60730cfe3c
Instead have ProcessStream transparently handle changes to the stream audio parameters (sample rate and channels). This removes two locks per 10 ms ProcessStream call taken by VoiceEngine (four total with the audio level indicator.) Also, prepare future improvements by having the splitting filter take a length parameter. This will allow it to work at different sample rates. Remove the useless splitting_filter wrapper. TESTED=voe_cmd_test with audio processing enabled and switching between codecs; unit tests. R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
141 lines
4.7 KiB
C++
141 lines
4.7 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include <list>
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#include <string>
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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class AudioBuffer;
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class CriticalSectionWrapper;
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class EchoCancellationImplWrapper;
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class EchoControlMobileImpl;
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class FileWrapper;
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class GainControlImpl;
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class HighPassFilterImpl;
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class LevelEstimatorImpl;
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class NoiseSuppressionImpl;
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class ProcessingComponent;
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class VoiceDetectionImpl;
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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namespace audioproc {
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class Event;
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} // namespace audioproc
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#endif
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class AudioProcessingImpl : public AudioProcessing {
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public:
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enum {
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kSampleRate8kHz = 8000,
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kSampleRate16kHz = 16000,
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kSampleRate32kHz = 32000
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};
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AudioProcessingImpl();
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virtual ~AudioProcessingImpl();
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CriticalSectionWrapper* crit() const;
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int split_sample_rate_hz() const;
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bool was_stream_delay_set() const;
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// AudioProcessing methods.
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virtual int Initialize() OVERRIDE;
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virtual void SetExtraOptions(const Config& config) OVERRIDE;
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virtual int EnableExperimentalNs(bool enable) OVERRIDE;
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virtual bool experimental_ns_enabled() const OVERRIDE {
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return false;
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}
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virtual int set_sample_rate_hz(int rate) OVERRIDE;
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virtual int sample_rate_hz() const OVERRIDE;
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virtual int set_num_channels(int input_channels,
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int output_channels) OVERRIDE;
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virtual int num_input_channels() const OVERRIDE;
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virtual int num_output_channels() const OVERRIDE;
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virtual int set_num_reverse_channels(int channels) OVERRIDE;
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virtual int num_reverse_channels() const OVERRIDE;
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virtual int ProcessStream(AudioFrame* frame) OVERRIDE;
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virtual int AnalyzeReverseStream(AudioFrame* frame) OVERRIDE;
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virtual int set_stream_delay_ms(int delay) OVERRIDE;
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virtual int stream_delay_ms() const OVERRIDE;
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virtual void set_delay_offset_ms(int offset) OVERRIDE;
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virtual int delay_offset_ms() const OVERRIDE;
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virtual int StartDebugRecording(
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const char filename[kMaxFilenameSize]) OVERRIDE;
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virtual int StartDebugRecording(FILE* handle) OVERRIDE;
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virtual int StopDebugRecording() OVERRIDE;
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virtual EchoCancellation* echo_cancellation() const OVERRIDE;
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virtual EchoControlMobile* echo_control_mobile() const OVERRIDE;
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virtual GainControl* gain_control() const OVERRIDE;
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virtual HighPassFilter* high_pass_filter() const OVERRIDE;
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virtual LevelEstimator* level_estimator() const OVERRIDE;
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virtual NoiseSuppression* noise_suppression() const OVERRIDE;
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virtual VoiceDetection* voice_detection() const OVERRIDE;
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// Module methods.
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virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE;
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protected:
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virtual int InitializeLocked();
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private:
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int MaybeInitializeLocked(int sample_rate_hz, int num_input_channels,
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int num_output_channels, int num_reverse_channels);
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bool is_data_processed() const;
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bool interleave_needed(bool is_data_processed) const;
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bool synthesis_needed(bool is_data_processed) const;
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bool analysis_needed(bool is_data_processed) const;
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EchoCancellationImplWrapper* echo_cancellation_;
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EchoControlMobileImpl* echo_control_mobile_;
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GainControlImpl* gain_control_;
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HighPassFilterImpl* high_pass_filter_;
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LevelEstimatorImpl* level_estimator_;
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NoiseSuppressionImpl* noise_suppression_;
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VoiceDetectionImpl* voice_detection_;
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std::list<ProcessingComponent*> component_list_;
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CriticalSectionWrapper* crit_;
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AudioBuffer* render_audio_;
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AudioBuffer* capture_audio_;
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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// TODO(andrew): make this more graceful. Ideally we would split this stuff
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// out into a separate class with an "enabled" and "disabled" implementation.
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int WriteMessageToDebugFile();
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int WriteInitMessage();
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scoped_ptr<FileWrapper> debug_file_;
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scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
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std::string event_str_; // Memory for protobuf serialization.
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#endif
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int sample_rate_hz_;
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int split_sample_rate_hz_;
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int samples_per_channel_;
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int stream_delay_ms_;
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int delay_offset_ms_;
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bool was_stream_delay_set_;
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int num_reverse_channels_;
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int num_input_channels_;
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int num_output_channels_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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