimeUs = timeUs; + info.mBuffer = buffer; + + mInputBuffers.push_back(info); + schedulePushIfNecessary(); +} + +void DirectRenderer::AudioRenderer::onMessageReceived( + const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatPushAudio: + { + onPushAudio(); + break; + } + + default: + break; + } +} + +void DirectRenderer::AudioRenderer::schedulePushIfNecessary() { + if (mPushPending || mInputBuffers.empty()) { + return; + } + + mPushPending = true; + + uint32_t numFramesPlayed; + CHECK_EQ(mAudioTrack->getPosition(&numFramesPlayed), + (status_t)OK); + + uint32_t numFramesPendingPlayout = mNumFramesWritten - numFramesPlayed; + + // This is how long the audio sink will have data to + // play back. + const float msecsPerFrame = 1000.0f / mAudioTrack->getSampleRate(); + + int64_t delayUs = + msecsPerFrame * numFramesPendingPlayout * 1000ll; + + // Let's give it more data after about half that time + // has elapsed. + (new AMessage(kWhatPushAudio, id()))->post(delayUs / 2); +} + +void DirectRenderer::AudioRenderer::onPushAudio() { + mPushPending = false; + + while (!mInputBuffers.empty()) { + const BufferInfo &info = *mInputBuffers.begin(); + + ssize_t n = writeNonBlocking( + info.mBuffer->data(), info.mBuffer->size()); + + if (n < (ssize_t)info.mBuffer->size()) { + CHECK_GE(n, 0); + + info.mBuffer->setRange( + info.mBuffer->offset() + n, info.mBuffer->size() - n); + break; + } + + mDecoderContext->releaseOutputBuffer(info.mIndex); + + mInputBuffers.erase(mInputBuffers.begin()); + } + + schedulePushIfNecessary(); +} + +ssize_t DirectRenderer::AudioRenderer::writeNonBlocking( + const uint8_t *data, size_t size) { + uint32_t numFramesPlayed; + status_t err = mAudioTrack->getPosition(&numFramesPlayed); + if (err != OK) { + return err; + } + + ssize_t numFramesAvailableToWrite = + mAudioTrack->frameCount() - (mNumFramesWritten - numFramesPlayed); + + size_t numBytesAvailableToWrite = + numFramesAvailableToWrite * mAudioTrack->frameSize(); + + if (size > numBytesAvailableToWrite) { + size = numBytesAvailableToWrite; + } + + CHECK_EQ(mAudioTrack->write(data, size), (ssize_t)size); + + size_t numFramesWritten = size / mAudioTrack->frameSize(); + mNumFramesWritten += numFramesWritten; + + return size; +} + +//////////////////////////////////////////////////////////////////////////////// + +DirectRenderer::DirectRenderer( + const sp<IGraphicBufferProducer> &bufferProducer) + : mSurfaceTex(bufferProducer), + mVideoRenderPending(false), + mNumFramesLate(0), + mNumFrames(0) { +} + +DirectRenderer::~DirectRenderer() { +} + +void DirectRenderer::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatDecoderNotify: + { + onDecoderNotify(msg); + break; + } + + case kWhatRenderVideo: + { + onRenderVideo(); + break; + } + + default: + TRESPASS(); + } +} + +void DirectRenderer::setFormat(size_t trackIndex, const sp<AMessage> &format) { + CHECK_LT(trackIndex, 2u); + + CHECK(mDecoderContext[trackIndex] == NULL); + + sp<AMessage> notify = new AMessage(kWhatDecoderNotify, id()); + notify->setSize("trackIndex", trackIndex); + + mDecoderContext[trackIndex] = new DecoderContext(notify); + looper()->registerHandler(mDecoderContext[trackIndex]); + + CHECK_EQ((status_t)OK, + mDecoderContext[trackIndex]->init( + format, trackIndex == 0 ? mSurfaceTex : NULL)); + + if (trackIndex == 1) { + // Audio + mAudioRenderer = new AudioRenderer(mDecoderContext[1]); + looper()->registerHandler(mAudioRenderer); + } +} + +void DirectRenderer::queueAccessUnit( + size_t trackIndex, const sp<ABuffer> &accessUnit) { + CHECK_LT(trackIndex, 2u); + + if (mDecoderContext[trackIndex] == NULL) { + CHECK_EQ(trackIndex, 0u); + + sp<AMessage> format = new AMessage; + format->setString("mime", "video/avc"); + format->setInt32("width", 640); + format->setInt32("height", 360); + + setFormat(trackIndex, format); + } + + mDecoderContext[trackIndex]->queueInputBuffer(accessUnit); +} + +void DirectRenderer::onDecoderNotify(const sp<AMessage> &msg) { + size_t trackIndex; + CHECK(msg->findSize("trackIndex", &trackIndex)); + + int32_t what; + CHECK(msg->findInt32("what", &what)); + + switch (what) { + case DecoderContext::kWhatOutputBufferReady: + { + size_t index; + CHECK(msg->findSize("index", &index)); + + int64_t timeUs; + CHECK(msg->findInt64("timeUs", &timeUs)); + + sp<ABuffer> buffer; + CHECK(msg->findBuffer("buffer", &buffer)); + + queueOutputBuffer(trackIndex, index, timeUs, buffer); + break; + } + + default: + TRESPASS(); + } +} + +void DirectRenderer::queueOutputBuffer( + size_t trackIndex, + size_t index, int64_t timeUs, const sp<ABuffer> &buffer) { + if (trackIndex == 1) { + // Audio + mAudioRenderer->queueInputBuffer(index, timeUs, buffer); + return; + } + + OutputInfo info; + info.mIndex = index; + info.mTimeUs = timeUs; + info.mBuffer = buffer; + mVideoOutputBuffers.push_back(info); + + scheduleVideoRenderIfNecessary(); +} + +void DirectRenderer::scheduleVideoRenderIfNecessary() { + if (mVideoRenderPending || mVideoOutputBuffers.empty()) { + return; + } + + mVideoRenderPending = true; + + int64_t timeUs = (*mVideoOutputBuffers.begin()).mTimeUs; + int64_t nowUs = ALooper::GetNowUs(); + + int64_t delayUs = timeUs - nowUs; + + (new AMessage(kWhatRenderVideo, id()))->post(delayUs); +} + +void DirectRenderer::onRenderVideo() { + mVideoRenderPending = false; + + int64_t nowUs = ALooper::GetNowUs(); + + while (!mVideoOutputBuffers.empty()) { + const OutputInfo &info = *mVideoOutputBuffers.begin(); + + if (info.mTimeUs > nowUs) { + break; + } + + if (info.mTimeUs + 15000ll < nowUs) { + ++mNumFramesLate; + } + ++mNumFrames; + + status_t err = + mDecoderContext[0]->renderOutputBufferAndRelease(info.mIndex); + CHECK_EQ(err, (status_t)OK); + + mVideoOutputBuffers.erase(mVideoOutputBuffers.begin()); + } + + scheduleVideoRenderIfNecessary(); +} + +} // namespace android + diff --git a/media/libstagefright/wifi-display/sink/DirectRenderer.h b/media/libstagefright/wifi-display/sink/DirectRenderer.h new file mode 100644 index 0000000..c5a4a83 --- /dev/null +++ b/media/libstagefright/wifi-display/sink/DirectRenderer.h @@ -0,0 +1,82 @@ +/* + * Copyright 2012, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef DIRECT_RENDERER_H_ + +#define DIRECT_RENDERER_H_ + +#include <media/stagefright/foundation/AHandler.h> + +namespace android { + +struct ABuffer; +struct AudioTrack; +struct IGraphicBufferProducer; +struct MediaCodec; + +// Renders audio and video data queued by calls to "queueAccessUnit". +struct DirectRenderer : public AHandler { + DirectRenderer(const sp<IGraphicBufferProducer> &bufferProducer); + + void setFormat(size_t trackIndex, const sp<AMessage> &format); + void queueAccessUnit(size_t trackIndex, const sp<ABuffer> &accessUnit); + +protected: + virtual void onMessageReceived(const sp<AMessage> &msg); + virtual ~DirectRenderer(); + +private: + struct DecoderContext; + struct AudioRenderer; + + enum { + kWhatDecoderNotify, + kWhatRenderVideo, + }; + + struct OutputInfo { + size_t mIndex; + int64_t mTimeUs; + sp<ABuffer> mBuffer; + }; + + sp<IGraphicBufferProducer> mSurfaceTex; + + sp<DecoderContext> mDecoderContext[2]; + List<OutputInfo> mVideoOutputBuffers; + + bool mVideoRenderPending; + + sp<AudioRenderer> mAudioRenderer; + + int32_t mNumFramesLate; + int32_t mNumFrames; + + void onDecoderNotify(const sp<AMessage> &msg); + + void queueOutputBuffer( + size_t trackIndex, + size_t index, int64_t timeUs, const sp<ABuffer> &buffer); + + void scheduleVideoRenderIfNecessary(); + void onRenderVideo(); + + DISALLOW_EVIL_CONSTRUCTORS(DirectRenderer); +}; + +} // namespace android + +#endif // DIRECT_RENDERER_H_ diff --git a/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp b/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp new file mode 100644 index 0000000..5db2099 --- /dev/null +++ b/media/libstagefright/wifi-display/sink/WifiDisplaySink.cpp @@ -0,0 +1,917 @@ +/* + * Copyright 2012, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NDEBUG 0 +#define LOG_TAG "WifiDisplaySink" +#include <utils/Log.h> + +#include "WifiDisplaySink.h" + +#include "DirectRenderer.h" +#include "MediaReceiver.h" +#include "ParsedMessage.h" +#include "TimeSyncer.h" + +#include <cutils/properties.h> +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/MediaErrors.h> +#include <media/stagefright/Utils.h> + +namespace android { + +// static +const AString WifiDisplaySink::sUserAgent = MakeUserAgent(); + +WifiDisplaySink::WifiDisplaySink( + uint32_t flags, + const sp<ANetworkSession> &netSession, + const sp<IGraphicBufferProducer> &bufferProducer, + const sp<AMessage> ¬ify) + : mState(UNDEFINED), + mFlags(flags), + mNetSession(netSession), + mSurfaceTex(bufferProducer), + mNotify(notify), + mUsingTCPTransport(false), + mUsingTCPInterleaving(false), + mSessionID(0), + mNextCSeq(1), + mIDRFrameRequestPending(false), + mTimeOffsetUs(0ll), + mTimeOffsetValid(false), + mSetupDeferred(false), + mLatencyCount(0), + mLatencySumUs(0ll), + mLatencyMaxUs(0ll), + mMaxDelayMs(-1ll) { + // We support any and all resolutions, but prefer 720p30 + mSinkSupportedVideoFormats.setNativeResolution( + VideoFormats::RESOLUTION_CEA, 5); // 1280 x 720 p30 + + mSinkSupportedVideoFormats.enableAll(); +} + +WifiDisplaySink::~WifiDisplaySink() { +} + +void WifiDisplaySink::start(const char *sourceHost, int32_t sourcePort) { + sp<AMessage> msg = new AMessage(kWhatStart, id()); + msg->setString("sourceHost", sourceHost); + msg->setInt32("sourcePort", sourcePort); + msg->post(); +} + +void WifiDisplaySink::start(const char *uri) { + sp<AMessage> msg = new AMessage(kWhatStart, id()); + msg->setString("setupURI", uri); + msg->post(); +} + +// static +bool WifiDisplaySink::ParseURL( + const char *url, AString *host, int32_t *port, AString *path, + AString *user, AString *pass) { + host->clear(); + *port = 0; + path->clear(); + user->clear(); + pass->clear(); + + if (strncasecmp("rtsp://", url, 7)) { + return false; + } + + const char *slashPos = strchr(&url[7], '/'); + + if (slashPos == NULL) { + host->setTo(&url[7]); + path->setTo("/"); + } else { + host->setTo(&url[7], slashPos - &url[7]); + path->setTo(slashPos); + } + + ssize_t atPos = host->find("@"); + + if (atPos >= 0) { + // Split of user:pass@ from hostname. + + AString userPass(*host, 0, atPos); + host->erase(0, atPos + 1); + + ssize_t colonPos = userPass.find(":"); + + if (colonPos < 0) { + *user = userPass; + } else { + user->setTo(userPass, 0, colonPos); + pass->setTo(userPass, colonPos + 1, userPass.size() - colonPos - 1); + } + } + + const char *colonPos = strchr(host->c_str(), ':'); + + if (colonPos != NULL) { + char *end; + unsigned long x = strtoul(colonPos + 1, &end, 10); + + if (end == colonPos + 1 || *end != '\0' || x >= 65536) { + return false; + } + + *port = x; + + size_t colonOffset = colonPos - host->c_str(); + size_t trailing = host->size() - colonOffset; + host->erase(colonOffset, trailing); + } else { + *port = 554; + } + + return true; +} + +void WifiDisplaySink::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatStart: + { + sleep(2); // XXX + + int32_t sourcePort; + CHECK(msg->findString("sourceHost", &mRTSPHost)); + CHECK(msg->findInt32("sourcePort", &sourcePort)); + + sp<AMessage> notify = new AMessage(kWhatRTSPNotify, id()); + + status_t err = mNetSession->createRTSPClient( + mRTSPHost.c_str(), sourcePort, notify, &mSessionID); + CHECK_EQ(err, (status_t)OK); + + mState = CONNECTING; + break; + } + + case kWhatRTSPNotify: + { + int32_t reason; + CHECK(msg->findInt32("reason", &reason)); + + switch (reason) { + case ANetworkSession::kWhatError: + { + int32_t sessionID; + CHECK(msg->findInt32("sessionID", &sessionID)); + + int32_t err; + CHECK(msg->findInt32("err", &err)); + + AString detail; + CHECK(msg->findString("detail", &detail)); + + ALOGE("An error occurred in session %d (%d, '%s/%s').", + sessionID, + err, + detail.c_str(), + strerror(-err)); + + if (sessionID == mSessionID) { + ALOGI("Lost control connection."); + + // The control connection is dead now. + mNetSession->destroySession(mSessionID); + mSessionID = 0; + + if (mNotify == NULL) { + looper()->stop(); + } else { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatDisconnected); + notify->post(); + } + } + break; + } + + case ANetworkSession::kWhatConnected: + { + ALOGI("We're now connected."); + mState = CONNECTED; + + if (mFlags & FLAG_SPECIAL_MODE) { + sp<AMessage> notify = new AMessage( + kWhatTimeSyncerNotify, id()); + + mTimeSyncer = new TimeSyncer(mNetSession, notify); + looper()->registerHandler(mTimeSyncer); + + mTimeSyncer->startClient(mRTSPHost.c_str(), 8123); + } + break; + } + + case ANetworkSession::kWhatData: + { + onReceiveClientData(msg); + break; + } + + default: + TRESPASS(); + } + break; + } + + case kWhatStop: + { + looper()->stop(); + break; + } + + case kWhatMediaReceiverNotify: + { + onMediaReceiverNotify(msg); + break; + } + + case kWhatTimeSyncerNotify: + { + int32_t what; + CHECK(msg->findInt32("what", &what)); + + if (what == TimeSyncer::kWhatTimeOffset) { + CHECK(msg->findInt64("offset", &mTimeOffsetUs)); + mTimeOffsetValid = true; + + if (mSetupDeferred) { + CHECK_EQ((status_t)OK, + sendSetup( + mSessionID, + "rtsp://x.x.x.x:x/wfd1.0/streamid=0")); + + mSetupDeferred = false; + } + } + break; + } + + case kWhatReportLateness: + { + if (mLatencyCount > 0) { + int64_t avgLatencyUs = mLatencySumUs / mLatencyCount; + + ALOGV("avg. latency = %lld ms (max %lld ms)", + avgLatencyUs / 1000ll, + mLatencyMaxUs / 1000ll); + + sp<AMessage> params = new AMessage; + params->setInt64("avgLatencyUs", avgLatencyUs); + params->setInt64("maxLatencyUs", mLatencyMaxUs); + mMediaReceiver->informSender(0 /* trackIndex */, params); + } + + mLatencyCount = 0; + mLatencySumUs = 0ll; + mLatencyMaxUs = 0ll; + + msg->post(kReportLatenessEveryUs); + break; + } + + default: + TRESPASS(); + } +} + +void WifiDisplaySink::dumpDelay(size_t trackIndex, int64_t timeUs) { + int64_t delayMs = (ALooper::GetNowUs() - timeUs) / 1000ll; + + if (delayMs > mMaxDelayMs) { + mMaxDelayMs = delayMs; + } + + static const int64_t kMinDelayMs = 0; + static const int64_t kMaxDelayMs = 300; + + const char *kPattern = "########################################"; + size_t kPatternSize = strlen(kPattern); + + int n = (kPatternSize * (delayMs - kMinDelayMs)) + / (kMaxDelayMs - kMinDelayMs); + + if (n < 0) { + n = 0; + } else if ((size_t)n > kPatternSize) { + n = kPatternSize; + } + + ALOGI("[%lld]: (%4lld ms / %4lld ms) %s", + timeUs / 1000, + delayMs, + mMaxDelayMs, + kPattern + kPatternSize - n); +} + +void WifiDisplaySink::onMediaReceiverNotify(const sp<AMessage> &msg) { + int32_t what; + CHECK(msg->findInt32("what", &what)); + + switch (what) { + case MediaReceiver::kWhatInitDone: + { + status_t err; + CHECK(msg->findInt32("err", &err)); + + ALOGI("MediaReceiver initialization completed w/ err %d", err); + break; + } + + case MediaReceiver::kWhatError: + { + status_t err; + CHECK(msg->findInt32("err", &err)); + + ALOGE("MediaReceiver signaled error %d", err); + break; + } + + case MediaReceiver::kWhatAccessUnit: + { + if (mRenderer == NULL) { + mRenderer = new DirectRenderer(mSurfaceTex); + looper()->registerHandler(mRenderer); + } + + sp<ABuffer> accessUnit; + CHECK(msg->findBuffer("accessUnit", &accessUnit)); + + int64_t timeUs; + CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); + + if (!mTimeOffsetValid && !