Wire up Beamformer in AudioProcessing
R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/38449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7969 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -180,6 +180,7 @@
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],
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}],
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['rtc_use_openmax_dl==1', {
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'defines': ['WEBRTC_BEAMFORMER'],
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'sources': [
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'beamformer/beamformer.cc',
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'beamformer/beamformer.h',
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@ -16,8 +16,8 @@
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
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#include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/beamformer/beamformer.h"
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#include "webrtc/modules/audio_processing/channel_buffer.h"
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#include "webrtc/modules/audio_processing/common.h"
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#include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
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@ -27,6 +27,7 @@
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#include "webrtc/modules/audio_processing/level_estimator_impl.h"
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#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
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#include "webrtc/modules/audio_processing/processing_component.h"
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#include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
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#include "webrtc/modules/audio_processing/voice_detection_impl.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/system_wrappers/interface/compile_assert.h"
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@ -183,7 +184,8 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config)
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#else
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use_new_agc_(config.Get<ExperimentalAgc>().enabled),
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#endif
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transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled) {
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transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled),
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beamformer_enabled_(config.Get<Beamforming>().enabled) {
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echo_cancellation_ = new EchoCancellationImpl(this, crit_);
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component_list_.push_back(echo_cancellation_);
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@ -265,6 +267,9 @@ int AudioProcessingImpl::Initialize(int input_sample_rate_hz,
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}
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int AudioProcessingImpl::InitializeLocked() {
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const int fwd_audio_buffer_channels = beamformer_enabled_ ?
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fwd_in_format_.num_channels() :
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fwd_out_format_.num_channels();
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render_audio_.reset(new AudioBuffer(rev_in_format_.samples_per_channel(),
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rev_in_format_.num_channels(),
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rev_proc_format_.samples_per_channel(),
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@ -273,7 +278,7 @@ int AudioProcessingImpl::InitializeLocked() {
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capture_audio_.reset(new AudioBuffer(fwd_in_format_.samples_per_channel(),
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fwd_in_format_.num_channels(),
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fwd_proc_format_.samples_per_channel(),
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fwd_out_format_.num_channels(),
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fwd_audio_buffer_channels,
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fwd_out_format_.samples_per_channel()));
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// Initialize all components.
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@ -295,6 +300,8 @@ int AudioProcessingImpl::InitializeLocked() {
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return err;
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}
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InitializeBeamformer();
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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if (debug_file_->Open()) {
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int err = WriteInitMessage();
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@ -392,7 +399,10 @@ int AudioProcessingImpl::MaybeInitializeLocked(int input_sample_rate_hz,
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num_reverse_channels == rev_in_format_.num_channels()) {
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return kNoError;
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}
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if (beamformer_enabled_ &&
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(num_input_channels < 2 || num_output_channels > 1)) {
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return kBadNumberChannelsError;
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}
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return InitializeLocked(input_sample_rate_hz,
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output_sample_rate_hz,
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reverse_sample_rate_hz,
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@ -593,6 +603,18 @@ int AudioProcessingImpl::ProcessStreamLocked() {
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ca->SplitIntoFrequencyBands();
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}
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#ifdef WEBRTC_BEAMFORMER
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if (beamformer_enabled_) {
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beamformer_->ProcessChunk(ca->split_channels_const_f(kBand0To8kHz),
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ca->split_channels_const_f(kBand8To16kHz),
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ca->num_channels(),
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ca->samples_per_split_channel(),
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ca->split_channels_f(kBand0To8kHz),
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ca->split_channels_f(kBand8To16kHz));
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ca->set_num_channels(1);
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}
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#endif
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RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(ca));
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RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(ca));
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RETURN_ON_ERR(noise_suppression_->AnalyzeCaptureAudio(ca));
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@ -894,6 +916,10 @@ VoiceDetection* AudioProcessingImpl::voice_detection() const {
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}
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bool AudioProcessingImpl::is_data_processed() const {
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if (beamformer_enabled_) {
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return true;
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}
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int enabled_count = 0;
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std::list<ProcessingComponent*>::const_iterator it;
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for (it = component_list_.begin(); it != component_list_.end(); it++) {
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@ -966,6 +992,20 @@ int AudioProcessingImpl::InitializeTransient() {
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return kNoError;
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}
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void AudioProcessingImpl::InitializeBeamformer() {
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if (beamformer_enabled_) {
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#ifdef WEBRTC_BEAMFORMER
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// TODO(aluebs): Don't use a hard-coded microphone spacing.
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beamformer_.reset(new Beamformer(kChunkSizeMs,
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split_rate_,
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fwd_in_format_.num_channels(),
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0.05f));
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#else
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assert(false);
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#endif
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}
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}
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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int AudioProcessingImpl::WriteMessageToDebugFile() {
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int32_t size = event_msg_->ByteSize();
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@ -23,6 +23,7 @@ namespace webrtc {
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class AgcManagerDirect;
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class AudioBuffer;
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class Beamformer;
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class CriticalSectionWrapper;
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class EchoCancellationImpl;
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class EchoControlMobileImpl;
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@ -168,6 +169,7 @@ class AudioProcessingImpl : public AudioProcessing {
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bool analysis_needed(bool is_data_processed) const;
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int InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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int InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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EchoCancellationImpl* echo_cancellation_;
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EchoControlMobileImpl* echo_control_mobile_;
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@ -215,6 +217,8 @@ class AudioProcessingImpl : public AudioProcessing {
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bool transient_suppressor_enabled_;
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scoped_ptr<TransientSuppressor> transient_suppressor_;
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const bool beamformer_enabled_;
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scoped_ptr<Beamformer> beamformer_;
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};
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} // namespace webrtc
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@ -82,6 +82,14 @@ struct ExperimentalNs {
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bool enabled;
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};
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// Use to enable beamforming. Must be provided through the constructor. It will
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// have no impact if used with AudioProcessing::SetExtraOptions().
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struct Beamforming {
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Beamforming() : enabled(false) {}
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explicit Beamforming(bool enabled) : enabled(enabled) {}
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bool enabled;
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};
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static const int kAudioProcMaxNativeSampleRateHz = 32000;
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// The Audio Processing Module (APM) provides a collection of voice processing
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