AudioConverter: Express delay in frames rather than msec.

chrome/renderer/media/cast_rtp_stream.cc

 // Copyright 2013 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 "chrome/renderer/media/cast_rtp_stream.h"
 
 #include <stdint.h>
 
 #include <algorithm>
 #include <memory>
@@ skipping 551 matching lines @@
             input_params_.sample_rate());
     converter_.reset(
         new media::AudioConverter(input_params_, output_params, false));
     converter_->AddInput(this);
     sample_frames_in_ = 0;
     sample_frames_out_ = 0;
   }
 
   // Called on real-time audio thread.
   double ProvideInput(media::AudioBus* audio_bus,
-                      base::TimeDelta buffer_delay) override {
+                      uint32_t frames_delayed) override {
     DCHECK(current_input_bus_);
     current_input_bus_->CopyTo(audio_bus);
     current_input_bus_ = nullptr;
     return 1.0;
   }
 
  private:
   const blink::WebMediaStreamTrack track_;
   const int output_channels_;
   const int output_sample_rate_;
@@ skipping 152 matching lines @@
   DCHECK(content::RenderThread::Get());
   DVLOG(1) << "CastRtpStream::DidEncounterError(" << message << ") = "
            << (IsAudio() ? "audio" : "video");
   // Save the WeakPtr first because the error callback might delete this object.
   base::WeakPtr<CastRtpStream> ptr = weak_factory_.GetWeakPtr();
   error_callback_.Run(message);
   base::ThreadTaskRunnerHandle::Get()->PostTask(
       FROM_HERE,
       base::Bind(&CastRtpStream::Stop, ptr));
 }