(mFlags & FLAG_SPECIAL_MODE)) { + mTimeOffsetUs = timeUs - ALooper::GetNowUs(); + mTimeOffsetValid = true; + } + + CHECK(mTimeOffsetValid); + + // We are the timesync _client_, + // client time = server time - time offset. + timeUs -= mTimeOffsetUs; + + size_t trackIndex; + CHECK(msg->findSize("trackIndex", &trackIndex)); + + int64_t nowUs = ALooper::GetNowUs(); + int64_t delayUs = nowUs - timeUs; + + mLatencySumUs += delayUs; + if (mLatencyCount == 0 || delayUs > mLatencyMaxUs) { + mLatencyMaxUs = delayUs; + } + ++mLatencyCount; + + // dumpDelay(trackIndex, timeUs); + + timeUs += 220000ll; // Assume 220 ms of latency + accessUnit->meta()->setInt64("timeUs", timeUs); + + sp<AMessage> format; + if (msg->findMessage("format", &format)) { + mRenderer->setFormat(trackIndex, format); + } + + mRenderer->queueAccessUnit(trackIndex, accessUnit); + break; + } + + case MediaReceiver::kWhatPacketLost: + { +#if 0 + if (!mIDRFrameRequestPending) { + ALOGI("requesting IDR frame"); + + sendIDRFrameRequest(mSessionID); + } +#endif + break; + } + + default: + TRESPASS(); + } +} + +void WifiDisplaySink::registerResponseHandler( + int32_t sessionID, int32_t cseq, HandleRTSPResponseFunc func) { + ResponseID id; + id.mSessionID = sessionID; + id.mCSeq = cseq; + mResponseHandlers.add(id, func); +} + +status_t WifiDisplaySink::sendM2(int32_t sessionID) { + AString request = "OPTIONS * RTSP/1.0\r\n"; + AppendCommonResponse(&request, mNextCSeq); + + request.append( + "Require: org.wfa.wfd1.0\r\n" + "\r\n"); + + status_t err = + mNetSession->sendRequest(sessionID, request.c_str(), request.size()); + + if (err != OK) { + return err; + } + + registerResponseHandler( + sessionID, mNextCSeq, &WifiDisplaySink::onReceiveM2Response); + + ++mNextCSeq; + + return OK; +} + +status_t WifiDisplaySink::onReceiveM2Response( + int32_t sessionID, const sp<ParsedMessage> &msg) { + int32_t statusCode; + if (!msg->getStatusCode(&statusCode)) { + return ERROR_MALFORMED; + } + + if (statusCode != 200) { + return ERROR_UNSUPPORTED; + } + + return OK; +} + +status_t WifiDisplaySink::onReceiveSetupResponse( + int32_t sessionID, const sp<ParsedMessage> &msg) { + int32_t statusCode; + if (!msg->getStatusCode(&statusCode)) { + return ERROR_MALFORMED; + } + + if (statusCode != 200) { + return ERROR_UNSUPPORTED; + } + + if (!msg->findString("session", &mPlaybackSessionID)) { + return ERROR_MALFORMED; + } + + if (!ParsedMessage::GetInt32Attribute( + mPlaybackSessionID.c_str(), + "timeout", + &mPlaybackSessionTimeoutSecs)) { + mPlaybackSessionTimeoutSecs = -1; + } + + ssize_t colonPos = mPlaybackSessionID.find(";"); + if (colonPos >= 0) { + // Strip any options from the returned session id. + mPlaybackSessionID.erase( + colonPos, mPlaybackSessionID.size() - colonPos); + } + + status_t err = configureTransport(msg); + + if (err != OK) { + return err; + } + + mState = PAUSED; + + return sendPlay( + sessionID, + "rtsp://x.x.x.x:x/wfd1.0/streamid=0"); +} + +status_t WifiDisplaySink::configureTransport(const sp<ParsedMessage> &msg) { + if (mUsingTCPTransport && !(mFlags & FLAG_SPECIAL_MODE)) { + // In "special" mode we still use a UDP RTCP back-channel that + // needs connecting. + return OK; + } + + AString transport; + if (!msg->findString("transport", &transport)) { + ALOGE("Missing 'transport' field in SETUP response."); + return ERROR_MALFORMED; + } + + AString sourceHost; + if (!ParsedMessage::GetAttribute( + transport.c_str(), "source", &sourceHost)) { + sourceHost = mRTSPHost; + } + + AString serverPortStr; + if (!ParsedMessage::GetAttribute( + transport.c_str(), "server_port", &serverPortStr)) { + ALOGE("Missing 'server_port' in Transport field."); + return ERROR_MALFORMED; + } + + int rtpPort, rtcpPort; + if (sscanf(serverPortStr.c_str(), "%d-%d", &rtpPort, &rtcpPort) != 2 + || rtpPort <= 0 || rtpPort > 65535 + || rtcpPort <=0 || rtcpPort > 65535 + || rtcpPort != rtpPort + 1) { + ALOGE("Invalid server_port description '%s'.", + serverPortStr.c_str()); + + return ERROR_MALFORMED; + } + + if (rtpPort & 1) { + ALOGW("Server picked an odd numbered RTP port."); + } + + return mMediaReceiver->connectTrack( + 0 /* trackIndex */, sourceHost.c_str(), rtpPort, rtcpPort); +} + +status_t WifiDisplaySink::onReceivePlayResponse( + int32_t sessionID, const sp<ParsedMessage> &msg) { + int32_t statusCode; + if (!msg->getStatusCode(&statusCode)) { + return ERROR_MALFORMED; + } + + if (statusCode != 200) { + return ERROR_UNSUPPORTED; + } + + mState = PLAYING; + + (new AMessage(kWhatReportLateness, id()))->post(kReportLatenessEveryUs); + + return OK; +} + +status_t WifiDisplaySink::onReceiveIDRFrameRequestResponse( + int32_t sessionID, const sp<ParsedMessage> &msg) { + CHECK(mIDRFrameRequestPending); + mIDRFrameRequestPending = false; + + return OK; +} + +void WifiDisplaySink::onReceiveClientData(const sp<AMessage> &msg) { + int32_t sessionID; + CHECK(msg->findInt32("sessionID", &sessionID)); + + sp<RefBase> obj; + CHECK(msg->findObject("data", &obj)); + + sp<ParsedMessage> data = + static_cast<ParsedMessage *>(obj.get()); + + ALOGV("session %d received '%s'", + sessionID, data->debugString().c_str()); + + AString method; + AString uri; + data->getRequestField(0, &method); + + int32_t cseq; + if (!data->findInt32("cseq", &cseq)) { + sendErrorResponse(sessionID, "400 Bad Request", -1 /* cseq */); + return; + } + + if (method.startsWith("RTSP/")) { + // This is a response. + + ResponseID id; + id.mSessionID = sessionID; + id.mCSeq = cseq; + + ssize_t index = mResponseHandlers.indexOfKey(id); + + if (index < 0) { + ALOGW("Received unsolicited server response, cseq %d", cseq); + return; + } + + HandleRTSPResponseFunc func = mResponseHandlers.valueAt(index); + mResponseHandlers.removeItemsAt(index); + + status_t err = (this->*func)(sessionID, data); + CHECK_EQ(err, (status_t)OK); + } else { + AString version; + data->getRequestField(2, &version); + if (!(version == AString("RTSP/1.0"))) { + sendErrorResponse(sessionID, "505 RTSP Version not supported", cseq); + return; + } + + if (method == "OPTIONS") { + onOptionsRequest(sessionID, cseq, data); + } else if (method == "GET_PARAMETER") { + onGetParameterRequest(sessionID, cseq, data); + } else if (method == "SET_PARAMETER") { + onSetParameterRequest(sessionID, cseq, data); + } else { + sendErrorResponse(sessionID, "405 Method Not Allowed", cseq); + } + } +} + +void WifiDisplaySink::onOptionsRequest( + int32_t sessionID, + int32_t cseq, + const sp<ParsedMessage> &data) { + AString response = "RTSP/1.0 200 OK\r\n"; + AppendCommonResponse(&response, cseq); + response.append("Public: org.wfa.wfd1.0, GET_PARAMETER, SET_PARAMETER\r\n"); + response.append("\r\n"); + + status_t err = mNetSession->sendRequest(sessionID, response.c_str()); + CHECK_EQ(err, (status_t)OK); + + err = sendM2(sessionID); + CHECK_EQ(err, (status_t)OK); +} + +void WifiDisplaySink::onGetParameterRequest( + int32_t sessionID, + int32_t cseq, + const sp<ParsedMessage> &data) { + AString body; + + if (mState == CONNECTED) { + mUsingTCPTransport = false; + mUsingTCPInterleaving = false; + + char val[PROPERTY_VALUE_MAX]; + if (property_get("media.wfd-sink.tcp-mode", val, NULL)) { + if (!strcasecmp("true", val) || !strcmp("1", val)) { + ALOGI("Using TCP unicast transport."); + mUsingTCPTransport = true; + mUsingTCPInterleaving = false; + } else if (!strcasecmp("interleaved", val)) { + ALOGI("Using TCP interleaved transport."); + mUsingTCPTransport = true; + mUsingTCPInterleaving = true; + } + } else if (mFlags & FLAG_SPECIAL_MODE) { + mUsingTCPTransport = true; + } + + body = "wfd_video_formats: "; + body.append(mSinkSupportedVideoFormats.getFormatSpec()); + + body.append( + "\r\nwfd_audio_codecs: AAC 0000000F 00\r\n" + "wfd_client_rtp_ports: RTP/AVP/"); + + if (mUsingTCPTransport) { + body.append("TCP;"); + if (mUsingTCPInterleaving) { + body.append("interleaved"); + } else { + body.append("unicast 19000 0"); + } + } else { + body.append("UDP;unicast 19000 0"); + } + + body.append(" mode=play\r\n"); + } + + AString response = "RTSP/1.0 200 OK\r\n"; + AppendCommonResponse(&response, cseq); + response.append("Content-Type: text/parameters\r\n"); + response.append(StringPrintf("Content-Length: %d\r\n", body.size())); + response.append("\r\n"); + response.append(body); + + status_t err = mNetSession->sendRequest(sessionID, response.c_str()); + CHECK_EQ(err, (status_t)OK); +} + +status_t WifiDisplaySink::sendSetup(int32_t sessionID, const char *uri) { + sp<AMessage> notify = new AMessage(kWhatMediaReceiverNotify, id()); + + mMediaReceiverLooper = new ALooper; + mMediaReceiverLooper->setName("media_receiver"); + + mMediaReceiverLooper->start( + false /* runOnCallingThread */, + false /* canCallJava */, + PRIORITY_AUDIO); + + mMediaReceiver = new MediaReceiver(mNetSession, notify); + mMediaReceiverLooper->registerHandler(mMediaReceiver); + + RTPReceiver::TransportMode rtpMode = RTPReceiver::TRANSPORT_UDP; + if (mUsingTCPTransport) { + if (mUsingTCPInterleaving) { + rtpMode = RTPReceiver::TRANSPORT_TCP_INTERLEAVED; + } else { + rtpMode = RTPReceiver::TRANSPORT_TCP; + } + } + + int32_t localRTPPort; + status_t err = mMediaReceiver->addTrack( + rtpMode, RTPReceiver::TRANSPORT_UDP /* rtcpMode */, &localRTPPort); + + if (err == OK) { + err = mMediaReceiver->initAsync(MediaReceiver::MODE_TRANSPORT_STREAM); + } + + if (err != OK) { + mMediaReceiverLooper->unregisterHandler(mMediaReceiver->id()); + mMediaReceiver.clear(); + + mMediaReceiverLooper->stop(); + mMediaReceiverLooper.clear(); + + return err; + } + + AString request = StringPrintf("SETUP %s RTSP/1.0\r\n", uri); + + AppendCommonResponse(&request, mNextCSeq); + + if (rtpMode == RTPReceiver::TRANSPORT_TCP_INTERLEAVED) { + request.append("Transport: RTP/AVP/TCP;interleaved=0-1\r\n"); + } else if (rtpMode == RTPReceiver::TRANSPORT_TCP) { + if (mFlags & FLAG_SPECIAL_MODE) { + // This isn't quite true, since the RTP connection is through TCP + // and the RTCP connection through UDP... + request.append( + StringPrintf( + "Transport: RTP/AVP/TCP;unicast;client_port=%d-%d\r\n", + localRTPPort, localRTPPort + 1)); + } else { + request.append( + StringPrintf( + "Transport: RTP/AVP/TCP;unicast;client_port=%d\r\n", + localRTPPort)); + } + } else { + request.append( + StringPrintf( + "Transport: RTP/AVP/UDP;unicast;client_port=%d-%d\r\n", + localRTPPort, + localRTPPort + 1)); + } + + request.append("\r\n"); + + ALOGV("request = '%s'", request.c_str()); + + err = mNetSession->sendRequest(sessionID, request.c_str(), request.size()); + + if (err != OK) { + return err; + } + + registerResponseHandler( + sessionID, mNextCSeq, &WifiDisplaySink::onReceiveSetupResponse); + + ++mNextCSeq; + + return OK; +} + +status_t WifiDisplaySink::sendPlay(int32_t sessionID, const char *uri) { + AString request = StringPrintf("PLAY %s RTSP/1.0\r\n", uri); + + AppendCommonResponse(&request, mNextCSeq); + + request.append(StringPrintf("Session: %s\r\n", mPlaybackSessionID.c_str())); + request.append("\r\n"); + + status_t err = + mNetSession->sendRequest(sessionID, request.c_str(), request.size()); + + if (err != OK) { + return err; + } + + registerResponseHandler( + sessionID, mNextCSeq, &WifiDisplaySink::onReceivePlayResponse); + + ++mNextCSeq; + + return OK; +} + +status_t WifiDisplaySink::sendIDRFrameRequest(int32_t sessionID) { + CHECK(!mIDRFrameRequestPending); + + AString request = "SET_PARAMETER rtsp://localhost/wfd1.0 RTSP/1.0\r\n"; + + AppendCommonResponse(&request, mNextCSeq); + + AString content = "wfd_idr_request\r\n"; + + request.append(StringPrintf("Session: %s\r\n", mPlaybackSessionID.c_str())); + request.append(StringPrintf("Content-Length: %d\r\n", content.size())); + request.append("\r\n"); + request.append(content); + + status_t err = + mNetSession->sendRequest(sessionID, request.c_str(), request.size()); + + if (err != OK) { + return err; + } + + registerResponseHandler( + sessionID, + mNextCSeq, + &WifiDisplaySink::onReceiveIDRFrameRequestResponse); + + ++mNextCSeq; + + mIDRFrameRequestPending = true; + + return OK; +} + +void WifiDisplaySink::onSetParameterRequest( + int32_t sessionID, + int32_t cseq, + const sp<ParsedMessage> &data) { + const char *content = data->getContent(); + + if (strstr(content, "wfd_trigger_method: SETUP\r\n") != NULL) { + if ((mFlags & FLAG_SPECIAL_MODE) && !mTimeOffsetValid) { + mSetupDeferred = true; + } else { + status_t err = + sendSetup( + sessionID, + "rtsp://x.x.x.x:x/wfd1.0/streamid=0"); + + CHECK_EQ(err, (status_t)OK); + } + } + + AString response = "RTSP/1.0 200 OK\r\n"; + AppendCommonResponse(&response, cseq); + response.append("\r\n"); + + status_t err = mNetSession->sendRequest(sessionID, response.c_str()); + CHECK_EQ(err, (status_t)OK); +} + +void WifiDisplaySink::sendErrorResponse( + int32_t sessionID, + const char *errorDetail, + int32_t cseq) { + AString response; + response.append("RTSP/1.0 "); + response.append(errorDetail); + response.append("\r\n"); + + AppendCommonResponse(&response, cseq); + + response.append("\r\n"); + + status_t err = mNetSession->sendRequest(sessionID, response.c_str()); + CHECK_EQ(err, (status_t)OK); +} + +// static +void WifiDisplaySink::AppendCommonResponse(AString *response, int32_t cseq) { + time_t now = time(NULL); + struct tm *now2 = gmtime(&now); + char buf[128]; + strftime(buf, sizeof(buf), "%a, %d %b %Y %H:%M:%S %z", now2); + + response->append("Date: "); + response->append(buf); + response->append("\r\n"); + + response->append(StringPrintf("User-Agent: %s\r\n", sUserAgent.c_str())); + + if (cseq >= 0) { + response->append(StringPrintf("CSeq: %d\r\n", cseq)); + } +} + +} // namespace android diff --git a/media/libstagefright/wifi-display/sink/WifiDisplaySink.h b/media/libstagefright/wifi-display/sink/WifiDisplaySink.h new file mode 100644 index 0000000..adb9d89 --- /dev/null +++ b/media/libstagefright/wifi-display/sink/WifiDisplaySink.h @@ -0,0 +1,196 @@ +/* + * Copyright 2012, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef WIFI_DISPLAY_SINK_H_ + +#define WIFI_DISPLAY_SINK_H_ + +#include "ANetworkSession.h" + +#include "VideoFormats.h" + +#include <gui/Surface.h> +#include <media/stagefright/foundation/AHandler.h> + +namespace android { + +struct AMessage; +struct DirectRenderer; +struct MediaReceiver; +struct ParsedMessage; +struct TimeSyncer; + +// Represents the RTSP client acting as a wifi display sink. +// Connects to a wifi display source and renders the incoming +// transport stream using a MediaPlayer instance. +struct WifiDisplaySink : public AHandler { + enum { + kWhatDisconnected, + }; + + enum Flags { + FLAG_SPECIAL_MODE = 1, + }; + + // If no notification message is specified (notify == NULL) + // the sink will stop its looper() once the session ends, + // otherwise it will post an appropriate notification but leave + // the looper() running. + WifiDisplaySink( + uint32_t flags, + const sp<ANetworkSession> &netSession, + const sp<IGraphicBufferProducer> &bufferProducer = NULL, + const sp<AMessage> ¬ify = NULL); + + void start(const char *sourceHost, int32_t sourcePort); + void start(const char *uri); + +protected: + virtual ~WifiDisplaySink(); + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum State { + UNDEFINED, + CONNECTING, + CONNECTED, + PAUSED, + PLAYING, + }; + + enum { + kWhatStart, + kWhatRTSPNotify, + kWhatStop, + kWhatMediaReceiverNotify, + kWhatTimeSyncerNotify, + kWhatReportLateness, + }; + + struct