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

media/base/audio_shifter.cc

 // 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.
 
 #include "media/base/audio_shifter.h"
 
 #include <algorithm>
 #include <cmath>
 #include <utility>
 
@@ skipping 69 matching lines @@
     scoped_ptr<AudioBus> audio_) :
     target_playout_time(target_playout_time_),
     audio(audio_.release()) {
 }
 
 AudioShifter::AudioQueueEntry::~AudioQueueEntry() {}
 
 AudioShifter::AudioShifter(base::TimeDelta max_buffer_size,
                            base::TimeDelta clock_accuracy,
                            base::TimeDelta adjustment_time,
-                           size_t rate,
-                           int channels) :
-    max_buffer_size_(max_buffer_size),
-    clock_accuracy_(clock_accuracy),
-    adjustment_time_(adjustment_time),
-    rate_(rate),
-    input_clock_smoother_(new ClockSmoother(clock_accuracy)),
-    output_clock_smoother_(new ClockSmoother(clock_accuracy)),
-    running_(false),
-    position_(0),
-    previous_requested_samples_(0),
-    resampler_(channels, 1.0, 96,
-               base::Bind(&AudioShifter::ResamplerCallback,
-                          base::Unretained(this))),
-    current_ratio_(1.0) {
-}
+                           int rate,
+                           int channels)
+    : max_buffer_size_(max_buffer_size),
+      clock_accuracy_(clock_accuracy),
+      adjustment_time_(adjustment_time),
+      rate_(rate),
+      channels_(channels),
+      input_clock_smoother_(new ClockSmoother(clock_accuracy)),
+      output_clock_smoother_(new ClockSmoother(clock_accuracy)),
+      running_(false),
+      position_(0),
+      previous_requested_samples_(0),
+      resampler_(
+          channels,
+          1.0,
+          96,
+          base::Bind(&AudioShifter::ResamplerCallback, base::Unretained(this))),
+      current_ratio_(1.0) {}
 
 AudioShifter::~AudioShifter() {}
 
 void AudioShifter::Push(scoped_ptr<AudioBus> input,
                         base::TimeTicks playout_time) {
   if (!queue_.empty()) {
     playout_time = input_clock_smoother_->Smooth(
         playout_time,
         base::TimeDelta::FromSeconds(queue_.back().audio->frames()) / rate_);
   }
@@ skipping 147 matching lines @@
   if (pos < destination->frames()) {
     // Underflow
     running_ = false;
     position_ = 0;
     previous_playout_time_ = base::TimeTicks();
     bias_ = base::TimeDelta();
     destination->ZeroFramesPartial(pos, destination->frames() - pos);
   }
 }
 
-void AudioShifter::Flush() {
-  resampler_.Flush();
-  position_ = 0;
-  queue_.clear();
-  running_ = false;
-  previous_playout_time_ = base::TimeTicks();
-  bias_ = base::TimeDelta();
-}
-
 void AudioShifter::Zero(AudioBus* output) {
   output->Zero();
   running_ = false;
   previous_playout_time_ = base::TimeTicks();
   bias_ = base::TimeDelta();
 }
 
 }  // namespace media