zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Wire up Beamformer in AudioProcessing

Browse Source 
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
This commit is contained in:
aluebs@webrtc.org
2014-12-19 19:57:34 +00:00
parent 8817256373
commit ae643ce280
4 changed files with 57 additions and 4 deletions
Download Patch File Download Diff File Expand all files Collapse all files

48
webrtc/modules/audio_processing/audio_processing_impl.cc
View File

@ -16,8 +16,8 @@
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
#include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/beamformer/beamformer.h"
#include "webrtc/modules/audio_processing/channel_buffer.h"
#include "webrtc/modules/audio_processing/common.h"
#include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
@ -27,6 +27,7 @@
#include "webrtc/modules/audio_processing/level_estimator_impl.h"
#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
#include "webrtc/modules/audio_processing/processing_component.h"
#include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
#include "webrtc/modules/audio_processing/voice_detection_impl.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/compile_assert.h"
@ -183,7 +184,8 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config)
#else
use_new_agc_(config.Get<ExperimentalAgc>().enabled),
#endif
transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled) {
transient_suppressor_enabled_(config.Get<ExperimentalNs>().enabled),
beamformer_enabled_(config.Get<Beamforming>().enabled) {
echo_cancellation_ = new EchoCancellationImpl(this, crit_);
component_list_.push_back(echo_cancellation_);
@ -265,6 +267,9 @@ int AudioProcessingImpl::Initialize(int input_sample_rate_hz,
}
int AudioProcessingImpl::InitializeLocked() {
const int fwd_audio_buffer_channels = beamformer_enabled_ ?
fwd_in_format_.num_channels() :
fwd_out_format_.num_channels();
render_audio_.reset(new AudioBuffer(rev_in_format_.samples_per_channel(),
rev_in_format_.num_channels(),
rev_proc_format_.samples_per_channel(),
@ -273,7 +278,7 @@ int AudioProcessingImpl::InitializeLocked() {
capture_audio_.reset(new AudioBuffer(fwd_in_format_.samples_per_channel(),
fwd_in_format_.num_channels(),
fwd_proc_format_.samples_per_channel(),
fwd_out_format_.num_channels(),
fwd_audio_buffer_channels,
fwd_out_format_.samples_per_channel()));
// Initialize all components.
@ -295,6 +300,8 @@ int AudioProcessingImpl::InitializeLocked() {
return err;
}
InitializeBeamformer();
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
int err = WriteInitMessage();
@ -392,7 +399,10 @@ int AudioProcessingImpl::MaybeInitializeLocked(int input_sample_rate_hz,
num_reverse_channels == rev_in_format_.num_channels()) {
return kNoError;
}
if (beamformer_enabled_ &&
(num_input_channels < 2 || num_output_channels > 1)) {
return kBadNumberChannelsError;
}
return InitializeLocked(input_sample_rate_hz,
output_sample_rate_hz,
reverse_sample_rate_hz,
@ -593,6 +603,18 @@ int AudioProcessingImpl::ProcessStreamLocked() {
ca->SplitIntoFrequencyBands();
}
#ifdef WEBRTC_BEAMFORMER
if (beamformer_enabled_) {
beamformer_->ProcessChunk(ca->split_channels_const_f(kBand0To8kHz),
ca->split_channels_const_f(kBand8To16kHz),
ca->num_channels(),
ca->samples_per_split_channel(),
ca->split_channels_f(kBand0To8kHz),
ca->split_channels_f(kBand8To16kHz));
ca->set_num_channels(1);
}
#endif
RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(ca));
RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(ca));
RETURN_ON_ERR(noise_suppression_->AnalyzeCaptureAudio(ca));
@ -894,6 +916,10 @@ VoiceDetection* AudioProcessingImpl::voice_detection() const {
}
bool AudioProcessingImpl::is_data_processed() const {
if (beamformer_enabled_) {
return true;
}
int enabled_count = 0;
std::list<ProcessingComponent*>::const_iterator it;
for (it = component_list_.begin(); it != component_list_.end(); it++) {
@ -966,6 +992,20 @@ int AudioProcessingImpl::InitializeTransient() {
return kNoError;
}
void AudioProcessingImpl::InitializeBeamformer() {
if (beamformer_enabled_) {
#ifdef WEBRTC_BEAMFORMER
// TODO(aluebs): Don't use a hard-coded microphone spacing.
beamformer_.reset(new Beamformer(kChunkSizeMs,
split_rate_,
fwd_in_format_.num_channels(),
0.05f));
#else
assert(false);
#endif
}
}
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
int AudioProcessingImpl::WriteMessageToDebugFile() {
int32_t size = event_msg_->ByteSize();