ResponseID { + int32_t mSessionID; + int32_t mCSeq; + + bool operator<(const ResponseID &other) const { + return mSessionID < other.mSessionID + || (mSessionID == other.mSessionID + && mCSeq < other.mCSeq); + } + }; + + typedef status_t (WifiDisplaySink::*HandleRTSPResponseFunc)( + int32_t sessionID, const sp<ParsedMessage> &msg); + + static const int64_t kReportLatenessEveryUs = 1000000ll; + + static const AString sUserAgent; + + State mState; + uint32_t mFlags; + VideoFormats mSinkSupportedVideoFormats; + sp<ANetworkSession> mNetSession; + sp<IGraphicBufferProducer> mSurfaceTex; + sp<AMessage> mNotify; + sp<TimeSyncer> mTimeSyncer; + bool mUsingTCPTransport; + bool mUsingTCPInterleaving; + AString mRTSPHost; + int32_t mSessionID; + + int32_t mNextCSeq; + + KeyedVector<ResponseID, HandleRTSPResponseFunc> mResponseHandlers; + + sp<ALooper> mMediaReceiverLooper; + sp<MediaReceiver> mMediaReceiver; + sp<DirectRenderer> mRenderer; + + AString mPlaybackSessionID; + int32_t mPlaybackSessionTimeoutSecs; + + bool mIDRFrameRequestPending; + + int64_t mTimeOffsetUs; + bool mTimeOffsetValid; + + bool mSetupDeferred; + + size_t mLatencyCount; + int64_t mLatencySumUs; + int64_t mLatencyMaxUs; + + int64_t mMaxDelayMs; + + status_t sendM2(int32_t sessionID); + status_t sendSetup(int32_t sessionID, const char *uri); + status_t sendPlay(int32_t sessionID, const char *uri); + status_t sendIDRFrameRequest(int32_t sessionID); + + status_t onReceiveM2Response( + int32_t sessionID, const sp<ParsedMessage> &msg); + + status_t onReceiveSetupResponse( + int32_t sessionID, const sp<ParsedMessage> &msg); + + status_t configureTransport(const sp<ParsedMessage> &msg); + + status_t onReceivePlayResponse( + int32_t sessionID, const sp<ParsedMessage> &msg); + + status_t onReceiveIDRFrameRequestResponse( + int32_t sessionID, const sp<ParsedMessage> &msg); + + void registerResponseHandler( + int32_t sessionID, int32_t cseq, HandleRTSPResponseFunc func); + + void onReceiveClientData(const sp<AMessage> &msg); + + void onOptionsRequest( + int32_t sessionID, + int32_t cseq, + const sp<ParsedMessage> &data); + + void onGetParameterRequest( + int32_t sessionID, + int32_t cseq, + const sp<ParsedMessage> &data); + + void onSetParameterRequest( + int32_t sessionID, + int32_t cseq, + const sp<ParsedMessage> &data); + + void onMediaReceiverNotify(const sp<AMessage> &msg); + + void sendErrorResponse( + int32_t sessionID, + const char *errorDetail, + int32_t cseq); + + static void AppendCommonResponse(AString *response, int32_t cseq); + + bool ParseURL( + const char *url, AString *host, int32_t *port, AString *path, + AString *user, AString *pass); + + void dumpDelay(size_t trackIndex, int64_t timeUs); + + DISALLOW_EVIL_CONSTRUCTORS(WifiDisplaySink); +}; + +} // namespace android + +#endif // WIFI_DISPLAY_SINK_H_ diff --git a/media/libstagefright/wifi-display/source/Converter.cpp b/media/libstagefright/wifi-display/source/Converter.cpp index 5344623..0214520 100644 --- a/media/libstagefright/wifi-display/source/Converter.cpp +++ b/media/libstagefright/wifi-display/source/Converter.cpp @@ -40,14 +40,13 @@ namespace android { Converter::Converter( const sp<AMessage> ¬ify, const sp<ALooper> &codecLooper, - const sp<AMessage> &format, - bool usePCMAudio) + const sp<AMessage> &outputFormat) : mInitCheck(NO_INIT), mNotify(notify), mCodecLooper(codecLooper), - mInputFormat(format), + mOutputFormat(outputFormat), mIsVideo(false), - mIsPCMAudio(usePCMAudio), + mIsPCMAudio(false), mNeedToManuallyPrependSPSPPS(false), mDoMoreWorkPending(false) #if ENABLE_SILENCE_DETECTION @@ -58,14 +57,14 @@ Converter::Converter( ,mNumFramesToDrop(0) { AString mime; - CHECK(mInputFormat->findString("mime", &mime)); + CHECK(mOutputFormat->findString("mime", &mime)); if (!strncasecmp("video/", mime.c_str(), 6)) { mIsVideo = true; + } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_RAW, mime.c_str())) { + mIsPCMAudio = true; } - CHECK(!usePCMAudio || !mIsVideo); - mInitCheck = initEncoder(); if (mInitCheck != OK) { @@ -152,23 +151,10 @@ int32_t Converter::GetInt32Property( } status_t Converter::initEncoder() { - AString inputMIME; - CHECK(mInputFormat->findString("mime", &inputMIME)); - AString outputMIME; - bool isAudio = false; - if (!strcasecmp(inputMIME.c_str(), MEDIA_MIMETYPE_AUDIO_RAW)) { - if (mIsPCMAudio) { - outputMIME = MEDIA_MIMETYPE_AUDIO_RAW; - } else { - outputMIME = MEDIA_MIMETYPE_AUDIO_AAC; - } - isAudio = true; - } else if (!strcasecmp(inputMIME.c_str(), MEDIA_MIMETYPE_VIDEO_RAW)) { - outputMIME = MEDIA_MIMETYPE_VIDEO_AVC; - } else { - TRESPASS(); - } + CHECK(mOutputFormat->findString("mime", &outputMIME)); + + bool isAudio = !strncasecmp(outputMIME.c_str(), "audio/", 6); if (!mIsPCMAudio) { mEncoder = MediaCodec::CreateByType( @@ -179,14 +165,10 @@ status_t Converter::initEncoder() { } } - mOutputFormat = mInputFormat->dup(); - if (mIsPCMAudio) { return OK; } - mOutputFormat->setString("mime", outputMIME.c_str()); - int32_t audioBitrate = GetInt32Property("media.wfd.audio-bitrate", 128000); int32_t videoBitrate = GetInt32Property("media.wfd.video-bitrate", 5000000); mPrevVideoBitrate = videoBitrate; @@ -427,7 +409,7 @@ void Converter::onMessageReceived(const sp<AMessage> &msg) { releaseEncoder(); AString mime; - CHECK(mInputFormat->findString("mime", &mime)); + CHECK(mOutputFormat->findString("mime", &mime)); ALOGI("encoder (%s) shut down.", mime.c_str()); break; } @@ -438,6 +420,17 @@ void Converter::onMessageReceived(const sp<AMessage> &msg) { break; } + case kWhatReleaseOutputBuffer: + { + if (mEncoder != NULL) { + size_t bufferIndex; + CHECK(msg->findInt32("bufferIndex", (int32_t*)&bufferIndex)); + CHECK(bufferIndex < mEncoderOutputBuffers.size()); + mEncoder->releaseOutputBuffer(bufferIndex); + } + break; + } + default: TRESPASS(); } @@ -645,6 +638,7 @@ status_t Converter::doMoreWork() { size_t size; int64_t timeUs; uint32_t flags; + native_handle_t* handle = NULL; err = mEncoder->dequeueOutputBuffer( &bufferIndex, &offset, &size, &timeUs, &flags); @@ -667,18 +661,52 @@ status_t Converter::doMoreWork() { notify->setInt32("what", kWhatEOS); notify->post(); } else { - sp<ABuffer> buffer = new ABuffer(size); +#if 0 + if (mIsVideo) { + int32_t videoBitrate = GetInt32Property( + "media.wfd.video-bitrate", 5000000); + + setVideoBitrate(videoBitrate); + } +#endif + + sp<ABuffer> buffer; + sp<ABuffer> outbuf = mEncoderOutputBuffers.itemAt(bufferIndex); + + if (outbuf->meta()->findPointer("handle", (void**)&handle) && + handle != NULL) { + int32_t rangeLength, rangeOffset; + CHECK(outbuf->meta()->findInt32("rangeOffset", &rangeOffset)); + CHECK(outbuf->meta()->findInt32("rangeLength", &rangeLength)); + outbuf->meta()->setPointer("handle", NULL); + + // MediaSender will post the following message when HDCP + // is done, to release the output buffer back to encoder. + sp<AMessage> notify(new AMessage( + kWhatReleaseOutputBuffer, id())); + notify->setInt32("bufferIndex", bufferIndex); + + buffer = new ABuffer( + rangeLength > (int32_t)size ? rangeLength : size); + buffer->meta()->setPointer("handle", handle); + buffer->meta()->setInt32("rangeOffset", rangeOffset); + buffer->meta()->setInt32("rangeLength", rangeLength); + buffer->meta()->setMessage("notify", notify); + } else { + buffer = new ABuffer(size); + } + buffer->meta()->setInt64("timeUs", timeUs); ALOGV("[%s] time %lld us (%.2f secs)", mIsVideo ? "video" : "audio", timeUs, timeUs / 1E6); - memcpy(buffer->data(), - mEncoderOutputBuffers.itemAt(bufferIndex)->base() + offset, - size); + memcpy(buffer->data(), outbuf->base() + offset, size); if (flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) { - mOutputFormat->setBuffer("csd-0", buffer); + if (!handle) { + mOutputFormat->setBuffer("csd-0", buffer); + } } else { sp<AMessage> notify = mNotify->dup(); notify->setInt32("what", kWhatAccessUnit); @@ -687,7 +715,9 @@ status_t Converter::doMoreWork() { } } - mEncoder->releaseOutputBuffer(bufferIndex); + if (!handle) { + mEncoder->releaseOutputBuffer(bufferIndex); + } if (flags & MediaCodec::BUFFER_FLAG_EOS) { break; diff --git a/media/libstagefright/wifi-display/source/Converter.h b/media/libstagefright/wifi-display/source/Converter.h index ba297c4..76c8b19 100644 --- a/media/libstagefright/wifi-display/source/Converter.h +++ b/media/libstagefright/wifi-display/source/Converter.h @@ -33,11 +33,9 @@ struct MediaCodec; // media access unit of a different format. // Right now this'll convert raw video into H.264 and raw audio into AAC. struct Converter : public AHandler { - Converter( - const sp<AMessage> ¬ify, - const sp<ALooper> &codecLooper, - const sp<AMessage> &format, - bool usePCMAudio); + Converter(const sp<AMessage> ¬ify, + const sp<ALooper> &codecLooper, + const sp<AMessage> &outputFormat); status_t initCheck() const; @@ -66,6 +64,7 @@ struct Converter : public AHandler { kWhatMediaPullerNotify, kWhatEncoderActivity, kWhatDropAFrame, + kWhatReleaseOutputBuffer, }; void shutdownAsync(); @@ -83,10 +82,9 @@ private: status_t mInitCheck; sp<AMessage> mNotify; sp<ALooper> mCodecLooper; - sp<AMessage> mInputFormat; + sp<AMessage> mOutputFormat; bool mIsVideo; bool mIsPCMAudio; - sp<AMessage> mOutputFormat; bool mNeedToManuallyPrependSPSPPS; sp<MediaCodec> mEncoder; diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.cpp b/media/libstagefright/wifi-display/source/PlaybackSession.cpp index 3d7b865..a15fbac 100644 --- a/media/libstagefright/wifi-display/source/PlaybackSession.cpp +++ b/media/libstagefright/wifi-display/source/PlaybackSession.cpp @@ -378,7 +378,9 @@ status_t WifiDisplaySource::PlaybackSession::init( bool usePCMAudio, bool enableVideo, VideoFormats::ResolutionType videoResolutionType, - size_t videoResolutionIndex) { + size_t videoResolutionIndex, + VideoFormats::ProfileType videoProfileType, + VideoFormats::LevelType videoLevelType) { sp<AMessage> notify = new AMessage(kWhatMediaSenderNotify, id()); mMediaSender = new MediaSender(mNetSession, notify); looper()->registerHandler(mMediaSender); @@ -390,7 +392,9 @@ status_t WifiDisplaySource::PlaybackSession::init( usePCMAudio, enableVideo, videoResolutionType, - videoResolutionIndex); + videoResolutionIndex, + videoProfileType, + videoLevelType); if (err == OK) { err = mMediaSender->initAsync( @@ -559,6 +563,8 @@ void WifiDisplaySource::PlaybackSession::onMessageReceived( converter->dropAFrame(); } } + } else if (what == MediaSender::kWhatInformSender) { + onSinkFeedback(msg); } else { TRESPASS(); } @@ -654,6 +660,89 @@ void WifiDisplaySource::PlaybackSession::onMessageReceived( } } +void WifiDisplaySource::PlaybackSession::onSinkFeedback(const sp<AMessage> &msg) { + int64_t avgLatencyUs; + CHECK(msg->findInt64("avgLatencyUs", &avgLatencyUs)); + + int64_t maxLatencyUs; + CHECK(msg->findInt64("maxLatencyUs", &maxLatencyUs)); + + ALOGI("sink reports avg. latency of %lld ms (max %lld ms)", + avgLatencyUs / 1000ll, + maxLatencyUs / 1000ll); + + if (mVideoTrackIndex >= 0) { + const sp<Track> &videoTrack = mTracks.valueFor(mVideoTrackIndex); + sp<Converter> converter = videoTrack->converter(); + + if (converter != NULL) { + int32_t videoBitrate = + Converter::GetInt32Property("media.wfd.video-bitrate", -1); + + char val[PROPERTY_VALUE_MAX]; + if (videoBitrate < 0 + && property_get("media.wfd.video-bitrate", val, NULL) + && !strcasecmp("adaptive", val)) { + videoBitrate = converter->getVideoBitrate(); + + if (avgLatencyUs > 300000ll) { + videoBitrate *= 0.6; + } else if (avgLatencyUs < 100000ll) { + videoBitrate *= 1.1; + } + } + + if (videoBitrate > 0) { + if (videoBitrate < 500000) { + videoBitrate = 500000; + } else if (videoBitrate > 10000000) { + videoBitrate = 10000000; + } + + if (videoBitrate != converter->getVideoBitrate()) { + ALOGI("setting video bitrate to %d bps", videoBitrate); + + converter->setVideoBitrate(videoBitrate); + } + } + } + + sp<RepeaterSource> repeaterSource = videoTrack->repeaterSource(); + if (repeaterSource != NULL) { + double rateHz = + Converter::GetInt32Property( + "media.wfd.video-framerate", -1); + + char val[PROPERTY_VALUE_MAX]; + if (rateHz < 0.0 + && property_get("media.wfd.video-framerate", val, NULL) + && !strcasecmp("adaptive", val)) { + rateHz = repeaterSource->getFrameRate(); + + if (avgLatencyUs > 300000ll) { + rateHz *= 0.9; + } else if (avgLatencyUs < 200000ll) { + rateHz *= 1.1; + } + } + + if (rateHz > 0) { + if (rateHz < 5.0) { + rateHz = 5.0; + } else if (rateHz > 30.0) { + rateHz = 30.0; + } + + if (rateHz != repeaterSource->getFrameRate()) { + ALOGI("setting frame rate to %.2f Hz", rateHz); + + repeaterSource->setFrameRate(rateHz); + } + } + } + } +} + status_t WifiDisplaySource::PlaybackSession::setupMediaPacketizer( bool enableAudio, bool enableVideo) { DataSource::RegisterDefaultSniffers(); @@ -785,7 +874,9 @@ status_t WifiDisplaySource::PlaybackSession::setupPacketizer( bool usePCMAudio, bool enableVideo, VideoFormats::ResolutionType videoResolutionType, - size_t videoResolutionIndex) { + size_t videoResolutionIndex, + VideoFormats::ProfileType videoProfileType, + VideoFormats::LevelType videoLevelType) { CHECK(enableAudio || enableVideo); if (!mMediaPath.empty()) { @@ -794,7 +885,8 @@ status_t WifiDisplaySource::PlaybackSession::setupPacketizer( if (enableVideo) { status_t err = addVideoSource( - videoResolutionType, videoResolutionIndex); + videoResolutionType, videoResolutionIndex, videoProfileType, + videoLevelType); if (err != OK) { return err; @@ -810,9 +902,13 @@ status_t WifiDisplaySource::PlaybackSession::setupPacketizer( status_t WifiDisplaySource::PlaybackSession::addSource( bool isVideo, const sp<MediaSource> &source, bool isRepeaterSource, - bool usePCMAudio, size_t *numInputBuffers) { + bool usePCMAudio, unsigned profileIdc, unsigned levelIdc, + unsigned constraintSet, size_t *numInputBuffers) { CHECK(!usePCMAudio || !isVideo); CHECK(!isRepeaterSource || isVideo); + CHECK(!profileIdc || isVideo); + CHECK(!levelIdc || isVideo); + CHECK(!constraintSet || isVideo); sp<ALooper> pullLooper = new ALooper; pullLooper->setName("pull_looper"); @@ -841,17 +937,25 @@ status_t WifiDisplaySource::PlaybackSession::addSource( CHECK_EQ(err, (status_t)OK); if (isVideo) { + format->setString("mime", MEDIA_MIMETYPE_VIDEO_AVC); format->setInt32("store-metadata-in-buffers", true); - + format->setInt32("store-metadata-in-buffers-output", (mHDCP != NULL)); format->setInt32( "color-format", OMX_COLOR_FormatAndroidOpaque); + format->setInt32("profile-idc", profileIdc); + format->setInt32("level-idc", levelIdc); + format->setInt32("constraint-set", constraintSet); + } else { + format->setString( + "mime", + usePCMAudio + ? MEDIA_MIMETYPE_AUDIO_RAW : MEDIA_MIMETYPE_AUDIO_AAC); } notify = new AMessage(kWhatConverterNotify, id()); notify->setSize("trackIndex", trackIndex); - sp<Converter> converter = - new Converter(notify, codecLooper, format, usePCMAudio); + sp<Converter> converter = new Converter(notify, codecLooper, format); err = converter->initCheck(); if (err != OK) { @@ -905,7 +1009,9 @@ status_t WifiDisplaySource::PlaybackSession::addSource( status_t WifiDisplaySource::PlaybackSession::addVideoSource( VideoFormats::ResolutionType videoResolutionType, - size_t videoResolutionIndex) { + size_t videoResolutionIndex, + VideoFormats::ProfileType videoProfileType, + VideoFormats::LevelType videoLevelType) { size_t