Add AudioBufferStream.

media/filters/ffmpeg_audio_decoder.cc

Index: media/filters/ffmpeg_audio_decoder.cc
diff --git a/media/filters/ffmpeg_audio_decoder.cc b/media/filters/ffmpeg_audio_decoder.cc
index 3ea70ff02407ac1df9d9b9b9ef5a39ae0cf85c07..58e1680940be63e89536cffe8ae36737e4106bb7 100644
--- a/media/filters/ffmpeg_audio_decoder.cc
+++ b/media/filters/ffmpeg_audio_decoder.cc
@@ -23,12 +23,6 @@
 
 namespace media {
 
-// Helper structure for managing multiple decoded audio frames per packet.
-struct QueuedAudioBuffer {
-  AudioDecoder::Status status;
-  scoped_refptr<AudioBuffer> buffer;
-};
-
 // Returns true if the decode result was end of stream.
 static inline bool IsEndOfStream(int result,
                                  int decoded_size,
@@ -73,63 +67,75 @@ FFmpegAudioDecoder::FFmpegAudioDecoder(
     const scoped_refptr<base::SingleThreadTaskRunner>& task_runner)
     : task_runner_(task_runner),
       weak_factory_(this),
-      demuxer_stream_(NULL),
+      state_(kUninitialized),
       bytes_per_channel_(0),
       channel_layout_(CHANNEL_LAYOUT_NONE),
       channels_(0),
       samples_per_second_(0),
       av_sample_format_(0),
       last_input_timestamp_(kNoTimestamp()),
-      output_frames_to_drop_(0) {
+      output_frames_to_drop_(0) {}
+
+FFmpegAudioDecoder::~FFmpegAudioDecoder() {
+  DCHECK_EQ(state_, kUninitialized);
+  DCHECK(!codec_context_);
+  DCHECK(!av_frame_);
 }
 
-void FFmpegAudioDecoder::Initialize(
-    DemuxerStream* stream,
-    const PipelineStatusCB& status_cb,
-    const StatisticsCB& statistics_cb) {
+void FFmpegAudioDecoder::Initialize(const AudioDecoderConfig& config,
+                                    const PipelineStatusCB& status_cb) {
   DCHECK(task_runner_->BelongsToCurrentThread());
-  PipelineStatusCB initialize_cb = BindToCurrentLoop(status_cb);
+  DCHECK(!config.is_encrypted());
 
   FFmpegGlue::InitializeFFmpeg();
-
-  if (demuxer_stream_) {
-    // TODO(scherkus): initialization currently happens more than once in
-    // PipelineIntegrationTest.BasicPlayback.
-    LOG(ERROR) << "Initialize has already been called.";
-    CHECK(false);
-  }
-
   weak_this_ = weak_factory_.GetWeakPtr();
-  demuxer_stream_ = stream;
 
-  if (!ConfigureDecoder()) {
-    status_cb.Run(DECODER_ERROR_NOT_SUPPORTED);
+  config_ = config;
+  PipelineStatusCB initialize_cb = BindToCurrentLoop(status_cb);
+
+  if (!config.IsValidConfig() || !ConfigureDecoder()) {
+    initialize_cb.Run(DECODER_ERROR_NOT_SUPPORTED);
     return;
   }
 
-  statistics_cb_ = statistics_cb;
+  // Success!
+  state_ = kNormal;
   initialize_cb.Run(PIPELINE_OK);
 }
 
-void FFmpegAudioDecoder::Read(const ReadCB& read_cb) {
+void FFmpegAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& buffer,
+                                const DecodeCB& decode_cb) {
   DCHECK(task_runner_->BelongsToCurrentThread());
-  DCHECK(!read_cb.is_null());
-  CHECK(read_cb_.is_null()) << "Overlapping decodes are not supported.";
-  DCHECK(reset_cb_.is_null());
-  DCHECK(stop_cb_.is_null());
+  DCHECK(!decode_cb.is_null());
+  CHECK_NE(state_, kUninitialized);
+  DecodeCB decode_cb_bound = BindToCurrentLoop(decode_cb);
 
