Changed mixing api and removed resampler

webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.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.
  */
 
 #include "webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h"
 
 #include <algorithm>
 #include <functional>
 
 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
 #include "webrtc/modules/audio_mixer/include/audio_mixer_defines.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
 namespace {
 
 class SourceFrame {
  public:
   SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before)
       : audio_source_(p),
         audio_frame_(a),
         muted_(m),
@@ skipping 150 matching lines @@
 
   if (_limiter->gain_control()->enable_limiter(true) != _limiter->kNoError)
     return false;
 
   if (_limiter->gain_control()->Enable(true) != _limiter->kNoError)
     return false;
 
   return true;
 }
 
-void NewAudioConferenceMixerImpl::Mix(AudioFrame* audio_frame_for_mixing) {
+void NewAudioConferenceMixerImpl::Mix(int sample_rate,
+                                      size_t number_of_channels,
+                                      void* audio_data) {
+  AudioFrame* audio_frame_for_mixing =
+      reinterpret_cast<AudioFrame*>(audio_data);

[aleloi, 2016/08/05 09:52:22]

It's still an AudioFrame on the inside yet...

   size_t remainingAudioSourcesAllowedToMix = kMaximumAmountOfMixedAudioSources;
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   AudioFrameList mixList;
   AudioFrameList additionalFramesList;
   std::map<int, MixerAudioSource*> mixedAudioSourcesMap;
   {
     CriticalSectionScoped cs(_cbCrit.get());
 
     int32_t lowFreq = GetLowestMixingFrequency();
     // SILK can run in 12 kHz and 24 kHz. These frequencies are not
@@ skipping 26 matching lines @@
           }
           break;
         case 48000:
           if (OutputFrequency() != kFbInHz) {
             SetOutputFrequency(kFbInHz);
           }
           break;
         default:
           RTC_NOTREACHED();
           return;
       }

[aleloi2, 2016/08/06 10:13:49]

Fingers have been itching to rewrite this for months now :)

     }
 
     mixList = UpdateToMix(remainingAudioSourcesAllowedToMix);
     remainingAudioSourcesAllowedToMix -= mixList.size();
     GetAdditionalAudio(&additionalFramesList);
   }
 
   // TODO(aleloi): it might be better to decide the number of channels
   //                with an API instead of dynamically.
 
@@ skipping 20 matching lines @@
 
     if (audio_frame_for_mixing->samples_per_channel_ == 0) {
       // Nothing was mixed, set the audio samples to silence.
       audio_frame_for_mixing->samples_per_channel_ = _sampleSize;
       audio_frame_for_mixing->Mute();
     } else {
       // Only call the limiter if we have something to mix.
       LimitMixedAudio(audio_frame_for_mixing);
     }
   }
+

[aleloi, 2016/08/05 09:52:21]

Resampling copied from OutputMixer. It would be nice to know how often the code
enters the if statement.

+  if (audio_frame_for_mixing->num_channels_ != number_of_channels ||
+      audio_frame_for_mixing->sample_rate_hz_ != sample_rate) {
+    audio_frame_for_mixing->num_channels_ = number_of_channels;
+    audio_frame_for_mixing->sample_rate_hz_ = sample_rate;
+    // TODO(andrew): Ideally the downmixing would occur much earlier, in
+    // AudioCodingModule.
+    voe::RemixAndResample(*audio_frame_for_mixing, &resampler_,

[aleloi, 2016/08/05 09:52:21]

I *think* it's OK for RemixAndResample to put data in the same buffer. I will
double check that some test covers it, and upload a new test in another patch
set if it's not.

+                          audio_frame_for_mixing);
+  }
   return;
 }
 
 int32_t NewAudioConferenceMixerImpl::SetOutputFrequency(
     const Frequency& frequency) {
   CriticalSectionScoped cs(_crit.get());
 
   _outputFrequency = frequency;
   _sampleSize =
       static_cast<size_t>((_outputFrequency * kProcessPeriodicityInMs) / 1000);
@@ skipping 373 matching lines @@
 
   if (error != _limiter->kNoError) {
     WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
                  "Error from AudioProcessing: %d", error);
     RTC_NOTREACHED();
     return false;
   }
   return true;
 }
 }  // namespace webrtc