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

media/base/audio_shifter.cc

Index: media/base/audio_shifter.cc
diff --git a/media/base/audio_shifter.cc b/media/base/audio_shifter.cc
index f5aee2c0e6b81c0e80af6e1a028a92743876ff5a..c88af3ac2304f0247e403ae91348deba03fb60bf 100644
--- a/media/base/audio_shifter.cc
+++ b/media/base/audio_shifter.cc
@@ -87,22 +87,24 @@ 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),

[miu, 2016/01/27 05:14:14]

'git cl format' did this when I added |channels_| to the initializer list.

[tommi (sloooow) - chröme, 2016/02/01 20:32:24]

Acknowledged.

+      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() {}
 
@@ -270,15 +272,6 @@ void AudioShifter::ResamplerCallback(int frame_delay, AudioBus* destination) {
   }
 }
 
-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;