commit 062e67a26e0553dd142be622821f493df541f0c6
author Phil Burk <philburk@google.com> Wed Feb 11 13:40:50 2015 -0800
committer Phil Burk <philburk@google.com> Tue Mar 24 13:24:18 2015 -0700
tree 125d28264adfc5b7bd993bb343569eea63bfb95d
parent 21b51b61ee52e6aa74d98b138d3dd4f0e17b1441

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>

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 @@
         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 @@
     } 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 @@
 }
 
 
-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 @@
         }
         // 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 @@
     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 @@
 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 @@
         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 @@
     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 @@
         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::flushHw_l()
 {
-    if (mOutput->stream->flush != NULL) {
-        mOutput->stream->flush(mOutput->stream);
-    }
+    mOutput->flush();
     mHwPaused = false;
 }
 
@@ -4868,7 +4864,7 @@
                     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);