-  read_cb_ = BindToCurrentLoop(read_cb);
+  if (state_ == kError) {
+    decode_cb_bound.Run(kDecodeError, NULL);
+    return;
+  }
 
-  // If we don't have any queued audio from the last packet we decoded, ask for
-  // more data from the demuxer to satisfy this read.
-  if (queued_audio_.empty()) {
-    ReadFromDemuxerStream();
+  // Return empty frames if decoding has finished.
+  if (state_ == kDecodeFinished) {
+    decode_cb_bound.Run(kOk, AudioBuffer::CreateEOSBuffer());
     return;
   }
 
-  base::ResetAndReturn(&read_cb_).Run(
-      queued_audio_.front().status, queued_audio_.front().buffer);
+  if (!buffer) {
+    decode_cb_bound.Run(kAborted, NULL);
+    return;
+  }
+
+  DecodeBuffer(buffer, decode_cb_bound);
+}
+
+scoped_refptr<AudioBuffer> FFmpegAudioDecoder::GetDecodeOutput() {
+  DCHECK(task_runner_->BelongsToCurrentThread());
+  if (queued_audio_.empty())
+    return NULL;
+  scoped_refptr<AudioBuffer> out = queued_audio_.front();
   queued_audio_.pop_front();
+  return out;
 }
 
 int FFmpegAudioDecoder::bits_per_channel() {
@@ -149,39 +155,34 @@ int FFmpegAudioDecoder::samples_per_second() {
 
 void FFmpegAudioDecoder::Reset(const base::Closure& closure) {
   DCHECK(task_runner_->BelongsToCurrentThread());
-  reset_cb_ = BindToCurrentLoop(closure);
-
-  // A demuxer read is pending, we'll wait until it finishes.
-  if (!read_cb_.is_null())
-    return;
 
-  DoReset();
+  avcodec_flush_buffers(codec_context_.get());
+  state_ = kNormal;
+  ResetTimestampState();
+  task_runner_->PostTask(FROM_HERE, closure);
 }
 
 void FFmpegAudioDecoder::Stop(const base::Closure& closure) {
   DCHECK(task_runner_->BelongsToCurrentThread());
-  stop_cb_ = BindToCurrentLoop(closure);
-
-  // A demuxer read is pending, we'll wait until it finishes.
-  if (!read_cb_.is_null())
-    return;
+  base::ScopedClosureRunner runner(BindToCurrentLoop(closure));
 
-  if (!reset_cb_.is_null()) {
-    DoReset();
+  if (state_ == kUninitialized)
     return;
-  }
 
-  DoStop();
+  ReleaseFFmpegResources();
+  ResetTimestampState();
+  state_ = kUninitialized;
 }
 
-FFmpegAudioDecoder::~FFmpegAudioDecoder() {}
-
+// Callback called from within FFmpeg to allocate a buffer based on

[DaleCurtis, 2014/03/04 01:01:33]

We put method comments in the .h file generally.  It doesn't look like this is
actually using any class members though, so you could just move it to the top of
the file as an internal static declaration.

[rileya (GONE FROM CHROMIUM), 2014/03/04 02:03:01]

Oh wow, I saw that the original version did this and didn't look closely, but
looks like this has no reason to be a member of FFmpegAudioDecoder. I hoisted
all of this up into the already-existing static function above (which previously
just called this on the decoder...).

