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Only adapt AGC when the desired signal is present

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Take the 50% quantile of the mask and compare it to certain threshold to determine if the desired signal is present. A hold is applied to avoid fast switching between states.
is_signal_present_ has been plotted and looks as expected. The AGC adaptation sounds promising, specially for the cases when the speaker fades in and out from the beam direction.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8078 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
aluebs@webrtc.org
2015-01-15 18:07:21 +00:00
parent 3e42a8a56a
commit d82f55d2a7
10 changed files with 310 additions and 115 deletions
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35
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -137,11 +137,16 @@ class GainControlForNewAgc : public GainControl, public VolumeCallbacks {
AudioProcessing* AudioProcessing::Create() {
Config config;
return Create(config);
return Create(config, nullptr);
}
AudioProcessing* AudioProcessing::Create(const Config& config) {
AudioProcessingImpl* apm = new AudioProcessingImpl(config);
return Create(config, nullptr);
}
AudioProcessing* AudioProcessing::Create(const Config& config,
Beamformer* beamformer) {
AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
if (apm->Initialize() != kNoError) {
delete apm;
apm = NULL;
@ -151,6 +156,10 @@ AudioProcessing* AudioProcessing::Create(const Config& config) {
}
AudioProcessingImpl::AudioProcessingImpl(const Config& config)
: AudioProcessingImpl(config, nullptr) {}
AudioProcessingImpl::AudioProcessingImpl(const Config& config,
Beamformer* beamformer)
: echo_cancellation_(NULL),
echo_control_mobile_(NULL),
gain_control_(NULL),
@ -181,6 +190,7 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config)
#endif
transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled),
beamformer_enabled_(config.Get<Beamforming>().enabled),
beamformer_(beamformer),
array_geometry_(config.Get<Beamforming>().array_geometry) {
echo_cancellation_ = new EchoCancellationImpl(this, crit_);
component_list_.push_back(echo_cancellation_);
@ -330,6 +340,11 @@ int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz,
num_reverse_channels > 2 || num_reverse_channels < 1) {
return kBadNumberChannelsError;
}
if (beamformer_enabled_ &&
(static_cast<size_t>(num_input_channels) != array_geometry_.size() ||
num_output_channels > 1)) {
return kBadNumberChannelsError;
}
fwd_in_format_.set(input_sample_rate_hz, num_input_channels);
fwd_out_format_.set(output_sample_rate_hz, num_output_channels);
@ -395,11 +410,6 @@ int AudioProcessingImpl::MaybeInitializeLocked(int input_sample_rate_hz,
num_reverse_channels == rev_in_format_.num_channels()) {
return kNoError;
}
if (beamformer_enabled_ &&
(static_cast<size_t>(num_input_channels) != array_geometry_.size() ||
num_output_channels > 1)) {
return kBadNumberChannelsError;
}
return InitializeLocked(input_sample_rate_hz,
output_sample_rate_hz,
reverse_sample_rate_hz,
@ -622,7 +632,9 @@ int AudioProcessingImpl::ProcessStreamLocked() {
RETURN_ON_ERR(echo_control_mobile_->ProcessCaptureAudio(ca));
RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(ca));
if (use_new_agc_ && gain_control_->is_enabled()) {
if (use_new_agc_ &&
gain_control_->is_enabled() &&
(!beamformer_enabled_ || beamformer_->is_target_present())) {
agc_manager_->Process(ca->split_bands_const(0)[kBand0To8kHz],
ca->samples_per_split_channel(),
split_rate_);
@ -990,9 +1002,10 @@ int AudioProcessingImpl::InitializeTransient() {
void AudioProcessingImpl::InitializeBeamformer() {
if (beamformer_enabled_) {
#ifdef WEBRTC_BEAMFORMER
beamformer_.reset(new Beamformer(kChunkSizeMs,
split_rate_,
array_geometry_));
if (!beamformer_) {
beamformer_.reset(new Beamformer(array_geometry_));
}
beamformer_->Initialize(kChunkSizeMs, split_rate_);
#else
assert(false);
#endif