14092e00f1
adding new AEC dump start interface for chrome. This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here: http://msdn.microsoft.com/en-us/library/ms235460.aspx Chromium bug:crbug/415935 TEST=bots R=bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7337 4adac7df-926f-26a2-2b94-8c16560cd09d
204 lines
7.1 KiB
C++
204 lines
7.1 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 AudioRate {
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public:
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explicit AudioRate(int sample_rate_hz)
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: rate_(sample_rate_hz),
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samples_per_channel_(AudioProcessing::kChunkSizeMs * rate_ / 1000) {}
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virtual ~AudioRate() {}
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void set(int rate) {
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rate_ = rate;
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samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
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}
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int rate() const { return rate_; }
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int samples_per_channel() const { return samples_per_channel_; }
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private:
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int rate_;
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int samples_per_channel_;
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};
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class AudioFormat : public AudioRate {
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public:
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AudioFormat(int sample_rate_hz, int num_channels)
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: AudioRate(sample_rate_hz),
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num_channels_(num_channels) {}
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virtual ~AudioFormat() {}
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void set(int rate, int num_channels) {
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AudioRate::set(rate);
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num_channels_ = num_channels;
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}
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int num_channels() const { return num_channels_; }
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private:
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int num_channels_;
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};
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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 input_sample_rate_hz,
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int output_sample_rate_hz,
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int reverse_sample_rate_hz,
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ChannelLayout input_layout,
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ChannelLayout output_layout,
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ChannelLayout reverse_layout) OVERRIDE;
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virtual void SetExtraOptions(const Config& config) OVERRIDE;
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virtual int set_sample_rate_hz(int rate) OVERRIDE;
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virtual int input_sample_rate_hz() const OVERRIDE;
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virtual int sample_rate_hz() const OVERRIDE;
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virtual int proc_sample_rate_hz() const OVERRIDE;
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virtual int proc_split_sample_rate_hz() const 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 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(const float* const* src,
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int samples_per_channel,
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int input_sample_rate_hz,
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ChannelLayout input_layout,
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int output_sample_rate_hz,
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ChannelLayout output_layout,
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float* const* dest) 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 StartDebugRecording(rtc::PlatformFile 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 input_sample_rate_hz,
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int output_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 input_sample_rate_hz,
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int output_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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scoped_ptr<AudioBuffer> render_audio_;
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scoped_ptr<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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AudioFormat fwd_in_format_;
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AudioFormat fwd_proc_format_;
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AudioRate fwd_out_format_;
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AudioFormat rev_in_format_;
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AudioFormat rev_proc_format_;
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int split_rate_;
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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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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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