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 d921a8a39d7d5752174bb3f08edc4f770ce71a6a..93a06aaaaa047f750cb379029491faa1a6cad9eb 100644
--- a/media/base/audio_buffer.cc
+++ b/media/base/audio_buffer.cc
@@ -49,6 +49,7 @@ AudioBuffer::AudioBuffer(SampleFormat sample_format,
                          int frame_count,
                          bool create_buffer,
                          const uint8_t* const* data,
+                         const size_t data_size,
                          const base::TimeDelta timestamp,
                          scoped_refptr<AudioBufferMemoryPool> pool)
     : sample_format_(sample_format),
@@ -61,7 +62,7 @@ AudioBuffer::AudioBuffer(SampleFormat sample_format,
       duration_(end_of_stream_
                     ? base::TimeDelta()
                     : CalculateDuration(adjusted_frame_count_, sample_rate_)),
-      data_size_(0),
+      data_size_(data_size),
       pool_(std::move(pool)) {
   CHECK_GE(channel_count_, 0);
   CHECK_LE(channel_count_, limits::kMaxChannels);
@@ -108,7 +109,8 @@ 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))
+    data_size_ = data_size_per_channel * channel_count_;

[chcunningham, 2017/05/05 01:14:12]

else DCHECK(data_size_ > 0)?

[AndyWu, 2017/05/06 01:46:48]

Done.

 
   if (pool_) {
     data_ = pool_->CreateBuffer(data_size_);
@@ -143,7 +145,26 @@ scoped_refptr<AudioBuffer> AudioBuffer::CopyFrom(
   CHECK(data[0]);
   return make_scoped_refptr(
       new AudioBuffer(sample_format, channel_layout, channel_count, sample_rate,
-                      frame_count, true, data, timestamp, std::move(pool)));
+                      frame_count, true, data, 0, timestamp, std::move(pool)));
+}
+
+// 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,
+    scoped_refptr<AudioBufferMemoryPool> pool) {
+  // 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, data_size, timestamp, std::move(pool)));
 }
 
 // static
@@ -157,7 +178,22 @@ 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, nullptr, kNoTimestamp, std::move(pool)));
+      true, nullptr, 0, kNoTimestamp, std::move(pool)));
+}
+
+// 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,
+    scoped_refptr<AudioBufferMemoryPool> pool) {
+  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, nullptr, data_size, kNoTimestamp, std::move(pool)));
 }
 
 // static
@@ -171,14 +207,14 @@ scoped_refptr<AudioBuffer> AudioBuffer::CreateEmptyBuffer(
   // Since data == nullptr, format doesn't matter.
   return make_scoped_refptr(new AudioBuffer(
       kSampleFormatF32, channel_layout, channel_count, sample_rate, frame_count,
-      false, nullptr, timestamp, nullptr));
+      false, nullptr, 0, timestamp, nullptr));
 }
 
 // static
 scoped_refptr<AudioBuffer> AudioBuffer::CreateEOSBuffer() {
   return make_scoped_refptr(new AudioBuffer(kUnknownSampleFormat,
                                             CHANNEL_LAYOUT_NONE, 0, 0, 0, false,
-                                            nullptr, kNoTimestamp, nullptr));
+                                            nullptr, 0, kNoTimestamp, nullptr));
 }
 
 // Convert int16_t values in the range [INT16_MIN, INT16_MAX] to [-1.0, 1.0].
@@ -205,6 +241,21 @@ 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_);
+
+  bool is_compressed_format = IsBitstream(sample_format_);
+  dest->set_is_compressed_format(is_compressed_format);

[chcunningham, 2017/05/05 01:14:11]

is_compressed_format is slightly different name from IsBitstream. Can you use
one or the other everywhere? 

[AndyWu, 2017/05/06 01:46:48]

Sorry, this change should not be included in this CL. Will implement it along
with AudioBus changes.

+
+  if (is_compressed_format) {
+    DCHECK(!source_frame_offset);
+    uint8_t* dest_data =
+        reinterpret_cast<uint8_t*>(dest->channel(0)) + dest->data_size();
+
+    memcpy(dest_data, channel_data_[0], data_size());
+    dest->set_data_size(dest_frame_offset + data_size());
+    dest->set_frames(dest->frames() + frame_count());
+    return;
+  }
+
   DCHECK_LE(dest_frame_offset + frames_to_copy, dest->frames());
 
   if (!data_) {
@@ -269,6 +320,7 @@ 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_));

[chcunningham, 2017/05/05 01:14:11]

These methods are called in frame_processor as part of append window trimming.
Remind me, what is the plan for that API? It obviously won't work, but AFIACT
people aren't prevented from trying to use it (and doing so will just mangle the
data). Perhaps we should turn this DCHECK into a LOG(ERROR) and make it a no-op?

[AndyWu, 2017/05/06 01:46:48]

Done. However, we really should prevent trimming calls on bitstream format.

[chcunningham, 2017/05/08 20:01:02]

They current MediaSource API doesn't really afford us a way to disable that
feature (other than internally making it a no-op). 

Perhaps (E)AC3 should be registered among the other MSE codecs and in its
registry documentation we can include a disclaimer that appendWindowStart/End
may not be supported. 

None of this blocks landing.

 
   TrimRange(0, frames_to_trim);
 }
@@ -276,6 +328,7 @@ 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;
@@ -285,6 +338,7 @@ 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);