Replace AudioDecoder::media_format() with AudioDecoderConfig.

Also includes some cleanup for AudioParameter users including a new helper method GetBytesPerSecond().

BUG=28206
TEST=media_unittests, test_shell_tests, etc...

Review URL: http://codereview.chromium.org/6903007

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@83098 0039d316-1c4b-4281-b951-d872f2087c98

16 files changed, 136 insertions, 95 deletions

diff --git a/media/audio/audio_parameters.cc b/media/audio/audio_parameters.cc
index d1d44a7..dfe2d07 100644
--- a/media/audio/audio_parameters.cc
+++ b/media/audio/audio_parameters.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -14,6 +14,14 @@ AudioParameters::AudioParameters()
       samples_per_packet(0) {
 }
 
+AudioParameters::AudioParameters(const media::AudioDecoderConfig& config)
+    : format(AUDIO_PCM_LINEAR),
+      channels(config.channels_per_sample),
+      sample_rate(config.sample_rate),
+      bits_per_sample(config.bits_per_channel),
+      samples_per_packet(0) {
+}
+
 AudioParameters::AudioParameters(Format format, int channels,
                                  int sample_rate, int bits_per_sample,
                                  int samples_per_packet)
@@ -38,6 +46,10 @@ int AudioParameters::GetPacketSize() const {
   return samples_per_packet * channels * bits_per_sample / 8;
 }
 
+int AudioParameters::GetBytesPerSecond() const {
+  return sample_rate * channels * bits_per_sample / 8;
+}
+
 bool AudioParameters::Compare::operator()(
     const AudioParameters& a,
     const AudioParameters& b) const {
diff --git a/media/audio/audio_parameters.h b/media/audio/audio_parameters.h
index 2842bc1..ac08067 100644
--- a/media/audio/audio_parameters.h
+++ b/media/audio/audio_parameters.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -6,6 +6,7 @@
 #define MEDIA_AUDIO_AUDIO_PARAMETERS_H_
 
 #include "base/basictypes.h"
+#include "media/base/audio_decoder_config.h"
 
 struct AudioParameters {
   // Compare is useful when AudioParameters is used as a key in std::map.
@@ -30,6 +31,8 @@ struct AudioParameters {
 
   AudioParameters();
 
+  explicit AudioParameters(const media::AudioDecoderConfig& config);
+
   AudioParameters(Format format, int channels, int sample_rate,
                   int bits_per_sample, int samples_per_packet);
 
@@ -40,6 +43,9 @@ struct AudioParameters {
   // Returns size of audio packets in bytes.
   int GetPacketSize() const;
 
+  // Returns the number of bytes representing one second of audio.
+  int GetBytesPerSecond() const;
+
   Format format;           // Format of the stream.
   int channels;            // Number of channels.
   int sample_rate;         // Sampling frequency/rate.
diff --git a/media/audio/audio_parameters_unittest.cc b/media/audio/audio_parameters_unittest.cc
index 61b8ad0..9f5bb2b 100644
--- a/media/audio/audio_parameters_unittest.cc
+++ b/media/audio/audio_parameters_unittest.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -7,6 +7,41 @@
 #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;
+  int expected_channels = 0;
+  int expected_rate = 0;
+  int expected_samples = 0;
+
+  AudioParameters params;
+
+  EXPECT_EQ(expected_format, params.format);
+  EXPECT_EQ(expected_bits, params.bits_per_sample);
+  EXPECT_EQ(expected_channels, params.channels);
+  EXPECT_EQ(expected_rate, params.sample_rate);
+  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;
+  int expected_rate = 44000;
+  int expected_samples = 0;
+
+  AudioDecoderConfig config(expected_bits, expected_channels, 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_rate, params.sample_rate);
+  EXPECT_EQ(expected_samples, params.samples_per_packet);
+}
+
 TEST(AudioParameters, GetPacketSize) {
   EXPECT_EQ(100, AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
                                  1, 1000,  8, 100).GetPacketSize());
@@ -20,6 +55,19 @@ TEST(AudioParameters, GetPacketSize) {
                                  2, 1000,  16, 200).GetPacketSize());
 }
 
