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 <philburk@google.com>

1 files changed, 13 insertions, 17 deletions

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, &timestamp.mTime);
+        int ret = mOutput->getPresentationPosition(&position64, &timestamp.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);