AudioConverter: Express delay in frames rather than msec.

content/renderer/media/webrtc_local_audio_source_provider.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 "content/renderer/media/webrtc_local_audio_source_provider.h"
 
 #include "base/logging.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/base/audio_fifo.h"
 #include "media/base/audio_hardware_config.h"
@@ skipping 112 matching lines @@
     output_wrapper_->SetChannelData(i, audio_data[i]);
 
   base::AutoLock auto_lock(lock_);
   if (!audio_converter_)
     return;
 
   is_enabled_ = true;
   audio_converter_->Convert(output_wrapper_.get());
 }
 
-double WebRtcLocalAudioSourceProvider::ProvideInput(
-    media::AudioBus* audio_bus, base::TimeDelta buffer_delay) {
+double WebRtcLocalAudioSourceProvider::ProvideInput(media::AudioBus* audio_bus,
+                                                    uint32_t frames_delayed) {
   if (fifo_->frames() >= audio_bus->frames()) {
     fifo_->Consume(audio_bus, 0, audio_bus->frames());
   } else {
     audio_bus->Zero();
     DVLOG(1) << "WARNING: Underrun, FIFO has data " << fifo_->frames()
              << " samples but " << audio_bus->frames()
              << " samples are needed";
   }
 
   return 1.0;
 }
 
 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
     const media::AudioParameters& sink_params) {
   sink_params_ = sink_params;
 }
 
 }  // namespace content