+// the dimensions of |codec_context|. See AVCodecContext.get_buffer2
+// documentation inside FFmpeg.
 int FFmpegAudioDecoder::GetAudioBuffer(AVCodecContext* codec,
                                        AVFrame* frame,
                                        int flags) {
   // Since this routine is called by FFmpeg when a buffer is required for audio
   // data, use the values supplied by FFmpeg (ignoring the current settings).
-  // RunDecodeLoop() gets to determine if the buffer is useable or not.
+  // FFmpegDecode() gets to determine if the buffer is useable or not.
   AVSampleFormat format = static_cast<AVSampleFormat>(frame->format);
   SampleFormat sample_format = AVSampleFormatToSampleFormat(format);
   int channels = DetermineChannels(frame);
@@ -241,241 +242,117 @@ int FFmpegAudioDecoder::GetAudioBuffer(AVCodecContext* codec,
   return 0;
 }
 
-void FFmpegAudioDecoder::DoStop() {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-  DCHECK(!stop_cb_.is_null());
-  DCHECK(read_cb_.is_null());
-  DCHECK(reset_cb_.is_null());
-
-  ResetTimestampState();
-  queued_audio_.clear();
-  ReleaseFFmpegResources();
-  base::ResetAndReturn(&stop_cb_).Run();
-}
-
-void FFmpegAudioDecoder::DoReset() {
-  DCHECK(task_runner_->BelongsToCurrentThread());
-  DCHECK(!reset_cb_.is_null());
-  DCHECK(read_cb_.is_null());
-
-  avcodec_flush_buffers(codec_context_.get());
-  ResetTimestampState();
-  queued_audio_.clear();
-  base::ResetAndReturn(&reset_cb_).Run();
-
-  if (!stop_cb_.is_null())
-    DoStop();
-}
-
-void FFmpegAudioDecoder::ReadFromDemuxerStream() {
-  DCHECK(!read_cb_.is_null());
-  demuxer_stream_->Read(base::Bind(
-      &FFmpegAudioDecoder::BufferReady, weak_this_));
-}
-
-void FFmpegAudioDecoder::BufferReady(
-    DemuxerStream::Status status,
-    const scoped_refptr<DecoderBuffer>& input) {
+void FFmpegAudioDecoder::DecodeBuffer(
+    const scoped_refptr<DecoderBuffer>& buffer,
+    const DecodeCB& decode_cb) {
   DCHECK(task_runner_->BelongsToCurrentThread());
-  DCHECK(!read_cb_.is_null());
-  DCHECK(queued_audio_.empty());
-  DCHECK_EQ(status != DemuxerStream::kOk, !input.get()) << status;
-
-  // Pending Reset: ignore the buffer we just got, send kAborted to |read_cb_|
-  // and carry out the Reset().
-  // If there happens to also be a pending Stop(), that will be handled at
-  // the end of DoReset().
-  if (!reset_cb_.is_null()) {
-    base::ResetAndReturn(&read_cb_).Run(kAborted, NULL);
-    DoReset();
-    return;
-  }
-
-  // Pending Stop: ignore the buffer we just got, send kAborted to |read_cb_|
-  // and carry out the Stop().
-  if (!stop_cb_.is_null()) {
-    base::ResetAndReturn(&read_cb_).Run(kAborted, NULL);
-    DoStop();
-    return;
-  }
-
-  if (status == DemuxerStream::kAborted) {
-    DCHECK(!input.get());
-    base::ResetAndReturn(&read_cb_).Run(kAborted, NULL);
-    return;
-  }
-
-  if (status == DemuxerStream::kConfigChanged) {
-    DCHECK(!input.get());
-
-    // Send a "end of stream" buffer to the decode loop
-    // to output any remaining data still in the decoder.
-    RunDecodeLoop(DecoderBuffer::CreateEOSBuffer(), true);
-
-    DVLOG(1) << "Config changed.";
-
-    if (!ConfigureDecoder()) {
-      base::ResetAndReturn(&read_cb_).Run(kDecodeError, NULL);
-      return;
-    }
-
-    ResetTimestampState();
-
-    if (queued_audio_.empty()) {
-      ReadFromDemuxerStream();
-      return;
-    }
-
-    base::ResetAndReturn(&read_cb_).Run(
-        queued_audio_.front().status, queued_audio_.front().buffer);
-    queued_audio_.pop_front();
-    return;
-  }
-
-  DCHECK_EQ(status, DemuxerStream::kOk);
-  DCHECK(input.get());
+  DCHECK_NE(state_, kUninitialized);
+  DCHECK_NE(state_, kDecodeFinished);
+  DCHECK_NE(state_, kError);
+
+  DCHECK(buffer);
+
+  // During decode, because reads are issued asynchronously, it is possible to

