Replace AudioDecoderConfig with simple accessors on AudioDecoder.

AudioDecoderConfig was being used as a struct for outputting format information to an AudioRenderer. This was confusing and in contrast to VideoDecoderConfig, which is is actually used as input to configure a video decoder!

Review URL: http://codereview.chromium.org/7796033

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@101563 0039d316-1c4b-4281-b951-d872f2087c98

17 files changed, 73 insertions, 103 deletions

diff --git a/media/audio/audio_parameters.cc b/media/audio/audio_parameters.cc
index 029f831..45c98d9 100644
--- a/media/audio/audio_parameters.cc
+++ b/media/audio/audio_parameters.cc
@@ -15,15 +15,6 @@ AudioParameters::AudioParameters()
       channels(0) {
 }
 
-AudioParameters::AudioParameters(const media::AudioDecoderConfig& config)
-    : format(AUDIO_PCM_LINEAR),
-      channel_layout(config.channel_layout),
-      sample_rate(config.sample_rate),
-      bits_per_sample(config.bits_per_channel),
-      samples_per_packet(0),
-      channels(ChannelLayoutToChannelCount(config.channel_layout)) {
-}
-
 AudioParameters::AudioParameters(Format format, ChannelLayout channel_layout,
                                  int sample_rate, int bits_per_sample,
                                  int samples_per_packet)
diff --git a/media/audio/audio_parameters.h b/media/audio/audio_parameters.h
index 5c86990..0de5f6a 100644
--- a/media/audio/audio_parameters.h
+++ b/media/audio/audio_parameters.h
@@ -6,7 +6,7 @@
 #define MEDIA_AUDIO_AUDIO_PARAMETERS_H_
 
 #include "base/basictypes.h"
-#include "media/base/audio_decoder_config.h"
+#include "media/base/channel_layout.h"
 #include "media/base/media_export.h"
 
 struct MEDIA_EXPORT AudioParameters {
@@ -31,9 +31,6 @@ struct MEDIA_EXPORT AudioParameters {
   static const uint32 kAudioDATSampleRate = 48000;
 
   AudioParameters();
-
-  explicit AudioParameters(const media::AudioDecoderConfig& config);
-
   AudioParameters(Format format, ChannelLayout channel_layout, int sample_rate,
                   int bits_per_sample, int samples_per_packet);
 
diff --git a/media/audio/audio_parameters_unittest.cc b/media/audio/audio_parameters_unittest.cc
index afcf6ff..da3e746 100644
--- a/media/audio/audio_parameters_unittest.cc
+++ b/media/audio/audio_parameters_unittest.cc
@@ -7,8 +7,6 @@
 #include "media/audio/audio_parameters.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
-using media::AudioDecoderConfig;
-
 TEST(AudioParameters, Constructor_Default) {
   AudioParameters::Format expected_format = AudioParameters::AUDIO_PCM_LINEAR;
   int expected_bits = 0;
@@ -27,26 +25,6 @@ TEST(AudioParameters, Constructor_Default) {
   EXPECT_EQ(expected_samples, params.samples_per_packet);
 }
 
