MediaStream audio rendering: Bypass audio processing for non-WebRTC cases.

media/base/audio_shifter.h

 // Copyright (c) 2014 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 MEDIA_BASE_AUDIO_SHIFTER_H
 #define MEDIA_BASE_AUDIO_SHIFTER_H
 
 #include <stddef.h>
 
 #include <deque>
@@ skipping 38 matching lines @@
   // seconds. The larger the value, the smoother and less audible
   // the transitions will be. (But it means that perfect audio
   // sync will take longer to achive.)
   // |rate| is audio frames per second, eg 48000.
   // |channels| is number of channels in input and output audio.
   // TODO(hubbe): Allow input rate and output rate to be different
   // since we're going to be resampling anyways.
   AudioShifter(base::TimeDelta max_buffer_size,
                base::TimeDelta clock_accuracy,
                base::TimeDelta adjustment_time,
-               size_t rate,
+               int rate,
                int channels);
   ~AudioShifter();
 
+  int sample_rate() const { return rate_; }
+  int channels() const { return channels_; }
+
   // Push Audio into the shifter. All inputs must have the same number of
   // channels, but bus size can vary. The playout time can be noisy and
   // does not have to line up perfectly with the number of samples pushed
   // so far. However, the playout_time in Push calls and Pull calls must
   // not diverge over time.
   // Given audio from an a microphone, a reasonable way to calculate
   // playout_time would be now + 30ms.
   // Ideally playout_time is some time in the future, in which case
   // the samples will be buffered until the approperiate time. If
   // playout_time is in the past, everything will still work, and we'll
   // try to keep the buffring to a minimum.
   void Push(scoped_ptr<AudioBus> input, base::TimeTicks playout_time);
 
   // Fills out |output| with samples. Tries to stretch/shrink the audio
   // to compensate for drift between input and output.
   // If called from an output device data pull, a reasonable way to
   // calculate playout_time would be now + audio pipeline delay.
   void Pull(AudioBus* output, base::TimeTicks playout_time);
 
-  // Flush audio (but leave timing info)
-  void Flush();
-
 private:
   void Zero(AudioBus* output);
   void ResamplerCallback(int frame_delay, AudioBus* destination);
 
   struct AudioQueueEntry {
     AudioQueueEntry(base::TimeTicks target_playout_time_,
                     scoped_ptr<AudioBus> audio_);
     ~AudioQueueEntry();
     base::TimeTicks target_playout_time;
     linked_ptr<AudioBus> audio;
   };
 
   typedef std::deque<AudioQueueEntry> AudioShifterQueue;
 
   // Set from constructor.
   const base::TimeDelta max_buffer_size_;
   const base::TimeDelta clock_accuracy_;
   const base::TimeDelta adjustment_time_;
-  const size_t rate_;
+  const int rate_;
+  const int channels_;
 
   // The clock smoothers are used to smooth out timestamps
   // and adjust for drift and inaccurate clocks.
   scoped_ptr<ClockSmoother> input_clock_smoother_;
   scoped_ptr<ClockSmoother> output_clock_smoother_;
 
   // Are we currently outputting data?
   bool running_;
 
   // Number of frames already consumed from |queue_|.
@@ skipping 18 matching lines @@
   // Resampler.
   MultiChannelResampler resampler_;
 
   // Current resampler ratio.
   double current_ratio_;
 };
 
 }  // namespace media
 
 #endif  // MEDIA_BASE_AUDIO_SHIFTER_H