diff options
26 files changed, 645 insertions, 543 deletions
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index 62f0c64..36430a3 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -60,7 +60,7 @@ public: size_t frameCount; // number of sample frames corresponding to size; // on input it is the number of frames available, // on output is the number of frames actually drained - // (currently ignored, but will make the primary field in future) + // (currently ignored but will make the primary field in future) size_t size; // input/output in bytes == frameCount * frameSize // FIXME this is redundant with respect to frameCount, @@ -444,7 +444,8 @@ private: // notification callback uint32_t mNotificationFramesAct; // actual number of frames between each // notification callback - bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 + bool mRefreshRemaining; // processAudioBuffer() should refresh + // mRemainingFrames and mRetryOnPartialBuffer // These are private to processAudioBuffer(), and are not protected by a lock uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index 225ef76..b96b8a1 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -155,7 +155,8 @@ public: class OutputDescriptor { public: OutputDescriptor() - : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) {} + : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) + {} uint32_t samplingRate; audio_format_t format; diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index 453c106..56bf49a 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -660,7 +660,6 @@ protected: size_t mReqFrameCount; // frame count to request the next time a new // IAudioTrack is needed - // constant after constructor or set() audio_format_t mFormat; // as requested by client, not forced to 16-bit audio_stream_type_t mStreamType; @@ -699,7 +698,8 @@ protected: uint32_t mNotificationFramesAct; // actual number of frames between each // notification callback, // at initial source sample rate - bool mRefreshRemaining; // processAudioBuffer() should refresh next 2 + bool mRefreshRemaining; // processAudioBuffer() should refresh + // mRemainingFrames and mRetryOnPartialBuffer // These are private to processAudioBuffer(), and are not protected by a lock uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h index ad7409d..d6e7ac4 100644 --- a/include/private/media/AudioTrackShared.h +++ b/include/private/media/AudioTrackShared.h @@ -48,7 +48,7 @@ namespace android { #define CBLK_STREAM_END_DONE 0x400 // set by server on render completion, cleared by client //EL_FIXME 20 seconds may not be enough and must be reconciled with new obtainBuffer implementation -#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 //assuming upto a maximum of 20 seconds of offloaded +#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 // assuming up to a maximum of 20 seconds of offloaded struct AudioTrackSharedStreaming { // similar to NBAIO MonoPipe diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index be818c6..22ad453 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -132,7 +132,7 @@ public: lStatus = reply.readInt32(); track = interface_cast<IAudioTrack>(reply.readStrongBinder()); } - if (status) { + if (status != NULL) { *status = lStatus; } return track; @@ -180,7 +180,7 @@ public: lStatus = reply.readInt32(); record = interface_cast<IAudioRecord>(reply.readStrongBinder()); } - if (status) { + if (status != NULL) { *status = lStatus; } return record; @@ -397,15 +397,25 @@ public: audio_io_handle_t output = (audio_io_handle_t) reply.readInt32(); ALOGV("openOutput() returned output, %d", output); devices = (audio_devices_t)reply.readInt32(); - if (pDevices != NULL) *pDevices = devices; + if (pDevices != NULL) { + *pDevices = devices; + } samplingRate = reply.readInt32(); - if (pSamplingRate != NULL) *pSamplingRate = samplingRate; + if (pSamplingRate != NULL) { + *pSamplingRate = samplingRate; + } format = (audio_format_t) reply.readInt32(); - if (pFormat != NULL) *pFormat = format; + if (pFormat != NULL) { + *pFormat = format; + } channelMask = (audio_channel_mask_t)reply.readInt32(); - if (pChannelMask != NULL) *pChannelMask = channelMask; + if (pChannelMask != NULL) { + *pChannelMask = channelMask; + } latency = reply.readInt32(); - if (pLatencyMs != NULL) *pLatencyMs = latency; + if (pLatencyMs != NULL) { + *pLatencyMs = latency; + } return output; } @@ -469,13 +479,21 @@ public: remote()->transact(OPEN_INPUT, data, &reply); audio_io_handle_t input = (audio_io_handle_t) reply.readInt32(); devices = (audio_devices_t)reply.readInt32(); - if (pDevices != NULL) *pDevices = devices; + if (pDevices != NULL) { + *pDevices = devices; + } samplingRate = reply.readInt32(); - if (pSamplingRate != NULL) *pSamplingRate = samplingRate; + if (pSamplingRate != NULL) { + *pSamplingRate = samplingRate; + } format = (audio_format_t) reply.readInt32(); - if (pFormat != NULL) *pFormat = format; + if (pFormat != NULL) { + *pFormat = format; + } channelMask = (audio_channel_mask_t)reply.readInt32(); - if (pChannelMask != NULL) *pChannelMask = channelMask; + if (pChannelMask != NULL) { + *pChannelMask = channelMask; + } return input; } @@ -517,11 +535,11 @@ public: status_t status = reply.readInt32(); if (status == NO_ERROR) { uint32_t tmp = reply.readInt32(); - if (halFrames) { + if (halFrames != NULL) { *halFrames = tmp; } tmp = reply.readInt32(); - if (dspFrames) { + if (dspFrames != NULL) { *dspFrames = tmp; } } @@ -639,7 +657,7 @@ public: if (pDesc == NULL) { return effect; - if (status) { + if (status != NULL) { *status = BAD_VALUE; } } @@ -657,7 +675,7 @@ public: } else { lStatus = reply.readInt32(); int tmp = reply.readInt32(); - if (id) { + if (id != NULL) { *id = tmp; } tmp = reply.readInt32(); @@ -667,7 +685,7 @@ public: effect = interface_cast<IEffect>(reply.readStrongBinder()); reply.read(pDesc, sizeof(effect_descriptor_t)); } - if (status) { + if (status != NULL) { *status = lStatus; } diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp index 1b20cbb..c9b5d26 100644 --- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp +++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp @@ -58,6 +58,8 @@ SoftAAC2::SoftAAC2( mIsADTS(false), mInputBufferCount(0), mSignalledError(false), + mSawInputEos(false), + mSignalledOutputEos(false), mAnchorTimeUs(0), mNumSamplesOutput(0), mOutputPortSettingsChange(NONE) { @@ -350,115 +352,83 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { return; } - while (!inQueue.empty() && !outQueue.empty()) { - BufferInfo *inInfo = *inQueue.begin(); - OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; + while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) { + BufferInfo *inInfo = NULL; + OMX_BUFFERHEADERTYPE *inHeader = NULL; + if (!inQueue.empty()) { + inInfo = *inQueue.begin(); + inHeader = inInfo->mHeader; + } BufferInfo *outInfo = *outQueue.begin(); OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; + outHeader->nFlags = 0; - if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { - inQueue.erase(inQueue.begin()); - inInfo->mOwnedByUs = false; - notifyEmptyBufferDone(inHeader); - - if (mDecoderHasData) { - // flush out the decoder's delayed data by calling DecodeFrame - // one more time, with the AACDEC_FLUSH flag set - INT_PCM *outBuffer = - reinterpret_cast<INT_PCM *>( - outHeader->pBuffer + outHeader->nOffset); - - AAC_DECODER_ERROR decoderErr = - aacDecoder_DecodeFrame(mAACDecoder, - outBuffer, - outHeader->nAllocLen, - AACDEC_FLUSH); - mDecoderHasData = false; - - if (decoderErr != AAC_DEC_OK) { - mSignalledError = true; - - notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, - NULL); - - return; - } - - outHeader->nFilledLen = - mStreamInfo->frameSize - * sizeof(int16_t) - * mStreamInfo->numChannels; - } else { - // we never submitted any data to the decoder, so there's nothing to flush out - outHeader->nFilledLen = 0; + if (inHeader) { + if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { + mSawInputEos = true; } - outHeader->nFlags = OMX_BUFFERFLAG_EOS; - - outQueue.erase(outQueue.begin()); - outInfo->mOwnedByUs = false; - notifyFillBufferDone(outHeader); - return; - } - - if (inHeader->nOffset == 0) { - mAnchorTimeUs = inHeader->nTimeStamp; - mNumSamplesOutput = 0; - } + if (inHeader->nOffset == 0 && inHeader->nFilledLen) { + mAnchorTimeUs = inHeader->nTimeStamp; + mNumSamplesOutput = 0; + } - size_t adtsHeaderSize = 0; - if (mIsADTS) { - // skip 30 bits, aac_frame_length follows. - // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll????? + if (mIsADTS) { + size_t adtsHeaderSize = 0; + // skip 30 bits, aac_frame_length follows. + // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll????? - const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset; + const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset; - bool signalError = false; - if (inHeader->nFilledLen < 7) { - ALOGE("Audio data too short to contain even the ADTS header. " - "Got %ld bytes.", inHeader->nFilledLen); - hexdump(adtsHeader, inHeader->nFilledLen); - signalError = true; - } else { - bool protectionAbsent = (adtsHeader[1] & 1); - - unsigned aac_frame_length = - ((adtsHeader[3] & 3) << 11) - | (adtsHeader[4] << 3) - | (adtsHeader[5] >> 5); - - if (inHeader->nFilledLen < aac_frame_length) { - ALOGE("Not enough audio data for the complete frame. " - "Got %ld bytes, frame size according to the ADTS " - "header is %u bytes.", - inHeader->nFilledLen, aac_frame_length); + bool signalError = false; + if (inHeader->nFilledLen < 7) { + ALOGE("Audio data too short to contain even the ADTS header. " + "Got %ld bytes.", inHeader->nFilledLen); hexdump(adtsHeader, inHeader->nFilledLen); signalError = true; } else { - adtsHeaderSize = (protectionAbsent ? 7 : 9); - - inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize; - inBufferLength[0] = aac_frame_length - adtsHeaderSize; - - inHeader->nOffset += adtsHeaderSize; - inHeader->nFilledLen -= adtsHeaderSize; + bool protectionAbsent = (adtsHeader[1] & 1); + + unsigned aac_frame_length = + ((adtsHeader[3] & 3) << 11) + | (adtsHeader[4] << 3) + | (adtsHeader[5] >> 5); + + if (inHeader->nFilledLen < aac_frame_length) { + ALOGE("Not enough audio data for the complete frame. " + "Got %ld bytes, frame size according to the ADTS " + "header is %u bytes.", + inHeader->nFilledLen, aac_frame_length); + hexdump(adtsHeader, inHeader->nFilledLen); + signalError = true; + } else { + adtsHeaderSize = (protectionAbsent ? 7 : 9); + + inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize; + inBufferLength[0] = aac_frame_length - adtsHeaderSize; + + inHeader->nOffset += adtsHeaderSize; + inHeader->nFilledLen -= adtsHeaderSize; + } } - } - if (signalError) { - mSignalledError = true; + if (signalError) { + mSignalledError = true; - notify(OMX_EventError, - OMX_ErrorStreamCorrupt, - ERROR_MALFORMED, - NULL); + notify(OMX_EventError, + OMX_ErrorStreamCorrupt, + ERROR_MALFORMED, + NULL); - return; + return; + } + } else { + inBuffer[0] = inHeader->pBuffer + inHeader->nOffset; + inBufferLength[0] = inHeader->nFilledLen; } } else { - inBuffer[0] = inHeader->pBuffer + inHeader->nOffset; - inBufferLength[0] = inHeader->nFilledLen; + inBufferLength[0] = 0; } // Fill and decode @@ -471,50 +441,66 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { int prevNumChannels = mStreamInfo->numChannels; AAC_DECODER_ERROR decoderErr = AAC_DEC_NOT_ENOUGH_BITS; - while (bytesValid[0] > 0 && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) { + while ((bytesValid[0] > 0 || mSawInputEos) && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) { + mDecoderHasData |= (bytesValid[0] > 0); aacDecoder_Fill(mAACDecoder, inBuffer, inBufferLength, bytesValid); - mDecoderHasData = true; decoderErr = aacDecoder_DecodeFrame(mAACDecoder, outBuffer, outHeader->nAllocLen, 0 /* flags */); - if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) { - ALOGW("Not enough bits, bytesValid %d", bytesValid[0]); + if (mSawInputEos && bytesValid[0] <= 0) { + if (mDecoderHasData) { + // flush out the decoder's delayed data by calling DecodeFrame + // one more time, with the AACDEC_FLUSH flag set + decoderErr = aacDecoder_DecodeFrame(mAACDecoder, + outBuffer, + outHeader->nAllocLen, + AACDEC_FLUSH); + mDecoderHasData = false; + } + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + mSignalledOutputEos = true; + break; + } else { + ALOGW("Not enough bits, bytesValid %d", bytesValid[0]); + } } } size_t numOutBytes = mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels; - if (decoderErr == AAC_DEC_OK) { - UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0]; - inHeader->nFilledLen -= inBufferUsedLength; - inHeader->nOffset += inBufferUsedLength; - } else { - ALOGW("AAC decoder returned error %d, substituting silence", - decoderErr); + if (inHeader) { + if (decoderErr == AAC_DEC_OK) { + UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0]; + inHeader->nFilledLen -= inBufferUsedLength; + inHeader->nOffset += inBufferUsedLength; + } else { + ALOGW("AAC decoder returned error %d, substituting silence", + decoderErr); - memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes); + memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes); - // Discard input buffer. - inHeader->nFilledLen = 0; + // Discard input buffer. + inHeader->nFilledLen = 0; - aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1); + aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1); - // fall through - } + // fall through + } - if (inHeader->nFilledLen == 0) { - inInfo->mOwnedByUs = false; - inQueue.erase(inQueue.begin()); - inInfo = NULL; - notifyEmptyBufferDone(inHeader); - inHeader = NULL; + if (inHeader->nFilledLen == 0) { + inInfo->mOwnedByUs = false; + inQueue.erase(inQueue.begin()); + inInfo = NULL; + notifyEmptyBufferDone(inHeader); + inHeader = NULL; + } } /* @@ -555,7 +541,6 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) { // we've previously decoded valid data, in the latter case // (decode failed) we'll output a silent frame. outHeader->nFilledLen = numOutBytes; - outHeader->nFlags = 0; outHeader->nTimeStamp = mAnchorTimeUs @@ -606,6 +591,8 @@ void SoftAAC2::onReset() { mStreamInfo->sampleRate = 0; mSignalledError = false; + mSawInputEos = false; + mSignalledOutputEos = false; mOutputPortSettingsChange = NONE; } diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h index 2d960ab..a7ea1e2 100644 --- a/media/libstagefright/codecs/aacdec/SoftAAC2.h +++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h @@ -55,6 +55,8 @@ private: bool mDecoderHasData; size_t mInputBufferCount; bool mSignalledError; + bool mSawInputEos; + bool mSignalledOutputEos; int64_t mAnchorTimeUs; int64_t mNumSamplesOutput; diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp index 7c382fb..877e3cb 100644 --- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp +++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp @@ -49,6 +49,8 @@ SoftMP3::SoftMP3( mNumChannels(2), mSamplingRate(44100), mSignalledError(false), + mSawInputEos(false), + mSignalledOutputEos(false), mOutputPortSettingsChange(NONE) { initPorts(); initDecoder(); @@ -194,48 +196,36 @@ void SoftMP3::onQueueFilled(OMX_U32 portIndex) { List<BufferInfo *> &inQueue = getPortQueue(0); List<BufferInfo *> &outQueue = getPortQueue(1); - while (!inQueue.empty() && !outQueue.empty()) { - BufferInfo *inInfo = *inQueue.begin(); - OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; + while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) { + BufferInfo *inInfo = NULL; + OMX_BUFFERHEADERTYPE *inHeader = NULL; + if (!inQueue.empty()) { + inInfo = *inQueue.begin(); + inHeader = inInfo->mHeader; + } BufferInfo *outInfo = *outQueue.begin(); OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; + outHeader->nFlags = 0; - if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { - inQueue.erase(inQueue.begin()); - inInfo->mOwnedByUs = false; - notifyEmptyBufferDone(inHeader); - - if (!mIsFirst) { - // pad the end of the stream with 529 samples, since that many samples - // were trimmed off the beginning when decoding started - outHeader->nFilledLen = - kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t); + if (inHeader) { + if (inHeader->nOffset == 0 && inHeader->nFilledLen) { + mAnchorTimeUs = inHeader->nTimeStamp; + mNumFramesOutput = 0; + } - memset(outHeader->pBuffer, 0, outHeader->nFilledLen); - } else { - // Since we never discarded frames from the start, we won't have - // to add any padding at the end either. - outHeader->nFilledLen = 0; + if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { + mSawInputEos = true; } - outHeader->nFlags = OMX_BUFFERFLAG_EOS; + mConfig->pInputBuffer = + inHeader->pBuffer + inHeader->nOffset; - outQueue.erase(outQueue.begin()); - outInfo->mOwnedByUs = false; - notifyFillBufferDone(outHeader); - return; - } - - if (inHeader->nOffset == 0) { - mAnchorTimeUs = inHeader->nTimeStamp; - mNumFramesOutput = 0; + mConfig->inputBufferCurrentLength = inHeader->nFilledLen; + } else { + mConfig->pInputBuffer = NULL; + mConfig->inputBufferCurrentLength = 0; } - - mConfig->pInputBuffer = - inHeader->pBuffer + inHeader->nOffset; - - mConfig->inputBufferCurrentLength = inHeader->nFilledLen; mConfig->inputBufferMaxLength = 0; mConfig->inputBufferUsedLength = 0; @@ -262,13 +252,28 @@ void SoftMP3::onQueueFilled(OMX_U32 portIndex) { mConfig->outputFrameSize = kOutputBufferSize / sizeof(int16_t); } - // This is recoverable, just ignore the current frame and - // play silence instead. - memset(outHeader->pBuffer, - 0, - mConfig->outputFrameSize * sizeof(int16_t)); - - mConfig->inputBufferUsedLength = inHeader->nFilledLen; + if (decoderErr == NO_ENOUGH_MAIN_DATA_ERROR && mSawInputEos) { + if (!mIsFirst) { + // pad the end of the stream with 529 samples, since that many samples + // were trimmed off the beginning when decoding started + outHeader->nOffset = 0; + outHeader->nFilledLen = kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t); + + memset(outHeader->pBuffer, 0, outHeader->nFilledLen); + } + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + mSignalledOutputEos = true; + } else { + // This is recoverable, just ignore the current frame and + // play silence instead. + memset(outHeader->pBuffer, + 