Mixing audio with different latency requirements

content/renderer/media/webrtc_audio_renderer.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include <stdint.h>
 
 #include <map>
@@ skipping 55 matching lines @@
     void set_volume(float volume) {
       DCHECK(CalledOnValidThread());
       volume_ = volume;
     }
 
    private:
     bool playing_;
     float volume_;
   };
 
-
-  // Returns platform specific optimal buffer size for rendering audio.
-  static int GetOptimalBufferSize(int sample_rate, int hardware_buffer_size);
-
   WebRtcAudioRenderer(
       const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread,
       const blink::WebMediaStream& media_stream,
       int source_render_frame_id,
       int session_id,
       const std::string& device_id,
       const url::Origin& security_origin);
 
   // Initialize function called by clients like WebRtcAudioDeviceImpl.
   // Stop() has to be called before |source| is deleted.
@@ skipping 167 matching lines @@
   // Used for triggering new UMA histogram. Counts number of render
   // callbacks modulo |kNumCallbacksBetweenRenderTimeHistograms|.
   int render_callback_count_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_