Support (E)AC3 passthrough

media/base/audio_parameters.h

 // Copyright (c) 2012 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.
 
 #ifndef MEDIA_BASE_AUDIO_PARAMETERS_H_
 #define MEDIA_BASE_AUDIO_PARAMETERS_H_
 
 #include <stdint.h>
 #include <string>
 
@@ skipping 23 matching lines @@
               "Audio buffer parameters struct alignment not same as AudioBus");
 struct MEDIA_EXPORT ALIGNAS(PARAMETERS_ALIGNMENT) AudioInputBufferParameters {
   double volume;
   uint32_t size;
   uint32_t hardware_delay_bytes;
   uint32_t id;
   bool key_pressed;
 };
 struct MEDIA_EXPORT ALIGNAS(PARAMETERS_ALIGNMENT) AudioOutputBufferParameters {
   uint32_t frames_skipped;
+  uint32_t frames;
+  uint32_t data_size;
 };
 #undef PARAMETERS_ALIGNMENT
 #if defined(OS_WIN)
 #pragma warning(pop)
 #endif
 
 static_assert(sizeof(AudioInputBufferParameters) %
                       AudioBus::kChannelAlignment ==
                   0,
               "AudioInputBufferParameters not aligned");
@@ skipping 11 matching lines @@
   int8_t audio[1];
 };
 
 class MEDIA_EXPORT AudioParameters {
  public:
   // TODO(miu): Rename this enum to something that correctly reflects its
   // semantics, such as "TransportScheme."
   enum Format {
     AUDIO_PCM_LINEAR = 0,            // PCM is 'raw' amplitude samples.
     AUDIO_PCM_LOW_LATENCY,           // Linear PCM, low latency requested.
+    AUDIO_RAW_AC3,                   // Raw compressed AC3 bitstream.
+    AUDIO_RAW_EAC3,                  // Raw compressed E-AC3 bitstream.
     AUDIO_FAKE,                      // Creates a fake AudioOutputStream object.
     AUDIO_FORMAT_LAST = AUDIO_FAKE,  // Only used for validation of format.
   };
 
   enum {
     // Telephone quality sample rate, mostly for speech-only audio.
     kTelephoneSampleRate = 8000,
     // CD sampling rate is 44.1 KHz or conveniently 2x2x3x3x5x5x7x7.
     kAudioCDSampleRate = 44100,
   };
@@ skipping 47 matching lines @@
   // can lead to audio/video sync drift.
   double GetMicrosecondsPerFrame() const;
 
   // Returns the duration of this buffer as calculated from frames_per_buffer()
   // and sample_rate().
   base::TimeDelta GetBufferDuration() const;
 
   // Comparison with other AudioParams.
   bool Equals(const AudioParameters& other) const;
 
+  // Return true if |format_| is raw compressed audio.
+  bool IsRawFormat() const {

[DaleCurtis, 2016/11/01 23:05:13]

Inline method must be hacker_style().

[AndyWu, 2016/11/04 18:04:24]

Done.

+    return format_ == AUDIO_RAW_AC3 || format_ == AUDIO_RAW_EAC3;
+  }
+
   void set_format(Format format) { format_ = format; }
   Format format() const { return format_; }
 
   // A setter for channel_layout_ is intentionally excluded.
   ChannelLayout channel_layout() const { return channel_layout_; }
 
   // The number of channels is usually computed from channel_layout_. Setting
   // this explictly is only required with CHANNEL_LAYOUT_DISCRETE.
   void set_channels_for_discrete(int channels) {
     DCHECK(channel_layout_ == CHANNEL_LAYOUT_DISCRETE ||
@@ skipping 71 matching lines @@
   if (a.sample_rate() != b.sample_rate())
     return a.sample_rate() < b.sample_rate();
   if (a.bits_per_sample() != b.bits_per_sample())
     return a.bits_per_sample() < b.bits_per_sample();
   return a.frames_per_buffer() < b.frames_per_buffer();
 }
 
 }  // namespace media
 
 #endif  // MEDIA_BASE_AUDIO_PARAMETERS_H_