diff options
23 files changed, 239 insertions, 67 deletions
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index f4ced52..45134c4 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -39,8 +39,12 @@ public: * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*. */ enum event_type { - EVENT_MORE_DATA = 0, // Request to read more data from PCM buffer. - EVENT_OVERRUN = 1, // PCM buffer overrun occurred. + EVENT_MORE_DATA = 0, // Request to read available data from buffer. + // If this event is delivered but the callback handler + // does not want to read the available data, the handler must + // explicitly + // ignore the event by setting frameCount to zero. + EVENT_OVERRUN = 1, // Buffer overrun occurred. EVENT_MARKER = 2, // Record head is at the specified marker position // (See setMarkerPosition()). EVENT_NEW_POS = 3, // Record head is at a new position @@ -63,6 +67,7 @@ public: // (currently ignored but will make the primary field in future) size_t size; // input/output in bytes == frameCount * frameSize + // on output is the number of bytes actually drained // FIXME this is redundant with respect to frameCount, // and TRANSFER_OBTAIN mode is broken for 8-bit data // since we don't define the frame format @@ -76,7 +81,7 @@ public: /* As a convenience, if a callback is supplied, a handler thread * is automatically created with the appropriate priority. This thread - * invokes the callback when a new buffer becomes ready or various conditions occur. + * invokes the callback when a new buffer becomes available or various conditions occur. * Parameters: * * event: type of event notified (see enum AudioRecord::event_type). @@ -99,6 +104,8 @@ public: * - NO_ERROR: successful operation * - NO_INIT: audio server or audio hardware not initialized * - BAD_VALUE: unsupported configuration + * frameCount is guaranteed to be non-zero if status is NO_ERROR, + * and is undefined otherwise. */ static status_t getMinFrameCount(size_t* frameCount, @@ -109,7 +116,7 @@ public: /* How data is transferred from AudioRecord */ enum transfer_type { - TRANSFER_DEFAULT, // not specified explicitly; determine from other parameters + TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters TRANSFER_CALLBACK, // callback EVENT_MORE_DATA TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer() TRANSFER_SYNC, // synchronous read() @@ -137,7 +144,7 @@ public: * be larger if the requested size is not compatible with current audio HAL * latency. Zero means to use a default value. * cbf: Callback function. If not null, this function is called periodically - * to consume new PCM data and inform of marker, position updates, etc. + * to consume new data and inform of marker, position updates, etc. * user: Context for use by the callback receiver. * notificationFrames: The callback function is called each time notificationFrames PCM * frames are ready in record track output buffer. @@ -171,9 +178,10 @@ public: * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful intialization * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use - * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) + * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) * - NO_INIT: audio server or audio hardware not initialized * - PERMISSION_DENIED: recording is not allowed for the requesting process + * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. * * Parameters not listed in the AudioRecord constructors above: * @@ -192,7 +200,7 @@ public: transfer_type transferType = TRANSFER_DEFAULT, audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE); - /* Result of constructing the AudioRecord. This must be checked + /* Result of constructing the AudioRecord. This must be checked for successful initialization * before using any AudioRecord API (except for set()), because using * an uninitialized AudioRecord produces undefined results. * See set() method above for possible return codes. @@ -221,7 +229,7 @@ public: status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, int triggerSession = 0); - /* Stop a track. If set, the callback will cease being called. Note that obtainBuffer() still + /* Stop a track. The callback will cease being called. Note that obtainBuffer() still * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. */ void stop(); @@ -236,7 +244,7 @@ public: * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition * with marker == 0 cancels marker notification callback. * To set a marker at a position which would compute as 0, - * a workaround is to the set the marker at a nearby position such as ~0 or 1. + * a workaround is to set the marker at a nearby position such as ~0 or 1. * If the AudioRecord has been opened with no callback function associated, * the operation will fail. * @@ -378,8 +386,10 @@ public: * returning the current value by this function call. Such loss typically occurs when the * user space process is blocked longer than the capacity of audio driver buffers. * Units: the number of input audio frames. + * FIXME The side-effect of resetting the counter may be incompatible with multi-client. + * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. */ - unsigned int getInputFramesLost() const; + uint32_t getInputFramesLost() const; private: /* copying audio record objects is not allowed */ @@ -426,6 +436,7 @@ private: nsecs_t processAudioBuffer(); // caller must hold lock on mLock for all _l methods + status_t openRecord_l(size_t epoch); // FIXME enum is faster than strcmp() for parameter 'from' @@ -477,7 +488,7 @@ private: audio_io_handle_t mInput; // returned by AudioSystem::getInput() - // may be changed if IAudioRecord object is re-created + // Next 3 fields may be changed if IAudioRecord is re-created, but always != 0 sp<IAudioRecord> mAudioRecord; sp<IMemory> mCblkMemory; audio_track_cblk_t* mCblk; // re-load after mLock.unlock() diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index ca9aaf7..5abab8a 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -138,7 +138,7 @@ public: audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid - static size_t getInputFramesLost(audio_io_handle_t ioHandle); + static uint32_t getInputFramesLost(audio_io_handle_t ioHandle); static int newAudioSessionId(); static void acquireAudioSessionId(int audioSession); @@ -200,12 +200,16 @@ public: static status_t setPhoneState(audio_mode_t state); static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); + + // Client must successfully hand off the handle reference to AudioFlinger via createTrack(), + // or release it with releaseOutput(). static audio_io_handle_t getOutput(audio_stream_type_t stream, uint32_t samplingRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, const audio_offload_info_t *offloadInfo = NULL); + static status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, int session); @@ -213,11 +217,15 @@ public: audio_stream_type_t stream, int session); static void releaseOutput(audio_io_handle_t output); + + // Client must successfully hand off the handle reference to AudioFlinger via openRecord(), + // or release it with releaseInput(). static audio_io_handle_t getInput(audio_source_t inputSource, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, int sessionId); + static status_t startInput(audio_io_handle_t input); static status_t stopInput(audio_io_handle_t input); static void releaseInput(audio_io_handle_t input); diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index c86a67b..644e55c 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -160,7 +160,7 @@ public: * sampleRate: Data source sampling rate in Hz. * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed * 16 bits per sample). - * channelMask: Channel mask. + * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true. * frameCount: Minimum size of track PCM buffer in frames. This defines the * application's contribution to the * latency of the track. The actual size selected by the AudioTrack could be @@ -338,7 +338,7 @@ public: */ status_t setSampleRate(uint32_t sampleRate); - /* Return current source sample rate in Hz, or 0 if unknown */ + /* Return current source sample rate in Hz */ uint32_t getSampleRate() const; /* Enables looping and sets the start and