width, height, framesPerSecond; bool interlaced; CHECK(VideoFormats::GetConfiguration( @@ -916,6 +1022,14 @@ status_t WifiDisplaySource::PlaybackSession::addVideoSource( &framesPerSecond, &interlaced)); + unsigned profileIdc, levelIdc, constraintSet; + CHECK(VideoFormats::GetProfileLevel( + videoProfileType, + videoLevelType, + &profileIdc, + &levelIdc, + &constraintSet)); + sp<SurfaceMediaSource> source = new SurfaceMediaSource(width, height); source->setUseAbsoluteTimestamps(); @@ -926,7 +1040,8 @@ status_t WifiDisplaySource::PlaybackSession::addVideoSource( size_t numInputBuffers; status_t err = addSource( true /* isVideo */, videoSource, true /* isRepeaterSource */, - false /* usePCMAudio */, &numInputBuffers); + false /* usePCMAudio */, profileIdc, levelIdc, constraintSet, + &numInputBuffers); if (err != OK) { return err; @@ -949,7 +1064,8 @@ status_t WifiDisplaySource::PlaybackSession::addAudioSource(bool usePCMAudio) { if (audioSource->initCheck() == OK) { return addSource( false /* isVideo */, audioSource, false /* isRepeaterSource */, - usePCMAudio, NULL /* numInputBuffers */); + usePCMAudio, 0 /* profileIdc */, 0 /* levelIdc */, + 0 /* constraintSet */, NULL /* numInputBuffers */); } ALOGW("Unable to instantiate audio source"); diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.h b/media/libstagefright/wifi-display/source/PlaybackSession.h index 39086a1..5c8ee94 100644 --- a/media/libstagefright/wifi-display/source/PlaybackSession.h +++ b/media/libstagefright/wifi-display/source/PlaybackSession.h @@ -53,7 +53,9 @@ struct WifiDisplaySource::PlaybackSession : public AHandler { bool usePCMAudio, bool enableVideo, VideoFormats::ResolutionType videoResolutionType, - size_t videoResolutionIndex); + size_t videoResolutionIndex, + VideoFormats::ProfileType videoProfileType, + VideoFormats::LevelType videoLevelType); void destroyAsync(); @@ -130,18 +132,25 @@ private: bool usePCMAudio, bool enableVideo, VideoFormats::ResolutionType videoResolutionType, - size_t videoResolutionIndex); + size_t videoResolutionIndex, + VideoFormats::ProfileType videoProfileType, + VideoFormats::LevelType videoLevelType); status_t addSource( bool isVideo, const sp<MediaSource> &source, bool isRepeaterSource, bool usePCMAudio, + unsigned profileIdc, + unsigned levelIdc, + unsigned contraintSet, size_t *numInputBuffers); status_t addVideoSource( VideoFormats::ResolutionType videoResolutionType, - size_t videoResolutionIndex); + size_t videoResolutionIndex, + VideoFormats::ProfileType videoProfileType, + VideoFormats::LevelType videoLevelType); status_t addAudioSource(bool usePCMAudio); diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp index 2c4a373..c674700 100644 --- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp +++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp @@ -261,12 +261,24 @@ void TSPacketizer::Track::finalize() { data[0] = 40; // descriptor_tag data[1] = 4; // descriptor_length - CHECK_GE(mCSD.size(), 1u); - const sp<ABuffer> &sps = mCSD.itemAt(0); - CHECK(!memcmp("\x00\x00\x00\x01", sps->data(), 4)); - CHECK_GE(sps->size(), 7u); - // profile_idc, constraint_set*, level_idc - memcpy(&data[2], sps->data() + 4, 3); + if (mCSD.size() > 0) { + CHECK_GE(mCSD.size(), 1u); + const sp<ABuffer> &sps = mCSD.itemAt(0); + CHECK(!memcmp("\x00\x00\x00\x01", sps->data(), 4)); + CHECK_GE(sps->size(), 7u); + // profile_idc, constraint_set*, level_idc + memcpy(&data[2], sps->data() + 4, 3); + } else { + int32_t profileIdc, levelIdc, constraintSet; + CHECK(mFormat->findInt32("profile-idc", &profileIdc)); + CHECK(mFormat->findInt32("level-idc", &levelIdc)); + CHECK(mFormat->findInt32("constraint-set", &constraintSet)); + CHECK_GE(profileIdc, 0u); + CHECK_GE(levelIdc, 0u); + data[2] = profileIdc; // profile_idc + data[3] = constraintSet; // constraint_set* + data[4] = levelIdc; // level_idc + } // AVC_still_present=0, AVC_24_hour_picture_flag=0, reserved data[5] = 0x3f; diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp index 22dd0b1..b421b35 100644 --- a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp +++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp @@ -23,6 +23,7 @@ #include "Parameters.h" #include "ParsedMessage.h" #include "rtp/RTPSender.h" +#include "TimeSyncer.h" #include <binder/IServiceManager.h> #include <gui/IGraphicBufferProducer.h> @@ -73,6 +74,12 @@ WifiDisplaySource::WifiDisplaySource( mSupportedSourceVideoFormats.setNativeResolution( VideoFormats::RESOLUTION_CEA, 5); // 1280x720 p30 + + // Enable all resolutions up to 1280x720p30 + mSupportedSourceVideoFormats.enableResolutionUpto( + VideoFormats::RESOLUTION_CEA, 5, + VideoFormats::PROFILE_CHP, // Constrained High Profile + VideoFormats::LEVEL_32); // Level 3.2 } WifiDisplaySource::~WifiDisplaySource() { @@ -164,6 +171,14 @@ void WifiDisplaySource::onMessageReceived(const sp<AMessage> &msg) { } else { err = -EINVAL; } + } + + if (err == OK) { + sp<AMessage> notify = new AMessage(kWhatTimeSyncerNotify, id()); + mTimeSyncer = new TimeSyncer(mNetSession, notify); + looper()->registerHandler(mTimeSyncer); + + mTimeSyncer->startServer(8123); mState = AWAITING_CLIENT_CONNECTION; } @@ -539,6 +554,11 @@ void WifiDisplaySource::onMessageReceived(const sp<AMessage> &msg) { break; } + case kWhatTimeSyncerNotify: + { + break; + } + default: TRESPASS(); } @@ -617,6 +637,9 @@ status_t WifiDisplaySource::sendM4(int32_t sessionID) { chosenVideoFormat.disableAll(); chosenVideoFormat.setNativeResolution( mChosenVideoResolutionType, mChosenVideoResolutionIndex); + chosenVideoFormat.setProfileLevel( + mChosenVideoResolutionType, mChosenVideoResolutionIndex, + mChosenVideoProfile, mChosenVideoLevel); body.append(chosenVideoFormat.getFormatSpec(true /* forM4Message */)); body.append("\r\n"); @@ -729,6 +752,8 @@ status_t WifiDisplaySource::sendM16(int32_t sessionID) { ++mNextCSeq; + scheduleKeepAlive(sessionID); + return OK; } @@ -845,7 +870,9 @@ status_t WifiDisplaySource::onReceiveM3Response( mSupportedSinkVideoFormats, mSupportedSourceVideoFormats, &mChosenVideoResolutionType, - &mChosenVideoResolutionIndex)) { + &mChosenVideoResolutionIndex, + &mChosenVideoProfile, + &mChosenVideoLevel)) { ALOGE("Sink and source share no commonly supported video " "formats."); @@ -864,6 +891,9 @@ status_t WifiDisplaySource::onReceiveM3Response( ALOGI("Picked video resolution %u x %u %c%u", width, height, interlaced ? 'i' : 'p', framesPerSecond); + + ALOGI("Picked AVC profile %d, level %d", + mChosenVideoProfile, mChosenVideoLevel); } else { ALOGI("Sink doesn't support video at all."); } @@ -994,8 +1024,6 @@ status_t WifiDisplaySource::onReceiveM16Response( if (mClientInfo.mPlaybackSession != NULL) { mClientInfo.mPlaybackSession->updateLiveness(); - - scheduleKeepAlive(sessionID); } return OK; @@ -1257,7 +1285,9 @@ status_t WifiDisplaySource::onSetupRequest( mUsingPCMAudio, mSinkSupportsVideo, mChosenVideoResolutionType, - mChosenVideoResolutionIndex); + mChosenVideoResolutionIndex, + mChosenVideoProfile, + mChosenVideoLevel); if (err != OK) { looper()->unregisterHandler(playbackSession->id()); @@ -1340,7 +1370,9 @@ status_t WifiDisplaySource::onPlayRequest( return ERROR_MALFORMED; } - if (mState != AWAITING_CLIENT_PLAY) { + if (mState != AWAITING_CLIENT_PLAY + && mState != PAUSED_TO_PLAYING + && mState != PAUSED) { ALOGW("Received PLAY request but we're in state %d", mState); sendErrorResponse( @@ -1367,7 +1399,7 @@ status_t WifiDisplaySource::onPlayRequest( return err; } - if (mState == PAUSED_TO_PLAYING) { + if (mState == PAUSED_TO_PLAYING || mPlaybackSessionEstablished) { mState = PLAYING; return OK; } @@ -1401,7 +1433,7 @@ status_t WifiDisplaySource::onPauseRequest( ALOGI("Received