-TEST(AudioParameters, Constructor_AudioDecoderConfig) {
-  AudioParameters::Format expected_format = AudioParameters::AUDIO_PCM_LINEAR;
-  int expected_bits = 8;
-  int expected_channels = 2;
-  ChannelLayout expected_channel_layout = CHANNEL_LAYOUT_STEREO;
-  int expected_rate = 44000;
-  int expected_samples = 0;
-
-  AudioDecoderConfig config(expected_bits, expected_channel_layout,
-                            expected_rate);
-  AudioParameters params(config);
-
-  EXPECT_EQ(expected_format, params.format);
-  EXPECT_EQ(expected_bits, params.bits_per_sample);
-  EXPECT_EQ(expected_channels, params.channels);
-  EXPECT_EQ(expected_channel_layout, params.channel_layout);
-  EXPECT_EQ(expected_rate, params.sample_rate);
-  EXPECT_EQ(expected_samples, params.samples_per_packet);
-}
-
 TEST(AudioParameters, Constructor_ParameterValues) {
   AudioParameters::Format expected_format =
       AudioParameters::AUDIO_PCM_LOW_LATENCY;
diff --git a/media/base/audio_decoder_config.h b/media/base/audio_decoder_config.h
deleted file mode 100644
index e2c848d..0000000
--- a/media/base/audio_decoder_config.h
+++ /dev/null
@@ -1,26 +0,0 @@
-// Copyright (c) 2011 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_DECODER_CONFIG_H_
-#define MEDIA_BASE_AUDIO_DECODER_CONFIG_H_
-
-#include "media/base/channel_layout.h"
-
-namespace media {
-
-struct AudioDecoderConfig {
-  AudioDecoderConfig(int bits, ChannelLayout layout, int rate)
-      : bits_per_channel(bits),
-        channel_layout(layout),
-        sample_rate(rate) {
-  }
-
-  int bits_per_channel;
-  ChannelLayout channel_layout;
-  int sample_rate;
-};
-
-}  // namespace media
-
-#endif  // MEDIA_BASE_AUDIO_DECODER_CONFIG_H_
diff --git a/media/base/filters.h b/media/base/filters.h
index 180739d..faafa4e 100644
--- a/media/base/filters.h
+++ b/media/base/filters.h
@@ -31,7 +31,7 @@
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/time.h"
-#include "media/base/audio_decoder_config.h"
+#include "media/base/channel_layout.h"
 #include "media/base/media_export.h"
 #include "media/base/pipeline_status.h"
 #include "media/base/video_frame.h"
@@ -215,8 +215,6 @@ class MEDIA_EXPORT AudioDecoder : public Filter {
   virtual void Initialize(DemuxerStream* stream, FilterCallback* callback,
                           StatisticsCallback* stats_callback) = 0;
 
-  virtual AudioDecoderConfig config() = 0;
-
   // Renderer provides an output buffer for Decoder to write to. These buffers
   // will be recycled to renderer via the permanent callback.
   //
@@ -230,6 +228,11 @@ class MEDIA_EXPORT AudioDecoder : public Filter {
     consume_audio_samples_callback_ = callback;
   }
 
+  // Returns various information about the decoded audio format.
+  virtual int bits_per_channel() = 0;
+  virtual ChannelLayout channel_layout() = 0;
+  virtual int sample_rate() = 0;
+
  protected:
   AudioDecoder();
   virtual ~AudioDecoder();
diff --git a/media/base/mock_filters.h b/media/base/mock_filters.h
index e215186..8abcbd3 100644
--- a/media/base/mock_filters.h
+++ b/media/base/mock_filters.h
@@ -212,8 +212,10 @@ class MockAudioDecoder : public AudioDecoder {
   MOCK_METHOD3(Initialize, void(DemuxerStream* stream,
                                 FilterCallback* callback,
                                 StatisticsCallback* stats_callback));
-  MOCK_METHOD0(config, AudioDecoderConfig());
   MOCK_METHOD1(ProduceAudioSamples, void(scoped_refptr<Buffer>));
+  MOCK_METHOD0(bits_per_channel, int(void));
+  MOCK_METHOD0(channel_layout, ChannelLayout(void));
+  MOCK_METHOD0(sample_rate, int(void));
 
   void ConsumeAudioSamplesForTest(scoped_refptr<Buffer> buffer) {
     AudioDecoder::ConsumeAudioSamples(buffer);
diff --git a/media/filters/audio_renderer_base.cc b/media/filters/audio_renderer_base.cc
index 2d6205e..f64f13f 100644
--- a/media/filters/audio_renderer_base.cc
+++ b/media/filters/audio_renderer_base.cc
@@ -104,15 +104,14 @@ void AudioRendererBase::Initialize(AudioDecoder* decoder,
 
   // Initialize our algorithm with media properties, initial playback rate,
   // and a callback to request more reads from the data source.
-  AudioDecoderConfig config = decoder_->config();
-  algorithm_->Initialize(ChannelLayoutToChannelCount(config.channel_layout),
-                         config.sample_rate,
-                         config.bits_per_channel,
-                         0.0f,
-                         cb);
+  ChannelLayout channel_layout = decoder_->channel_layout();
+  int channels = ChannelLayoutToChannelCount(channel_layout);
+  int bits_per_channel = decoder_->bits_per_channel();
+  int sample_rate = decoder_->sample_rate();
+  algorithm_->Initialize(channels, sample_rate, bits_per_channel, 0.0f, cb);
 
