Changed mixing api and removed resampler

webrtc/modules/audio_mixer/audio_mixer.cc

 /*
  *  Copyright (c) 2012 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 141 matching lines @@
 int32_t AudioMixer::SetMixabilityStatus(MixerAudioSource& audio_source,
                                         bool mixable) {
   return _mixerModule.SetMixabilityStatus(&audio_source, mixable);
 }
 
 int32_t AudioMixer::SetAnonymousMixabilityStatus(MixerAudioSource& audio_source,
                                                  bool mixable) {
   return _mixerModule.SetAnonymousMixabilityStatus(&audio_source, mixable);
 }
 
-int32_t AudioMixer::MixActiveChannels() {
-  _mixerModule.Mix(&_audioFrame);
-  return 0;
-}
-
 int AudioMixer::GetSpeechOutputLevel(uint32_t& level) {
   int8_t currentLevel = _audioLevel.Level();
   level = static_cast<uint32_t>(currentLevel);
   WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
                "GetSpeechOutputLevel() => level=%u", level);
   return 0;
 }
 
 int AudioMixer::GetSpeechOutputLevelFullRange(uint32_t& level) {
   int16_t currentLevel = _audioLevel.LevelFullRange();
@@ skipping 187 matching lines @@
       "AudioMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%" PRIuS ")",
       sample_rate_hz, num_channels);
 
   // --- Record playout if enabled
   {
     rtc::CritScope cs(&_fileCritSect);
     if (_outputFileRecording && _outputFileRecorderPtr)
       _outputFileRecorderPtr->RecordAudioToFile(_audioFrame);
   }
 
-  frame->num_channels_ = num_channels;
-  frame->sample_rate_hz_ = sample_rate_hz;
-  // TODO(andrew): Ideally the downmixing would occur much earlier, in
-  // AudioCodingModule.
-  RemixAndResample(_audioFrame, &resampler_, frame);
+  _mixerModule.Mix(sample_rate_hz, num_channels, frame);
+
   return 0;
 }
 
 int32_t AudioMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm) {
   if (_audioFrame.sample_rate_hz_ != _mixingFrequencyHz) {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
                  "AudioMixer::DoOperationsOnCombinedSignal() => "
                  "mixing frequency = %d",
                  _audioFrame.sample_rate_hz_);
     _mixingFrequencyHz = _audioFrame.sample_rate_hz_;
@@ skipping 35 matching lines @@
     }
   }
 
   // --- Measure audio level (0-9) for the combined signal
   _audioLevel.ComputeLevel(_audioFrame);
 
   return 0;
 }
 }  // namespace voe
 }  // namespace webrtc