+TEST(AudioParameters, GetBytesPerSecond) {
+  EXPECT_EQ(0, AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
+                               0, 0, 0, 0).GetBytesPerSecond());
+  EXPECT_EQ(0, AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
+                               2, 0, 0, 0).GetBytesPerSecond());
+  EXPECT_EQ(0, AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
+                               0, 100, 0, 0).GetBytesPerSecond());
+  EXPECT_EQ(0, AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
+                               0, 0, 8, 0).GetBytesPerSecond());
+  EXPECT_EQ(200, AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
+                                 2, 100, 8, 0).GetBytesPerSecond());
+}
+
 TEST(AudioParameters, Compare) {
   AudioParameters values[] = {
     AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, 1, 1000,  8, 100),
diff --git a/media/base/audio_decoder_config.h b/media/base/audio_decoder_config.h
new file mode 100644
index 0000000..791d2cd
--- /dev/null
+++ b/media/base/audio_decoder_config.h
@@ -0,0 +1,24 @@
+// 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_
+
+namespace media {
+
+struct AudioDecoderConfig {
+  AudioDecoderConfig(int bits, int channels, int rate)
+      : bits_per_channel(bits),
+        channels_per_sample(channels),
+        sample_rate(rate) {
+  }
+
+  int bits_per_channel;
+  int channels_per_sample;
+  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 7d56114..f290fec 100644
--- a/media/base/filters.h
+++ b/media/base/filters.h
@@ -30,6 +30,7 @@
 #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/media_format.h"
 #include "media/base/video_frame.h"
 
@@ -238,6 +239,8 @@ class AudioDecoder : public Filter {
   virtual void Initialize(DemuxerStream* stream, FilterCallback* callback,
                           StatisticsCallback* stats_callback) = 0;
 
+  virtual AudioDecoderConfig config() = 0;
+
   // |set_fill_buffer_done_callback| install permanent callback from downstream
   // filter (i.e. Renderer). The callback is used to deliver buffers at
   // runtime to downstream filter.
@@ -255,9 +258,6 @@ class AudioDecoder : public Filter {
   // We could also pass empty pointer here to let decoder provide buffers pool.
   virtual void ProduceAudioSamples(scoped_refptr<Buffer> buffer) = 0;
 
-  // Returns the media format produced by this decoder.
-  virtual const MediaFormat& media_format() = 0;
-
  protected:
   AudioDecoder();
   ~AudioDecoder();
diff --git a/media/base/mock_filters.h b/media/base/mock_filters.h
index 636b60f..7a58095 100644
--- a/media/base/mock_filters.h
+++ b/media/base/mock_filters.h
@@ -214,7 +214,7 @@ class MockAudioDecoder : public AudioDecoder {
   MOCK_METHOD3(Initialize, void(DemuxerStream* stream,
                                 FilterCallback* callback,
                                 StatisticsCallback* stats_callback));
-  MOCK_METHOD0(media_format, const MediaFormat&());
+  MOCK_METHOD0(config, AudioDecoderConfig());
   MOCK_METHOD1(ProduceAudioSamples, void(scoped_refptr<Buffer>));
 
   // change to public to allow unittest for access;
diff --git a/media/filters/audio_renderer_base.cc b/media/filters/audio_renderer_base.cc
index 4dec0a4..7e40ba1 100644
--- a/media/filters/audio_renderer_base.cc
+++ b/media/filters/audio_renderer_base.cc
@@ -90,16 +90,6 @@ void AudioRendererBase::Initialize(AudioDecoder* decoder,
 
   decoder_->set_consume_audio_samples_callback(
       NewCallback(this, &AudioRendererBase::ConsumeAudioSamples));
-  // Get the media properties to initialize our algorithms.
-  int channels = 0;
-  int sample_rate = 0;
-  int sample_bits = 0;
-  if (!ParseMediaFormat(decoder_->media_format(), &channels, &sample_rate,
-                        &sample_bits)) {
-    host()->SetError(PIPELINE_ERROR_INITIALIZATION_FAILED);
-    callback->Run();
-    return;
-  }
 
   // Create a callback so our algorithm can request more reads.
   AudioRendererAlgorithmBase::RequestReadCallback* cb =
@@ -110,14 +100,15 @@ 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.
-  algorithm_->Initialize(channels,
-                         sample_rate,
-                         sample_bits,
+  AudioDecoderConfig config = decoder_->config();
+  algorithm_->Initialize(config.channels_per_sample,
+                         config.sample_rate,
+                         config.bits_per_channel,
                          0.0f,
                          cb);
 
