/* ** ** Copyright 2008, 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 "AudioRecord" #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) #define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) namespace android { // --------------------------------------------------------------------------- // static status_t AudioRecord::getMinFrameCount( int* frameCount, uint32_t sampleRate, int format, int channelCount) { size_t size = 0; if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &size) != NO_ERROR) { LOGE("AudioSystem could not query the input buffer size."); return NO_INIT; } if (size == 0) { LOGE("Unsupported configuration: sampleRate %d, format %d, channelCount %d", sampleRate, format, channelCount); return BAD_VALUE; } // We double the size of input buffer for ping pong use of record buffer. size <<= 1; if (audio_is_linear_pcm(format)) { size /= channelCount * audio_bytes_per_sample(format); } *frameCount = size; return NO_ERROR; } // --------------------------------------------------------------------------- AudioRecord::AudioRecord() : mStatus(NO_INIT), mSessionId(0) { } AudioRecord::AudioRecord( int inputSource, uint32_t sampleRate, int format, uint32_t channelMask, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId) : mStatus(NO_INIT), mSessionId(0) { mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, flags, cbf, user, notificationFrames, sessionId); } AudioRecord::~AudioRecord() { if (mStatus == NO_ERROR) { // Make sure that callback function exits in the case where // it is looping on buffer empty condition in obtainBuffer(). // Otherwise the callback thread will never exit. stop(); if (mClientRecordThread != 0) { mClientRecordThread->requestExitAndWait(); mClientRecordThread.clear(); } mAudioRecord.clear(); IPCThreadState::self()->flushCommands(); AudioSystem::releaseAudioSessionId(mSessionId); } } status_t AudioRecord::set( int inputSource, uint32_t sampleRate, int format, uint32_t channelMask, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, bool threadCanCallJava, int sessionId) { ALOGV("set(): sampleRate %d, channelMask %d, frameCount %d",sampleRate, channelMask, frameCount); AutoMutex lock(mLock); if (mAudioRecord != 0) { return INVALID_OPERATION; } if (inputSource == AUDIO_SOURCE_DEFAULT) { inputSource = AUDIO_SOURCE_MIC; } if (sampleRate == 0) { sampleRate = DEFAULT_SAMPLE_RATE; } // these below should probably come from the audioFlinger too... if (format == 0) { format = AUDIO_FORMAT_PCM_16_BIT; } // validate parameters if (!audio_is_valid_format(format)) { LOGE("Invalid format"); return BAD_VALUE; } if (!audio_is_input_channel(channelMask)) { return BAD_VALUE; } int channelCount = popcount(channelMask); if (sessionId == 0 ) { mSessionId = AudioSystem::newAudioSessionId(); } else { mSessionId = sessionId; } ALOGV("set(): mSessionId %d", mSessionId); audio_io_handle_t input = AudioSystem::getInput(inputSource, sampleRate, format, channelMask, (audio_in_acoustics_t)flags, mSessionId); if (input == 0) { LOGE("Could not get audio input for record source %d", inputSource); return BAD_VALUE; } // validate framecount int minFrameCount = 0; status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelCount); if (status != NO_ERROR) { return status; } ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount); if (frameCount == 0) { frameCount = minFrameCount; } else if (frameCount < minFrameCount) { return BAD_VALUE; } if (notificationFrames == 0) { notificationFrames = frameCount/2; } // create the IAudioRecord status = openRecord_l(sampleRate, format, channelMask, frameCount, flags, input); if (status != NO_ERROR) { return status; } if (cbf != 0) { mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava); if (mClientRecordThread == 0) { return NO_INIT; } } mStatus = NO_ERROR; mFormat = format; // Update buffer size in case it has been