Use passthrough decoder for (E)AC3 formats

media/base/audio_buffer.cc

Index: media/base/audio_buffer.cc
diff --git a/media/base/audio_buffer.cc b/media/base/audio_buffer.cc
index 8b029930dec53cd304b834365d4c3562f0a8240c..cf997a94fb6b1d24befbe192ecff6d56c7be3296 100644
--- a/media/base/audio_buffer.cc
+++ b/media/base/audio_buffer.cc
@@ -26,6 +26,7 @@ AudioBuffer::AudioBuffer(SampleFormat sample_format,
                          int frame_count,
                          bool create_buffer,
                          const uint8_t* const* data,
+                         size_t data_size,
                          const base::TimeDelta timestamp)
     : sample_format_(sample_format),
       channel_layout_(channel_layout),
@@ -37,7 +38,7 @@ AudioBuffer::AudioBuffer(SampleFormat sample_format,
       duration_(end_of_stream_
                     ? base::TimeDelta()
                     : CalculateDuration(adjusted_frame_count_, sample_rate_)),
-      data_size_(0) {
+      data_size_(data_size) {
   CHECK_GE(channel_count_, 0);
   CHECK_LE(channel_count_, limits::kMaxChannels);
   CHECK_GE(frame_count, 0);
@@ -79,7 +80,8 @@ AudioBuffer::AudioBuffer(SampleFormat sample_format,
   DCHECK(IsInterleaved(sample_format)) << sample_format_;
   // Allocate our own buffer and copy the supplied data into it. Buffer must
   // contain the data for all channels.
-  data_size_ = data_size_per_channel * channel_count_;
+  if (!IsBitstream(sample_format))

[chcunningham, 2016/12/16 22:03:14]

I feel like the interleaved code path is the only one that makes sense for the
compressed bitstream. I'm not sure what sample formats are available in AC3, but
lets say the format is planar instead of interleaved. I expect that the attempt
to copy specific channel data based on standard PCM planar layout will go
horribly wrong for encoded AC3? If thats right, we want to re-work this to
always skip to  interleaved setup, even if the layout is actually planar once
decoded. 

[AndyWu, 2017/05/02 23:41:03]

I do make (E)AC3 as interleaved formats:
https://cs.chromium.org/chromium/src/media/base/sample_format.cc?l=91


On 2016/12/16 22:03:14, chcunningham wrote:

+    data_size_ = data_size_per_channel * channel_count_;
   data_.reset(
       static_cast<uint8_t*>(base::AlignedAlloc(data_size_, kChannelAlignment)));
   channel_data_.reserve(1);
@@ -102,14 +104,27 @@ scoped_refptr<AudioBuffer> AudioBuffer::CopyFrom(
   // If you hit this CHECK you likely have a bug in a demuxer. Go fix it.
   CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
   CHECK(data[0]);
-  return make_scoped_refptr(new AudioBuffer(sample_format,
-                                            channel_layout,
-                                            channel_count,
-                                            sample_rate,
-                                            frame_count,
-                                            true,
-                                            data,
-                                            timestamp));
+  return make_scoped_refptr(
+      new AudioBuffer(sample_format, channel_layout, channel_count, sample_rate,
+                      frame_count, true, data, 0, timestamp));
+}
+
+// static
+scoped_refptr<AudioBuffer> AudioBuffer::CopyBitstreamFrom(
+    SampleFormat sample_format,
+    ChannelLayout channel_layout,
+    int channel_count,
+    int sample_rate,
+    int frame_count,
+    const uint8_t* const* data,
+    const size_t data_size,
+    const base::TimeDelta timestamp) {
+  // If you hit this CHECK you likely have a bug in a demuxer. Go fix it.
+  CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
+  CHECK(data[0]);

[chcunningham, 2016/12/16 22:03:14]

Are we confident that data[0] would never return 0 for a bitstream format? Never
any leading padding? For all bistream formats that we might encounter now/later?

[AndyWu, 2017/05/02 23:41:03]

data[0] is a pointer; null pointer is unexpected.