[DaleCurtis, 2014/03/04 01:01:33]

Instead of having this similar / same comment in both the Audio and Video
decoders, is there a place we can put in the DecoderStream?

[rileya (GONE FROM CHROMIUM), 2014/03/04 02:03:01]

The problem is that this logic is only shared by the FFmpeg Audio/VideoDecoders,
Opus doesn't even track states, and DecryptingAudioDecoder has a somewhat
different set of states.

There's probably some refactoring potential in here somewhere (FFmpeg audio and
video decoders are pretty similar for one, and most decoders use similar but
subtly different state machines... perhaps a common base for the various
Decoders to keep some of the state transitions and stuff consistent would be
useful...).

+  // receive multiple end of stream buffers since each decode is acked. When the
+  // first end of stream buffer is read, FFmpeg may still have frames queued
+  // up in the decoder so we need to go through the decode loop until it stops
+  // giving sensible data.  After that, the decoder should output empty
+  // frames.  There are three states the decoder can be in:
+  //
+  //   kNormal: This is the starting state. Buffers are decoded. Decode errors
+  //            are discarded.
+  //   kFlushCodec: There isn't any more input data. Call avcodec_decode_audio4
+  //                until no more data is returned to flush out remaining
+  //                frames. The input buffer is ignored at this point.
+  //   kDecodeFinished: All calls return empty frames.
+  //   kError: Unexpected error happened.
+  //
+  // These are the possible state transitions.
+  //
+  // kNormal -> kFlushCodec:
+  //     When buffer->end_of_stream() is first true.
+  // kNormal -> kError:
+  //     A decoding error occurs and decoding needs to stop.
+  // kFlushCodec -> kDecodeFinished:
+  //     When avcodec_decode_audio4() returns 0 data.
+  // kFlushCodec -> kError:
+  //     When avcodec_decode_audio4() errors out.
+  // (any state) -> kNormal:
+  //     Any time Reset() is called.
 
   // Make sure we are notified if http://crbug.com/49709 returns.  Issue also
   // occurs with some damaged files.
-  if (!input->end_of_stream() && input->timestamp() == kNoTimestamp() &&
+  if (!buffer->end_of_stream() && buffer->timestamp() == kNoTimestamp() &&
       output_timestamp_helper_->base_timestamp() == kNoTimestamp()) {
     DVLOG(1) << "Received a buffer without timestamps!";
-    base::ResetAndReturn(&read_cb_).Run(kDecodeError, NULL);
+    decode_cb.Run(kDecodeError, NULL);
     return;
   }
 
-  if (!input->end_of_stream()) {
+  if (!buffer->end_of_stream()) {
     if (last_input_timestamp_ == kNoTimestamp() &&
         codec_context_->codec_id == AV_CODEC_ID_VORBIS &&
-        input->timestamp() < base::TimeDelta()) {
+        buffer->timestamp() < base::TimeDelta()) {
       // Dropping frames for negative timestamps as outlined in section A.2
       // in the Vorbis spec. http://xiph.org/vorbis/doc/Vorbis_I_spec.html
       output_frames_to_drop_ = floor(
-          0.5 + -input->timestamp().InSecondsF() * samples_per_second_);
+          0.5 + -buffer->timestamp().InSecondsF() * samples_per_second_);
     } else {
       if (last_input_timestamp_ != kNoTimestamp() &&
-          input->timestamp() < last_input_timestamp_) {
-        const base::TimeDelta diff = input->timestamp() - last_input_timestamp_;
+          buffer->timestamp() < last_input_timestamp_) {
+        const base::TimeDelta diff =
+            buffer->timestamp() - last_input_timestamp_;
         DLOG(WARNING)
             << "Input timestamps are not monotonically increasing! "
-            << " ts " << input->timestamp().InMicroseconds() << " us"
+            << " ts " << buffer->timestamp().InMicroseconds() << " us"
             << " diff " << diff.InMicroseconds() << " us";
       }
 
