Guard against midstream audio configuration changes.

media/filters/audio_file_reader.cc

Index: media/filters/audio_file_reader.cc
diff --git a/media/filters/audio_file_reader.cc b/media/filters/audio_file_reader.cc
index 58c4501b43f62b74d4a8021e64ab72708e4565a7..32627b04b10ad558bab4b4facc386da0a3c4552d 100644
--- a/media/filters/audio_file_reader.cc
+++ b/media/filters/audio_file_reader.cc
@@ -15,21 +15,16 @@ namespace media {
 AudioFileReader::AudioFileReader(FFmpegURLProtocol* protocol)
     : codec_context_(NULL),
       stream_index_(0),
-      protocol_(protocol) {
+      protocol_(protocol),
+      channels_(0),
+      sample_rate_(0),
+      av_sample_format_(0) {
 }
 
 AudioFileReader::~AudioFileReader() {
   Close();
 }
 
-int AudioFileReader::channels() const {
-  return codec_context_->channels;
-}
-
-int AudioFileReader::sample_rate() const {
-  return codec_context_->sample_rate;
-}
-
 base::TimeDelta AudioFileReader::duration() const {
   const AVRational av_time_base = {1, AV_TIME_BASE};
 
@@ -110,6 +105,11 @@ bool AudioFileReader::Open() {
     return false;
   }
 
+  // Store initial values to guard against midstream configuration changes.
+  channels_ = codec_context_->channels;
+  sample_rate_ = codec_context_->sample_rate;
+  av_sample_format_ = codec_context_->sample_fmt;
+
   return true;
 }
 
@@ -179,6 +179,22 @@ int AudioFileReader::Read(AudioBus* audio_bus) {
         break;
       }
 
+      if (av_frame->sample_rate != sample_rate_ ||
+          av_frame->channels != channels_ ||
+          av_frame->format != av_sample_format_) {
+        DLOG(ERROR) << "Unsupported midstream configuration change!"
+                    << " Sample Rate: " << av_frame->sample_rate << " vs "
+                    << sample_rate_
+                    << ", Channels: " << av_frame->channels << " vs "
+                    << channels_
+                    << ", Sample Format: " << av_frame->format << " vs "
+                    << av_sample_format_;
+
+        // This is an unrecoverable error, so bail out.
+        continue_decoding = false;
+        break;
+      }
+
       // Truncate, if necessary, if the destination isn't big enough.
       if (current_frame + frames_read > audio_bus->frames())
         frames_read = audio_bus->frames() - current_frame;