Less lock acquisitions for AudioMixer.

webrtc/modules/audio_mixer/test/audio_mixer_unittest.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 194 matching lines @@
 TEST(AudioMixer, ParticipantSampleRate) {
   const std::unique_ptr<AudioMixer> mixer(AudioMixer::Create(kId));
 
   MockMixerAudioSource participant;
   ResetFrame(participant.fake_frame());
 
   EXPECT_EQ(0, mixer->SetMixabilityStatus(&participant, true));
   for (auto frequency : {8000, 16000, 32000, 48000}) {
     EXPECT_CALL(participant, GetAudioFrameWithMuted(_, frequency))
         .Times(Exactly(1));
+    participant.fake_frame()->sample_rate_hz_ = frequency;
+    participant.fake_frame()->samples_per_channel_ = frequency / 100;

[aleloi, 2016/08/30 15:13:34]

There are now RTC_DCHECKs for receiving correct rate and
samples/channel from audio sources.

     mixer->Mix(frequency, 1, &frame_for_mixing);
     EXPECT_EQ(frequency, frame_for_mixing.sample_rate_hz_);
   }
 }
 
 TEST(AudioMixer, ParticipantNumberOfChannels) {
   const std::unique_ptr<AudioMixer> mixer(AudioMixer::Create(kId));
 
   MockMixerAudioSource participant;
   ResetFrame(participant.fake_frame());
@@ skipping 181 matching lines @@
       kAudioSources, MixerAudioSource::AudioFrameInfo::kNormal);
   frame_info[0] = MixerAudioSource::AudioFrameInfo::kMuted;
   std::fill(frames[0].data_, frames[0].data_ + kDefaultSampleRateHz / 100,
             std::numeric_limits<int16_t>::max());
   std::vector<bool> expected_status(kAudioSources, true);
   expected_status[0] = false;
 
   MixAndCompare(frames, frame_info, expected_status);
 }
 }  // namespace webrtc