Deprecate LogCB in favor of using MediaLog

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/callback_helpers.h"
 #include "base/single_thread_task_runner.h"
 #include "media/base/audio_buffer.h"
 #include "media/base/audio_bus.h"
@@ skipping 108 matching lines @@
   // reference to the AudioBuffer object.
   void* opaque = NULL;
   buffer.swap(reinterpret_cast<AudioBuffer**>(&opaque));
   frame->buf[0] = av_buffer_create(
       frame->data[0], buffer_size_in_bytes, ReleaseAudioBufferImpl, opaque, 0);
   return 0;
 }
 
 FFmpegAudioDecoder::FFmpegAudioDecoder(
     const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
-    const LogCB& log_cb)
+    const scoped_refptr<MediaLog>& media_log)
     : task_runner_(task_runner),
       state_(kUninitialized),
       av_sample_format_(0),
-      log_cb_(log_cb) {
+      media_log_(media_log) {
 }
 
 FFmpegAudioDecoder::~FFmpegAudioDecoder() {
   DCHECK(task_runner_->BelongsToCurrentThread());
 
   if (state_ != kUninitialized) {
     ReleaseFFmpegResources();
     ResetTimestampState();
   }
 }
@@ skipping 111 matching lines @@
     int frame_decoded = 0;
     const int result = avcodec_decode_audio4(
         codec_context_.get(), av_frame_.get(), &frame_decoded, &packet);
 
     if (result < 0) {
       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.";
 
-      MEDIA_LOG(DEBUG, log_cb_)
+      MEDIA_LOG(DEBUG, media_log_)
           << "Dropping audio frame which failed decode with timestamp: "
           << buffer->timestamp().InMicroseconds()
           << " us, duration: " << buffer->duration().InMicroseconds()
           << " us, packet size: " << buffer->data_size() << " bytes";
 
       break;
     }
 
     // Update packet size and data pointer in case we need to call the decoder
     // with the remaining bytes from this packet.
     packet.size -= result;
     packet.data += result;
 
     scoped_refptr<AudioBuffer> output;
     const int channels = DetermineChannels(av_frame_.get());
     if (frame_decoded) {
       if (av_frame_->sample_rate != config_.samples_per_second() ||
           channels != ChannelLayoutToChannelCount(config_.channel_layout()) ||
           av_frame_->format != av_sample_format_) {
         DLOG(ERROR) << "Unsupported midstream configuration change!"
                     << " Sample Rate: " << av_frame_->sample_rate << " vs "
                     << config_.samples_per_second()
                     << ", Channels: " << channels << " vs "
                     << ChannelLayoutToChannelCount(config_.channel_layout())
                     << ", Sample Format: " << av_frame_->format << " vs "
                     << av_sample_format_;
 
         if (config_.codec() == kCodecAAC &&
             av_frame_->sample_rate == 2 * config_.samples_per_second()) {
-          MEDIA_LOG(DEBUG, log_cb_) << "Implicit HE-AAC signalling is being"
-                                    << " used. Please use mp4a.40.5 instead of"
-                                    << " mp4a.40.2 in the mimetype.";
+          MEDIA_LOG(DEBUG, media_log_)
+              << "Implicit HE-AAC signalling is being"
+              << " used. Please use mp4a.40.5 instead of"
+              << " mp4a.40.2 in the mimetype.";
         }
         // This is an unrecoverable error, so bail out.
         av_frame_unref(av_frame_.get());
         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]));
@@ skipping 79 matching lines @@
 
   ResetTimestampState();
   return true;
 }
 
 void FFmpegAudioDecoder::ResetTimestampState() {
   discard_helper_->Reset(config_.codec_delay());
 }
 
 }  // namespace media