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
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index a2c26a3..af6a7ed 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -180,6 +180,7 @@
],
}],
['rtc_use_openmax_dl==1', {
+ 'defines': ['WEBRTC_BEAMFORMER'],
'sources': [
'beamformer/beamformer.cc',
'beamformer/beamformer.h',
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 3ce84fb..086380e 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -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 @@
#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::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 @@
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 @@
return err;
}
+ InitializeBeamformer();
+
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
int err = WriteInitMessage();
@@ -392,7 +399,10 @@
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 @@
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 @@
}
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 @@
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();
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index be70273..152d64c 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -23,6 +23,7 @@
class AgcManagerDirect;
class AudioBuffer;
+class Beamformer;
class CriticalSectionWrapper;
class EchoCancellationImpl;
class EchoControlMobileImpl;
@@ -168,6 +169,7 @@
bool analysis_needed(bool is_data_processed) const;
int InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
int InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+ void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
EchoCancellationImpl* echo_cancellation_;
EchoControlMobileImpl* echo_control_mobile_;
@@ -215,6 +217,8 @@
bool transient_suppressor_enabled_;
scoped_ptr<TransientSuppressor> transient_suppressor_;
+ const bool beamformer_enabled_;
+ scoped_ptr<Beamformer> beamformer_;
};
} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 1cca0ed..79340ae 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -82,6 +82,14 @@
bool enabled;
};
+// Use to enable beamforming. Must be provided through the constructor. It will
+// have no impact if used with AudioProcessing::SetExtraOptions().
+struct Beamforming {
+ Beamforming() : enabled(false) {}
+ explicit Beamforming(bool enabled) : enabled(enabled) {}
+ bool enabled;
+};
+
static const int kAudioProcMaxNativeSampleRateHz = 32000;
// The Audio Processing Module (APM) provides a collection of voice processing
