a8b97373d5
- Add an Initialize() overload to allow specification of format parameters. This is mainly useful for testing, but could be used in the cases where a consumer knows the format before the streams arrive. - Add a reverse_sample_rate_hz_ parameter to prepare for mismatched capture and render rates. There is no functional change as it is currently constrained to match the capture rate. - Fix a bug in the float dump: we need to use add_ rather than set_. - Add a debug dump test for both int and float interfaces. - Enable unpacking of float dumps. - Enable audioproc to read float dumps. - Move more shared functionality to test_utils.h, and generally tidy up a bit by consolidating repeated code. BUG=2894 TESTED=Verified that the output produced by the float debug dump test is correct. Processed the resulting debug dump file with audioproc and ensured that we get identical output. (This is crucial, as we need to be able to exactly reproduce online results offline.) R=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5676 4adac7df-926f-26a2-2b94-8c16560cd09d
164 lines
6.1 KiB
C++
164 lines
6.1 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 EchoCancellationImpl;
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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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explicit AudioProcessingImpl(const Config& config);
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virtual ~AudioProcessingImpl();
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// AudioProcessing methods.
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virtual int Initialize() OVERRIDE;
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virtual int Initialize(int sample_rate_hz,
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int reverse_sample_rate_hz,
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int num_input_channels,
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int num_output_channels,
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int num_reverse_channels) 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 split_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 void set_output_will_be_muted(bool muted) OVERRIDE;
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virtual bool output_will_be_muted() const OVERRIDE;
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virtual int ProcessStream(AudioFrame* frame) OVERRIDE;
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virtual int ProcessStream(float* const* data,
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int samples_per_channel,
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int sample_rate_hz,
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ChannelLayout input_layout,
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ChannelLayout output_layout) OVERRIDE;
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virtual int AnalyzeReverseStream(AudioFrame* frame) OVERRIDE;
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virtual int AnalyzeReverseStream(const float* const* data,
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int samples_per_channel,
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int sample_rate_hz,
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ChannelLayout layout) 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 bool was_stream_delay_set() 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 void set_stream_key_pressed(bool key_pressed) OVERRIDE;
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virtual bool stream_key_pressed() 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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protected:
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// Overridden in a mock.
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virtual int InitializeLocked();
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private:
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int InitializeLocked(int sample_rate_hz,
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int reverse_sample_rate_hz,
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int num_input_channels,
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int num_output_channels,
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int num_reverse_channels);
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int MaybeInitializeLocked(int sample_rate_hz,
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int reverse_sample_rate_hz,
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int num_input_channels,
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int num_output_channels,
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int num_reverse_channels);
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int ProcessStreamLocked();
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int AnalyzeReverseStreamLocked();
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bool is_data_processed() const;
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bool output_copy_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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EchoCancellationImpl* 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 reverse_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 reverse_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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bool output_will_be_muted_;
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bool key_pressed_;
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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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