   // Give the subclass an opportunity to initialize itself.
-  if (!OnInitialize(decoder_->media_format())) {
+  if (!OnInitialize(config)) {
     host()->SetError(PIPELINE_ERROR_INITIALIZATION_FAILED);
     callback->Run();
     return;
@@ -253,17 +244,6 @@ void AudioRendererBase::ScheduleRead_Locked() {
   decoder_->ProduceAudioSamples(buffer);
 }
 
-// static
-bool AudioRendererBase::ParseMediaFormat(const MediaFormat& media_format,
-                                         int* channels_out,
-                                         int* sample_rate_out,
-                                         int* sample_bits_out) {
-  // TODO(scherkus): might be handy to support NULL parameters.
-  return media_format.GetAsInteger(MediaFormat::kChannels, channels_out) &&
-      media_format.GetAsInteger(MediaFormat::kSampleRate, sample_rate_out) &&
-      media_format.GetAsInteger(MediaFormat::kSampleBits, sample_bits_out);
-}
-
 void AudioRendererBase::SetPlaybackRate(float playback_rate) {
   algorithm_->set_playback_rate(playback_rate);
 }
diff --git a/media/filters/audio_renderer_base.h b/media/filters/audio_renderer_base.h
index 6c77acb..0b13e02a 100644
--- a/media/filters/audio_renderer_base.h
+++ b/media/filters/audio_renderer_base.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -44,10 +44,9 @@ class AudioRendererBase : public AudioRenderer {
   virtual bool HasEnded();
 
  protected:
-  // Called by Initialize().  |media_format| is the format of the AudioDecoder.
   // Subclasses should return true if they were able to initialize, false
   // otherwise.
-  virtual bool OnInitialize(const MediaFormat& media_format) = 0;
+  virtual bool OnInitialize(const AudioDecoderConfig& config) = 0;
 
   // Called by Stop().  Subclasses should perform any necessary cleanup during
   // this time, such as stopping any running threads.
@@ -79,12 +78,6 @@ class AudioRendererBase : public AudioRenderer {
                     const base::TimeDelta& playback_delay,
                     bool buffers_empty);
 
-  // Helper to parse a media format and return whether we were successful
-  // retrieving all the information we care about.
-  static bool ParseMediaFormat(const MediaFormat& media_format,
-                               int* channels_out, int* sample_rate_out,
-                               int* sample_bits_out);
-
   // Get/Set the playback rate of |algorithm_|.
   virtual void SetPlaybackRate(float playback_rate);
   virtual float GetPlaybackRate();
diff --git a/media/filters/audio_renderer_base_unittest.cc b/media/filters/audio_renderer_base_unittest.cc
index c1af178..f63bad9 100644
--- a/media/filters/audio_renderer_base_unittest.cc
+++ b/media/filters/audio_renderer_base_unittest.cc
@@ -16,7 +16,6 @@ using ::testing::InSequence;
 using ::testing::Invoke;
 using ::testing::NotNull;
 using ::testing::Return;
-using ::testing::ReturnRef;
 using ::testing::StrictMock;
 
 namespace media {
@@ -32,7 +31,7 @@ class MockAudioRendererBase : public AudioRendererBase {
   MOCK_METHOD1(SetVolume, void(float volume));
 
   // AudioRendererBase implementation.
-  MOCK_METHOD1(OnInitialize, bool(const MediaFormat& media_format));
+  MOCK_METHOD1(OnInitialize, bool(const AudioDecoderConfig& config));
   MOCK_METHOD0(OnStop, void());
 
   // Used for verifying check points during tests.
@@ -57,12 +56,9 @@ class AudioRendererBaseTest : public ::testing::Test {
     EXPECT_CALL(*decoder_, ProduceAudioSamples(_))
         .WillRepeatedly(Invoke(this, &AudioRendererBaseTest::EnqueueCallback));
 
-    // Sets the essential media format keys for this decoder.
-    decoder_media_format_.SetAsInteger(MediaFormat::kChannels, 1);
-    decoder_media_format_.SetAsInteger(MediaFormat::kSampleRate, 44100);
-    decoder_media_format_.SetAsInteger(MediaFormat::kSampleBits, 16);
-    EXPECT_CALL(*decoder_, media_format())
-        .WillRepeatedly(ReturnRef(decoder_media_format_));
+    // Set up audio properties.
+    ON_CALL(*decoder_, config())
+        .WillByDefault(Return(AudioDecoderConfig(16, 1, 44100)));
   }
 
   virtual ~AudioRendererBaseTest() {
@@ -78,7 +74,6 @@ class AudioRendererBaseTest : public ::testing::Test {
   scoped_refptr<MockAudioRendererBase> renderer_;
   scoped_refptr<MockAudioDecoder> decoder_;
   StrictMock<MockFilterHost> host_;
-  MediaFormat decoder_media_format_;
 
