Use passthrough decoder for (E)AC3 formats

media/base/audio_buffer.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/base/audio_buffer.h"
 
 #include <cmath>
 
 #include "base/logging.h"
 #include "media/base/audio_bus.h"
 #include "media/base/limits.h"
 #include "media/base/timestamp_constants.h"
 
 namespace media {
 
 static base::TimeDelta CalculateDuration(int frames, double sample_rate) {
   DCHECK_GT(sample_rate, 0);
   return base::TimeDelta::FromMicroseconds(
       frames * base::Time::kMicrosecondsPerSecond / sample_rate);
 }
 
 AudioBuffer::AudioBuffer(SampleFormat sample_format,
                          ChannelLayout channel_layout,
                          int channel_count,
                          int sample_rate,
                          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),
       channel_count_(channel_count),
       sample_rate_(sample_rate),
       adjusted_frame_count_(frame_count),
       end_of_stream_(!create_buffer && data == NULL && frame_count == 0),
       timestamp_(timestamp),
       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);
   DCHECK(channel_layout == CHANNEL_LAYOUT_DISCRETE ||
          ChannelLayoutToChannelCount(channel_layout) == channel_count);
 
   int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format);
   DCHECK_LE(bytes_per_channel, kChannelAlignment);
 
   // Empty buffer?
@@ skipping 21 matching lines @@
       if (data)
         memcpy(channel_data_[i], data[i], data_size_per_channel);
     }
     return;
   }
 
   // Remaining formats are interleaved data.
   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);
   channel_data_.push_back(data_.get());
   if (data)
     memcpy(data_.get(), data[0], data_size_);
 }
 
 AudioBuffer::~AudioBuffer() {}
 
 // static
 scoped_refptr<AudioBuffer> AudioBuffer::CopyFrom(
     SampleFormat sample_format,
     ChannelLayout channel_layout,
     int channel_count,
     int sample_rate,
     int frame_count,
     const uint8_t* const* data,
     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]);
-  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
 scoped_refptr<AudioBuffer> AudioBuffer::CreateBuffer(
     SampleFormat sample_format,
     ChannelLayout channel_layout,
     int channel_count,
     int sample_rate,
     int frame_count) {
   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
 scoped_refptr<AudioBuffer> AudioBuffer::CreateEmptyBuffer(
     ChannelLayout channel_layout,
     int channel_count,
     int sample_rate,
     int frame_count,
     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].
 inline float ConvertSample(int16_t value) {
   return value * (value < 0 ? -1.0f / std::numeric_limits<int16_t>::min()
                             : 1.0f / std::numeric_limits<int16_t>::max());
 }
 
 void AudioBuffer::AdjustSampleRate(int sample_rate) {
   DCHECK(!end_of_stream_);
   sample_rate_ = sample_rate;
   duration_ = CalculateDuration(adjusted_frame_count_, sample_rate_);
 }
 
 void AudioBuffer::ReadFrames(int frames_to_copy,
                              int source_frame_offset,
                              int dest_frame_offset,
                              AudioBus* dest) {
   // Deinterleave each channel (if necessary) and convert to 32bit
   // floating-point with nominal range -1.0 -> +1.0 (if necessary).
 
   // |dest| must have the same number of channels, and the number of frames
   // specified must be in range.
   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_) {
     // Special case for an empty buffer.
     dest->ZeroFramesPartial(dest_frame_offset, frames_to_copy);
     return;
   }
 
   if (sample_format_ == kSampleFormatPlanarF32) {
     // Format is planar float32. Copy the data from each channel as a block.
@@ skipping 45 matching lines @@
   int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format_);
   int frame_size = channel_count_ * bytes_per_channel;
   const uint8_t* source_data = data_.get() + source_frame_offset * frame_size;
   dest->FromInterleavedPartial(
       source_data, dest_frame_offset, frames_to_copy, bytes_per_channel);
 }
 
 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);
 }
 
 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;
   duration_ = CalculateDuration(adjusted_frame_count_, sample_rate_);
 }
 
 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);
   CHECK_LE(frames_to_trim, adjusted_frame_count_);
 
   const int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format_);
   const int frames_to_copy = adjusted_frame_count_ - end;
   if (frames_to_copy > 0) {
     switch (sample_format_) {
       case kSampleFormatPlanarS16:
@@ skipping 25 matching lines @@
     }
   } else {
     CHECK_EQ(frames_to_copy, 0);
   }
 
   // Trim the leftover data off the end of the buffer and update duration.
   TrimEnd(frames_to_trim);
 }
 
 }  // namespace media