From 062e67a26e0553dd142be622821f493df541f0c6 Mon Sep 17 00:00:00 2001 From: Phil Burk Date: Wed, 11 Feb 2015 13:40:50 -0800 Subject: AudioFlinger: call SPDIF wrapper from AudioFlinger Create an interface layer between the AudioFlinger and the HAL that manages the wrapping and format conversion. Removed unnecessary includes. Handle rate conversion in getRenderPosition(). Try to open HAL with encoded format before wrapping with SPDIF. Bug: 17566660 Change-Id: I00ad888ca15ff0f85b85efb8167c7f5ea761a244 Signed-off-by: Phil Burk --- services/audioflinger/Threads.cpp | 30 +++++++++++++----------------- 1 file changed, 13 insertions(+), 17 deletions(-) (limited to 'services/audioflinger/Threads.cpp') diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index d0b825c..54e0043 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -2009,7 +2009,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l() LOG_FATAL("HAL format %#x not supported for mixed output", mFormat); } - mFrameSize = audio_stream_out_frame_size(mOutput->stream); + mFrameSize = mOutput->getFrameSize(); mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); mFrameCount = mBufferSize / mFrameSize; if (mFrameCount & 15) { @@ -2160,7 +2160,7 @@ status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, ui } else { status_t status; uint32_t frames; - status = mOutput->stream->get_render_position(mOutput->stream, &frames); + status = mOutput->getRenderPosition(&frames); *dspFrames = (size_t)frames; return status; } @@ -2202,13 +2202,13 @@ uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) } -AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const +AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const { Mutex::Autolock _l(mLock); return mOutput; } -AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() +AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() { Mutex::Autolock _l(mLock); AudioStreamOut *output = mOutput; @@ -2354,8 +2354,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write() } // FIXME We should have an implementation of timestamps for direct output threads. // They are used e.g for multichannel PCM playback over HDMI. - bytesWritten = mOutput->stream->write(mOutput->stream, - (char *)mSinkBuffer + offset, mBytesRemaining); + bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); if (mUseAsyncWrite && ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { // do not wait for async callback in case of error of full write @@ -2908,8 +2907,7 @@ status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) if ((mType == OFFLOAD || mType == DIRECT) && mOutput != NULL && mOutput->stream->get_presentation_position) { uint64_t position64; - int ret = mOutput->stream->get_presentation_position( - mOutput->stream, &position64, ×tamp.mTime); + int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); if (ret == 0) { timestamp.mPosition = (uint32_t)position64; return NO_ERROR; @@ -3289,7 +3287,7 @@ bool AudioFlinger::PlaybackThread::waitingAsyncCallback() void AudioFlinger::PlaybackThread::threadLoop_standby() { ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); - mOutput->stream->common.standby(&mOutput->stream->common); + mOutput->standby(); if (mUseAsyncWrite != 0) { // discard any pending drain or write ack by incrementing sequence mWriteAckSequence = (mWriteAckSequence + 2) & ~1; @@ -4058,7 +4056,7 @@ bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePa status = mOutput->stream->common.set_parameters(&mOutput->stream->common, keyValuePair.string()); if (!mStandby && status == INVALID_OPERATION) { - mOutput->stream->common.standby(&mOutput->stream->common); + mOutput->standby(); mStandby = true; mBytesWritten = 0; status = mOutput->stream->common.set_parameters(&mOutput->stream->common, @@ -4400,8 +4398,8 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix() while (frameCount) { AudioBufferProvider::Buffer buffer; buffer.frameCount = frameCount; - mActiveTrack->getNextBuffer(&buffer); - if (buffer.raw == NULL) { + status_t status = mActiveTrack->getNextBuffer(&buffer); + if (status != NO_ERROR || buffer.raw == NULL) { memset(curBuf, 0, frameCount * mFrameSize); break; } @@ -4513,7 +4511,7 @@ bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& key status = mOutput->stream->common.set_parameters(&mOutput->stream->common, keyValuePair.string()); if (!mStandby && status == INVALID_OPERATION) { - mOutput->stream->common.standby(&mOutput->stream->common); + mOutput->standby(); mStandby = true; mBytesWritten = 0; status = mOutput->stream->common.set_parameters(&mOutput->stream->common, @@ -4576,9 +4574,7 @@ void AudioFlinger::DirectOutputThread::cacheParameters_l() void AudioFlinger::DirectOutputThread::flushHw_l() { - if (mOutput->stream->flush != NULL) { - mOutput->stream->flush(mOutput->stream); - } + mOutput->flush(); mHwPaused = false; } @@ -4868,7 +4864,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTr size_t audioHALFrames = (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; size_t framesWritten = - mBytesWritten / audio_stream_out_frame_size(mOutput->stream); + mBytesWritten / mOutput->getFrameSize(); track->presentationComplete(framesWritten, audioHALFrames); track->reset(); tracksToRemove->add(track); -- cgit v1.1