end points of looping. @@ -363,7 +363,7 @@ public: /* Sets marker position. When playback reaches the number of frames specified, a callback with * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker * notification callback. To set a marker at a position which would compute as 0, - * a workaround is to the set the marker at a nearby position such as ~0 or 1. + * a workaround is to set the marker at a nearby position such as ~0 or 1. * If the AudioTrack has been opened with no callback function associated, the operation will * fail. * @@ -661,6 +661,7 @@ protected: audio_track_cblk_t* mCblk; // re-load after mLock.unlock() sp<AudioTrackThread> mAudioTrackThread; + float mVolume[2]; float mSendLevel; mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it. @@ -674,7 +675,10 @@ protected: audio_stream_type_t mStreamType; uint32_t mChannelCount; audio_channel_mask_t mChannelMask; + sp<IMemory> mSharedBuffer; transfer_type mTransfer; + audio_offload_info_t mOffloadInfoCopy; + const audio_offload_info_t* mOffloadInfo; // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer. @@ -715,8 +719,8 @@ protected: bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() uint32_t mObservedSequence; // last observed value of mSequence - sp<IMemory> mSharedBuffer; uint32_t mLoopPeriod; // in frames, zero means looping is disabled + uint32_t mMarkerPosition; // in wrapping (overflow) frame units bool mMarkerReached; uint32_t mNewPosition; // in frames diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h index 899d79f..a794e87 100644 --- a/include/media/IAudioFlinger.h +++ b/include/media/IAudioFlinger.h @@ -67,6 +67,9 @@ public: size_t frameCount, track_flags_t *flags, const sp<IMemory>& sharedBuffer, + // On successful return, AudioFlinger takes over the handle + // reference and will release it when the track is destroyed. + // However on failure, the client is responsible for release. audio_io_handle_t output, pid_t tid, // -1 means unused, otherwise must be valid non-0 int *sessionId, @@ -78,6 +81,9 @@ public: status_t *status) = 0; virtual sp<IAudioRecord> openRecord( + // On successful return, AudioFlinger takes over the handle + // reference and will release it when the track is destroyed. + // However on failure, the client is responsible for release. audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, @@ -170,7 +176,7 @@ public: virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, audio_io_handle_t output) const = 0; - virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0; + virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0; virtual int newAudioSessionId() = 0; @@ -188,6 +194,7 @@ public: effect_descriptor_t *pDesc, const sp<IEffectClient>& client, int32_t priority, + // AudioFlinger doesn't take over handle reference from client audio_io_handle_t output, int sessionId, status_t *status, diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h index 3a87474..db8216b 100644 --- a/include/media/stagefright/MetaData.h +++ b/include/media/stagefright/MetaData.h @@ -215,6 +215,8 @@ public: bool findData(uint32_t key, uint32_t *type, const void **data, size_t *size) const; + bool hasData(uint32_t key) const; + void dumpToLog() const; protected: diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 076404b..6565a16 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -289,6 +289,9 @@ status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession) // reset current position as seen by client to 0 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); + // force refresh of remaining frames by processAudioBuffer() as last + // read before stop could be partial. + mRefreshRemaining = true; mNewPosition = mProxy->getPosition() + mUpdatePeriod; int32_t flags = android_atomic_acquire_load(&mCblk->mFlags); @@ -352,6 +355,7 @@ bool AudioRecord::stopped() const status_t AudioRecord::setMarkerPosition(uint32_t marker) { + // The only purpose of setting marker position is to get a callback if (mCbf == NULL) { return INVALID_OPERATION; } @@ -377,6 +381,7 @@ status_t AudioRecord::getMarkerPosition(uint32_t *marker) const