Guard against midstream audio configuration changes.

webkit/media/crypto/ppapi/ffmpeg_cdm_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 "webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.h"
 
 #include <algorithm>
 
 #include "base/logging.h"
 #include "media/base/audio_bus.h"
@@ skipping 69 matching lines @@
   }
 }
 
 FFmpegCdmAudioDecoder::FFmpegCdmAudioDecoder(cdm::Allocator* allocator)
     : is_initialized_(false),
       allocator_(allocator),
       codec_context_(NULL),
       av_frame_(NULL),
       bits_per_channel_(0),
       samples_per_second_(0),
+      channels_(0),
+      av_sample_format_(0),
       bytes_per_frame_(0),
       last_input_timestamp_(media::kNoTimestamp()),
       output_bytes_to_drop_(0) {
 }
 
 FFmpegCdmAudioDecoder::~FFmpegCdmAudioDecoder() {
   ReleaseFFmpegResources();
 }
 
 bool FFmpegCdmAudioDecoder::Initialize(const cdm::AudioDecoderConfig& config) {
@@ skipping 47 matching lines @@
   // Success!
   av_frame_ = avcodec_alloc_frame();
   bits_per_channel_ = config.bits_per_channel;
   samples_per_second_ = config.samples_per_second;
   bytes_per_frame_ = codec_context_->channels * bits_per_channel_ / 8;
   output_timestamp_helper_.reset(new media::AudioTimestampHelper(
       bytes_per_frame_, config.samples_per_second));
   serialized_audio_frames_.reserve(bytes_per_frame_ * samples_per_second_);
   is_initialized_ = true;
 
+  // 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 FFmpegCdmAudioDecoder::Deinitialize() {
   DVLOG(1) << "Deinitialize()";
   ReleaseFFmpegResources();
   is_initialized_ = false;
   ResetTimestampState();
 }
 
@@ skipping 95 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(timestamp);
       }
     }
 
     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_;
         return cdm::kDecodeError;
       }
 
       decoded_audio_size = av_samples_get_buffer_size(
           NULL, codec_context_->channels, av_frame_->nb_samples,
           codec_context_->sample_fmt, 1);
       // If we're decoding into float, adjust audio size.
       if (converter_bus_ && bits_per_channel_ / 8 != sizeof(float)) {
         DCHECK(codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT ||
                codec_context_->sample_fmt == AV_SAMPLE_FMT_FLTP);
@@ skipping 103 matching lines @@
   }
 }
 
 void FFmpegCdmAudioDecoder::SerializeInt64(int64 value) {
   int previous_size = serialized_audio_frames_.size();
   serialized_audio_frames_.resize(previous_size + sizeof(value));
   memcpy(&serialized_audio_frames_[0] + previous_size, &value, sizeof(value));
 }
 
 }  // namespace webkit_media