0, + mConfig->outputFrameSize * sizeof(int16_t)); + + if (inHeader) { + mConfig->inputBufferUsedLength = inHeader->nFilledLen; + } + } } else if (mConfig->samplingRate != mSamplingRate || mConfig->num_channels != mNumChannels) { mSamplingRate = mConfig->samplingRate; @@ -289,7 +294,7 @@ void SoftMP3::onQueueFilled(OMX_U32 portIndex) { outHeader->nFilledLen = mConfig->outputFrameSize * sizeof(int16_t) - outHeader->nOffset; - } else { + } else if (!mSignalledOutputEos) { outHeader->nOffset = 0; outHeader->nFilledLen = mConfig->outputFrameSize * sizeof(int16_t); } @@ -298,23 +303,24 @@ void SoftMP3::onQueueFilled(OMX_U32 portIndex) { mAnchorTimeUs + (mNumFramesOutput * 1000000ll) / mConfig->samplingRate; - outHeader->nFlags = 0; - - CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength); + if (inHeader) { + CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength); - inHeader->nOffset += mConfig->inputBufferUsedLength; - inHeader->nFilledLen -= mConfig->inputBufferUsedLength; + inHeader->nOffset += mConfig->inputBufferUsedLength; + inHeader->nFilledLen -= mConfig->inputBufferUsedLength; - mNumFramesOutput += mConfig->outputFrameSize / mNumChannels; - if (inHeader->nFilledLen == 0) { - inInfo->mOwnedByUs = false; - inQueue.erase(inQueue.begin()); - inInfo = NULL; - notifyEmptyBufferDone(inHeader); - inHeader = NULL; + if (inHeader->nFilledLen == 0) { + inInfo->mOwnedByUs = false; + inQueue.erase(inQueue.begin()); + inInfo = NULL; + notifyEmptyBufferDone(inHeader); + inHeader = NULL; + } } + mNumFramesOutput += mConfig->outputFrameSize / mNumChannels; + outInfo->mOwnedByUs = false; outQueue.erase(outQueue.begin()); outInfo = NULL; @@ -362,6 +368,8 @@ void SoftMP3::onReset() { pvmp3_InitDecoder(mConfig, mDecoderBuf); mIsFirst = true; mSignalledError = false; + mSawInputEos = false; + mSignalledOutputEos = false; mOutputPortSettingsChange = NONE; } diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.h b/media/libstagefright/codecs/mp3dec/SoftMP3.h index 4af91ea..f9e7b53 100644 --- a/media/libstagefright/codecs/mp3dec/SoftMP3.h +++ b/media/libstagefright/codecs/mp3dec/SoftMP3.h @@ -61,6 +61,8 @@ private: bool mIsFirst; bool mSignalledError; + bool mSawInputEos; + bool mSignalledOutputEos; enum { NONE, diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp index 51bb958..515e4d3 100644 --- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp +++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp @@ -54,6 +54,8 @@ SoftVorbis::SoftVorbis( mAnchorTimeUs(0), mNumFramesOutput(0), mNumFramesLeftOnPage(-1), + mSawInputEos(false), + mSignalledOutputEos(false), mOutputPortSettingsChange(NONE) { initPorts(); CHECK_EQ(initDecoder(), (status_t)OK); @@ -290,48 +292,47 @@ void SoftVorbis::onQueueFilled(OMX_U32 portIndex) { return; } - while (!inQueue.empty() && !outQueue.empty()) { - BufferInfo *inInfo = *inQueue.begin(); - OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader; + while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) { + BufferInfo *inInfo = NULL; + OMX_BUFFERHEADERTYPE *inHeader = NULL; + if (!inQueue.empty()) { + inInfo = *inQueue.begin(); + inHeader = inInfo->mHeader; + } BufferInfo *outInfo = *outQueue.begin(); OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader; - if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { - inQueue.erase(inQueue.begin()); - inInfo->mOwnedByUs = false; - notifyEmptyBufferDone(inHeader); + int32_t numPageSamples = 0; - outHeader->nFilledLen = 0; - outHeader->nFlags = OMX_BUFFERFLAG_EOS; + if (inHeader) { + if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) { + mSawInputEos = true; + } - outQueue.erase(outQueue.begin()); - outInfo->mOwnedByUs = false; - notifyFillBufferDone(outHeader); - return; - } + if (inHeader->nFilledLen || !mSawInputEos) { + CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples)); + memcpy(&numPageSamples, + inHeader->pBuffer + + inHeader->nOffset + inHeader->nFilledLen - 4, + sizeof(numPageSamples)); - int32_t numPageSamples; - CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples)); - memcpy(&numPageSamples, - inHeader->pBuffer - + inHeader->nOffset + inHeader->nFilledLen - 4, - sizeof(numPageSamples)); + if (inHeader->nOffset == 0) { + mAnchorTimeUs = inHeader->nTimeStamp; + mNumFramesOutput = 0; + } - if (numPageSamples >= 0) { - mNumFramesLeftOnPage = numPageSamples; + inHeader->nFilledLen -= sizeof(numPageSamples);; + } } - if (inHeader->nOffset == 0) { - mAnchorTimeUs = inHeader->nTimeStamp; - mNumFramesOutput = 0; + if (numPageSamples >= 0) { + mNumFramesLeftOnPage = numPageSamples; } - inHeader->nFilledLen -= sizeof(numPageSamples);; - ogg_buffer buf; - buf.data = inHeader->pBuffer + inHeader->nOffset; - buf.size = inHeader->nFilledLen; + buf.data = inHeader ? inHeader->pBuffer + inHeader->nOffset : NULL; + buf.size = inHeader ? inHeader->nFilledLen : 0; buf.refcount = 1; buf.ptr.owner = NULL; @@ -351,6 +352,7 @@ void SoftVorbis::onQueueFilled(OMX_U32 portIndex) { int numFrames = 0; + outHeader->nFlags = 0; int err = vorbis_dsp_synthesis(mState, &pack, 1); if (err != 0) { ALOGW("vorbis_dsp_synthesis returned %d", err); @@ -370,13 +372,16 @@ void SoftVorbis::onQueueFilled(OMX_U32 portIndex) { ALOGV("discarding %d frames at end of page", numFrames - mNumFramesLeftOnPage); numFrames = mNumFramesLeftOnPage; + if (mSawInputEos) { + outHeader->nFlags = OMX_BUFFERFLAG_EOS; + mSignalledOutputEos = true; + } } mNumFramesLeftOnPage -= numFrames; } outHeader->nFilledLen = numFrames * sizeof(int16_t) * mVi->channels; outHeader->nOffset = 0; - outHeader->nFlags = 0; outHeader->nTimeStamp = mAnchorTimeUs @@ -384,11 +389,13 @@ void SoftVorbis::onQueueFilled(OMX_U32 portIndex) { mNumFramesOutput += numFrames; - inInfo->mOwnedByUs = false; - inQueue.erase(inQueue.begin()); - inInfo = NULL; - notifyEmptyBufferDone(inHeader); - inHeader = NULL; + if (inHeader) { + inInfo->mOwnedByUs = false; + inQueue.erase(inQueue.begin()); + inInfo = NULL; + notifyEmptyBufferDone(inHeader); + inHeader = NULL; + } outInfo->mOwnedByUs = false; outQueue.erase(outQueue.begin()); @@ -425,6 +432,8 @@ void SoftVorbis::onReset() { mVi = NULL; } + mSawInputEos = false; + mSignalledOutputEos = false; mOutputPortSettingsChange = NONE; } diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h index cb628a0..1d00816 100644 --- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h +++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h @@ -59,6 +59,8 @@ private: int64_t mAnchorTimeUs; int64_t mNumFramesOutput; int32_t mNumFramesLeftOnPage; + bool mSawInputEos; + bool mSignalledOutputEos; enum { NONE, diff --git a/media/libstagefright/timedtext/test/Android.mk b/media/libstagefright/timedtext/test/Android.mk index a5e7ba2..9a9fde2 100644 --- a/media/libstagefright/timedtext/test/Android.mk +++ b/media/libstagefright/timedtext/test/Android.mk @@ -2,7 +2,6 @@ LOCAL_PATH:= $(call my-dir) # ================================================================ # Unit tests for libstagefright_timedtext -# See also /development/testrunner/test_defs.xml # ================================================================ # ================================================================ @@ -18,10 +17,13 @@ LOCAL_SRC_FILES := TimedTextSRTSource_test.cpp LOCAL_C_INCLUDES := \ $(TOP)/external/expat/lib \ - $(TOP)/frameworks/base/media/libstagefright/timedtext + $(TOP)/frameworks/av/media/libstagefright/timedtext LOCAL_SHARED_LIBRARIES := \ + libbinder \ libexpat \ - libstagefright + libstagefright \ + libstagefright_foundation \ + libutils include $(BUILD_NATIVE_TEST) diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 626b5c2..fc98891 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -163,12 +163,15 @@ AudioFlinger::AudioFlinger() (void) property_get("af.tee", value, "0"); teeEnabled = atoi(value); } - if (teeEnabled & 1) + if (teeEnabled & 1) { mTeeSinkInputEnabled = true; - if (teeEnabled & 2) + } + if (teeEnabled & 2) { mTeeSinkOutputEnabled = true; - if (teeEnabled & 4) + } + if (teeEnabled & 4) { mTeeSinkTrackEnabled = true; + } #endif } @@ -506,10 +509,12 @@ sp<IAudioTrack> AudioFlinger::createTrack( track = thread->createTrack_l(client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); + // 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 // for a track to be created if (lStatus == NO_ERROR && effectThread != NULL) { + // no risk of deadlock because AudioFlinger::mLock is held Mutex::Autolock _dl(thread->mLock); Mutex::Autolock _sl(effectThread->mLock); moveEffectChain_l(lSessionId, effectThread, thread, true); @@ -529,7 +534,9 @@ sp<IAudioTrack> AudioFlinger::createTrack( } } } + } + if (lStatus == NO_ERROR) { // s for server's pid, n for normal mixer name, f for fast index name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, @@ -543,9 +550,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( } Exit: - if (status != NULL) { - *status = lStatus; - } + *status = lStatus; return trackHandle; } @@ -1279,6 +1284,7 @@ sp<IAudioRecord> AudioFlinger::openRecord( recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, frameCount, lSessionId, flags, tid, &lStatus); } + if (lStatus != NO_ERROR) { // remove local strong reference to Client before deleting the RecordTrack so that the // Client destructor is called by the TrackBase destructor with mLock held @@ -1287,14 +1293,11 @@ sp<IAudioRecord> AudioFlinger::openRecord( goto Exit; } - // return to handle to client + // return handle to client recordHandle = new RecordHandle(recordTrack); - lStatus = NO_ERROR; Exit: - if (status) { - *status = lStatus; - } + *status = lStatus; return recordHandle; } @@ -1435,18 +1438,15 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo) { - PlaybackThread *thread = NULL; struct audio_config config; + memset(&config, 0, sizeof(config)); config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; - if (offloadInfo) { + if (offloadInfo != NULL) { config.offload_info = *offloadInfo; } - audio_stream_out_t *outStream = NULL; - AudioHwDevice *outHwDev; - ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", module, (pDevices != NULL) ? *pDevices : 0, @@ -1455,7 +1455,7 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, config.channel_mask, flags); ALOGV("openOutput(), offloadInfo %p version 0x%04x", - offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version ); + offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version); if (pDevices == NULL || *pDevices == 0) { return 0; @@ -1463,15 +1463,17 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, Mutex::Autolock _l(mLock); - outHwDev = findSuitableHwDev_l(module, *pDevices); - if (outHwDev == NULL) + AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices); + if (outHwDev == NULL) { return 0; + } audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); audio_io_handle_t id = nextUniqueId(); mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; + audio_stream_out_t *outStream = NULL; status_t status = hwDevHal->open_output_stream(hwDevHal, id, *pDevices, @@ -1491,6 +1493,7 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, if (status == NO_ERROR && outStream != NULL) { AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); + PlaybackThread *thread; if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { thread = new OffloadThread(this, output, id, *pDevices); ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); @@ -1658,18 +1661,15 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask) { - status_t status; - RecordThread *thread = NULL; struct audio_config config; + memset(&config, 0, sizeof(config)); config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; uint32_t reqSamplingRate = config.sample_rate; audio_format_t reqFormat = config.format; - audio_channel_mask_t reqChannels = config.channel_mask; - audio_stream_in_t *inStream = NULL; - AudioHwDevice *inHwDev; + audio_channel_mask_t reqChannelMask = config.channel_mask; if (pDevices == NULL || *pDevices == 0) { return 0; @@ -1677,14 +1677,16 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, Mutex::Autolock _l(mLock); - inHwDev = findSuitableHwDev_l(module, *pDevices); - if (inHwDev == NULL) + AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices); + if (inHwDev == NULL) { return 0; + } audio_hw_device_t *inHwHal = inHwDev->hwDevice(); audio_io_handle_t id = nextUniqueId(); - status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, + audio_stream_in_t *inStream = NULL; + status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " "status %d", @@ -1700,7 +1702,7 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, if (status == BAD_VALUE && reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && (config.sample_rate <= 2 * reqSamplingRate) && - (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { + (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) { ALOGV("openInput() reopening with proposed sampling rate and channel mask"); inStream = NULL; status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); @@ -1762,10 +1764,10 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, // Start record thread // RecordThread requires both input and output device indication to forward to audio // pre processing modules - thread = new RecordThread(this, + RecordThread *thread = new RecordThread(this, input, reqSamplingRate, - reqChannels, + reqChannelMask, id, primaryOutputDevice_l(), *pDevices @@ -1782,7 +1784,7 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, *pFormat = config.format; } if (pChannelMask != NULL) { - *pChannelMask = reqChannels; + *pChannelMask = reqChannelMask; } // notify client processes of the new input creation @@ -1928,7 +1930,7 @@ void AudioFlinger::purgeStaleEffects_l() { } } if (!found) { - Mutex::Autolock _l (t->mLock); + Mutex::Autolock _l(t->mLock); // remove all effects from the chain while (ec->mEffects.size()) { sp<EffectModule> effect = ec->mEffects[0]; @@ -2225,9 +2227,7 @@ sp<IEffect> AudioFlinger::createEffect( } Exit: - if (status != NULL) { - *status = lStatus; - } + *status = lStatus; return handle; } diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 0992308..39cdec0 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -109,7 +109,7 @@ public: pid_t tid, int *sessionId, String8& name, - status_t *status); + status_t *status /*non-NULL*/); virtual sp<IAudioRecord> openRecord( audio_io_handle_t input, @@ -120,7 +120,7 @@ public: IAudioFlinger::track_flags_t *flags, pid_t tid, int *sessionId, - status_t *status); + status_t *status /*non-NULL*/); virtual uint32_t sampleRate(audio_io_handle_t output) const; virtual int channelCount(audio_io_handle_t output) const; @@ -209,7 +209,7 @@ public: int32_t priority, audio_io_handle_t io, int sessionId, - status_t *status, + status_t *status /*non-NULL*/, int *id, int *enabled); diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index df4e029..91aedbb 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -229,7 +229,7 @@ int AudioMixer::getTrackName(audio_channel_mask_t channelMask, int sessionId) void AudioMixer::invalidateState(uint32_t mask) { - if (mask) { + if (mask != 0) { mState.needsChanged |= mask; mState.hook = process__validate; } @@ -709,7 +709,7 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) // select the processing hooks state->hook = process__nop; - if (countActiveTracks) { + if (countActiveTracks > 0) { if (resampling) { if (!state->outputTemp) { state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; @@ -745,15 +745,14 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) // Now that the volume ramp has been done, set optimal state and // track hooks for subsequent mixer process - if (countActiveTracks) { + if (countActiveTracks > 0) { bool allMuted = true; uint32_t en = state->enabledTracks; while (en) { const int i = 31 - __builtin_clz(en); en &= ~(1<<i); track_t& t = state->tracks[i]; - if (!t.doesResample() && t.volumeRL == 0) - { + if (!t.doesResample() && t.volumeRL == 0) { t.needs |= NEEDS_MUTE_ENABLED; t.hook = track__nop; } else { @@ -1124,8 +1123,9 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) t.in = t.buffer.raw; // t.in == NULL can happen if the track was flushed just after having // been enabled for mixing. - if (t.in == NULL) + if (t.in == NULL) { enabledTracks &= ~(1<<i); + } } e0 = enabledTracks; @@ -1162,7 +1162,7 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) } while (outFrames) { size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; - if (inFrames) { + if (inFrames > 0) { t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux); t.frameCount -= inFrames; @@ -1445,8 +1445,9 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state, int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS, int outputFrameIndex) { - if (AudioBufferProvider::kInvalidPTS == basePTS) + if (AudioBufferProvider::kInvalidPTS == basePTS) { return AudioBufferProvider::kInvalidPTS; + } return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate); } diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp index 900b411..d6ea758 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audioflinger/AudioPolicyService.cpp @@ -77,24 +77,28 @@ AudioPolicyService::AudioPolicyService() mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this); /* instantiate the audio policy manager */ rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module); - if (rc) + if (rc) { return; + } rc = audio_policy_dev_open(module, &mpAudioPolicyDev); ALOGE_IF(rc, "couldn't open audio policy device (%s)", strerror(-rc)); - if (rc) + if (rc) { return; + } rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this, &mpAudioPolicy); ALOGE_IF(rc, "couldn't create audio policy (%s)", strerror(-rc)); - if (rc) + if (rc) { return; + } rc = mpAudioPolicy->init_check(mpAudioPolicy); ALOGE_IF(rc, "couldn't init_check the audio policy (%s)", strerror(-rc)); - if (rc) + if (rc) { return; + } ALOGI("Loaded audio policy from %s (%s)", module->name, module->id); @@ -126,10 +130,12 @@ AudioPolicyService::~AudioPolicyService() } mInputs.clear(); - if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) + if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) { mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy); - if (mpAudioPolicyDev != NULL) + } + if (mpAudioPolicyDev != NULL) { audio_policy_dev_close(mpAudioPolicyDev); + } } status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, @@ -1105,11 +1111,13 @@ int AudioPolicyService::setStreamVolume(audio_stream_type_t stream, int AudioPolicyService::startTone(audio_policy_tone_t tone, audio_stream_type_t stream) { - if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) + if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) { ALOGE("startTone: illegal tone requested (%d)", tone); - if (stream != AUDIO_STREAM_VOICE_CALL) + } + if (stream != AUDIO_STREAM_VOICE_CALL) { ALOGE("startTone: illegal stream (%d) requested for tone %d", stream, tone); + } mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING, AUDIO_STREAM_VOICE_CALL); return 0; @@ -1500,8 +1508,9 @@ static audio_io_handle_t aps_open_dup_output(void *service, static int aps_close_output(void *service, audio_io_handle_t output) { sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) + if (af == 0) { return PERMISSION_DENIED; + } return af->closeOutput(output); } @@ -1564,8 +1573,9 @@ static audio_io_handle_t aps_open_input_on_module(void *service, static int aps_close_input(void *service, audio_io_handle_t input) { sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) + if (af == 0) { return PERMISSION_DENIED; + } return af->closeInput(input); } @@ -1574,8 +1584,9 @@ static int aps_set_stream_output(void *service, audio_stream_type_t stream, audio_io_handle_t output) { sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) + if (af == 0) { return PERMISSION_DENIED; + } return af->setStreamOutput(stream, output); } @@ -1585,8 +1596,9 @@ static int aps_move_effects(void *service, int session, audio_io_handle_t dst_output) { sp<IAudioFlinger> af = AudioSystem::get_audio_flinger(); - if (af == 0) + if (af == 0) { return PERMISSION_DENIED; + } return af->moveEffects(session, src_output, dst_output); } diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index 2c3c719..323f1a4 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -339,8 +339,9 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction); out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction); Advance(&inputIndex, &phaseFraction, phaseIncrement); - if (outputIndex == outputSampleCount) + if (outputIndex == outputSampleCount) { break; + } } // process input samples @@ -434,8 +435,9 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, out[outputIndex++] += vl * sample; out[outputIndex++] += vr * sample; Advance(&inputIndex, &phaseFraction, phaseIncrement); - if (outputIndex == outputSampleCount) + if (outputIndex == outputSampleCount) { break; + } } // process input samples diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp index 18e59e9..1f9714b 100644 --- a/services/audioflinger/AudioResamplerCubic.cpp +++ b/services/audioflinger/AudioResamplerCubic.cpp @@ -66,8 +66,9 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, if (mBuffer.frameCount == 0) { mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, mPTS); - if (mBuffer.raw == NULL) + if (mBuffer.raw == NULL) { return; + } // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); } int16_t *in = mBuffer.i16; @@ -97,8 +98,9 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, calculateOutputPTS(outputIndex / 2)); - if (mBuffer.raw == NULL) + if (mBuffer.raw == NULL) { goto save_state; // ugly, but efficient + } in = mBuffer.i16; // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); } @@ -132,8 +134,9 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, if (mBuffer.frameCount == 0) { mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, mPTS); - if (mBuffer.raw == NULL) + if (mBuffer.raw == NULL) { return; + } // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); } int16_t *in = mBuffer.i16; @@ -163,8 +166,9 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, mBuffer.frameCount = inFrameCount; provider->getNextBuffer(&mBuffer, calculateOutputPTS(outputIndex / 2)); - if (mBuffer.raw == NULL) + if (mBuffer.raw == NULL) { goto save_state; // ugly, but efficient + } // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); in = mBuffer.i16; } diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp index 86671a9..fc45caa 100644 --- a/services/audioflinger/Effects.cpp +++ b/services/audioflinger/Effects.cpp @@ -116,8 +116,9 @@ status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) continue; } // first non destroyed handle is considered in control - if (controlHandle == NULL) + if (controlHandle == NULL) { controlHandle = h; + } if (h->priority() <= priority) { break; } diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp index f27ea17..7126e92 100644 --- a/services/audioflinger/FastMixer.cpp +++ b/services/audioflinger/FastMixer.cpp @@ -459,8 +459,9 @@ bool FastMixer::threadLoop() } int64_t pts; - if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts))) + if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts))) { pts = AudioBufferProvider::kInvalidPTS; + } // process() is CPU-bound mixer->process(pts); diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h index 0308b99..829b5d6 100644 --- a/services/audioflinger/PlaybackTracks.h +++ b/services/audioflinger/PlaybackTracks.h @@ -33,6 +33,7 @@ public: int sessionId, IAudioFlinger::track_flags_t flags); virtual ~Track(); + virtual status_t initCheck() const; static void appendDumpHeader(String8& result); void dump(char* buffer, size_t size); diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h index ffe3e9f..6c0d1d3 100644 --- a/services/audioflinger/RecordTracks.h +++ b/services/audioflinger/RecordTracks.h @@ -57,5 +57,4 @@ private: // releaseBuffer() not overridden bool mOverflow; // overflow on most recent attempt to fill client buffer - AudioRecordServerProxy* mAudioRecordServerProxy; }; diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index e35f47e..f2f2d6b 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -266,8 +266,8 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio : Thread(false /*canCallJava*/), mType(type), mAudioFlinger(audioFlinger), - // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are - // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() + // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize + // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() mParamStatus(NO_ERROR), mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), @@ -293,6 +293,17 @@ AudioFlinger::ThreadBase::~ThreadBase() } } +status_t AudioFlinger::ThreadBase::readyToRun() +{ + status_t status = initCheck(); + if (status == NO_ERROR) { + ALOGI("AudioFlinger's thread %p ready to run", this); + } else { + ALOGE("No working audio driver found."); + } + return status; +} + void AudioFlinger::ThreadBase::exit() { ALOGV("ThreadBase::exit"); @@ -365,7 +376,13 @@ void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32 void AudioFlinger::ThreadBase::processConfigEvents() { - mLock.lock(); + Mutex::Autolock _l(mLock); + processConfigEvents_l(); +} + +// post condition: mConfigEvents.isEmpty() +void AudioFlinger::ThreadBase::processConfigEvents_l() +{ while (!mConfigEvents.isEmpty()) { ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); ConfigEvent *event = mConfigEvents[0]; @@ -373,32 +390,31 @@ void AudioFlinger::ThreadBase::processConfigEvents() // release mLock before locking AudioFlinger mLock: lock order is always // AudioFlinger then ThreadBase to avoid cross deadlock mLock.unlock(); - switch(event->type()) { - case CFG_EVENT_PRIO: { - PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); - // FIXME Need to understand why this has be done asynchronously - int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), - true /*asynchronous*/); - if (err != 0) { - ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " - "error %d", - prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); - } - } break; - case CFG_EVENT_IO: { - IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); - mAudioFlinger->mLock.lock(); + switch (event->type()) { + case CFG_EVENT_PRIO: { + PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); + // FIXME Need to understand why this has be done asynchronously + int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), + true /*asynchronous*/); + if (err != 0) { + ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", + prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); + } + } break; + case CFG_EVENT_IO: { + IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); + { + Mutex::Autolock _l(mAudioFlinger->mLock); audioConfigChanged_l(ioEvent->event(), ioEvent->param()); - mAudioFlinger->mLock.unlock(); - } break; - default: - ALOGE("processConfigEvents() unknown event type %d", event->type()); - break; + } + } break; + default: + ALOGE("processConfigEvents() unknown event type %d", event->type()); + break; } delete event; mLock.lock(); } - mLock.unlock(); } void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) @@ -423,6 +439,8 