status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod) { + // The only purpose of setting position update period is to get a callback if (mCbf == NULL) { return INVALID_OPERATION; } @@ -412,7 +417,7 @@ status_t AudioRecord::getPosition(uint32_t *position) const return NO_ERROR; } -unsigned int AudioRecord::getInputFramesLost() const +uint32_t AudioRecord::getInputFramesLost() const { // no need to check mActive, because if inactive this will return 0, which is what we want return AudioSystem::getInputFramesLost(getInput()); @@ -591,6 +596,9 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *r if (newSequence == oldSequence) { status = restoreRecord_l("obtainBuffer"); if (status != NO_ERROR) { + buffer.mFrameCount = 0; + buffer.mRaw = NULL; + buffer.mNonContig = 0; break; } } @@ -770,7 +778,7 @@ nsecs_t AudioRecord::processAudioBuffer() uint32_t sequence = mSequence; // These fields don't need to be cached, because they are assigned only by set(): - // mTransfer, mCbf, mUserData, mSampleRate + // mTransfer, mCbf, mUserData, mSampleRate, mFrameSize mLock.unlock(); @@ -844,8 +852,8 @@ nsecs_t AudioRecord::processAudioBuffer() "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); + ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", + mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); if (err != NO_ERROR) { if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { break; diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp index 69d9273..dcb72f8 100644 --- a/media/libmedia/AudioSystem.cpp +++ b/media/libmedia/AudioSystem.cpp @@ -413,9 +413,9 @@ status_t AudioSystem::getRenderPosition(audio_io_handle_t output, size_t *halFra return af->getRenderPosition(halFrames, dspFrames, output); } -size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) { +uint32_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); - unsigned int result = 0; + uint32_t result = 0; if (af == 0) return result; if (ioHandle == 0) return result; diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index d5e3e67..72be5ca 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -54,16 +54,22 @@ status_t AudioTrack::getMinFrameCount( status_t status; status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); if (status != NO_ERROR) { + ALOGE("Unable to query output sample rate for stream type %d; status %d", + streamType, status); return status; } size_t afFrameCount; status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); if (status != NO_ERROR) { + ALOGE("Unable to query output frame count for stream type %d; status %d", + streamType, status); return status; } uint32_t afLatency; status = AudioSystem::getOutputLatency(&afLatency, streamType); if (status != NO_ERROR) { + ALOGE("Unable to query output latency for stream type %d; status %d", + streamType, status); return status; } @@ -216,6 +222,7 @@ status_t AudioTrack::set( ALOGE("Invalid transfer type %d", transferType); return BAD_VALUE; } + mSharedBuffer = sharedBuffer; mTransfer = transferType; // FIXME "int" here is legacy and will be replaced by size_t later @@ -244,6 +251,11 @@ status_t AudioTrack::set( if (streamType == AUDIO_STREAM_DEFAULT) { streamType = AUDIO_STREAM_MUSIC; } + if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { + ALOGE("Invalid stream type %d", streamType); + return BAD_VALUE; + } + mStreamType = streamType; status_t status; if (sampleRate == 0) { @@ -266,6 +278,7 @@ status_t AudioTrack::set( ALOGE("Invalid format %d", format); return BAD_VALUE; } + mFormat = format; if (!audio_is_output_channel(channelMask)) { ALOGE("Invalid channel mask %#x", channelMask); @@ -317,6 +330,16 @@ status_t AudioTrack::set( return BAD_VALUE; } + // Make copy of input parameter offloadInfo so that in the future: + // (a) createTrack_l doesn't need it as an input parameter + // (b) we can support re-creation of offloaded tracks + if (offloadInfo != NULL) { + mOffloadInfoCopy = *offloadInfo; + mOffloadInfo = &mOffloadInfoCopy; + } else { + mOffloadInfo = NULL; + } + mVolume[LEFT] = 1.0f; mVolume[RIGHT] = 1.0f; mSendLevel = 0.0f; @@ -355,7 +378,7 @@ status_t