   // Give the subclass an opportunity to initialize itself.
-  if (!OnInitialize(config)) {
+  if (!OnInitialize(bits_per_channel, channel_layout, sample_rate)) {
     host()->SetError(PIPELINE_ERROR_INITIALIZATION_FAILED);
     callback->Run();
     return;
diff --git a/media/filters/audio_renderer_base.h b/media/filters/audio_renderer_base.h
index 5305c7b..cf73924 100644
--- a/media/filters/audio_renderer_base.h
+++ b/media/filters/audio_renderer_base.h
@@ -46,7 +46,9 @@ class MEDIA_EXPORT AudioRendererBase : public AudioRenderer {
  protected:
   // Subclasses should return true if they were able to initialize, false
   // otherwise.
-  virtual bool OnInitialize(const AudioDecoderConfig& config) = 0;
+  virtual bool OnInitialize(int bits_per_channel,
+                            ChannelLayout channel_layout,
+                            int sample_rate) = 0;
 
   // Called by Stop().  Subclasses should perform any necessary cleanup during
   // this time, such as stopping any running threads.
diff --git a/media/filters/audio_renderer_base_unittest.cc b/media/filters/audio_renderer_base_unittest.cc
index b0be7ae..87faab3 100644
--- a/media/filters/audio_renderer_base_unittest.cc
+++ b/media/filters/audio_renderer_base_unittest.cc
@@ -31,7 +31,7 @@ class MockAudioRendererBase : public AudioRendererBase {
   MOCK_METHOD1(SetVolume, void(float volume));
 
   // AudioRendererBase implementation.
-  MOCK_METHOD1(OnInitialize, bool(const AudioDecoderConfig& config));
+  MOCK_METHOD3(OnInitialize, bool(int, ChannelLayout, int));
   MOCK_METHOD0(OnStop, void());
 
   // Used for verifying check points during tests.
@@ -57,9 +57,12 @@ class AudioRendererBaseTest : public ::testing::Test {
         .WillRepeatedly(Invoke(this, &AudioRendererBaseTest::EnqueueCallback));
 
     // Set up audio properties.
-    ON_CALL(*decoder_, config())
-        .WillByDefault(Return(AudioDecoderConfig(16, CHANNEL_LAYOUT_MONO,
-                                                 44100)));
+    ON_CALL(*decoder_, bits_per_channel())
+        .WillByDefault(Return(16));
+    ON_CALL(*decoder_, channel_layout())
+        .WillByDefault(Return(CHANNEL_LAYOUT_MONO));
+    ON_CALL(*decoder_, sample_rate())
+        .WillByDefault(Return(44100));
   }
 
   virtual ~AudioRendererBaseTest() {
@@ -93,7 +96,7 @@ TEST_F(AudioRendererBaseTest, Initialize_Failed) {
   InSequence s;
 
   // Our subclass will fail when asked to initialize.
-  EXPECT_CALL(*renderer_, OnInitialize(_))
+  EXPECT_CALL(*renderer_, OnInitialize(_, _, _))
       .WillOnce(Return(false));
 
   // We expect to receive an error.
@@ -108,7 +111,7 @@ TEST_F(AudioRendererBaseTest, Initialize_Successful) {
   InSequence s;
 
   // Then our subclass will be asked to initialize.
-  EXPECT_CALL(*renderer_, OnInitialize(_))
+  EXPECT_CALL(*renderer_, OnInitialize(_, _, _))
       .WillOnce(Return(true));
 
   // Initialize, we shouldn't have any reads.
@@ -136,7 +139,7 @@ TEST_F(AudioRendererBaseTest, OneCompleteReadCycle) {
   InSequence s;
 
   // Then our subclass will be asked to initialize.
-  EXPECT_CALL(*renderer_, OnInitialize(_))
+  EXPECT_CALL(*renderer_, OnInitialize(_, _, _))
       .WillOnce(Return(true));
 
   // Initialize, we shouldn't have any reads.
diff --git a/media/filters/audio_renderer_impl.cc b/media/filters/audio_renderer_impl.cc
index ae91993..84fb645 100644
--- a/media/filters/audio_renderer_impl.cc
+++ b/media/filters/audio_renderer_impl.cc
@@ -72,9 +72,11 @@ void AudioRendererImpl::OnError(AudioOutputStream* stream, int code) {
   NOTIMPLEMENTED();
 }
 