-      last_input_timestamp_ = input->timestamp();
+      last_input_timestamp_ = buffer->timestamp();
     }
   }
 
-  RunDecodeLoop(input, false);
-
-  // We exhausted the provided packet, but it wasn't enough for a frame.  Ask
-  // for more data in order to fulfill this read.
-  if (queued_audio_.empty()) {
-    ReadFromDemuxerStream();
-    return;
-  }
-
-  // Execute callback to return the first frame we decoded.
-  base::ResetAndReturn(&read_cb_).Run(
-      queued_audio_.front().status, queued_audio_.front().buffer);
-  queued_audio_.pop_front();
-}
-
-bool FFmpegAudioDecoder::ConfigureDecoder() {
-  const AudioDecoderConfig& config = demuxer_stream_->audio_decoder_config();
-
-  if (!config.IsValidConfig()) {
-    DLOG(ERROR) << "Invalid audio stream -"
-                << " codec: " << config.codec()
-                << " channel layout: " << config.channel_layout()
-                << " bits per channel: " << config.bits_per_channel()
-                << " samples per second: " << config.samples_per_second();
-    return false;
-  }
-
-  if (config.is_encrypted()) {
-    DLOG(ERROR) << "Encrypted audio stream not supported";
-    return false;
+  // Transition to kFlushCodec on the first end of stream buffer.
+  if (state_ == kNormal && buffer->end_of_stream()) {
+    state_ = kFlushCodec;
   }
 
-  if (codec_context_.get() &&
-      (bytes_per_channel_ != config.bytes_per_channel() ||
-       channel_layout_ != config.channel_layout() ||
-       samples_per_second_ != config.samples_per_second())) {
-    DVLOG(1) << "Unsupported config change :";
-    DVLOG(1) << "\tbytes_per_channel : " << bytes_per_channel_
-             << " -> " << config.bytes_per_channel();
-    DVLOG(1) << "\tchannel_layout : " << channel_layout_
-             << " -> " << config.channel_layout();
-    DVLOG(1) << "\tsample_rate : " << samples_per_second_
-             << " -> " << config.samples_per_second();
-    return false;
-  }
-
-  // Release existing decoder resources if necessary.
-  ReleaseFFmpegResources();
-
-  // Initialize AVCodecContext structure.
-  codec_context_.reset(avcodec_alloc_context3(NULL));
-  AudioDecoderConfigToAVCodecContext(config, codec_context_.get());
-
-  codec_context_->opaque = this;
-  codec_context_->get_buffer2 = GetAudioBufferImpl;
-  codec_context_->refcounted_frames = 1;
-
-  AVCodec* codec = avcodec_find_decoder(codec_context_->codec_id);
-  if (!codec || avcodec_open2(codec_context_.get(), codec, NULL) < 0) {
-    DLOG(ERROR) << "Could not initialize audio decoder: "
-                << codec_context_->codec_id;
-    return false;
+  scoped_refptr<AudioBuffer> audio_buffer;

[DaleCurtis, 2014/03/04 01:01:33]

Unused? You could just set result/audio_buffer= output and have one
decode_cb.Run() call below.

[rileya (GONE FROM CHROMIUM), 2014/03/04 02:03:01]

Whoops, yup. I ended up just removing it, moving the callback call to the bottom
made the control flow harder to follow.

