1 /* 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_ 13 14 #include <list> 15 16 #include "audio_processing.h" 17 18 namespace webrtc { 19 class CriticalSectionWrapper; 20 class FileWrapper; 21 22 class AudioBuffer; 23 class EchoCancellationImpl; 24 class EchoControlMobileImpl; 25 class GainControlImpl; 26 class HighPassFilterImpl; 27 class LevelEstimatorImpl; 28 class NoiseSuppressionImpl; 29 class ProcessingComponent; 30 class VoiceDetectionImpl; 31 32 class AudioProcessingImpl : public AudioProcessing { 33 public: 34 enum { 35 kSampleRate8kHz = 8000, 36 kSampleRate16kHz = 16000, 37 kSampleRate32kHz = 32000 38 }; 39 40 explicit AudioProcessingImpl(int id); 41 virtual ~AudioProcessingImpl(); 42 43 CriticalSectionWrapper* crit() const; 44 45 int split_sample_rate_hz() const; 46 bool was_stream_delay_set() const; 47 48 // AudioProcessing methods. 49 virtual int Initialize(); 50 virtual int InitializeLocked(); 51 virtual int set_sample_rate_hz(int rate); 52 virtual int sample_rate_hz() const; 53 virtual int set_num_channels(int input_channels, int output_channels); 54 virtual int num_input_channels() const; 55 virtual int num_output_channels() const; 56 virtual int set_num_reverse_channels(int channels); 57 virtual int num_reverse_channels() const; 58 virtual int ProcessStream(AudioFrame* frame); 59 virtual int AnalyzeReverseStream(AudioFrame* frame); 60 virtual int set_stream_delay_ms(int delay); 61 virtual int stream_delay_ms() const; 62 virtual int StartDebugRecording(const char filename[kMaxFilenameSize]); 63 virtual int StopDebugRecording(); 64 virtual EchoCancellation* echo_cancellation() const; 65 virtual EchoControlMobile* echo_control_mobile() const; 66 virtual GainControl* gain_control() const; 67 virtual HighPassFilter* high_pass_filter() const; 68 virtual LevelEstimator* level_estimator() const; 69 virtual NoiseSuppression* noise_suppression() const; 70 virtual VoiceDetection* voice_detection() const; 71 72 // Module methods. 73 virtual WebRtc_Word32 Version(WebRtc_Word8* version, 74 WebRtc_UWord32& remainingBufferInBytes, 75 WebRtc_UWord32& position) const; 76 virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id); 77 78 private: 79 int id_; 80 81 EchoCancellationImpl* echo_cancellation_; 82 EchoControlMobileImpl* echo_control_mobile_; 83 GainControlImpl* gain_control_; 84 HighPassFilterImpl* high_pass_filter_; 85 LevelEstimatorImpl* level_estimator_; 86 NoiseSuppressionImpl* noise_suppression_; 87 VoiceDetectionImpl* voice_detection_; 88 89 std::list<ProcessingComponent*> component_list_; 90 91 FileWrapper* debug_file_; 92 CriticalSectionWrapper* crit_; 93 94 AudioBuffer* render_audio_; 95 AudioBuffer* capture_audio_; 96 97 int sample_rate_hz_; 98 int split_sample_rate_hz_; 99 int samples_per_channel_; 100 int stream_delay_ms_; 101 bool was_stream_delay_set_; 102 103 int num_render_input_channels_; 104 int num_capture_input_channels_; 105 int num_capture_output_channels_; 106 }; 107 } // namespace webrtc 108 109 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_ 110