limited by AudioFlinger during track creation mFrameCount = mCblk->frameCount; mChannelCount = (uint8_t)channelCount; mChannelMask = channelMask; mActive = 0; mCbf = cbf; mNotificationFrames = notificationFrames; mRemainingFrames = notificationFrames; mUserData = user; // TODO: add audio hardware input latency here mLatency = (1000*mFrameCount) / sampleRate; mMarkerPosition = 0; mMarkerReached = false; mNewPosition = 0; mUpdatePeriod = 0; mInputSource = (uint8_t)inputSource; mFlags = flags; mInput = input; AudioSystem::acquireAudioSessionId(mSessionId); return NO_ERROR; } status_t AudioRecord::initCheck() const { return mStatus; } // ------------------------------------------------------------------------- uint32_t AudioRecord::latency() const { return mLatency; } int AudioRecord::format() const { return mFormat; } int AudioRecord::channelCount() const { return mChannelCount; } uint32_t AudioRecord::frameCount() const { return mFrameCount; } int AudioRecord::frameSize() const { if (audio_is_linear_pcm(mFormat)) { return channelCount()*audio_bytes_per_sample(mFormat); } else { return sizeof(uint8_t); } } int AudioRecord::inputSource() const { return (int)mInputSource; } // ------------------------------------------------------------------------- status_t AudioRecord::start() { status_t ret = NO_ERROR; sp t = mClientRecordThread; ALOGV("start"); if (t != 0) { if (t->exitPending()) { if (t->requestExitAndWait() == WOULD_BLOCK) { LOGE("AudioRecord::start called from thread"); return WOULD_BLOCK; } } t->mLock.lock(); } 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 audioRecord = mAudioRecord; sp iMem = mCblkMemory; audio_track_cblk_t* cblk = mCblk; if (mActive == 0) { mActive = 1; cblk->lock.lock(); if (!(cblk->flags & CBLK_INVALID_MSK)) { cblk->lock.unlock(); ret = mAudioRecord->start(); cblk->lock.lock(); if (ret == DEAD_OBJECT) { android_atomic_or(CBLK_INVALID_ON, &cblk->flags); } } if (cblk->flags & CBLK_INVALID_MSK) { ret = restoreRecord_l(cblk); } cblk->lock.unlock(); if (ret == NO_ERROR) { mNewPosition = cblk->user + mUpdatePeriod; cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; cblk->waitTimeMs = 0; if (t != 0) { t->run("ClientRecordThread", ANDROID_PRIORITY_AUDIO); } else { setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO); } } else { mActive = 0; } } if (t != 0) { t->mLock.unlock(); } return ret; } status_t AudioRecord::stop() { sp t = mClientRecordThread; ALOGV("stop"); if (t != 0) { t->mLock.lock(); } AutoMutex lock(mLock); if (mActive == 1) { mActive = 0; 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->requestExit(); } else { setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL); } } if (t != 0) { t->mLock.unlock(); } return NO_ERROR; } bool AudioRecord::stopped() const { return !mActive; } uint32_t AudioRecord::getSampleRate() { AutoMutex lock(mLock); return mCblk->sampleRate; } status_t AudioRecord::setMarkerPosition(uint32_t marker) { if (mCbf == 0) return INVALID_OPERATION; mMarkerPosition = marker; mMarkerReached = false; return NO_ERROR; } status_t AudioRecord::getMarkerPosition(uint32_t *marker) { if (marker == 0) return BAD_VALUE; *marker = mMarkerPosition; return NO_ERROR; } status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) { if (mCbf == 0) return INVALID_OPERATION; uint32_t curPosition; getPosition(&curPosition); mNewPosition = curPosition + updatePeriod; mUpdatePeriod = updatePeriod; return NO_ERROR; } status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) { if (updatePeriod == 0) return BAD_VALUE; *updatePeriod = mUpdatePeriod; return NO_ERROR; } status_t AudioRecord::getPosition(uint32_t *position) { if (position == 0) return BAD_VALUE; AutoMutex lock(mLock); *position = mCblk->user; return NO_ERROR; } unsigned int AudioRecord::getInputFramesLost() { if (mActive) return AudioSystem::getInputFramesLost(mInput); else return 0; } // ------------------------------------------------------------------------- // must be called with mLock held status_t AudioRecord::openRecord_l( uint32_t sampleRate, uint32_t format, uint32_t channelMask, int frameCount, uint32_t flags, audio_io_handle_t input) { status_t status; const sp& audioFlinger = AudioSystem::get_audio_flinger(); if (audioFlinger == 0) { return NO_INIT; } sp record = audioFlinger->openRecord(getpid(), input, sampleRate, format, channelMask, frameCount, ((uint16_t)flags) << 16, &mSessionId, &status); if (record == 0) { LOGE("AudioFlinger could not create record track, status: %d", status); return status; } sp cblk = record->getCblk(); if (cblk == 0) { LOGE("Could not get control block"); return NO_INIT; } mAudioRecord.clear(); mAudioRecord = record; mCblkMemory.clear(); mCblkMemory = cblk; mCblk = static_cast(cblk->pointer()); mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); android_atomic_and(~CBLK_DIRECTION_MSK, &mCblk->flags); mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; mCblk->waitTimeMs = 0; return NO_ERROR; } status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) { AutoMutex lock(mLock); int 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; uint32_t framesReady = cblk->framesReady(); if (framesReady == 0) { cblk->lock.lock(); goto start_loop_here; while (framesReady == 0) { active = mActive; if (UNLIKELY(!active)) { cblk->lock.unlock(); return NO_MORE_BUFFERS; } if (UNLIKELY(!waitCount)) { cblk->lock.unlock(); return WOULD_BLOCK; } if (!(cblk->flags & CBLK_INVALID_MSK)) { mLock.unlock(); result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); cblk->lock.unlock(); mLock.lock(); if (mActive == 0) { return status_t(STOPPED); } cblk->lock.lock(); } if (cblk->flags & CBLK_INVALID_MSK) { goto create_new_record; } if (__builtin_expect(result!=NO_ERROR, false)) { cblk->waitTimeMs += waitTimeMs; if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { LOGW( "obtainBuffer timed out (is the CPU pegged?) " "user=%08x, server=%08x", cblk->user, cblk->server); cblk->lock.unlock(); result = mAudioRecord->start(); cblk->lock.lock(); if (result == DEAD_OBJECT) { android_atomic_or(CBLK_INVALID_ON, &cblk->flags); create_new_record: result = AudioRecord::restoreRecord_l(cblk); } if (result != NO_ERROR) { LOGW("obtainBuffer create Track error %d", result); cblk->lock.unlock(); return result; } cblk->waitTimeMs = 0; } if (--waitCount == 0) { cblk->lock.unlock(); return TIMED_OUT; } } // read the server count again start_loop_here: framesReady = cblk->framesReady(); } cblk->lock.unlock(); } cblk->waitTimeMs = 0; if (framesReq > framesReady) { framesReq = framesReady; } uint32_t u = cblk->user; uint32_t bufferEnd = cblk->userBase + cblk->frameCount; if (u + framesReq > bufferEnd) { framesReq = bufferEnd - u; } audioBuffer->flags = 0; audioBuffer->channelCount= mChannelCount; audioBuffer->format = mFormat; audioBuffer->frameCount = framesReq; audioBuffer->size = framesReq*cblk->frameSize; audioBuffer->raw = (int8_t*)cblk->buffer(u); active = mActive; return active ? status_t(NO_ERROR) : status_t(STOPPED); } void AudioRecord::releaseBuffer(Buffer* audioBuffer) { AutoMutex lock(mLock); mCblk->stepUser(audioBuffer->frameCount); } audio_io_handle_t AudioRecord::getInput() { AutoMutex lock(mLock); return mInput; } // must be called with mLock held audio_io_handle_t AudioRecord::getInput_l() { mInput = AudioSystem::getInput(mInputSource, mCblk->sampleRate, mFormat, mChannelMask, (audio_in_acoustics_t)mFlags, mSessionId); return mInput; } int AudioRecord::getSessionId() { return mSessionId; } // ------------------------------------------------------------------------- ssize_t AudioRecord::read(void* buffer, size_t userSize) { ssize_t read = 0; Buffer audioBuffer; int8_t *dst = static_cast(buffer); if (ssize_t(userSize) < 0) { // sanity-check. user is most-likely passing an error code. LOGE("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 audioRecord = mAudioRecord; sp iMem = mCblkMemory; mLock.unlock(); do { audioBuffer.frameCount = userSize/frameSize(); // 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)); if (err < 0) { // out of buffers, return #bytes written if (err == status_t(NO_MORE_BUFFERS)) break; if (err == status_t(TIMED_OUT)) err = 0; return ssize_t(err); } size_t bytesRead = audioBuffer.size; memcpy(dst, audioBuffer.i8, bytesRead); dst += bytesRead; userSize -= bytesRead; read += bytesRead; releaseBuffer(&audioBuffer); } while (userSize); return read; } // ------------------------------------------------------------------------- bool AudioRecord::processAudioBuffer(const sp& 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 audioRecord = mAudioRecord; sp iMem = mCblkMemory; audio_track_cblk_t* cblk = mCblk; mLock.unlock(); // Manage marker callback if (!mMarkerReached && (mMarkerPosition > 0)) { if (cblk->user >= mMarkerPosition) { mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); mMarkerReached = true; } } // Manage new position callback if (mUpdatePeriod > 0) { while (cblk->user >= mNewPosition) { mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); mNewPosition += mUpdatePeriod; } } 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) { LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); return false; } break; } if (err == status_t(STOPPED)) return false; size_t reqSize = audioBuffer.size; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); readSize = audioBuffer.size; // Sanity check on returned size if (ssize_t(readSize) <= 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 (readSize > reqSize) readSize = reqSize; audioBuffer.size = readSize; audioBuffer.frameCount = readSize/frameSize(); frames -= audioBuffer.frameCount; releaseBuffer(&audioBuffer); } while (frames); // Manage overrun callback if (mActive && (cblk->framesAvailable() == 0)) { ALOGV("Overrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { mCbf(EVENT_OVERRUN, mUserData, 0); } } if (frames == 0) { mRemainingFrames = mNotificationFrames; } else { mRemainingFrames = frames; } return true; } // 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*& cblk) { status_t result; if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { LOGW("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(cblk->sampleRate, mFormat, mChannelMask, mFrameCount, mFlags, getInput_l()); if (result == NO_ERROR) { result = mAudioRecord->start(); } if (result != NO_ERROR) { mActive = false; } // signal old cblk condition for other threads waiting for restore completion android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); cblk->cv.broadcast(); } else { if (!(cblk->flags & CBLK_RESTORED_MSK)) { LOGW("dead IAudioRecord, waiting for a new one to be created"); mLock.unlock(); result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); cblk->lock.unlock(); mLock.lock(); } else { LOGW("dead IAudioRecord, already restored"); result = NO_ERROR; cblk->lock.unlock(); } if (result != NO_ERROR || mActive == 0) { result = status_t(STOPPED); } } ALOGV("restoreRecord_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); if (result == NO_ERROR) { // from now on we switch to the newly created cblk cblk = mCblk; } cblk->lock.lock(); LOGW_IF(result != NO_ERROR, "restoreRecord_l() error %d", result); return result; } // ========================================================================= AudioRecord::ClientRecordThread::ClientRecordThread(AudioRecord& receiver, bool bCanCallJava) : Thread(bCanCallJava), mReceiver(receiver) { } bool AudioRecord::ClientRecordThread::threadLoop() { return mReceiver.processAudioBuffer(this); } // ------------------------------------------------------------------------- }; // namespace android