+  return make_scoped_refptr(
+      new AudioBuffer(sample_format, channel_layout, channel_count, sample_rate,
+                      frame_count, true, data, data_size, timestamp));
 }
 
 // static
@@ -122,7 +137,21 @@ scoped_refptr<AudioBuffer> AudioBuffer::CreateBuffer(
   CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
   return make_scoped_refptr(
       new AudioBuffer(sample_format, channel_layout, channel_count, sample_rate,
-                      frame_count, true, NULL, kNoTimestamp));
+                      frame_count, true, NULL, 0, kNoTimestamp));
+}
+
+// static
+scoped_refptr<AudioBuffer> AudioBuffer::CreateBitstreamBuffer(
+    SampleFormat sample_format,
+    ChannelLayout channel_layout,
+    int channel_count,
+    int sample_rate,
+    int frame_count,
+    size_t data_size) {
+  CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
+  return make_scoped_refptr(
+      new AudioBuffer(sample_format, channel_layout, channel_count, sample_rate,
+                      frame_count, true, NULL, data_size, kNoTimestamp));
 }
 
 // static
@@ -134,21 +163,16 @@ scoped_refptr<AudioBuffer> AudioBuffer::CreateEmptyBuffer(
     const base::TimeDelta timestamp) {
   CHECK_GT(frame_count, 0);  // Otherwise looks like an EOF buffer.
   // Since data == NULL, format doesn't matter.
-  return make_scoped_refptr(new AudioBuffer(kSampleFormatF32,
-                                            channel_layout,
-                                            channel_count,
-                                            sample_rate,
-                                            frame_count,
-                                            false,
-                                            NULL,
-                                            timestamp));
+  return make_scoped_refptr(
+      new AudioBuffer(kSampleFormatF32, channel_layout, channel_count,
+                      sample_rate, frame_count, false, NULL, 0, timestamp));
 }
 
 // static
 scoped_refptr<AudioBuffer> AudioBuffer::CreateEOSBuffer() {
   return make_scoped_refptr(new AudioBuffer(kUnknownSampleFormat,
                                             CHANNEL_LAYOUT_NONE, 0, 0, 0, false,
-                                            NULL, kNoTimestamp));
+                                            NULL, 0, kNoTimestamp));
 }
 
 // Convert int16_t values in the range [INT16_MIN, INT16_MAX] to [-1.0, 1.0].
@@ -175,6 +199,14 @@ void AudioBuffer::ReadFrames(int frames_to_copy,
   DCHECK(!end_of_stream());
   DCHECK_EQ(dest->channels(), channel_count_);
   DCHECK_LE(source_frame_offset + frames_to_copy, adjusted_frame_count_);
+
+  const bool is_bitstream_format = IsBitstream(sample_format_);
+
+  if (is_bitstream_format) {
+    // TODO(tsunghung): Implement it along with AudioBus changes.
+    NOTREACHED() << "Invalid sample format!";
+  }
+
   DCHECK_LE(dest_frame_offset + frames_to_copy, dest->frames());
 
   if (!data_) {
@@ -240,6 +272,7 @@ void AudioBuffer::ReadFrames(int frames_to_copy,
 void AudioBuffer::TrimStart(int frames_to_trim) {
   CHECK_GE(frames_to_trim, 0);
   CHECK_LE(frames_to_trim, adjusted_frame_count_);
+  DCHECK(!IsBitstream(sample_format_));
 
   TrimRange(0, frames_to_trim);
 }
@@ -247,6 +280,7 @@ void AudioBuffer::TrimStart(int frames_to_trim) {
 void AudioBuffer::TrimEnd(int frames_to_trim) {
   CHECK_GE(frames_to_trim, 0);
   CHECK_LE(frames_to_trim, adjusted_frame_count_);
+  DCHECK(!IsBitstream(sample_format_));
 
   // Adjust the number of frames and duration for this buffer.
   adjusted_frame_count_ -= frames_to_trim;
@@ -256,6 +290,7 @@ void AudioBuffer::TrimEnd(int frames_to_trim) {
 void AudioBuffer::TrimRange(int start, int end) {
   CHECK_GE(start, 0);
   CHECK_LE(end, adjusted_frame_count_);
+  DCHECK(!IsBitstream(sample_format_));
 
   const int frames_to_trim = end - start;
   CHECK_GE(frames_to_trim, 0);