Add client side support for more AUDIO_FORMAT_PCM_*

As much as possible, the checks for specific formats are done on server
side.  The only exception is 8-bit, which needs a client side workaround.

Change-Id: I52fe086c039460c8dac009de03b67eb54c56a836

diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 2c48bbf..79db323 100644 (file)
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -154,8 +154,9 @@ public:
      * streamType:         Select the type of audio stream this track is attached to
      *                     (e.g. AUDIO_STREAM_MUSIC).
      * sampleRate:         Data source sampling rate in Hz.
-     * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
-     *                     16 bits per sample).
+     * format:             Audio format.  For mixed tracks, any PCM format supported by server is OK
+     *                     or AUDIO_FORMAT_PCM_8_BIT which is handled on client side.  For direct
+     *                     and offloaded tracks, the possible format(s) depends on the output sink.
      * channelMask:        Channel mask, such that audio_is_output_channel(channelMask) is true.
      * frameCount:         Minimum size of track PCM buffer in frames. This defines the
      *                     application's contribution to the

diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index dc4f90e..aaaa3f1 100644 (file)
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -315,12 +315,20 @@ status_t AudioTrack::set(
         flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
     }
 
-    if (audio_is_linear_pcm(format)) {
-        mFrameSize = channelCount * audio_bytes_per_sample(format);
-        mFrameSizeAF = channelCount * sizeof(int16_t);
+    if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
+        if (audio_is_linear_pcm(format)) {
+            mFrameSize = channelCount * audio_bytes_per_sample(format);
+        } else {
+            mFrameSize = sizeof(uint8_t);
+        }
+        mFrameSizeAF = mFrameSize;
     } else {
-        mFrameSize = sizeof(uint8_t);
-        mFrameSizeAF = sizeof(uint8_t);
+        ALOG_ASSERT(audio_is_linear_pcm(format));
+        mFrameSize = channelCount * audio_bytes_per_sample(format);
+        mFrameSizeAF = channelCount * audio_bytes_per_sample(
+                format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format);
+        // createTrack will return an error if PCM format is not supported by server,
+        // so no need to check for specific PCM formats here
     }
 
     // Make copy of input parameter offloadInfo so that in the future:
@@ -931,7 +939,11 @@ status_t AudioTrack::createTrack_l(size_t epoch)
 
         // Ensure that buffer alignment matches channel count
         // 8-bit data in shared memory is not currently supported by AudioFlinger
-        size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
+        size_t alignment = audio_bytes_per_sample(
+                mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat);
+        if (alignment & 1) {
+            alignment = 1;
+        }
         if (mChannelCount > 1) {
             // More than 2 channels does not require stronger alignment than stereo
             alignment <<= 1;
@@ -947,7 +959,7 @@ status_t AudioTrack::createTrack_l(size_t epoch)
         // there's no frameCount parameter.
         // But when initializing a shared buffer AudioTrack via set(),
         // there _is_ a frameCount parameter.  We silently ignore it.
-        frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t);
+        frameCount = mSharedBuffer->size() / mFrameSizeAF;
 
     } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {