Test upload to see if the ADM is passed correctly to audio_state.

webrtc/audio/audio_state.cc

 /*
  *  Copyright (c) 2015 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/audio/audio_state.h"
 
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/modules/audio_device/include/audio_device_defines.h"
 
 namespace webrtc {
+
+namespace {
+class MiMAudioTransport : public AudioTransport {
+ public:
+  MiMAudioTransport(AudioTransport* voe_audio_transport)
+      : voe_audio_transport_(voe_audio_transport) {}
+
+  virtual ~MiMAudioTransport() {}
+
+  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 {
+    // Used by native apps to PUSH data to VoE (those that do not
+    // supply their own ADM).
+
+    // Just pass through.
+    return voe_audio_transport_->RecordedDataIsAvailable(
+        audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
+        totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
+  }
+  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 {
+    return voe_audio_transport_->NeedMorePlayData(
+        nSamples, nBytesPerSample, nChannels, samplesPerSec, audioSamples,
+        nSamplesOut, elapsed_time_ms, ntp_time_ms);
+  }
+
+  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 {
+    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 {
+    return voe_audio_transport_->PullRenderData(
+        bits_per_sample, sample_rate, number_of_channels, number_of_frames,
+        audio_data, elapsed_time_ms, ntp_time_ms);
+  }
+
+ private:
+  AudioTransport* voe_audio_transport_;
+};
+}  // namespace
+
 namespace internal {
 
 AudioState::AudioState(const AudioState::Config& config)
     : config_(config), voe_base_(config.voice_engine) {
   process_thread_checker_.DetachFromThread();
   // Only one AudioState should be created per VoiceEngine.
   RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
+
+  if (voe_base_->audio_transport()) {
+    mim_transport_ = new MiMAudioTransport(voe_base_->audio_transport());
+
+    // first reset the old one (which is VoE->audio_transport())
+    audio_device()->RegisterAudioCallback(nullptr);
+    audio_device()->RegisterAudioCallback(mim_transport_);
+  } else {
+    LOG(LS_INFO) << "fail fail fail!";
+  }
 }
 
 AudioState::~AudioState() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  // voe_base_->audio_device_module()->RegisterAudioCallback(nullptr);
   voe_base_->DeRegisterVoiceEngineObserver();
+  if (mim_transport_) {
+    delete static_cast<MiMAudioTransport*>(mim_transport_);
+  }
 }
 
 VoiceEngine* AudioState::voice_engine() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_.voice_engine;
 }
 
+AudioDeviceModule* AudioState::audio_device() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  // if (config_.audio_device_module) {
+  //   return config_.audio_device_module;
+  // }
+  RTC_CHECK(voe_base_->audio_device_module());
+  return voe_base_->audio_device_module();
+}
+
 bool AudioState::typing_noise_detected() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   rtc::CritScope lock(&crit_sect_);
   return typing_noise_detected_;
 }
 
 // Reference count; implementation copied from rtc::RefCountedObject.
 int AudioState::AddRef() const {
   return rtc::AtomicOps::Increment(&ref_count_);
 }
@@ skipping 22 matching lines @@
     typing_noise_detected_ = false;
   }
 }
 }  // namespace internal
 
 rtc::scoped_refptr<AudioState> AudioState::Create(
     const AudioState::Config& config) {
   return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
 }
 }  // namespace webrtc