Added a new OnMoreData() interface which will not feed the playout data to APM.

BUG=3147
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11059005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5895 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_device/include/audio_device_defines.h

@@ -101,13 +101,32 @@ public:
     // Method to pass the captured audio data to the specific VoE channel.
     // |voe_channel| is the id of the VoE channel which is the sink to the
     // capture data.
-    // TODO(xians): Make the interface pure virtual after libjingle
+    // TODO(xians): Remove this interface after Libjingle switches to
-    // has its implementation.
+    // PushCaptureData().
     virtual void OnData(int voe_channel, const void* audio_data,
                         int bits_per_sample, int sample_rate,
                         int number_of_channels,
                         int number_of_frames) {}
 
+    // Method to push the captured audio data to the specific VoE channel.
+    // The data will not undergo audio processing.
+    // |voe_channel| is the id of the VoE channel which is the sink to the
+    // capture data.
+    // TODO(xians): Make the interface pure virtual after Libjingle
+    // has its implementation.
+    virtual void PushCaptureData(int voe_channel, const void* audio_data,
+                                 int bits_per_sample, int sample_rate,
+                                 int number_of_channels,
+                                 int number_of_frames) {}
+
+    // Method to pull mixed render audio data from all active VoE channels.
+    // The data will not be passed as reference for audio processing internally.
+    // TODO(xians): Support getting the unmixed render data from specific VoE
+    // channel.
+    virtual void PullRenderData(int bits_per_sample, int sample_rate,
+                                int number_of_channels, int number_of_frames,
+                                void* audio_data) {}
+
 protected:
     virtual ~AudioTransport() {}
 };

webrtc/modules/audio_device/test/audio_device_test_api.cc

@@ -142,10 +142,14 @@ class AudioTransportAPI: public AudioTransport {
     return 0;
   }
 
-  virtual void OnData(int voe_channel, const void* audio_data,
+  virtual void PushCaptureData(int voe_channel, const void* audio_data,
-                      int bits_per_sample, int sample_rate,
+                               int bits_per_sample, int sample_rate,
-                      int number_of_channels,
+                               int number_of_channels,
-                      int number_of_frames) {}
+                               int number_of_frames) {}
+
+  virtual void PullRenderData(int bits_per_sample, int sample_rate,
+                              int number_of_channels, int number_of_frames,
+                              void* audio_data) {}
  private:
   uint32_t rec_count_;
   uint32_t play_count_;

webrtc/modules/audio_device/test/func_test_manager.cc

@@ -542,11 +542,16 @@ int AudioTransportImpl::OnDataAvailable(const int voe_channels[],
   return 0;
 }
 
-void AudioTransportImpl::OnData(int voe_channel,
+void AudioTransportImpl::PushCaptureData(int voe_channel,
-                                const void* audio_data,
+                                         const void* audio_data,
-                                int bits_per_sample, int sample_rate,
+                                         int bits_per_sample, int sample_rate,
-                                int number_of_channels,
+                                         int number_of_channels,
-                                int number_of_frames) {}
+                                         int number_of_frames) {}
+
+void AudioTransportImpl::PullRenderData(int bits_per_sample, int sample_rate,
+                                        int number_of_channels,
+                                        int number_of_frames,
+                                        void* audio_data) {}
 
 FuncTestManager::FuncTestManager() :
     _processThread(NULL),

webrtc/modules/audio_device/test/func_test_manager.h

@@ -131,10 +131,14 @@ public:
                                 bool key_pressed,
                                 bool need_audio_processing);
 
-    virtual void OnData(int voe_channel, const void* audio_data,
+    virtual void PushCaptureData(int voe_channel, const void* audio_data,
-                        int bits_per_sample, int sample_rate,
+                                 int bits_per_sample, int sample_rate,
-                        int number_of_channels,
+                                 int number_of_channels,
-                        int number_of_frames);
+                                 int number_of_frames);
+
+    virtual void PullRenderData(int bits_per_sample, int sample_rate,
+                                int number_of_channels, int number_of_frames,
+                                void* audio_data);
 
     AudioTransportImpl(AudioDeviceModule* audioDevice);
     ~AudioTransportImpl();

webrtc/voice_engine/output_mixer.cc

@@ -532,7 +532,7 @@ int OutputMixer::GetMixedAudio(int sample_rate_hz,
 }
 
 int32_t
-OutputMixer::DoOperationsOnCombinedSignal()
+OutputMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm)
 {
     if (_audioFrame.sample_rate_hz_ != _mixingFrequencyHz)
     {
@@ -565,10 +565,8 @@ OutputMixer::DoOperationsOnCombinedSignal()
     }
 
     // --- Far-end Voice Quality Enhancement (AudioProcessing Module)
-    // TODO(ajm): Check with VoEBase if |need_audio_processing| is false.
+    if (feed_data_to_apm)
-    // If so, we don't need to call this method and can avoid the subsequent
+      APMAnalyzeReverseStream();
-    // resampling. See: https://code.google.com/p/webrtc/issues/detail?id=3147
-    APMAnalyzeReverseStream();
 
     // --- External media processing
     {

webrtc/voice_engine/output_mixer.h

@@ -60,7 +60,7 @@ public:
 
     int32_t MixActiveChannels();
 
-    int32_t DoOperationsOnCombinedSignal();
+    int32_t DoOperationsOnCombinedSignal(bool feed_data_to_apm);
 
     int32_t SetMixabilityStatus(MixerParticipant& participant,
                                 bool mixable);

webrtc/voice_engine/voe_base_impl.cc

@@ -150,39 +150,18 @@ int32_t VoEBaseImpl::NeedMorePlayData(
         void* audioSamples,
         uint32_t& nSamplesOut)
 {
-    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
+  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "VoEBaseImpl::NeedMorePlayData(nSamples=%u, "
+               "VoEBaseImpl::NeedMorePlayData(nSamples=%u, "
-                     "nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)",
+               "nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)",
-                 nSamples, nBytesPerSample, nChannels, samplesPerSec);
+               nSamples, nBytesPerSample, nChannels, samplesPerSec);
 
-    assert(_shared->output_mixer() != NULL);
+  GetPlayoutData(static_cast<int>(samplesPerSec),
+                 static_cast<int>(nChannels),
+                 static_cast<int>(nSamples), true, audioSamples);
 
-    // TODO(andrew): if the device is running in mono, we should tell the mixer
+  nSamplesOut = _audioFrame.samples_per_channel_;
-    // here so that it will only request mono from AudioCodingModule.
-    // Perform mixing of all active participants (channel-based mixing)
-    _shared->output_mixer()->MixActiveChannels();
 
-    // Additional operations on the combined signal
+  return 0;
-    _shared->output_mixer()->DoOperationsOnCombinedSignal();
-
-    // Retrieve the final output mix (resampled to match the ADM)
-    _shared->output_mixer()->GetMixedAudio(samplesPerSec, nChannels,
-        &_audioFrame);
-
-    assert(static_cast<int>(nSamples) == _audioFrame.samples_per_channel_);
-    assert(samplesPerSec ==
-        static_cast<uint32_t>(_audioFrame.sample_rate_hz_));
-
-    // Deliver audio (PCM) samples to the ADM
-    memcpy(
-           (int16_t*) audioSamples,
-           (const int16_t*) _audioFrame.data_,
-           sizeof(int16_t) * (_audioFrame.samples_per_channel_
-                   * _audioFrame.num_channels_));
-
-    nSamplesOut = _audioFrame.samples_per_channel_;
-
-    return 0;
 }
 
 int VoEBaseImpl::OnDataAvailable(const int voe_channels[],
@@ -219,8 +198,8 @@ int VoEBaseImpl::OnDataAvailable(const int voe_channels[],
     // TODO(ajm): In the case where multiple channels are using the same codec
     // rate, this path needlessly does extra conversions. We should convert once
     // and share between channels.
-    OnData(voe_channels[i], audio_data, 16, sample_rate,
+    PushCaptureData(voe_channels[i], audio_data, 16, sample_rate,
-           number_of_channels, number_of_frames);
+                    number_of_channels, number_of_frames);
   }
 
   // Return 0 to indicate no need to change the volume.
@@ -231,6 +210,14 @@ void VoEBaseImpl::OnData(int voe_channel, const void* audio_data,
                          int bits_per_sample, int sample_rate,
                          int number_of_channels,
                          int number_of_frames) {
+  PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate,
+                  number_of_channels, number_of_frames);
+}
+
+void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data,
+                                  int bits_per_sample, int sample_rate,
+                                  int number_of_channels,
+                                  int number_of_frames) {
   voe::ChannelOwner ch = _shared->channel_manager().GetChannel(voe_channel);
   voe::Channel* channel_ptr = ch.channel();
   if (!channel_ptr)
@@ -246,6 +233,16 @@ void VoEBaseImpl::OnData(int voe_channel, const void* audio_data,
   }
 }
 
+void VoEBaseImpl::PullRenderData(int bits_per_sample, int sample_rate,
+                                 int number_of_channels, int number_of_frames,
+                                 void* audio_data) {
+  assert(bits_per_sample == 16);
+  assert(number_of_frames == static_cast<int>(sample_rate / 100));
+
+  GetPlayoutData(sample_rate, number_of_channels, number_of_frames, false,
+                 audio_data);
+}
+
 int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
@@ -1188,4 +1185,29 @@ int VoEBaseImpl::ProcessRecordedDataWithAPM(
   return 0;
 }
 
+void VoEBaseImpl::GetPlayoutData(int sample_rate, int number_of_channels,
+                                 int number_of_frames, bool feed_data_to_apm,
+                                 void* audio_data) {
+  assert(_shared->output_mixer() != NULL);
+
+  // TODO(andrew): if the device is running in mono, we should tell the mixer
+  // here so that it will only request mono from AudioCodingModule.
+  // Perform mixing of all active participants (channel-based mixing)
+  _shared->output_mixer()->MixActiveChannels();
+
+  // Additional operations on the combined signal
+  _shared->output_mixer()->DoOperationsOnCombinedSignal(feed_data_to_apm);
+
+  // Retrieve the final output mix (resampled to match the ADM)
+  _shared->output_mixer()->GetMixedAudio(sample_rate, number_of_channels,
+                                         &_audioFrame);
+
+  assert(number_of_frames == _audioFrame.samples_per_channel_);
+  assert(sample_rate == _audioFrame.sample_rate_hz_);
+
+  // Deliver audio (PCM) samples to the ADM
+  memcpy(audio_data, _audioFrame.data_,
+         sizeof(int16_t) * number_of_frames * number_of_channels);
+}
+
 }  // namespace webrtc

webrtc/voice_engine/voe_base_impl.h

@@ -106,6 +106,14 @@ public:
                         int bits_per_sample, int sample_rate,
                         int number_of_channels, int number_of_frames);
 
+    virtual void PushCaptureData(int voe_channel, const void* audio_data,
+                                 int bits_per_sample, int sample_rate,
+                                 int number_of_channels, int number_of_frames);
+
+    virtual void PullRenderData(int bits_per_sample, int sample_rate,
+                                int number_of_channels, int number_of_frames,
+                                void* audio_data);
+
     // AudioDeviceObserver
     virtual void OnErrorIsReported(ErrorCode error);
     virtual void OnWarningIsReported(WarningCode warning);
@@ -138,6 +146,10 @@ private:
                                    uint32_t volume,
                                    bool key_pressed);
 
+    void GetPlayoutData(int sample_rate, int number_of_channels,
+                        int number_of_frames, bool feed_data_to_apm,
+                        void* audio_data);
+
     int32_t AddBuildInfo(char* str) const;
     int32_t AddVoEVersion(char* str) const;
 

webrtc/voice_engine/voe_external_media_impl.cc

@@ -320,7 +320,7 @@ int VoEExternalMediaImpl::ExternalPlayoutGetData(
 
     // Retrieve mixed output at the specified rate
     shared_->output_mixer()->MixActiveChannels();
-    shared_->output_mixer()->DoOperationsOnCombinedSignal();
+    shared_->output_mixer()->DoOperationsOnCombinedSignal(true);
     shared_->output_mixer()->GetMixedAudio(samplingFreqHz, 1, &audioFrame);
 
     // Deliver audio (PCM) samples to the external sink