media-internals: Differentiate error, info, and debug log messages

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 245 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(log_cb_)
+      MEDIA_LOG(DEBUG, log_cb_)
           << "Dropping audio frame which failed decode with timestamp: "
-          << buffer->timestamp().InMicroseconds() << " us, duration: "
-          << buffer->duration().InMicroseconds() << " us, packet size: "
-          << buffer->data_size() << " bytes";
+          << 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(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, log_cb_) << "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