+  if (!FFmpegDecode(buffer)) {
+    state_ = kError;
+    decode_cb.Run(kDecodeError, NULL);
+    return;
   }
 
-  // Success!
-  av_frame_.reset(av_frame_alloc());
-  channel_layout_ = config.channel_layout();
-  samples_per_second_ = config.samples_per_second();
-  output_timestamp_helper_.reset(
-      new AudioTimestampHelper(config.samples_per_second()));
+  if (queued_audio_.empty()) {
+    if (state_ == kFlushCodec) {
+      DCHECK(buffer->end_of_stream());
+      state_ = kDecodeFinished;
+      decode_cb.Run(kOk, AudioBuffer::CreateEOSBuffer());
+      return;
+    }
 
-  // Store initial values to guard against midstream configuration changes.
-  channels_ = codec_context_->channels;
-  if (channels_ != ChannelLayoutToChannelCount(channel_layout_)) {
-    DLOG(ERROR) << "Audio configuration specified "
-                << ChannelLayoutToChannelCount(channel_layout_)
-                << " channels, but FFmpeg thinks the file contains "
-                << channels_ << " channels";
-    return false;
+    decode_cb.Run(kNotEnoughData, NULL);
+    return;
   }
-  av_sample_format_ = codec_context_->sample_fmt;
-  sample_format_ = AVSampleFormatToSampleFormat(
-      static_cast<AVSampleFormat>(av_sample_format_));
-  bytes_per_channel_ = SampleFormatToBytesPerChannel(sample_format_);
 
-  return true;
+  decode_cb.Run(kOk, queued_audio_.front());
+  queued_audio_.pop_front();
 }
 
-void FFmpegAudioDecoder::ReleaseFFmpegResources() {
-  codec_context_.reset();
-  av_frame_.reset();
-}
+bool FFmpegAudioDecoder::FFmpegDecode(
+    const scoped_refptr<DecoderBuffer>& buffer) {
 
-void FFmpegAudioDecoder::ResetTimestampState() {
-  output_timestamp_helper_->SetBaseTimestamp(kNoTimestamp());
-  last_input_timestamp_ = kNoTimestamp();
-  output_frames_to_drop_ = 0;
-}
+  DCHECK(queued_audio_.empty());
 
-void FFmpegAudioDecoder::RunDecodeLoop(
-    const scoped_refptr<DecoderBuffer>& input,
-    bool skip_eos_append) {
   AVPacket packet;
   av_init_packet(&packet);
-  if (input->end_of_stream()) {
+  if (buffer->end_of_stream()) {
     packet.data = NULL;
     packet.size = 0;
   } else {
-    packet.data = const_cast<uint8*>(input->data());
-    packet.size = input->data_size();
+    packet.data = const_cast<uint8*>(buffer->data());
+    packet.size = buffer->data_size();
   }
 
   // Each audio packet may contain several frames, so we must call the decoder
@@ -488,16 +365,16 @@ void FFmpegAudioDecoder::RunDecodeLoop(
         codec_context_.get(), av_frame_.get(), &frame_decoded, &packet);
 
     if (result < 0) {
-      DCHECK(!input->end_of_stream())
+      DCHECK(!buffer->end_of_stream())
           << "End of stream buffer produced an error! "
           << "This is quite possibly a bug in the audio decoder not handling "
           << "end of stream AVPackets correctly.";
 
