Merge AudioRendererImpl and AudioRendererBase; add NullAudioSink

content/renderer/media/webrtc_audio_device_impl.cc

Index: content/renderer/media/webrtc_audio_device_impl.cc
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index d37d0cf1156b9cfeb3ccd4c99599b4a763714ac7..4f1eb34c687fc56126eac9488785c097c5ec5c84 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -70,10 +70,10 @@ int32_t WebRtcAudioDeviceImpl::Release() {
   return ret;
 }
 
-size_t WebRtcAudioDeviceImpl::Render(
+int WebRtcAudioDeviceImpl::Render(
     const std::vector<float*>& audio_data,
-    size_t number_of_frames,
-    size_t audio_delay_milliseconds) {
+    int number_of_frames,
+    int audio_delay_milliseconds) {
   DCHECK_LE(number_of_frames, output_buffer_size());
 
   {
@@ -90,12 +90,12 @@ size_t WebRtcAudioDeviceImpl::Render(
     // Even if the hardware runs at 44.1kHz, we use 44.0 internally.
     samples_per_sec = 44000;
   }
-  uint32_t samples_per_10_msec = (samples_per_sec / 100);
+  int samples_per_10_msec = (samples_per_sec / 100);
   const int bytes_per_10_msec =
       channels * samples_per_10_msec * bytes_per_sample_;
 
   uint32_t num_audio_samples = 0;
-  size_t accumulated_audio_samples = 0;
+  int accumulated_audio_samples = 0;
 
   char* audio_byte_buffer = reinterpret_cast<char*>(output_buffer_.get());
 
@@ -136,8 +136,8 @@ void WebRtcAudioDeviceImpl::OnRenderError() {
 
 void WebRtcAudioDeviceImpl::Capture(
     const std::vector<float*>& audio_data,
-    size_t number_of_frames,
-    size_t audio_delay_milliseconds) {
+    int number_of_frames,
+    int audio_delay_milliseconds) {
   DCHECK_LE(number_of_frames, input_buffer_size());
 
   int output_delay_ms = 0;
@@ -166,7 +166,7 @@ void WebRtcAudioDeviceImpl::Capture(
   const int samples_per_10_msec = (samples_per_sec / 100);
   const int bytes_per_10_msec =
       channels * samples_per_10_msec * bytes_per_sample_;
-  size_t accumulated_audio_samples = 0;
+  int accumulated_audio_samples = 0;
 
   char* audio_byte_buffer = reinterpret_cast<char*>(input_buffer_.get());
 
@@ -333,8 +333,8 @@ int32_t WebRtcAudioDeviceImpl::Init() {
 
   ChannelLayout out_channel_layout = CHANNEL_LAYOUT_MONO;
   AudioParameters::Format in_format = AudioParameters::AUDIO_PCM_LINEAR;
-  size_t in_buffer_size = 0;
-  size_t out_buffer_size = 0;
+  int in_buffer_size = 0;
+  int out_buffer_size = 0;
 
   // TODO(henrika): factor out all platform specific parts in separate
   // functions. Code is a bit messy right now.