17e40641b3
This is mainly to support the native audio format in Chrome. Although this implementation just moves the float->int conversion under the hood, we will transition AudioProcessing towards supporting this format throughout. - Add a test which verifies we get identical output with the float and int interfaces. - The float and int wrappers are tasked with conversion to the AudioBuffer format. A new shared Process/Analyze method does most of the work. - Add a new field to the debug.proto to hold deinterleaved data. - Add helpers to audio_utils.cc, and start using numeric_limits. - Note that there was no performance difference between numeric_limits and a literal value when measured on Linux using gcc or clang. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, henrikg@webrtc.org, tommi@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5641 4adac7df-926f-26a2-2b94-8c16560cd09d
148 lines
5.4 KiB
C++
148 lines
5.4 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 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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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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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 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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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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