MEDIA_LOG for large encoded timestamp gaps in decoder stream.

media/base/audio_converter.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.
 //
 // AudioConverter implementation.  Uses MultiChannelSincResampler for resampling
 // audio, ChannelMixer for channel mixing, and AudioPullFifo for buffering.
 //
 // Delay estimates are provided to InputCallbacks based on the frame delay
 // information reported via the resampler and FIFO units.
 
@@ skipping 172 matching lines @@
     CreateUnmixedAudioIfNecessary(dest->frames());
 
   AudioBus* const temp_dest = needs_downmix ? unmixed_audio_.get() : dest;
 
   // Sanity check our inputs.
   DCHECK_EQ(temp_dest->frames(), mixer_input_audio_bus_->frames());
   DCHECK_EQ(temp_dest->channels(), mixer_input_audio_bus_->channels());
 
   // Calculate the buffer delay for this callback.
   base::TimeDelta buffer_delay = initial_delay_;
+  LOG(ERROR) << "initial delay:" << initial_delay_.InMicroseconds();
   if (resampler_) {
-    buffer_delay += base::TimeDelta::FromMicroseconds(
+    base::TimeDelta resampler_delay = base::TimeDelta::FromMicroseconds(
         resampler_frame_delay_ * output_frame_duration_.InMicroseconds());
+    LOG(ERROR) << "resampler delay:" << resampler_delay.InMicroseconds()
+               << " frames:" << resampler_frame_delay_
+               << " output_dur:" << output_frame_duration_.InMicroseconds();
+    buffer_delay += resampler_delay;
   }
   if (audio_fifo_) {
-    buffer_delay += base::TimeDelta::FromMicroseconds(
+    base::TimeDelta fifo_delay = base::TimeDelta::FromMicroseconds(
         fifo_frame_delay * input_frame_duration_.InMicroseconds());
+    LOG(ERROR) << "fifo delay:" << fifo_delay.InMicroseconds()
+               << " frames:" << fifo_frame_delay
+               << " input_dur:" << input_frame_duration_.InMicroseconds();
+    buffer_delay += fifo_delay;
   }
 
   // If we only have a single input, avoid an extra copy.
   AudioBus* const provide_input_dest =
       transform_inputs_.size() == 1 ? temp_dest : mixer_input_audio_bus_.get();
 
   // Have each mixer render its data into an output buffer then mix the result.
   for (auto* input : transform_inputs_) {
     const float volume = input->ProvideInput(provide_input_dest, buffer_delay);
 
@@ skipping 39 matching lines @@
   else
     SourceCallback(0, dest);
 }
 
 void AudioConverter::CreateUnmixedAudioIfNecessary(int frames) {
   if (!unmixed_audio_ || unmixed_audio_->frames() != frames)
     unmixed_audio_ = AudioBus::Create(input_channel_count_, frames);
 }
 
 }  // namespace media