AudioTrack::set( mAudioTrackThread->requestExitAndWait(); mAudioTrackThread.clear(); } - //Use of direct and offloaded output streams is ref counted by audio policy manager. + // Use of direct and offloaded output streams is ref counted by audio policy manager. // As getOutput was called above and resulted in an output stream to be opened, // we need to release it. AudioSystem::releaseOutput(output); @@ -363,9 +386,6 @@ status_t AudioTrack::set( } mStatus = NO_ERROR; - mStreamType = streamType; - mFormat = format; - mSharedBuffer = sharedBuffer; mState = STATE_STOPPED; mUserData = user; mLoopPeriod = 0; @@ -698,6 +718,7 @@ status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) AutoMutex lock(mLock); mNewPosition = mProxy->getPosition() + updatePeriod; mUpdatePeriod = updatePeriod; + return NO_ERROR; } @@ -1004,6 +1025,11 @@ status_t AudioTrack::createTrack_l( ALOGE("Could not get control block"); return NO_INIT; } + void *iMemPointer = iMem->pointer(); + if (iMemPointer == NULL) { + ALOGE("Could not get control block pointer"); + return NO_INIT; + } // invariant that mAudioTrack != 0 is true only after set() returns successfully if (mAudioTrack != 0) { mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); @@ -1011,7 +1037,7 @@ status_t AudioTrack::createTrack_l( } mAudioTrack = track; mCblkMemory = iMem; - audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); + audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); mCblk = cblk; size_t temp = cblk->frameCount_; if (temp < frameCount || (frameCount == 0 && temp == 0)) { @@ -1610,7 +1636,6 @@ nsecs_t AudioTrack::processAudioBuffer() size_t reqSize = audioBuffer.size; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); size_t writtenSize = audioBuffer.size; - size_t writtenFrames = writtenSize / mFrameSize; // Sanity check on returned size if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { @@ -1735,7 +1760,7 @@ status_t AudioTrack::restoreTrack_l(const char *from) } } if (result != NO_ERROR) { - //Use of direct and offloaded output streams is ref counted by audio policy manager. + // Use of direct and offloaded output streams is ref counted by audio policy manager. // As getOutput was called above and resulted in an output stream to be opened, // we need to release it. AudioSystem::releaseOutput(output); diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index c9c8d58..86a4d74 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -138,6 +138,17 @@ public: name = reply.readString8(); lStatus = reply.readInt32(); track = interface_cast<IAudioTrack>(reply.readStrongBinder()); + if (lStatus == NO_ERROR) { + if (track == 0) { + ALOGE("createTrack should have returned an IAudioTrack"); + lStatus = UNKNOWN_ERROR; + } + } else { + if (track != 0) { + ALOGE("createTrack returned an IAudioTrack but with status %d", lStatus); + track.clear(); + } + } } if (status != NULL) { *status = lStatus; @@ -564,13 +575,16 @@ public: return status; } - virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const + virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); data.writeInt32((int32_t) ioHandle); - remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply); - return reply.readInt32(); + status_t status = remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply); + if (status != NO_ERROR) { + return 0; + } + return (uint32_t) reply.readInt32(); } virtual int newAudioSessionId() @@ -795,6 +809,7 @@ status_t BnAudioFlinger::onTransact( (audio_stream_type_t) streamType, sampleRate, format, channelMask, frameCount, &flags, buffer, output, tid, &sessionId, name, clientUid, &status); + LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR)); } reply->writeInt32(flags); reply->writeInt32(sessionId); @@ -1044,7 +1059,7 @@ status_t BnAudioFlinger::onTransact( case GET_INPUT_FRAMES_LOST: { CHECK_INTERFACE(IAudioFlinger, data, reply); audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32(); - reply->writeInt32(getInputFramesLost(ioHandle)); + reply->writeInt32((int32_t) getInputFramesLost(ioHandle)); return NO_ERROR; } break; case NEW_AUDIO_SESSION_ID: { diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp index 92b9a92..bf5271e 100644 --- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp +++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp @@ -422,7 +422,9 @@ void NuPlayer::Renderer::onDrainVideoQueue() { ALOGV("video late by %lld us (%.2f secs)", mVideoLateByUs, mVideoLateByUs / 1E6); } else { - ALOGV("rendering video at media time %.2f secs", mediaTimeUs / 1E6); + ALOGV("rendering video at media time %.2f secs", + (mFlags & FLAG_REAL_TIME ? realTimeUs : + (realTimeUs + mAnchorTimeMediaUs - mAnchorTimeRealUs)) / 1E6); if (mSoftRenderer != NULL) { mSoftRenderer->render(entry->mBuffer->data(), entry->mBuffer->size(), NULL); } diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp index d7223d9..cadadc8 100644 --- a/media/libstagefright/AudioSource.cpp +++ b/media/libstagefright/AudioSource.cpp @@ -278,7 +278,7 @@ status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) { // Drop retrieved and previously lost audio data. if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) { - mRecord->getInputFramesLost(); + (void) mRecord->getInputFramesLost(); ALOGV("Drop audio data at %lld/%lld us", timeUs, mStartTimeUs); return OK; } diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp index 9c89e82..dfb5c04 100644 --- a/media/libstagefright/MPEG4Extractor.cpp +++ b/media/libstagefright/MPEG4Extractor.cpp @@ -39,6 +39,7 @@ #include <utils/String8.h> #include <byteswap.h> +#include "include/ID3.h" namespace android { @@ -1787,6 +1788,18 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) { break; } + case FOURCC('I', 'D', '3', '2'): + { + if (chunk_data_size < 6) { + return ERROR_MALFORMED; + } + + parseID3v2MetaData(data_offset + 6); + + *offset += chunk_size; + break; + } + case FOURCC('-', '-', '-', '-'): { mLastCommentMean.clear(); @@ -2167,7 +2180,7 @@ status_t MPEG4Extractor::parseITunesMetaData(off64_t offset, size_t size) { break; } - if (size >= 8 && metadataKey) { + if (size >= 8 && metadataKey && !mFileMetaData->hasData(metadataKey)) { if (metadataKey == kKeyAlbumArt) { mFileMetaData->setData( kKeyAlbumArt, MetaData::TYPE_NONE, @@ -2316,6 +2329,62 @@ status_t MPEG4Extractor::parse3GPPMetaData(off64_t offset, size_t size, int dept return OK; } +void MPEG4Extractor::parseID3v2MetaData(off64_t offset) { + ID3 id3(mDataSource, true /* ignorev1 */, offset); + + if (id3.isValid()) { + struct Map { + int key; + const char *tag1; + const char *tag2; + }; + static const Map kMap[] = { + { kKeyAlbum, "TALB", "TAL" }, + { kKeyArtist, "TPE1", "TP1" }, + { kKeyAlbumArtist, "TPE2", "TP2" }, + { kKeyComposer, "TCOM", "TCM" }, + { kKeyGenre, "TCON", "TCO" }, + { kKeyTitle, "TIT2", "TT2" }, + { kKeyYear, "TYE", "TYER" }, + { kKeyAuthor, "TXT", "TEXT" }, + { kKeyCDTrackNumber, "TRK", "TRCK" }, + { kKeyDiscNumber, "TPA", "TPOS" }, + { kKeyCompilation, "TCP", "TCMP" }, + }; + static const size_t kNumMapEntries = sizeof(kMap) / sizeof(kMap[0]); + + for (size_t i = 0; i < kNumMapEntries; ++i) { + if (!mFileMetaData->hasData(kMap[i].key)) { + ID3::Iterator *it = new ID3::Iterator(id3, kMap[i].tag1); + if (it->done()) { + delete it; + it = new ID3::Iterator(id3, kMap[i].tag2); + } + + if (it->done()) { + delete it; + continue; + } + + String8 s; + it->getString(&s); + delete it; + + mFileMetaData->setCString(kMap[i].key, s); + } + } + + size_t dataSize; + String8 mime; + const void *data = id3.getAlbumArt(&dataSize, &mime); + + if (data) { + mFileMetaData->setData(kKeyAlbumArt, MetaData::TYPE_NONE, data, dataSize); + mFileMetaData->setCString(kKeyAlbumArtMIME, mime.string()); + } + } +} + sp<MediaSource> MPEG4Extractor::getTrack(size_t index) { status_t err; if ((err = readMetaData()) != OK) { diff --git a/media/libstagefright/MetaData.cpp b/media/libstagefright/MetaData.cpp index 7b60afc..1daead7 100644 --- a/media/libstagefright/MetaData.cpp +++ b/media/libstagefright/MetaData.cpp @@ -221,6 +221,16 @@ bool MetaData::findData(uint32_t key, uint32_t *type, return true; } +bool MetaData::hasData(uint32_t key) const { + ssize_t i = mItems.indexOfKey(key); + + if (i < 0) { + return