-bool AudioRendererImpl::OnInitialize(const AudioDecoderConfig& config) {
-  AudioParameters params(config);
-  params.samples_per_packet = kSamplesPerBuffer;
+bool AudioRendererImpl::OnInitialize(int bits_per_channel,
+                                     ChannelLayout channel_layout,
+                                     int sample_rate) {
+  AudioParameters params(AudioParameters::AUDIO_PCM_LINEAR, channel_layout,
+                         sample_rate, bits_per_channel, kSamplesPerBuffer);
 
   bytes_per_second_ = params.GetBytesPerSecond();
 
diff --git a/media/filters/audio_renderer_impl.h b/media/filters/audio_renderer_impl.h
index f52913d..238f59c 100644
--- a/media/filters/audio_renderer_impl.h
+++ b/media/filters/audio_renderer_impl.h
@@ -45,7 +45,9 @@ class MEDIA_EXPORT AudioRendererImpl
 
  protected:
   // AudioRendererBase implementation.
-  virtual bool OnInitialize(const AudioDecoderConfig& config);
+  virtual bool OnInitialize(int bits_per_channel,
+                            ChannelLayout channel_layout,
+                            int sample_rate);
   virtual void OnStop();
 
  private:
diff --git a/media/filters/ffmpeg_audio_decoder.cc b/media/filters/ffmpeg_audio_decoder.cc
index 22c5685..50335d3 100644
--- a/media/filters/ffmpeg_audio_decoder.cc
+++ b/media/filters/ffmpeg_audio_decoder.cc
@@ -29,7 +29,9 @@ const size_t FFmpegAudioDecoder::kOutputBufferSize =
 FFmpegAudioDecoder::FFmpegAudioDecoder(MessageLoop* message_loop)
     : DecoderBase<AudioDecoder, Buffer>(message_loop),
       codec_context_(NULL),
-      config_(0, CHANNEL_LAYOUT_NONE, 0),
+      bits_per_channel_(0),
+      channel_layout_(CHANNEL_LAYOUT_NONE),
+      sample_rate_(0),
       estimated_next_timestamp_(kNoTimestamp) {
 }
 
@@ -70,12 +72,11 @@ void FFmpegAudioDecoder::DoInitialize(DemuxerStream* demuxer_stream,
     return;
   }
 
-  config_.bits_per_channel =
-      av_get_bits_per_sample_fmt(codec_context_->sample_fmt);
-  config_.channel_layout =
+  bits_per_channel_ = av_get_bits_per_sample_fmt(codec_context_->sample_fmt);
+  channel_layout_ =
       ChannelLayoutToChromeChannelLayout(codec_context_->channel_layout,
                                          codec_context_->channels);
-  config_.sample_rate = codec_context_->sample_rate;
+  sample_rate_ = codec_context_->sample_rate;
 
   // Prepare the output buffer.
   output_buffer_.reset(static_cast<uint8*>(av_malloc(kOutputBufferSize)));
@@ -86,14 +87,22 @@ void FFmpegAudioDecoder::DoInitialize(DemuxerStream* demuxer_stream,
   *success = true;
 }
 
-AudioDecoderConfig FFmpegAudioDecoder::config() {
-  return config_;
-}
-
 void FFmpegAudioDecoder::ProduceAudioSamples(scoped_refptr<Buffer> output) {
   DecoderBase<AudioDecoder, Buffer>::PostReadTaskHack(output);
 }
 
+int FFmpegAudioDecoder::bits_per_channel() {
+  return bits_per_channel_;
+}
+
+ChannelLayout FFmpegAudioDecoder::channel_layout() {
+  return channel_layout_;
+}
+
+int FFmpegAudioDecoder::sample_rate() {
+  return sample_rate_;
+}
+
 void FFmpegAudioDecoder::DoSeek(base::TimeDelta time, Task* done_cb) {
   avcodec_flush_buffers(codec_context_);
   estimated_next_timestamp_ = kNoTimestamp;
diff --git a/media/filters/ffmpeg_audio_decoder.h b/media/filters/ffmpeg_audio_decoder.h
index 1025083..e92a851 100644
--- a/media/filters/ffmpeg_audio_decoder.h
+++ b/media/filters/ffmpeg_audio_decoder.h
@@ -21,8 +21,10 @@ class MEDIA_EXPORT FFmpegAudioDecoder
   virtual ~FFmpegAudioDecoder();
 