   // Number of asynchronous read requests sent to |decoder_|.
   size_t pending_reads_;
diff --git a/media/filters/audio_renderer_impl.cc b/media/filters/audio_renderer_impl.cc
index b0dcaf1..ae91993 100644
--- a/media/filters/audio_renderer_impl.cc
+++ b/media/filters/audio_renderer_impl.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -72,19 +72,11 @@ void AudioRendererImpl::OnError(AudioOutputStream* stream, int code) {
   NOTIMPLEMENTED();
 }
 
-bool AudioRendererImpl::OnInitialize(const MediaFormat& media_format) {
-  // Parse out audio parameters.
-  AudioParameters params;
-  if (!ParseMediaFormat(media_format, &params.channels,
-                        &params.sample_rate, &params.bits_per_sample)) {
-    return false;
-  }
-
-  // Set packet size.
+bool AudioRendererImpl::OnInitialize(const AudioDecoderConfig& config) {
+  AudioParameters params(config);
   params.samples_per_packet = kSamplesPerBuffer;
 
-  bytes_per_second_ = params.sample_rate * params.channels *
-      params.bits_per_sample / 8;
+  bytes_per_second_ = params.GetBytesPerSecond();
 
   // Create our audio stream.
   stream_ = AudioManager::GetAudioManager()->MakeAudioOutputStream(params);
diff --git a/media/filters/audio_renderer_impl.h b/media/filters/audio_renderer_impl.h
index 3e19992..b0ffc87 100644
--- a/media/filters/audio_renderer_impl.h
+++ b/media/filters/audio_renderer_impl.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -44,7 +44,7 @@ class AudioRendererImpl : public AudioRendererBase,
 
  protected:
   // AudioRendererBase implementation.
-  virtual bool OnInitialize(const MediaFormat& media_format);
+  virtual bool OnInitialize(const AudioDecoderConfig& config);
   virtual void OnStop();
 
  private:
diff --git a/media/filters/decoder_base.h b/media/filters/decoder_base.h
index 086aa96..280c0bf 100644
--- a/media/filters/decoder_base.h
+++ b/media/filters/decoder_base.h
@@ -53,8 +53,6 @@ class DecoderBase : public Decoder {
                           callback));
   }
 
-  virtual const MediaFormat& media_format() { return media_format_; }
-
   // Audio decoder.
   // Note that this class is only used by the audio decoder, this will
   // eventually be merged into FFmpegAudioDecoder.
@@ -97,8 +95,7 @@ class DecoderBase : public Decoder {
   // the demuxer stream.  Returns true if successful, otherwise false indicates
   // a fatal error.  The derived class should NOT call the filter host's
   // InitializationComplete() method.  If this method returns true, then the
-  // base class will call the host to complete initialization.  During this
-  // call, the derived class must fill in the media_format_ member.
+  // base class will call the host to complete initialization.
   virtual void DoInitialize(DemuxerStream* demuxer_stream, bool* success,
                             Task* done_cb) = 0;
 
@@ -118,8 +115,6 @@ class DecoderBase : public Decoder {
   // the EnequeueResult() method from within this method.
   virtual void DoDecode(Buffer* input) = 0;
 
-  MediaFormat media_format_;
-
   void OnDecodeComplete(const PipelineStatistics& statistics) {
     statistics_callback_->Run(statistics);
 
diff --git a/media/filters/ffmpeg_audio_decoder.cc b/media/filters/ffmpeg_audio_decoder.cc
index 74085a7..0dfc783d 100644
--- a/media/filters/ffmpeg_audio_decoder.cc
+++ b/media/filters/ffmpeg_audio_decoder.cc
@@ -30,6 +30,7 @@ const size_t FFmpegAudioDecoder::kOutputBufferSize =
 FFmpegAudioDecoder::FFmpegAudioDecoder(MessageLoop* message_loop)
     : DecoderBase<AudioDecoder, Buffer>(message_loop),
       codec_context_(NULL),
+      config_(0, 0, 0),
       estimated_next_timestamp_(kNoTimestamp) {
 }
 
