Cast: Basic cast_receiver API for chrome.

content/renderer/media/webrtc_audio_capturer.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 "content/renderer/media/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
@@ skipping 183 matching lines @@
   media::AudioSampleRate asr;
   if (media::ToAudioSampleRate(device_info_.device.input.sample_rate, &asr)) {
     UMA_HISTOGRAM_ENUMERATION(
         "WebRTC.AudioInputSampleRate", asr, media::kAudioSampleRateMax + 1);
   } else {
     UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputSampleRateUnexpected",
                          device_info_.device.input.sample_rate);
   }
 
   // Create and configure the default audio capturing source.
-  SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id_),
-                    channel_layout,
-                    static_cast<float>(device_info_.device.input.sample_rate));
+  SetCapturerSourceInternal(
+      AudioDeviceFactory::NewInputDevice(render_view_id_),
+      channel_layout,
+      static_cast<float>(device_info_.device.input.sample_rate));
 
   // Add the capturer to the WebRtcAudioDeviceImpl since it needs some hardware
   // information from the capturer.
   if (audio_device_)
     audio_device_->AddAudioCapturer(this);
 
   return true;
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer(
@@ skipping 61 matching lines @@
   if (stop_source) {
     // Since WebRtcAudioCapturer does not inherit MediaStreamAudioSource,
     // and instead MediaStreamAudioSource is composed of a WebRtcAudioCapturer,
     // we have to call StopSource on the MediaStreamSource. This will call
     // MediaStreamAudioSource::DoStopSource which in turn call
     // WebRtcAudioCapturerer::Stop();
     audio_source_->StopSource();
   }
 }
 
-void WebRtcAudioCapturer::SetCapturerSource(
+void WebRtcAudioCapturer::SetCapturerSourceInternal(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::ChannelLayout channel_layout,
     float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
            << "sample_rate=" << sample_rate << ")";
   scoped_refptr<media::AudioCapturerSource> old_source;
   {
     base::AutoLock auto_lock(lock_);
     if (source_.get() == source.get())
@@ skipping 58 matching lines @@
   }
 
   // Do nothing if the current buffer size is the WebRtc native buffer size.
   if (GetBufferSize(input_params.sample_rate()) ==
           input_params.frames_per_buffer()) {
     return;
   }
 
   // Create a new audio stream as source which will open the hardware using
   // WebRtc native buffer size.
-  SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
-                    input_params.channel_layout(),
-                    static_cast<float>(input_params.sample_rate()));
+  SetCapturerSourceInternal(AudioDeviceFactory::NewInputDevice(render_view_id),
+                            input_params.channel_layout(),
+                            static_cast<float>(input_params.sample_rate()));
 }
 
 void WebRtcAudioCapturer::Start() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::Start()";
   base::AutoLock auto_lock(lock_);
   if (running_ || !source_.get())
     return;
 
   // Start the data source, i.e., start capturing data from the current source.
@@ skipping 198 matching lines @@
       hardware_buffer_size <= peer_connection_buffer_size &&
       !audio_processor_->has_audio_processing()) {
     DVLOG(1) << "WebRtcAudioCapturer is using hardware buffer size "
              << hardware_buffer_size;
     return hardware_buffer_size;
   }
 
   return (sample_rate / 100);
 }
 
-void WebRtcAudioCapturer::SetCapturerSourceForTesting(
+void WebRtcAudioCapturer::SetCapturerSource(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::AudioParameters params) {
   // Create a new audio stream as source which uses the new source.
-  SetCapturerSource(source, params.channel_layout(),
-                    static_cast<float>(params.sample_rate()));
+  SetCapturerSourceInternal(source, params.channel_layout(),
+                            static_cast<float>(params.sample_rate()));
 }
 
 }  // namespace content