Guard against midstream audio configuration changes.

media/filters/ffmpeg_audio_decoder.cc

 // 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.
 
 #include "media/filters/ffmpeg_audio_decoder.h"
 
 #include "base/bind.h"
 #include "base/callback_helpers.h"
 #include "base/location.h"
 #include "base/message_loop_proxy.h"
@@ skipping 25 matching lines @@
   // 3. An end of stream buffer is received.
   return result == 0 && decoded_size == 0 && input->IsEndOfStream();
 }
 
 FFmpegAudioDecoder::FFmpegAudioDecoder(
     const scoped_refptr<base::MessageLoopProxy>& message_loop)
     : message_loop_(message_loop),
       codec_context_(NULL),
       bits_per_channel_(0),
       channel_layout_(CHANNEL_LAYOUT_NONE),
+      channels_(0),
       samples_per_second_(0),
+      av_sample_format_(0),
       bytes_per_frame_(0),
       last_input_timestamp_(kNoTimestamp()),
       output_bytes_to_drop_(0),
       av_frame_(NULL) {
 }
 
 void FFmpegAudioDecoder::Initialize(
     const scoped_refptr<DemuxerStream>& stream,
     const PipelineStatusCB& status_cb,
     const StatisticsCB& statistics_cb) {
@@ skipping 239 matching lines @@
   }
 
   // Success!
   av_frame_ = avcodec_alloc_frame();
   bits_per_channel_ = config.bits_per_channel();
   channel_layout_ = config.channel_layout();
   samples_per_second_ = config.samples_per_second();
   output_timestamp_helper_.reset(new AudioTimestampHelper(
       config.bytes_per_frame(), config.samples_per_second()));
   bytes_per_frame_ = config.bytes_per_frame();
+
+  // Store initial values to guard against mid-frame configuration changes.

[scherkus (not reviewing), 2013/02/12 02:16:56]

consistent terminology nit: s/mid-frame/mid-stream/

[DaleCurtis, 2013/02/12 02:41:24]

Done.

+  channels_ = codec_context_->channels;
+  av_sample_format_ = codec_context_->sample_fmt;
+
   return true;
 }
 
 void FFmpegAudioDecoder::ReleaseFFmpegResources() {
   if (codec_context_) {
     av_free(codec_context_->extradata);
     avcodec_close(codec_context_);
     av_free(codec_context_);
   }
 
@@ skipping 64 matching lines @@
         // If we have to drop samples it always means the timeline starts at 0.
         DCHECK_EQ(codec_context_->codec_id, CODEC_ID_VORBIS);
         output_timestamp_helper_->SetBaseTimestamp(base::TimeDelta());
       } else {
         output_timestamp_helper_->SetBaseTimestamp(input->GetTimestamp());
       }
     }
 
     int decoded_audio_size = 0;
     if (frame_decoded) {
-      int output_sample_rate = av_frame_->sample_rate;
-      if (output_sample_rate != samples_per_second_) {
-        DLOG(ERROR) << "Output sample rate (" << output_sample_rate
-                    << ") doesn't match expected rate " << samples_per_second_;
+      if (av_frame_->sample_rate != samples_per_second_ ||
+          av_frame_->channels != channels_ ||
+          av_frame_->format != av_sample_format_) {
+        DLOG(ERROR) << "Unsupported mid-frame configuration change!"

[scherkus (not reviewing), 2013/02/12 02:16:56]

consistent terminology nit: s/mid-frame/mid-stream/

[DaleCurtis, 2013/02/12 02:41:24]

Done.

+                    << " Sample Rate: " << av_frame_->sample_rate << " vs "
+                    << samples_per_second_
+                    << ", Channels: " << av_frame_->channels << " vs "
+                    << channels_
+                    << ", Sample Format: " << av_frame_->format << " vs "
+                    << av_sample_format_;
 
         // This is an unrecoverable error, so bail out.
         QueuedAudioBuffer queue_entry = { kDecodeError, NULL };
         queued_audio_.push_back(queue_entry);
         break;
       }
 
       decoded_audio_size = av_samples_get_buffer_size(
           NULL, codec_context_->channels, av_frame_->nb_samples,
           codec_context_->sample_fmt, 1);
@@ skipping 70 matching lines @@
     // Decoding finished successfully, update statistics.
     if (result > 0) {
       PipelineStatistics statistics;
       statistics.audio_bytes_decoded = result;
       statistics_cb_.Run(statistics);
     }
   } while (packet.size > 0);
 }
 
 }  // namespace media