Replace AudioDecoderConfig with simple accessors on AudioDecoder.

media/filters/ffmpeg_audio_decoder.cc

 // 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.
 
 #include "media/filters/ffmpeg_audio_decoder.h"
 
 #include "media/base/data_buffer.h"
 #include "media/base/limits.h"
 #include "media/ffmpeg/ffmpeg_common.h"
 #include "media/filters/ffmpeg_demuxer.h"
@@ skipping 11 matching lines @@
 
 namespace media {
 
 // Size of the decoded audio buffer.
 const size_t FFmpegAudioDecoder::kOutputBufferSize =
     AVCODEC_MAX_AUDIO_FRAME_SIZE;
 
 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) {
 }
 
 FFmpegAudioDecoder::~FFmpegAudioDecoder() {
 }
 
 void FFmpegAudioDecoder::DoInitialize(DemuxerStream* demuxer_stream,
                                       bool* success,
                                       Task* done_cb) {
   base::ScopedTaskRunner done_runner(done_cb);
@@ skipping 20 matching lines @@
         << " sample rate: " << codec_context_->sample_rate;
     return;
   }
 
   // Serialize calls to avcodec_open().
   AVCodec* codec = avcodec_find_decoder(codec_context_->codec_id);
   if (!codec || avcodec_open(codec_context_, codec) < 0) {
     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)));
   if (!output_buffer_.get()) {
     host()->SetError(PIPELINE_ERROR_OUT_OF_MEMORY);
     return;
   }
   *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;
   done_cb->Run();
   delete done_cb;
 }
 
 // ConvertAudioF32ToS32() converts float audio (F32) to int (S32) in place.
 // This is a temporary solution.
 // The purpose of this short term fix is to enable WMApro, which decodes to
@@ skipping 145 matching lines @@
 base::TimeDelta FFmpegAudioDecoder::CalculateDuration(size_t size) {
   int64 denominator = codec_context_->channels *
       av_get_bits_per_sample_fmt(codec_context_->sample_fmt) / 8 *
       codec_context_->sample_rate;
   double microseconds = size /
       (denominator / static_cast<double>(base::Time::kMicrosecondsPerSecond));
   return base::TimeDelta::FromMicroseconds(static_cast<int64>(microseconds));
 }
 
 }  // namespace