       DLOG(WARNING)
           << "Failed to decode an audio frame with timestamp: "
-          << input->timestamp().InMicroseconds() << " us, duration: "
-          << input->duration().InMicroseconds() << " us, packet size: "
-          << input->data_size() << " bytes";
+          << buffer->timestamp().InMicroseconds() << " us, duration: "
+          << buffer->duration().InMicroseconds() << " us, packet size: "
+          << buffer->data_size() << " bytes";
 
       break;
     }
@@ -508,15 +385,15 @@ void FFmpegAudioDecoder::RunDecodeLoop(
     packet.data += result;
 
     if (output_timestamp_helper_->base_timestamp() == kNoTimestamp() &&
-        !input->end_of_stream()) {
-      DCHECK(input->timestamp() != kNoTimestamp());
+        !buffer->end_of_stream()) {
+      DCHECK(buffer->timestamp() != kNoTimestamp());
       if (output_frames_to_drop_ > 0) {
         // Currently Vorbis is the only codec that causes us to drop samples.
         // If we have to drop samples it always means the timeline starts at 0.
         DCHECK_EQ(codec_context_->codec_id, AV_CODEC_ID_VORBIS);
         output_timestamp_helper_->SetBaseTimestamp(base::TimeDelta());
       } else {
-        output_timestamp_helper_->SetBaseTimestamp(input->timestamp());
+        output_timestamp_helper_->SetBaseTimestamp(buffer->timestamp());
       }
     }
 
@@ -525,7 +402,10 @@ void FFmpegAudioDecoder::RunDecodeLoop(
     int original_frames = 0;
     int channels = DetermineChannels(av_frame_.get());
     if (frame_decoded) {
-      if (av_frame_->sample_rate != samples_per_second_ ||
+
+      // TODO(rileya) Remove this check once we properly support midstream audio
+      // config changes.
+      if (av_frame_->sample_rate != config_.samples_per_second() ||
           channels != channels_ ||
           av_frame_->format != av_sample_format_) {
         DLOG(ERROR) << "Unsupported midstream configuration change!"
@@ -537,16 +417,16 @@ void FFmpegAudioDecoder::RunDecodeLoop(
                     << av_sample_format_;
 
         // This is an unrecoverable error, so bail out.
-        QueuedAudioBuffer queue_entry = { kDecodeError, NULL };
-        queued_audio_.push_back(queue_entry);
+        queued_audio_.clear();

[rileya (GONE FROM CHROMIUM), 2014/03/04 00:14:27]

Previously we could queue up buffers which were not kOk here, meaning we needed
to return a struct with the buffer and the status.

Instead I empty the queue and return an error, in theory there might be some
case where we throw out a valid buffer because the next buffer in the same
packet has a config change, but we'll support config changes soon enough
anyway...

         av_frame_unref(av_frame_.get());
-        break;
+        return false;
       }
 
       // Get the AudioBuffer that the data was decoded into. Adjust the number
       // of frames, in case fewer than requested were actually decoded.
       output = reinterpret_cast<AudioBuffer*>(
           av_buffer_get_opaque(av_frame_->buf[0]));
+
       DCHECK_EQ(channels_, output->channel_count());
       original_frames = av_frame_->nb_samples;
       int unread_frames = output->frame_count() - original_frames;
@@ -574,8 +454,7 @@ void FFmpegAudioDecoder::RunDecodeLoop(
       output->set_duration(
           output_timestamp_helper_->GetFrameDuration(decoded_frames));
       output_timestamp_helper_->AddFrames(decoded_frames);
-    } else if (IsEndOfStream(result, original_frames, input) &&
-               !skip_eos_append) {
+    } else if (IsEndOfStream(result, original_frames, buffer)) {
       DCHECK_EQ(packet.size, 0);
       output = AudioBuffer::CreateEOSBuffer();
     } else {
@@ -583,18 +462,98 @@ void FFmpegAudioDecoder::RunDecodeLoop(
       output = NULL;
     }
 
-    if (output.get()) {
-      QueuedAudioBuffer queue_entry = { kOk, output };
-      queued_audio_.push_back(queue_entry);
-    }
+    if (output.get())
+      queued_audio_.push_back(output);
 
