Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/audio_transport_proxy.h

Index: webrtc/audio/audio_transport_proxy.h
diff --git a/webrtc/audio/audio_transport_proxy.h b/webrtc/audio/audio_transport_proxy.h
new file mode 100644
index 0000000000000000000000000000000000000000..12576bd2f7ed4ab8511e38d1cf474f489515b476
--- /dev/null
+++ b/webrtc/audio/audio_transport_proxy.h
@@ -0,0 +1,119 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_
+#define WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_
+
+#include <algorithm>
+
+#include "webrtc/api/audio/audio_mixer.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/audio_device/include/audio_device_defines.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+class AudioTransportProxy : public AudioTransport {
+ public:
+  AudioTransportProxy(AudioTransport* voe_audio_transport,
+                      AudioProcessing* apm,
+                      AudioMixer* mixer)
+      : voe_audio_transport_(voe_audio_transport), apm_(apm), mixer_(mixer) {
+    RTC_DCHECK(mixer_);
+  }
+
+  virtual ~AudioTransportProxy() {}

[the sun, 2016/10/27 10:06:46]

override

+
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  const size_t nSamples,
+                                  const size_t nBytesPerSample,
+                                  const size_t nChannels,
+                                  const uint32_t samplesPerSec,
+                                  const uint32_t totalDelayMS,
+                                  const int32_t clockDrift,
+                                  const uint32_t currentMicLevel,
+                                  const bool keyPressed,
+                                  uint32_t& newMicLevel) override {
+    // Pass call through to original audio transport instance.
+    if (voe_audio_transport_) {
+      return voe_audio_transport_->RecordedDataIsAvailable(
+          audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
+          totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
+    } else {
+      LOG(LS_ERROR)
+          << "AudioTransport proxy doesn't know where to send recorded data: "
+             "no Audio Transport provided.";
+    }
+
+    return -1;
+  }
+  int32_t NeedMorePlayData(const size_t nSamples,
+                           const size_t nBytesPerSample,
+                           const size_t nChannels,
+                           const uint32_t samplesPerSec,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override {

[the sun, 2016/10/27 10:06:46]

RTC_DCHECK_EQ(2u, nBytesPerSample);
RTC_DCHECK(audioSamples);
etc
- Everything you can check, such as the requested nSamples and nChannels not
causing us to try reading outside the AudioFrame buffer.

(Similarly for all methods here.)

+    mixer_->Mix(static_cast<int>(samplesPerSec), static_cast<int>(nChannels),
+                &frame_for_mixing_);
+    *elapsed_time_ms = frame_for_mixing_.elapsed_time_ms_;
+    *ntp_time_ms = frame_for_mixing_.ntp_time_ms_;
+
+    if (apm_) {
+      apm_->ProcessReverseStream(&frame_for_mixing_);
+    } else {
+      LOG(LS_ERROR) << "NeedMorePlayData called, but no APM provided.";

[the sun, 2016/10/27 10:06:46]

In what context does this happen? Is the right way to fix the test/mock that
gives you a nullptr in audio_processing()?

+    }
+
+    // Deliver audio (PCM) samples to the ADM.
+    std::copy(frame_for_mixing_.data_,
+              frame_for_mixing_.data_ + nSamples * nChannels,
+              static_cast<int16_t*>(audioSamples));
+    nSamplesOut = frame_for_mixing_.samples_per_channel_;
+
+    return 0;
+  }
+
+  void PushCaptureData(int voe_channel,
+                       const void* audio_data,
+                       int bits_per_sample,
+                       int sample_rate,
+                       size_t number_of_channels,
+                       size_t number_of_frames) override {

[the sun, 2016/10/27 10:06:46]

Add comment here why this path is unreachable.

+    RTC_NOTREACHED();
+  }
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override {
+    mixer_->Mix(static_cast<int>(sample_rate),
+                static_cast<int>(number_of_channels), &frame_for_mixing_);
+    *elapsed_time_ms = frame_for_mixing_.elapsed_time_ms_;
+    *ntp_time_ms = frame_for_mixing_.ntp_time_ms_;
+
+    // Deliver audio (PCM) samples to the ADM.
+    std::copy(frame_for_mixing_.data_,
+              frame_for_mixing_.data_ + number_of_frames * number_of_channels,
+              static_cast<int16_t*>(audio_data));
+  }
+
+ private:
+  AudioTransport* voe_audio_transport_;
+  AudioProcessing* apm_;
+  AudioMixer* mixer_;
+  AudioFrame frame_for_mixing_;

[the sun, 2016/10/27 10:06:46]

DISALLOW_IMPLICIT...

+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_AUDIO_TRANSPORT_PROXY_H_