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Merge audio_processing changes.

Browse Source 
R=aluebs@webrtc.org, bjornv@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2014-12-15 09:41:24 +00:00
parent fb108b5a28
commit 788acd17ad
108 changed files with 7822 additions and 51 deletions
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32
webrtc/modules/audio_processing/audio_processing_impl.h
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@ -8,28 +8,32 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include <list>
#include <string>
#include "webrtc/base/thread_annotations.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class AgcManagerDirect;
class AudioBuffer;
class CriticalSectionWrapper;
class EchoCancellationImpl;
class EchoControlMobileImpl;
class FileWrapper;
class GainControlImpl;
class GainControlForNewAgc;
class HighPassFilterImpl;
class LevelEstimatorImpl;
class NoiseSuppressionImpl;
class ProcessingComponent;
class TransientSuppressor;
class VoiceDetectionImpl;
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
@ -138,7 +142,7 @@ class AudioProcessingImpl : public AudioProcessing {
protected:
// Overridden in a mock.
virtual int InitializeLocked();
virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
private:
int InitializeLocked(int input_sample_rate_hz,
@ -146,20 +150,24 @@ class AudioProcessingImpl : public AudioProcessing {
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels);
int num_reverse_channels)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
int MaybeInitializeLocked(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels);
int ProcessStreamLocked();
int AnalyzeReverseStreamLocked();
int num_reverse_channels)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
bool is_data_processed() const;
bool output_copy_needed(bool is_data_processed) const;
bool synthesis_needed(bool is_data_processed) const;
bool analysis_needed(bool is_data_processed) const;
int InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
int InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
EchoCancellationImpl* echo_cancellation_;
EchoControlMobileImpl* echo_control_mobile_;
@ -168,6 +176,7 @@ class AudioProcessingImpl : public AudioProcessing {
LevelEstimatorImpl* level_estimator_;
NoiseSuppressionImpl* noise_suppression_;
VoiceDetectionImpl* voice_detection_;
scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
std::list<ProcessingComponent*> component_list_;
CriticalSectionWrapper* crit_;
@ -199,8 +208,15 @@ class AudioProcessingImpl : public AudioProcessing {
bool output_will_be_muted_;
bool key_pressed_;
// Only set through the constructor's Config parameter.
const bool use_new_agc_;
scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
bool transient_suppressor_enabled_;
scoped_ptr<TransientSuppressor> transient_suppressor_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_