@@ -68,17 +69,10 @@ void FFmpegAudioDecoder::DoInitialize(DemuxerStream* demuxer_stream,
     return;
   }
 
-  // When calling avcodec_find_decoder(), |codec_context_| might be altered by
-  // the decoder to give more accurate values of the output format of the
-  // decoder. So set the media format after a decoder is allocated.
-  // TODO(hclam): Reuse the information provided by the demuxer for now, we may
-  // need to wait until the first buffer is decoded to know the correct
-  // information.
-  media_format_.SetAsInteger(MediaFormat::kChannels, codec_context_->channels);
-  media_format_.SetAsInteger(MediaFormat::kSampleBits,
-      av_get_bits_per_sample_fmt(codec_context_->sample_fmt));
-  media_format_.SetAsInteger(MediaFormat::kSampleRate,
-      codec_context_->sample_rate);
+  config_.bits_per_channel =
+      av_get_bits_per_sample_fmt(codec_context_->sample_fmt);
+  config_.channels_per_sample = codec_context_->channels;
+  config_.sample_rate = codec_context_->sample_rate;
 
   // Prepare the output buffer.
   output_buffer_.reset(static_cast<uint8*>(av_malloc(kOutputBufferSize)));
@@ -89,6 +83,10 @@ void FFmpegAudioDecoder::DoInitialize(DemuxerStream* demuxer_stream,
   *success = true;
 }
 
+AudioDecoderConfig FFmpegAudioDecoder::config() {
+  return config_;
+}
+
 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 20bc22e..c3ca801 100644
--- a/media/filters/ffmpeg_audio_decoder.h
+++ b/media/filters/ffmpeg_audio_decoder.h
@@ -1,4 +1,4 @@
-// Copyright (c) 2009 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -19,6 +19,9 @@ class FFmpegAudioDecoder : public DecoderBase<AudioDecoder, Buffer> {
   explicit FFmpegAudioDecoder(MessageLoop* message_loop);
   virtual ~FFmpegAudioDecoder();
 
+  // AudioDecoder implementation.
+  virtual AudioDecoderConfig config();
+
  protected:
   virtual void DoInitialize(DemuxerStream* demuxer_stream, bool* success,
                             Task* done_cb);
@@ -36,6 +39,8 @@ class FFmpegAudioDecoder : public DecoderBase<AudioDecoder, Buffer> {
   // is initialized in OnInitialize().
   AVCodecContext* codec_context_;
 
+  AudioDecoderConfig config_;
+
   // Estimated timestamp for next packet. Useful for packets without timestamps.
   base::TimeDelta estimated_next_timestamp_;
 
diff --git a/media/filters/null_audio_renderer.cc b/media/filters/null_audio_renderer.cc
index 15fd41b..de04f0a 100644
--- a/media/filters/null_audio_renderer.cc
+++ b/media/filters/null_audio_renderer.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -60,18 +60,11 @@ void NullAudioRenderer::ThreadMain() {
   }
 }
 
-bool NullAudioRenderer::OnInitialize(const MediaFormat& media_format) {
-  // Parse out audio parameters.
-  int channels;
-  int sample_rate;
-  int sample_bits;
-  if (!ParseMediaFormat(media_format, &channels, &sample_rate, &sample_bits)) {
-    return false;
-  }
-
+bool NullAudioRenderer::OnInitialize(const AudioDecoderConfig& config) {
   // Calculate our bytes per millisecond value and allocate our buffer.
-  bytes_per_millisecond_ = (channels * sample_rate * sample_bits / 8)
-      / base::Time::kMillisecondsPerSecond;
+  bytes_per_millisecond_ =
+      (config.channels_per_sample * config.sample_rate *
+       config.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 db57276..28da99b 100644
--- a/media/filters/null_audio_renderer.h
+++ b/media/filters/null_audio_renderer.h
@@ -41,7 +41,7 @@ class NullAudioRenderer : public AudioRendererBase,
 
  protected:
   // AudioRendererBase implementation.
-  virtual bool OnInitialize(const MediaFormat& media_format);
+  virtual bool OnInitialize(const AudioDecoderConfig& config);
   virtual void OnStop();
 
  private: