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-rw-r--r--webrtc/modules/audio_device/include/audio_device_defines.h23
-rw-r--r--webrtc/modules/audio_device/test/audio_device_test_api.cc12
-rw-r--r--webrtc/modules/audio_device/test/func_test_manager.cc15
-rw-r--r--webrtc/modules/audio_device/test/func_test_manager.h12
-rw-r--r--webrtc/voice_engine/output_mixer.cc8
-rw-r--r--webrtc/voice_engine/output_mixer.h2
-rw-r--r--webrtc/voice_engine/voe_base_impl.cc86
-rw-r--r--webrtc/voice_engine/voe_base_impl.h12
-rw-r--r--webrtc/voice_engine/voe_external_media_impl.cc2
9 files changed, 118 insertions, 54 deletions
diff --git a/webrtc/modules/audio_device/include/audio_device_defines.h b/webrtc/modules/audio_device/include/audio_device_defines.h
index 9f3e24b10a..0704ea8321 100644
--- a/webrtc/modules/audio_device/include/audio_device_defines.h
+++ b/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
- // has its implementation.
+ // TODO(xians): Remove this interface after Libjingle switches to
+ // 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() {}
};
diff --git a/webrtc/modules/audio_device/test/audio_device_test_api.cc b/webrtc/modules/audio_device/test/audio_device_test_api.cc
index 7692b12a46..fb00e0a4e4 100644
--- a/webrtc/modules/audio_device/test/audio_device_test_api.cc
+++ b/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,
- 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) {}
private:
uint32_t rec_count_;
uint32_t play_count_;
diff --git a/webrtc/modules/audio_device/test/func_test_manager.cc b/webrtc/modules/audio_device/test/func_test_manager.cc
index 7024e2c6e4..30123d1ba8 100644
--- a/webrtc/modules/audio_device/test/func_test_manager.cc
+++ b/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,
- const void* audio_data,
- int bits_per_sample, int sample_rate,
- int number_of_channels,
- int number_of_frames) {}
+void AudioTransportImpl::PushCaptureData(int voe_channel,
+ const void* audio_data,
+ int bits_per_sample, int sample_rate,
+ int number_of_channels,
+ 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),
diff --git a/webrtc/modules/audio_device/test/func_test_manager.h b/webrtc/modules/audio_device/test/func_test_manager.h
index 6e21466e67..bd32f627ae 100644
--- a/webrtc/modules/audio_device/test/func_test_manager.h
+++ b/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,
- 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);
AudioTransportImpl(AudioDeviceModule* audioDevice);
~AudioTransportImpl();
diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc
index 1648a9d975..f5bf51e353 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/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 so, we don't need to call this method and can avoid the subsequent
- // resampling. See: https://code.google.com/p/webrtc/issues/detail?id=3147
- APMAnalyzeReverseStream();
+ if (feed_data_to_apm)
+ APMAnalyzeReverseStream();
// --- External media processing
{
diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h
index fc9afd8c54..bd17c3beca 100644
--- a/webrtc/voice_engine/output_mixer.h
+++ b/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);
diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index 1b6e14491e..41b230ea19 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/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),
- "VoEBaseImpl::NeedMorePlayData(nSamples=%u, "
- "nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)",
- nSamples, nBytesPerSample, nChannels, samplesPerSec);
-
- 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();
-
- // 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_));
+ WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
+ "VoEBaseImpl::NeedMorePlayData(nSamples=%u, "
+ "nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)",
+ nSamples, nBytesPerSample, nChannels, samplesPerSec);
- // 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_));
+ GetPlayoutData(static_cast<int>(samplesPerSec),
+ static_cast<int>(nChannels),
+ static_cast<int>(nSamples), true, audioSamples);
- nSamplesOut = _audioFrame.samples_per_channel_;
+ nSamplesOut = _audioFrame.samples_per_channel_;
- return 0;
+ 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,
- number_of_channels, number_of_frames);
+ PushCaptureData(voe_channels[i], audio_data, 16, sample_rate,
+ 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
diff --git a/webrtc/voice_engine/voe_base_impl.h b/webrtc/voice_engine/voe_base_impl.h
index df406eb93e..8a3973a894 100644
--- a/webrtc/voice_engine/voe_base_impl.h
+++ b/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;
diff --git a/webrtc/voice_engine/voe_external_media_impl.cc b/webrtc/voice_engine/voe_external_media_impl.cc
index c76c280b05..28d7c1a96b 100644
--- a/webrtc/voice_engine/voe_external_media_impl.cc
+++ b/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
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-30 13:21:36 +0000