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) result.append(buffer); snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); result.append(buffer); + snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); + result.append(buffer); snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); result.append(buffer); snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); @@ -682,8 +700,7 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( int sessionId, effect_descriptor_t *desc, int *enabled, - status_t *status - ) + status_t *status) { sp<EffectModule> effect; sp<EffectHandle> handle; @@ -783,9 +800,7 @@ Exit: handle.clear(); } - if (status != NULL) { - *status = lStatus; - } + *status = lStatus; return handle; } @@ -929,7 +944,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge type_t type) : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), mNormalFrameCount(0), mMixBuffer(NULL), - mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), + mSuspended(0), mBytesWritten(0), // mStreamTypes[] initialized in constructor body mOutput(output), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), @@ -985,7 +1000,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge AudioFlinger::PlaybackThread::~PlaybackThread() { mAudioFlinger->unregisterWriter(mNBLogWriter); - delete [] mAllocMixBuffer; + delete[] mMixBuffer; } void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) @@ -1075,16 +1090,6 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& } // Thread virtuals -status_t AudioFlinger::PlaybackThread::readyToRun() -{ - status_t status = initCheck(); - if (status == NO_ERROR) { - ALOGI("AudioFlinger's thread %p ready to run", this); - } else { - ALOGE("No working audio driver found."); - } - return status; -} void AudioFlinger::PlaybackThread::onFirstRef() { @@ -1250,8 +1255,12 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac track = TimedTrack::create(this, client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId); } - if (track == 0 || track->getCblk() == NULL || track->name() < 0) { - lStatus = NO_MEMORY; + + // new Track always returns non-NULL, + // 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) { + track.clear(); goto Exit; } @@ -1276,9 +1285,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac lStatus = NO_ERROR; Exit: - if (status) { - *status = lStatus; - } + *status = lStatus; return track; } @@ -1563,7 +1570,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() mFormat); } mFrameSize = audio_stream_frame_size(&mOutput->stream->common); - mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; + mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); + mFrameCount = mBufferSize / mFrameSize; if (mFrameCount & 15) { ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", mFrameCount); @@ -1619,11 +1627,11 @@ void AudioFlinger::PlaybackThread::readOutputParameters() ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); - delete[] mAllocMixBuffer; - size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; - mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; - mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); - memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); + delete[] mMixBuffer; + size_t normalBufferSize = mNormalFrameCount * mFrameSize; + // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) + mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; + memset(mMixBuffer, 0, normalBufferSize); // force reconfiguration of effect chains and engines to take new buffer size and audio // parameters into account @@ -1757,7 +1765,7 @@ void AudioFlinger::PlaybackThread::threadLoop_removeTracks( const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); - if (count) { + if (count > 0) { for (size_t i = 0 ; i < count ; i++) { const sp<Track>& track = tracksToRemove.itemAt(i); if (!track->isOutputTrack()) { @@ -2304,7 +2312,7 @@ if (mType == MIXER) { void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); - if (count) { + if (count > 0) { for (size_t i=0 ; i<count ; i++) { const sp<Track>& track = tracksToRemove.itemAt(i); mActiveTracks.remove(track); @@ -2679,7 +2687,7 @@ void AudioFlinger::MixerThread::threadLoop_sleepTime() sleepTime = idleSleepTime; } } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { - memset (mMixBuffer, 0, mixBufferSize); + memset(mMixBuffer, 0, mixBufferSize); sleepTime = 0; ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), "anticipated start"); @@ -2905,7 +2913,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // +1 for rounding and +1 for additional sample needed for interpolation desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; // add frames already consumed but not yet released by the resampler - // because cblk->framesReady() will include these frames + // because mAudioTrackServerProxy->framesReady() will include these frames desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); // the minimum track buffer size is normally twice the number of frames necessary // to fill one buffer and the resampler should not leave more than one buffer worth @@ -3243,6 +3251,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { status = BAD_VALUE; } else { + // no need to save value, since it's constant reconfig = true; } } @@ -3250,6 +3259,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { status = BAD_VALUE; } else { + // no need to save value, since it's constant reconfig = true; } } @@ -4207,7 +4217,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, ) : ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), - // mRsmpInIndex and mBufferSize set by readInputParameters() + // mRsmpInIndex set by readInputParameters() mReqChannelCount(popcount(channelMask)), mReqSampleRate(sampleRate) // mBytesRead is only meaningful while active, and so is cleared in start() @@ -4235,18 +4245,9 @@ void AudioFlinger::RecordThread::onFirstRef() run(mName, PRIORITY_URGENT_AUDIO); } -status_t AudioFlinger::RecordThread::readyToRun() -{ - status_t status = initCheck(); - ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); - return status; -} - bool AudioFlinger::RecordThread::threadLoop() { AudioBufferProvider::Buffer buffer; - sp<RecordTrack> activeTrack; - Vector< sp<EffectChain> > effectChains; nsecs_t lastWarning = 0; @@ -4256,21 +4257,34 @@ bool AudioFlinger::RecordThread::threadLoop() // used to verify we've read at least once before evaluating how many bytes were read bool readOnce = false; + // used to request a deferred sleep, to be executed later while mutex is unlocked + bool doSleep = false; + // start recording - while (!exitPending()) { + for (;;) { + sp<RecordTrack> activeTrack; + TrackBase::track_state activeTrackState; + Vector< sp<EffectChain> > effectChains; - processConfigEvents(); + // sleep with mutex unlocked + if (doSleep) { + doSleep = false; + usleep(kRecordThreadSleepUs); + } { // scope for mLock Mutex::Autolock _l(mLock); - checkForNewParameters_l(); - if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { + if (exitPending()) { + break; + } + processConfigEvents_l(); + // return value 'reconfig' is currently unused + bool reconfig = checkForNewParameters_l(); + // make a stable copy of mActiveTrack + activeTrack = mActiveTrack; + if (activeTrack == 0) { standby(); - - if (exitPending()) { - break; - } - + // exitPending() can't become true here releaseWakeLock_l(); ALOGV("RecordThread: loop stopping"); // go to sleep @@ -4279,172 +4293,191 @@ bool AudioFlinger::RecordThread::threadLoop() acquireWakeLock_l(); continue; } - if (mActiveTrack != 0) { - if (mActiveTrack->isTerminated()) { - removeTrack_l(mActiveTrack); - mActiveTrack.clear(); - } else if (mActiveTrack->mState == TrackBase::PAUSING) { - standby(); + + if (activeTrack->isTerminated()) { + removeTrack_l(activeTrack); + mActiveTrack.clear(); + continue; + } + + activeTrackState = activeTrack->mState; + switch (activeTrackState) { + case TrackBase::PAUSING: + standby(); + mActiveTrack.clear(); + mStartStopCond.broadcast(); + doSleep = true; + continue; + + case TrackBase::RESUMING: + mStandby = false; + if (mReqChannelCount != activeTrack->channelCount()) { mActiveTrack.clear(); mStartStopCond.broadcast(); - } else if (mActiveTrack->mState == TrackBase::RESUMING) { - if (mReqChannelCount != mActiveTrack->channelCount()) { + continue; + } + if (readOnce) { + mStartStopCond.broadcast(); + // record start succeeds only if first read from audio input succeeds + if (mBytesRead < 0) { mActiveTrack.clear(); - mStartStopCond.broadcast(); - } else if (readOnce) { - // record start succeeds only if first read from audio input - // succeeds - if (mBytesRead >= 0) { - mActiveTrack->mState = TrackBase::ACTIVE; - } else { - mActiveTrack.clear(); - } - mStartStopCond.broadcast(); + continue; } - mStandby = false; + activeTrack->mState = TrackBase::ACTIVE; } + break; + + case TrackBase::ACTIVE: + break; + + case TrackBase::IDLE: + doSleep = true; + continue; + + default: + LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); } + lockEffectChains_l(effectChains); } - if (mActiveTrack != 0) { - if (mActiveTrack->mState != TrackBase::ACTIVE && - mActiveTrack->mState != TrackBase::RESUMING) { - unlockEffectChains(effectChains); - usleep(kRecordThreadSleepUs); - continue; - } - for (size_t i = 0; i < effectChains.size(); i ++) { - effectChains[i]->process_l(); - } + // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable + // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING - buffer.frameCount = mFrameCount; - status_t status = mActiveTrack->getNextBuffer(&buffer); - if (status == NO_ERROR) { - readOnce = true; - size_t framesOut = buffer.frameCount; - if (mResampler == NULL) { - // no resampling - while (framesOut) { - size_t framesIn = mFrameCount - mRsmpInIndex; - if (framesIn) { - int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; - int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * - mActiveTrack->mFrameSize; - if (framesIn > framesOut) - framesIn = framesOut; - mRsmpInIndex += framesIn; - framesOut -= framesIn; - if (mChannelCount == mReqChannelCount) { - memcpy(dst, src, framesIn * mFrameSize); - } else { - if (mChannelCount == 1) { - upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, - (int16_t *)src, framesIn); - } else { - downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, - (int16_t *)src, framesIn); - } - } + for (size_t i = 0; i < effectChains.size(); i ++) { + // thread mutex is not locked, but effect chain is locked + effectChains[i]->process_l(); + } + + buffer.frameCount = mFrameCount; + status_t status = activeTrack->getNextBuffer(&buffer); + if (status == NO_ERROR) { + readOnce = true; + size_t framesOut = buffer.frameCount; + if (mResampler == NULL) { + // no resampling + while (framesOut) { + size_t framesIn = mFrameCount - mRsmpInIndex; + if (framesIn > 0) { + int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; + int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * + activeTrack->mFrameSize; + if (framesIn > framesOut) { + framesIn = framesOut; } - if (framesOut && mFrameCount == mRsmpInIndex) { - void *readInto; - if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { - readInto = buffer.raw; - framesOut = 0; + mRsmpInIndex += framesIn; + framesOut -= framesIn; + if (mChannelCount == mReqChannelCount) { + memcpy(dst, src, framesIn * mFrameSize); + } else { + if (mChannelCount == 1) { + upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, + (int16_t *)src, framesIn); } else { - readInto = mRsmpInBuffer; - mRsmpInIndex = 0; + downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, + (int16_t *)src, framesIn); } - mBytesRead = mInput->stream->read(mInput->stream, readInto, - mBufferSize); - if (mBytesRead <= 0) { - if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) - { - ALOGE("Error reading audio input"); - // Force input into standby so that it tries to - // recover at next read attempt - inputStandBy(); - usleep(kRecordThreadSleepUs); - } - mRsmpInIndex = mFrameCount; - framesOut = 0; - buffer.frameCount = 0; + } + } + if (framesOut > 0 && mFrameCount == mRsmpInIndex) { + void *readInto; + if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { + readInto = buffer.raw; + framesOut = 0; + } else { + readInto = mRsmpInBuffer; + mRsmpInIndex = 0; + } + mBytesRead = mInput->stream->read(mInput->stream, readInto, + mBufferSize); + if (mBytesRead <= 0) { + // TODO: verify that it's benign to use a stale track state + if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) + { + ALOGE("Error reading audio input"); + // Force input into standby so that it tries to + // recover at next read attempt + inputStandBy(); + doSleep = true; } + mRsmpInIndex = mFrameCount; + framesOut = 0; + buffer.frameCount = 0; + } #ifdef TEE_SINK - else if (mTeeSink != 0) { - (void) mTeeSink->write(readInto, - mBytesRead >> Format_frameBitShift(mTeeSink->format())); - } -#endif + else if (mTeeSink != 0) { + (void) mTeeSink->write(readInto, + mBytesRead >> Format_frameBitShift(mTeeSink->format())); } +#endif } + } + } else { + // resampling + + // resampler accumulates, but we only have one source track + memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); + // alter output frame count as if we were expecting stereo samples + if (mChannelCount == 1 && mReqChannelCount == 1) { + framesOut >>= 1; + } + mResampler->resample(mRsmpOutBuffer, framesOut, + this /* AudioBufferProvider* */); + // ditherAndClamp() works as long as all buffers returned by + // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. + if (mChannelCount == 2 && mReqChannelCount == 1) { + // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t + ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); + // the resampler always outputs stereo samples: + // do post stereo to mono conversion + downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, + framesOut); } else { - // resampling + ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); + } + // now done with mRsmpOutBuffer - // resampler accumulates, but we only have one source track - memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); - // alter output frame count as if we were expecting stereo samples - if (mChannelCount == 1 && mReqChannelCount == 1) { - framesOut >>= 1; - } - mResampler->resample(mRsmpOutBuffer, framesOut, - this /* AudioBufferProvider* */); - // ditherAndClamp() works as long as all buffers returned by - // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. - if (mChannelCount == 2 && mReqChannelCount == 1) { - // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t - ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); - // the resampler always outputs stereo samples: - // do post stereo to mono conversion - downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, - framesOut); - } else { - ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); + } + if (mFramestoDrop == 0) { + activeTrack->releaseBuffer(&buffer); + } else { + if (mFramestoDrop > 0) { + mFramestoDrop -= buffer.frameCount; + if (mFramestoDrop <= 0) { + clearSyncStartEvent(); } - // now done with mRsmpOutBuffer - - } - if (mFramestoDrop == 0) { - mActiveTrack->releaseBuffer(&buffer); } else { - if (mFramestoDrop > 0) { - mFramestoDrop -= buffer.frameCount; - if (mFramestoDrop <= 0) { - clearSyncStartEvent(); - } - } else { - mFramestoDrop += buffer.frameCount; - if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || - mSyncStartEvent->isCancelled()) { - ALOGW("Synced record %s, session %d, trigger session %d", - (mFramestoDrop >= 0) ? "timed out" : "cancelled", - mActiveTrack->sessionId(), - (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); - clearSyncStartEvent(); - } + mFramestoDrop += buffer.frameCount; + if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || + mSyncStartEvent->isCancelled()) { + ALOGW("Synced record %s, session %d, trigger session %d", + (mFramestoDrop >= 0) ? "timed out" : "cancelled", + activeTrack->sessionId(), + (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); + clearSyncStartEvent(); } } - mActiveTrack->clearOverflow(); } - // client isn't retrieving buffers fast enough - else { - if (!mActiveTrack->setOverflow()) { - nsecs_t now = systemTime(); - if ((now - lastWarning) > kWarningThrottleNs) { - ALOGW("RecordThread: buffer overflow"); - lastWarning = now; - } + activeTrack->clearOverflow(); + } + // client isn't retrieving buffers fast enough + else { + if (!activeTrack->setOverflow()) { + nsecs_t now = systemTime(); + if ((now - lastWarning) > kWarningThrottleNs) { + ALOGW("RecordThread: buffer overflow"); + lastWarning = now; } - // Release the processor for a while before asking for a new buffer. - // This will give the application more chance to read from the buffer and - // clear the overflow. - usleep(kRecordThreadSleepUs); } + // Release the processor for a while before asking for a new buffer. + // This will give the application more chance to read from the buffer and + // clear the overflow. + doSleep = true; } + // enable changes in effect chain unlockEffectChains(effectChains); - effectChains.clear(); + // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end } standby(); @@ -4474,7 +4507,7 @@ void AudioFlinger::RecordThread::inputStandBy() mInput->stream->common.standby(&mInput->stream->common); } -sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( +sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, audio_format_t format, @@ -4553,8 +4586,9 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR track = new RecordTrack(this, client, sampleRate, format, channelMask, frameCount, sessionId); - if (track->getCblk() == 0) { - lStatus = NO_MEMORY; + lStatus = track->initCheck(); + if (lStatus != NO_ERROR) { + track.clear(); goto Exit; } mTracks.add(track); @@ -4575,9 +4609,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR lStatus = NO_ERROR; Exit: - if (status) { - *status = lStatus; - } + *status = lStatus; return track; } @@ -4608,6 +4640,7 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac } { + // This section is a rendezvous between binder thread executing start() and RecordThread AutoMutex lock(mLock); if (mActiveTrack != 0) { if (recordTrack != mActiveTrack.get()) { @@ -4618,11 +4651,13 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac return status; } + // FIXME why? already set in constructor, 'STARTING_1' would be more accurate recordTrack->mState = TrackBase::IDLE; mActiveTrack = recordTrack; mLock.unlock(); status_t status = AudioSystem::startInput(mId); mLock.lock(); + // FIXME should verify that mActiveTrack is still == recordTrack if (status != NO_ERROR) { mActiveTrack.clear(); clearSyncStartEvent(); @@ -4633,6 +4668,8 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac if (mResampler != NULL) { mResampler->reset(); } + // FIXME hijacking a playback track state name which was intended for start after pause; + // here 'STARTING_2' would be more accurate mActiveTrack->mState = TrackBase::RESUMING; // signal thread to start ALOGV("Signal record thread"); @@ -4643,6 +4680,7 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac status = INVALID_OPERATION; goto startError; } + // FIXME incorrect usage of wait: no explicit predicate or loop mStartStopCond.wait(mLock); if (mActiveTrack == 0) { ALOGV("Record failed to start"); @@ -4693,11 +4731,13 @@ bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { return false; } + // note that threadLoop may still be processing the track at this point [without lock] recordTrack->mState = TrackBase::PAUSING; // do not wait for mStartStopCond if exiting if (exitPending()) { return true; } + // FIXME incorrect usage of wait: no explicit predicate or loop mStartStopCond.wait(mLock); // if we have been restarted, recordTrack == mActiveTrack.get() here if (exitPending() || recordTrack != mActiveTrack.get()) { @@ -4833,6 +4873,7 @@ status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* // Force input into standby so that it tries to // recover at next read attempt inputStandBy(); + // FIXME an awkward place to sleep, consider using doSleep when this is pulled up usleep(kRecordThreadSleepUs); } buffer->raw = NULL; @@ -4875,7 +4916,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() int value; audio_format_t reqFormat = mFormat; uint32_t reqSamplingRate = mReqSampleRate; - uint32_t reqChannelCount = mReqChannelCount; + audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { reqSamplingRate = value; @@ -4890,8 +4931,13 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() } } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { - reqChannelCount = popcount(value); - reconfig = true; + audio_channel_mask_t mask = (audio_channel_mask_t) value; + if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { + status = BAD_VALUE; + } else { + reqChannelMask = mask; + reconfig = true; + } } if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { // do not accept frame count changes if tracks are open as the track buffer @@ -4940,6 +4986,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() } mAudioSource = (audio_source_t)value; } + if (status == NO_ERROR) { status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); @@ -4956,7 +5003,8 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() <= (2 * reqSamplingRate)) && popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && - (reqChannelCount <= FCC_2)) { + (reqChannelMask == AUDIO_CHANNEL_IN_MONO || + reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { status = NO_ERROR; } if (status == NO_ERROR) { @@ -5033,8 +5081,7 @@ void AudioFlinger::RecordThread::readInputParameters() mFrameCount = mBufferSize / mFrameSize; mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; - if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) - { + if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { int channelCount; // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid // stereo to mono post process as the resampler always outputs stereo. diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index 3fe470c..7999436 100644 --- a/services/audioflinger/Threads.h +++ b/services/audioflinger/Threads.h @@ -36,6 +36,8 @@ public: audio_devices_t outDevice, audio_devices_t inDevice, type_t type); virtual ~ThreadBase(); + virtual status_t readyToRun(); + void dumpBase(int fd, const Vector<String16>& args); void dumpEffectChains(int fd, const Vector<String16>& args); @@ -141,6 +143,7 @@ public: void sendIoConfigEvent_l(int event, int param = 0); void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); void processConfigEvents(); + void processConfigEvents_l(); // see note at declaration of mStandby, mOutDevice and mInDevice bool standby() const { return mStandby; } @@ -156,7 +159,7 @@ public: int sessionId, effect_descriptor_t *desc, int *enabled, - status_t *status); + status_t *status /*non-NULL*/); void disconnectEffect(const sp< EffectModule>& effect, EffectHandle *handle, bool unpinIfLast); @@ -270,6 +273,7 @@ protected: uint32_t mChannelCount; size_t mFrameSize; audio_format_t mFormat; + size_t mBufferSize; // HAL buffer size for read() or write() // Parameter sequence by client: binder thread calling setParameters(): // 1. Lock mLock @@ -353,7 +357,6 @@ public: void dump(int fd, const Vector<String16>& args); // Thread virtuals - virtual status_t readyToRun(); virtual bool threadLoop(); // RefBase @@ -419,7 +422,7 @@ public: int sessionId, IAudioFlinger::track_flags_t *flags, pid_t tid, - status_t *status); + status_t *status /*non-NULL*/); AudioStreamOut* getOutput() const; AudioStreamOut* clearOutput(); @@ -471,7 +474,6 @@ protected: size_t mNormalFrameCount; // normal mixer and effects int16_t* mMixBuffer; // frame size aligned mix buffer - int8_t* mAllocMixBuffer; // mixer buffer allocation address // suspend count, > 0 means suspended. While suspended, the thread continues to pull from // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle @@ -854,12 +856,12 @@ public: // Thread virtuals virtual bool threadLoop(); - virtual status_t readyToRun(); // RefBase virtual void onFirstRef(); virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } + sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, @@ -869,7 +871,7 @@ public: int sessionId, IAudioFlinger::track_flags_t *flags, pid_t tid, - status_t *status); + status_t *status /*non-NULL*/); status_t start(RecordTrack* recordTrack, AudioSystem::sync_event_t event, @@ -912,13 +914,13 @@ public: bool hasFastRecorder() const { return false; } private: - void clearSyncStartEvent(); + void clearSyncStartEvent(); // Enter standby if not already in standby, and set mStandby flag - void standby(); + void standby(); // Call the HAL standby method unconditionally, and don't change mStandby flag - void inputStandBy(); + void inputStandBy(); AudioStreamIn *mInput; SortedVector < sp<RecordTrack> > mTracks; @@ -933,7 +935,6 @@ private: int32_t *mRsmpOutBuffer; int16_t *mRsmpInBuffer; // [mFrameCount * mChannelCount] size_t mRsmpInIndex; - size_t mBufferSize; // stream buffer size for read() const uint32_t mReqChannelCount; const uint32_t mReqSampleRate; ssize_t mBytesRead; diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h index 523e4b2..00a91b7 100644 --- a/services/audioflinger/TrackBase.h +++ b/services/audioflinger/TrackBase.h @@ -47,6 +47,7 @@ public: int sessionId, bool isOut); virtual ~TrackBase(); + virtual status_t initCheck() const { return getCblk() != 0 ? NO_ERROR : NO_MEMORY; } virtual status_t start(AudioSystem::sync_event_t event, int triggerSession) = 0; @@ -76,15 +77,6 @@ protected: virtual uint32_t sampleRate() const { return mSampleRate; } - // Return a pointer to the start of a contiguous slice of the track buffer. - // Parameter 'offset' is the requested start position, expressed in - // monotonically increasing frame units relative to the track epoch. - // Parameter 'frames' is the requested length, also in frame units. - // Always returns non-NULL. It is the caller's responsibility to - // verify that this will be successful; the result of calling this - // function with invalid 'offset' or 'frames' is undefined. - void* getBuffer(uint32_t offset, uint32_t frames) const; - bool isStopped() const { return (mState == STOPPED || mState == FLUSHED); } diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index 6002aa3..7f58300 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -366,6 +366,15 @@ AudioFlinger::PlaybackThread::Track::~Track() ALOGV("PlaybackThread::Track destructor"); } +status_t AudioFlinger::PlaybackThread::Track::initCheck() const +{ + status_t status = TrackBase::initCheck(); + if (status == NO_ERROR && mName < 0) { + status = NO_MEMORY; + } + return status; +} + void AudioFlinger::PlaybackThread::Track::destroy() { // NOTE: destroyTrack_l() can remove a strong reference to this Track @@ -985,15 +994,17 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, "AudioFlingerTimed"); - if (mTimedMemoryDealer == NULL) + if (mTimedMemoryDealer == NULL) { return NO_MEMORY; + } } sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); if (newBuffer == NULL) { newBuffer = mTimedMemoryDealer->allocate(size); - if (newBuffer == NULL) + if (newBuffer == NULL) { return NO_MEMORY; + } } *buffer = newBuffer; @@ -1702,9 +1713,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( { ALOGV("RecordTrack constructor"); if (mCblk != NULL) { - mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, - mFrameSize); - mServerProxy = mAudioRecordServerProxy; + mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); } } |