   // AudioDecoder implementation.
-  virtual AudioDecoderConfig config();
   virtual void ProduceAudioSamples(scoped_refptr<Buffer> output);
+  virtual int bits_per_channel();
+  virtual ChannelLayout channel_layout();
+  virtual int sample_rate();
 
  protected:
   virtual void DoInitialize(DemuxerStream* demuxer_stream, bool* success,
@@ -41,7 +43,10 @@ class MEDIA_EXPORT FFmpegAudioDecoder
   // is initialized in OnInitialize().
   AVCodecContext* codec_context_;
 
-  AudioDecoderConfig config_;
+  // Decoded audio format.
+  int bits_per_channel_;
+  ChannelLayout channel_layout_;
+  int sample_rate_;
 
   // Estimated timestamp for next packet. Useful for packets without timestamps.
   base::TimeDelta estimated_next_timestamp_;
diff --git a/media/filters/ffmpeg_audio_decoder_unittest.cc b/media/filters/ffmpeg_audio_decoder_unittest.cc
index 885ef5a..9edc85a 100644
--- a/media/filters/ffmpeg_audio_decoder_unittest.cc
+++ b/media/filters/ffmpeg_audio_decoder_unittest.cc
@@ -141,9 +141,9 @@ class FFmpegAudioDecoderTest : public testing::Test {
 TEST_F(FFmpegAudioDecoderTest, Initialize) {
   Initialize();
 
-  EXPECT_EQ(16, decoder_->config().bits_per_channel);
-  EXPECT_EQ(CHANNEL_LAYOUT_STEREO, decoder_->config().channel_layout);
-  EXPECT_EQ(44100, decoder_->config().sample_rate);
+  EXPECT_EQ(16, decoder_->bits_per_channel());
+  EXPECT_EQ(CHANNEL_LAYOUT_STEREO, decoder_->channel_layout());
+  EXPECT_EQ(44100, decoder_->sample_rate());
 
   Stop();
 }
diff --git a/media/filters/null_audio_renderer.cc b/media/filters/null_audio_renderer.cc
index 21b3e07..0d5daf5 100644
--- a/media/filters/null_audio_renderer.cc
+++ b/media/filters/null_audio_renderer.cc
@@ -60,11 +60,13 @@ void NullAudioRenderer::ThreadMain() {
   }
 }
 
-bool NullAudioRenderer::OnInitialize(const AudioDecoderConfig& config) {
+bool NullAudioRenderer::OnInitialize(int bits_per_channel,
+                                     ChannelLayout channel_layout,
+                                     int sample_rate) {
   // Calculate our bytes per millisecond value and allocate our buffer.
   bytes_per_millisecond_ =
-      (ChannelLayoutToChannelCount(config.channel_layout) * config.sample_rate *
-       config.bits_per_channel / 8) / base::Time::kMillisecondsPerSecond;
+      (ChannelLayoutToChannelCount(channel_layout) * sample_rate *
+       bits_per_channel / 8) / base::Time::kMillisecondsPerSecond;
   buffer_size_ = bytes_per_millisecond_ * kBufferSizeInMilliseconds;
   buffer_.reset(new uint8[buffer_size_]);
   DCHECK(buffer_.get());
diff --git a/media/filters/null_audio_renderer.h b/media/filters/null_audio_renderer.h
index f330c77..2f9d062 100644
--- a/media/filters/null_audio_renderer.h
+++ b/media/filters/null_audio_renderer.h
@@ -38,7 +38,9 @@ class MEDIA_EXPORT NullAudioRenderer
 
  protected:
   // AudioRendererBase implementation.
-  virtual bool OnInitialize(const AudioDecoderConfig& config);
+  virtual bool OnInitialize(int bits_per_channel,
+                            ChannelLayout channel_layout,
+                            int sample_rate);
   virtual void OnStop();
 
  private:
diff --git a/media/media.gyp b/media/media.gyp
index 7af199e..55133ec6 100644
--- a/media/media.gyp
+++ b/media/media.gyp
@@ -84,7 +84,6 @@
         'audio/win/waveout_output_win.h',
         'base/async_filter_factory_base.cc',
         'base/async_filter_factory_base.h',
-        'base/audio_decoder_config.h',
         'base/bitstream_buffer.h',
         'base/buffers.cc',
         'base/buffers.h',