-    // Decoding finished successfully, update statistics.
-    if (result > 0) {
-      PipelineStatistics statistics;
-      statistics.audio_bytes_decoded = result;
-      statistics_cb_.Run(statistics);
-    }
   } while (packet.size > 0);
+
+  return true;
+}
+
+void FFmpegAudioDecoder::ReleaseFFmpegResources() {
+  codec_context_.reset();
+  av_frame_.reset();
+}
+
+bool FFmpegAudioDecoder::ConfigureDecoder() {
+  if (!config_.IsValidConfig()) {
+    DLOG(ERROR) << "Invalid audio stream -"
+                << " codec: " << config_.codec()
+                << " channel layout: " << config_.channel_layout()
+                << " bits per channel: " << config_.bits_per_channel()
+                << " samples per second: " << config_.samples_per_second();
+    return false;
+  }
+
+  if (config_.is_encrypted()) {
+    DLOG(ERROR) << "Encrypted audio stream not supported";
+    return false;
+  }
+
+  // TODO(rileya) Remove this check once we properly support midstream audio

[DaleCurtis, 2014/03/04 01:01:33]

Note: Some of these were added to avoid ffmpeg security issues, so be sure to
run ffmpeg_regression_tests when you do remove them.

+  // config changes.
+  if (codec_context_.get() &&
+      (bytes_per_channel_ != config_.bytes_per_channel() ||
+       channel_layout_ != config_.channel_layout() ||
+       samples_per_second_ != config_.samples_per_second())) {
+    DVLOG(1) << "Unsupported config change :";
+    DVLOG(1) << "\tbytes_per_channel : " << bytes_per_channel_
+             << " -> " << config_.bytes_per_channel();
+    DVLOG(1) << "\tchannel_layout : " << channel_layout_
+             << " -> " << config_.channel_layout();
+    DVLOG(1) << "\tsample_rate : " << samples_per_second_
+             << " -> " << config_.samples_per_second();
+    return false;
+  }
+
+  // Release existing decoder resources if necessary.
+  ReleaseFFmpegResources();
+
+  // Initialize AVCodecContext structure.
+  codec_context_.reset(avcodec_alloc_context3(NULL));
+  AudioDecoderConfigToAVCodecContext(config_, codec_context_.get());
+
+  codec_context_->opaque = this;
+  codec_context_->get_buffer2 = GetAudioBufferImpl;
+  codec_context_->refcounted_frames = 1;
+
+  AVCodec* codec = avcodec_find_decoder(codec_context_->codec_id);
+  if (!codec || avcodec_open2(codec_context_.get(), codec, NULL) < 0) {
+    DLOG(ERROR) << "Could not initialize audio decoder: "
+                << codec_context_->codec_id;
+    ReleaseFFmpegResources();
+    state_ = kUninitialized;
+    return false;
+  }
+
+  // Success!
+  av_frame_.reset(av_frame_alloc());
+  channel_layout_ = config_.channel_layout();
+  samples_per_second_ = config_.samples_per_second();
+  output_timestamp_helper_.reset(
+      new AudioTimestampHelper(config_.samples_per_second()));
+
+  // Store initial values to guard against midstream configuration changes.
+  channels_ = codec_context_->channels;
+  if (channels_ != ChannelLayoutToChannelCount(channel_layout_)) {
+    DLOG(ERROR) << "Audio configuration specified "
+                << ChannelLayoutToChannelCount(channel_layout_)
+                << " channels, but FFmpeg thinks the file contains "
+                << channels_ << " channels";
+    return false;
+  }
+  av_sample_format_ = codec_context_->sample_fmt;
+  sample_format_ = AVSampleFormatToSampleFormat(
+      static_cast<AVSampleFormat>(av_sample_format_));
+  bytes_per_channel_ = SampleFormatToBytesPerChannel(sample_format_);
+
+  return true;
+}
+
+void FFmpegAudioDecoder::ResetTimestampState() {
+  output_timestamp_helper_->SetBaseTimestamp(kNoTimestamp());
+  last_input_timestamp_ = kNoTimestamp();
+  output_frames_to_drop_ = 0;
 }
 
 }  // namespace media