false; + } + + return true; +} + MetaData::typed_data::typed_data() : mType(0), mSize(0) { diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp index a486522..f0f203c 100644 --- a/media/libstagefright/id3/ID3.cpp +++ b/media/libstagefright/id3/ID3.cpp @@ -56,14 +56,14 @@ private: DISALLOW_EVIL_CONSTRUCTORS(MemorySource); }; -ID3::ID3(const sp<DataSource> &source, bool ignoreV1) +ID3::ID3(const sp<DataSource> &source, bool ignoreV1, off64_t offset) : mIsValid(false), mData(NULL), mSize(0), mFirstFrameOffset(0), mVersion(ID3_UNKNOWN), mRawSize(0) { - mIsValid = parseV2(source); + mIsValid = parseV2(source, offset); if (!mIsValid && !ignoreV1) { mIsValid = parseV1(source); @@ -79,7 +79,7 @@ ID3::ID3(const uint8_t *data, size_t size, bool ignoreV1) mRawSize(0) { sp<MemorySource> source = new MemorySource(data, size); - mIsValid = parseV2(source); + mIsValid = parseV2(source, 0); if (!mIsValid && !ignoreV1) { mIsValid = parseV1(source); @@ -115,7 +115,7 @@ bool ID3::ParseSyncsafeInteger(const uint8_t encoded[4], size_t *x) { return true; } -bool ID3::parseV2(const sp<DataSource> &source) { +bool ID3::parseV2(const sp<DataSource> &source, off64_t offset) { struct id3_header { char id[3]; uint8_t version_major; @@ -126,7 +126,7 @@ struct id3_header { id3_header header; if (source->readAt( - 0, &header, sizeof(header)) != (ssize_t)sizeof(header)) { + offset, &header, sizeof(header)) != (ssize_t)sizeof(header)) { return false; } @@ -185,7 +185,7 @@ struct id3_header { mSize = size; mRawSize = mSize + sizeof(header); - if (source->readAt(sizeof(header), mData, mSize) != (ssize_t)mSize) { + if (source->readAt(offset + sizeof(header), mData, mSize) != (ssize_t)mSize) { free(mData); mData = NULL; diff --git a/media/libstagefright/include/ID3.h b/media/libstagefright/include/ID3.h index cca83ab..e83f3ef 100644 --- a/media/libstagefright/include/ID3.h +++ b/media/libstagefright/include/ID3.h @@ -35,7 +35,7 @@ struct ID3 { ID3_V2_4, }; - ID3(const sp<DataSource> &source, bool ignoreV1 = false); + ID3(const sp<DataSource> &source, bool ignoreV1 = false, off64_t offset = 0); ID3(const uint8_t *data, size_t size, bool ignoreV1 = false); ~ID3(); @@ -86,7 +86,7 @@ private: size_t mRawSize; bool parseV1(const sp<DataSource> &source); - bool parseV2(const sp<DataSource> &source); + bool parseV2(const sp<DataSource> &source, off64_t offset); void removeUnsynchronization(); bool removeUnsynchronizationV2_4(bool iTunesHack); diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h index bd5e4b9..7b4bc6d 100644 --- a/media/libstagefright/include/MPEG4Extractor.h +++ b/media/libstagefright/include/MPEG4Extractor.h @@ -97,6 +97,7 @@ private: status_t parseChunk(off64_t *offset, int depth); status_t parseITunesMetaData(off64_t offset, size_t size); status_t parse3GPPMetaData(off64_t offset, size_t size, int depth); + void parseID3v2MetaData(off64_t offset); status_t updateAudioTrackInfoFromESDS_MPEG4Audio( const void *esds_data, size_t esds_size); diff --git a/media/mtp/MtpDataPacket.cpp b/media/mtp/MtpDataPacket.cpp index 930f0b0..c4f87a0 100644 --- a/media/mtp/MtpDataPacket.cpp +++ b/media/mtp/MtpDataPacket.cpp @@ -331,7 +331,7 @@ void MtpDataPacket::putString(const char* s) { void MtpDataPacket::putString(const uint16_t* string) { int count = 0; - for (int i = 0; i < 256; i++) { + for (int i = 0; i <= MTP_STRING_MAX_CHARACTER_NUMBER; i++) { if (string[i]) count++; else diff --git a/media/mtp/MtpStringBuffer.cpp b/media/mtp/MtpStringBuffer.cpp index fe8cf04..f3420a4 100644 --- a/media/mtp/MtpStringBuffer.cpp +++ b/media/mtp/MtpStringBuffer.cpp @@ -56,42 +56,47 @@ MtpStringBuffer::~MtpStringBuffer() { } void MtpStringBuffer::set(const char* src) { - int length = strlen(src); - if (length >= sizeof(mBuffer)) - length = sizeof(mBuffer) - 1; - memcpy(mBuffer, src, length); - // count the characters int count = 0; char ch; - while ((ch = *src++) != 0) { + char* dest = (char*)mBuffer; + + while ((ch = *src++) != 0 && count < MTP_STRING_MAX_CHARACTER_NUMBER) { if ((ch & 0x80) == 0) { // single byte character + *dest++ = ch; } else if ((ch & 0xE0) == 0xC0) { // two byte character - if (! *src++) { + char ch1 = *src++; + if (! ch1) { // last character was truncated, so ignore last byte - length--; break; } + + *dest++ = ch; + *dest++ = ch1; } else if ((ch & 0xF0) == 0xE0) { // 3 byte char - if (! *src++) { + char ch1 = *src++; + if (! ch1) { // last character was truncated, so ignore last byte - length--; break; } - if (! *src++) { - // last character was truncated, so ignore last two bytes - length -= 2; + char ch2 = *src++; + if (! ch2) { + // last character was truncated, so ignore last byte break; } + + *dest++ = ch; + *dest++ = ch1; + *dest++ = ch2; } count++; } - mByteCount = length + 1; - mBuffer[length] = 0; + *dest++ = 0; + mByteCount = dest - (char*)mBuffer; mCharCount = count; } @@ -100,7 +105,7 @@ void MtpStringBuffer::set(const uint16_t* src) { uint16_t ch; uint8_t* dest = mBuffer; - while ((ch = *src++) != 0 && count < 255) { + while ((ch = *src++) != 0 && count < MTP_STRING_MAX_CHARACTER_NUMBER) { if (ch >= 0x0800) { *dest++ = (uint8_t)(0xE0 | (ch >> 12)); *dest++ = (uint8_t)(0x80 | ((ch >> 6) & 0x3F)); diff --git a/media/mtp/MtpStringBuffer.h b/media/mtp/MtpStringBuffer.h index cbc8307..e5150df 100644 --- a/media/mtp/MtpStringBuffer.h +++ b/media/mtp/MtpStringBuffer.h @@ -19,6 +19,9 @@ #include <stdint.h> +// Max Character number of a MTP String +#define MTP_STRING_MAX_CHARACTER_NUMBER 255 + namespace android { class MtpDataPacket; @@ -29,7 +32,7 @@ class MtpStringBuffer { private: // mBuffer contains string in UTF8 format // maximum 3 bytes/character, with 1 extra for zero termination - uint8_t mBuffer[255 * 3 + 1]; + uint8_t mBuffer[MTP_STRING_MAX_CHARACTER_NUMBER * 3 + 1]; int mCharCount; int mByteCount; diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 34811a7..1257161 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -559,6 +559,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( track = thread->createTrack_l(client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); + LOG_ALWAYS_FATAL_IF((track != 0) != (lStatus == NO_ERROR)); // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless // move effect chain to this output thread if an effect on same session was waiting @@ -1058,7 +1059,7 @@ size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t form return size; } -unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const +uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const { Mutex::Autolock _l(mLock); diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 066d5d5..0ab43e0 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -189,7 +189,7 @@ public: virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, audio_io_handle_t output) const; - virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; + virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; virtual int newAudioSessionId(); diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 01b90a8..d5a0e21 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -1339,6 +1339,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac // but TimedTrack::create() is a factory that could fail by returning NULL lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; if (lStatus != NO_ERROR) { + ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); track.clear(); goto Exit; } @@ -5371,7 +5372,7 @@ void AudioFlinger::RecordThread::readInputParameters() mRsmpInIndex = mFrameCount; } -unsigned int AudioFlinger::RecordThread::getInputFramesLost() +uint32_t AudioFlinger::RecordThread::getInputFramesLost() { Mutex::Autolock _l(mLock); if (initCheck() != NO_ERROR) { diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index 6b81c38..2b749fa 100644 --- a/services/audioflinger/Threads.h +++ b/services/audioflinger/Threads.h @@ -907,7 +907,7 @@ public: virtual String8 getParameters(const String8& keys); virtual void audioConfigChanged_l(int event, int param = 0); void readInputParameters(); - virtual unsigned int getInputFramesLost(); + virtual uint32_t getInputFramesLost(); virtual status_t addEffectChain_l(const sp<EffectChain>& chain); virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); |