PAUSE request."); - if (mState != PLAYING_TO_PAUSED) { + if (mState != PLAYING_TO_PAUSED && mState != PLAYING) { return INVALID_OPERATION; } diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.h b/media/libstagefright/wifi-display/source/WifiDisplaySource.h index 44d3e4d..64186fc 100644 --- a/media/libstagefright/wifi-display/source/WifiDisplaySource.h +++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.h @@ -30,6 +30,7 @@ namespace android { struct IHDCP; struct IRemoteDisplayClient; struct ParsedMessage; +struct TimeSyncer; // Represents the RTSP server acting as a wifi display source. // Manages incoming connections, sets up Playback sessions as necessary. @@ -82,6 +83,7 @@ private: kWhatHDCPNotify, kWhatFinishStop2, kWhatTeardownTriggerTimedOut, + kWhatTimeSyncerNotify, }; struct ResponseID { @@ -118,6 +120,7 @@ private: sp<ANetworkSession> mNetSession; sp<IRemoteDisplayClient> mClient; AString mMediaPath; + sp<TimeSyncer> mTimeSyncer; struct in_addr mInterfaceAddr; int32_t mSessionID; @@ -131,6 +134,8 @@ private: VideoFormats::ResolutionType mChosenVideoResolutionType; size_t mChosenVideoResolutionIndex; + VideoFormats::ProfileType mChosenVideoProfile; + VideoFormats::LevelType mChosenVideoLevel; bool mSinkSupportsAudio; diff --git a/media/libstagefright/wifi-display/udptest.cpp b/media/libstagefright/wifi-display/udptest.cpp new file mode 100644 index 0000000..111846d --- /dev/null +++ b/media/libstagefright/wifi-display/udptest.cpp @@ -0,0 +1,116 @@ +/* + * Copyright 2012, The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +//#define LOG_NEBUG 0 +#define LOG_TAG "udptest" +#include <utils/Log.h> + +#include "ANetworkSession.h" +#include "TimeSyncer.h" + +#include <binder/ProcessState.h> +#include <media/stagefright/foundation/AMessage.h> + +namespace android { + +} // namespace android + +static void usage(const char *me) { + fprintf(stderr, + "usage: %s -c host[:port]\tconnect to test server\n" + " -l \tcreate a test server\n", + me); +} + +int main(int argc, char **argv) { + using namespace android; + + ProcessState::self()->startThreadPool(); + + int32_t localPort = -1; + int32_t connectToPort = -1; + AString connectToHost; + + int res; + while ((res = getopt(argc, argv, "hc:l:")) >= 0) { + switch (res) { + case 'c': + { + const char *colonPos = strrchr(optarg, ':'); + + if (colonPos == NULL) { + connectToHost = optarg; + connectToPort = 49152; + } else { + connectToHost.setTo(optarg, colonPos - optarg); + + char *end; + connectToPort = strtol(colonPos + 1, &end, 10); + + if (*end != '\0' || end == colonPos + 1 + || connectToPort < 1 || connectToPort > 65535) { + fprintf(stderr, "Illegal port specified.\n"); + exit(1); + } + } + break; + } + + case 'l': + { + char *end; + localPort = strtol(optarg, &end, 10); + + if (*end != '\0' || end == optarg + || localPort < 1 || localPort > 65535) { + fprintf(stderr, "Illegal port specified.\n"); + exit(1); + } + break; + } + + case '?': + case 'h': + usage(argv[0]); + exit(1); + } + } + + if (localPort < 0 && connectToPort < 0) { + fprintf(stderr, + "You need to select either client or server mode.\n"); + exit(1); + } + + sp<ANetworkSession> netSession = new ANetworkSession; + netSession->start(); + + sp<ALooper> looper = new ALooper; + + sp<TimeSyncer> handler = new TimeSyncer(netSession, NULL /* notify */); + looper->registerHandler(handler); + + if (localPort >= 0) { + handler->startServer(localPort); + } else { + handler->startClient(connectToHost.c_str(), connectToPort); + } + + looper->start(true /* runOnCallingThread */); + + return 0; +} + diff --git a/media/libstagefright/wifi-display/wfd.cpp b/media/libstagefright/wifi-display/wfd.cpp index c947765..9fee4d0 100644 --- a/media/libstagefright/wifi-display/wfd.cpp +++ b/media/libstagefright/wifi-display/wfd.cpp @@ -18,6 +18,7 @@ #define LOG_TAG "wfd" #include <utils/Log.h> +#include "sink/WifiDisplaySink.h" #include "source/WifiDisplaySource.h" #include <binder/ProcessState.h> @@ -38,8 +39,12 @@ namespace android { static void usage(const char *me) { fprintf(stderr, "usage:\n" - " %s -l iface[:port]\tcreate a wifi display source\n" - " -f(ilename) \tstream media\n", + " %s -c host[:port]\tconnect to wifi source\n" + " -u uri \tconnect to an rtsp uri\n" + " -l ip[:port] \tlisten on the specified port " + " -f(ilename) \tstream media " + "(create a sink)\n" + " -s(pecial) \trun in 'special' mode\n", me); } @@ -209,14 +214,48 @@ int main(int argc, char **argv) { DataSource::RegisterDefaultSniffers(); + AString connectToHost; + int32_t connectToPort = -1; + AString uri; + AString listenOnAddr; int32_t listenOnPort = -1; AString path; + bool specialMode = false; + int res; - while ((res = getopt(argc, argv, "hl:f:")) >= 0) { + while ((res = getopt(argc, argv, "hc:l:u:f:s")) >= 0) { switch (res) { + case 'c': + { + const char *colonPos = strrchr(optarg, ':'); + + if (colonPos == NULL) { + connectToHost = optarg; + connectToPort = WifiDisplaySource::kWifiDisplayDefaultPort; + } else { + connectToHost.setTo(optarg, colonPos - optarg); + + char *end; + connectToPort = strtol(colonPos + 1, &end, 10); + + if (*end != '\0' || end == colonPos + 1 + || connectToPort < 1 || connectToPort > 65535) { + fprintf(stderr, "Illegal port specified.\n"); + exit(1); + } + } + break; + } + + case 'u': + { + uri = optarg; + break; + } + case 'f': { path = optarg; @@ -245,6 +284,12 @@ int main(int argc, char **argv) { break; } + case 's': + { + specialMode = true; + break; + } + case '?': case 'h': default: @@ -253,6 +298,13 @@ int main(int argc, char **argv) { } } + if (connectToPort >= 0 && listenOnPort >= 0) { + fprintf(stderr, + "You can connect to a source or create one, " + "but not both at the same time.\n"); + exit(1); + } + if (listenOnPort >= 0) { if (path.empty()) { createSource(listenOnAddr, listenOnPort); @@ -263,7 +315,72 @@ int main(int argc, char **argv) { exit(0); } - usage(argv[0]); + if (connectToPort < 0 && uri.empty()) { + fprintf(stderr, + "You need to select either source host or uri.\n"); + + exit(1); + } + + if (connectToPort >= 0 && !uri.empty()) { + fprintf(stderr, + "You need to either connect to a wfd host or an rtsp url, " + "not both.\n"); + exit(1); + } + + sp<SurfaceComposerClient> composerClient = new SurfaceComposerClient; + CHECK_EQ(composerClient->initCheck(), (status_t)OK); + + sp<IBinder> display(SurfaceComposerClient::getBuiltInDisplay( + ISurfaceComposer::eDisplayIdMain)); + DisplayInfo info; + SurfaceComposerClient::getDisplayInfo(display, &info); + ssize_t displayWidth = info.w; + ssize_t displayHeight = info.h; + + ALOGV("display is %d x %d\n", displayWidth, displayHeight); + + sp<SurfaceControl> control = + composerClient->createSurface( + String8("A Surface"), + displayWidth, + displayHeight, + PIXEL_FORMAT_RGB_565, + 0); + + CHECK(control != NULL); + CHECK(control->isValid()); + + SurfaceComposerClient::openGlobalTransaction(); + CHECK_EQ(control->setLayer(INT_MAX), (status_t)OK); + CHECK_EQ(control->show(), (status_t)OK); + SurfaceComposerClient::closeGlobalTransaction(); + + sp<Surface> surface = control->getSurface(); + CHECK(surface != NULL); + + sp<ANetworkSession> session = new ANetworkSession; + session->start(); + + sp<ALooper> looper = new ALooper; + + sp<WifiDisplaySink> sink = new WifiDisplaySink( + specialMode ? WifiDisplaySink::FLAG_SPECIAL_MODE : 0 /* flags */, + session, + surface->getIGraphicBufferProducer()); + + looper->registerHandler(sink); + + if (connectToPort >= 0) { + sink->start(connectToHost.c_str(), connectToPort); + } else { + sink->start(uri.c_str()); + } + + looper->start(true /* runOnCallingThread */); + + composerClient->dispose(); return 0; } |