Roll FFmpeg for M37.

media/filters/audio_file_reader.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/audio_file_reader.h"
 
 #include <cmath>
 
 #include "base/logging.h"
 #include "base/time/time.h"
@@ skipping 113 matching lines @@
   AVPacket packet;
   int current_frame = 0;
   bool continue_decoding = true;
 
   while (current_frame < audio_bus->frames() && continue_decoding &&
          ReadPacket(&packet)) {
     // Make a shallow copy of packet so we can slide packet.data as frames are
     // decoded from the packet; otherwise av_free_packet() will corrupt memory.
     AVPacket packet_temp = packet;
     do {
-      avcodec_get_frame_defaults(av_frame.get());
       int frame_decoded = 0;
       int result = avcodec_decode_audio4(
           codec_context_, av_frame.get(), &frame_decoded, &packet_temp);
 
       if (result < 0) {
         DLOG(WARNING)
             << "AudioFileReader::Read() : error in avcodec_decode_audio4() -"
             << result;
         break;
       }
 
       // Update packet size and data pointer in case we need to call the decoder
       // with the remaining bytes from this packet.
       packet_temp.size -= result;
       packet_temp.data += result;
 
       if (!frame_decoded)
         continue;
 
       // Determine the number of sample-frames we just decoded.  Check overflow.
       int frames_read = av_frame->nb_samples;
       if (frames_read < 0) {
         continue_decoding = false;
+        av_frame_unref(av_frame.get());
         break;
       }
 
 #ifdef CHROMIUM_NO_AVFRAME_CHANNELS
       int channels = av_get_channel_layout_nb_channels(
           av_frame->channel_layout);
 #else
       int channels = av_frame->channels;
 #endif
       if (av_frame->sample_rate != sample_rate_ ||
           channels != channels_ ||
           av_frame->format != av_sample_format_) {
         DLOG(ERROR) << "Unsupported midstream configuration change!"
                     << " Sample Rate: " << av_frame->sample_rate << " vs "
                     << sample_rate_
                     << ", Channels: " << channels << " vs "
                     << channels_
                     << ", Sample Format: " << av_frame->format << " vs "
                     << av_sample_format_;
 
         // This is an unrecoverable error, so bail out.
         continue_decoding = false;
+        av_frame_unref(av_frame.get());
         break;
       }
 
       // Truncate, if necessary, if the destination isn't big enough.
       if (current_frame + frames_read > audio_bus->frames()) {
         DLOG(ERROR) << "Truncating decoded data due to output size.";
         frames_read = audio_bus->frames() - current_frame;
       }
 
       // Deinterleave each channel and convert to 32bit floating-point with
@@ skipping 13 matching lines @@
         for (int ch = 0; ch < audio_bus->channels(); ++ch) {
           memcpy(audio_bus->channel(ch) + current_frame,
                  av_frame->extended_data[ch], sizeof(float) * frames_read);
         }
       } else {
         audio_bus->FromInterleavedPartial(
             av_frame->data[0], current_frame, frames_read, bytes_per_sample);
       }
 
       current_frame += frames_read;
+      av_frame_unref(av_frame.get());
     } while (packet_temp.size > 0);
     av_free_packet(&packet);
   }
 
   // Zero any remaining frames.
   audio_bus->ZeroFramesPartial(
       current_frame, audio_bus->frames() - current_frame);
 
   // Returns the actual number of sample-frames decoded.
   // Ideally this represents the "true" exact length of the file.
@@ skipping 26 matching lines @@
     if (output_packet->stream_index != stream_index_) {
       av_free_packet(output_packet);
       continue;
     }
     return true;
   }
   return false;
 }
 
 }  // namespace media