Use the AudioProcessing float interface in MediaStreamAudioProcessor.

content/renderer/media/webrtc_audio_capturer_unittest.cc

 // Copyright 2013 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 "base/command_line.h"
 #include "base/logging.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
@@ skipping 59 matching lines @@
 };
 
 }  // namespace
 
 class WebRtcAudioCapturerTest : public testing::Test {
  protected:
   WebRtcAudioCapturerTest()
 #if defined(OS_ANDROID)
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) {
     // Android works with a buffer size bigger than 20ms.

[ajm, 2014/08/14 03:33:05]

Android using 20ms frames.

 #else
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) {
 #endif
   }
 
   void DisableAudioTrackProcessing() {
     CommandLine::ForCurrentProcess()->AppendSwitch(
         switches::kDisableAudioTrackProcessing);
   }
@@ skipping 32 matching lines @@
     scoped_ptr<media::AudioBus> audio_bus = media::AudioBus::Create(params_);
     audio_bus->Zero();
 
     media::AudioCapturerSource::CaptureCallback* callback =
         static_cast<media::AudioCapturerSource::CaptureCallback*>(capturer_);
 
     // Verify the sink is getting the correct values.
     EXPECT_CALL(*sink, FormatIsSet());
     EXPECT_CALL(*sink,
                 OnDataCallback(_, _, delay_ms, expected_volume_value,
-                               need_audio_processing, key_pressed));
+                               need_audio_processing, key_pressed))
+        .Times(AtLeast(1));

[ajm, 2014/08/14 03:33:05]

The error is that this was getting called more than once on Android. I think
that's as it should be actually and the previous behavior is suboptimal. This
test on Android happens to be using 20ms frames (see above). Since
MediaStreamAudioProcessor is designed to divide the input into 10ms chunks, we
should actually be getting two 10ms callbacks on Android.

I think this wasn't happening before due to buffering in the resampler. On the
first pass, it won't provide as much output as it gets input, because it
requires a bit of delay for the filter kernel. MSAP is now just using a simple
FIFO and doesn't require the delay.

     callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed);
 
     // Verify the cached values in the capturer fits what we expect.
     base::TimeDelta cached_delay;
     int cached_volume = !expected_volume_value;
     bool cached_key_pressed = !key_pressed;
     capturer_->GetAudioProcessingParams(&cached_delay, &cached_volume,
                                         &cached_key_pressed);
     EXPECT_EQ(cached_delay.InMilliseconds(), delay_ms);
     EXPECT_EQ(cached_volume, expected_volume_value);
@@ skipping 38 matching lines @@
                                "", "", params_.sample_rate(),
                                params_.channel_layout(),
                                params_.frames_per_buffer()),
           constraint_factory.CreateWebMediaConstraints(), NULL, NULL)
   );
   EXPECT_TRUE(capturer == NULL);
 }
 
 
 }  // namespace content