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 "audio_processing.h" 15 16 #include <list> 17 #include <string> 18 19 #include "scoped_ptr.h" 20 21 namespace webrtc { 22 class AudioBuffer; 23 class CriticalSectionWrapper; 24 class EchoCancellationImpl; 25 class EchoControlMobileImpl; 26 class FileWrapper; 27 class GainControlImpl; 28 class HighPassFilterImpl; 29 class LevelEstimatorImpl; 30 class NoiseSuppressionImpl; 31 class ProcessingComponent; 32 class VoiceDetectionImpl; 33 34 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 35 namespace audioproc { 36 37 class Event; 38 39 } // namespace audioproc 40 #endif 41 42 class AudioProcessingImpl : public AudioProcessing { 43 public: 44 enum { 45 kSampleRate8kHz = 8000, 46 kSampleRate16kHz = 16000, 47 kSampleRate32kHz = 32000 48 }; 49 50 explicit AudioProcessingImpl(int id); 51 virtual ~AudioProcessingImpl(); 52 53 CriticalSectionWrapper* crit() const; 54 55 int split_sample_rate_hz() const; 56 bool was_stream_delay_set() const; 57 58 // AudioProcessing methods. 59 virtual int Initialize(); 60 virtual int InitializeLocked(); 61 virtual int set_sample_rate_hz(int rate); 62 virtual int sample_rate_hz() const; 63 virtual int set_num_channels(int input_channels, int output_channels); 64 virtual int num_input_channels() const; 65 virtual int num_output_channels() const; 66 virtual int set_num_reverse_channels(int channels); 67 virtual int num_reverse_channels() const; 68 virtual int ProcessStream(AudioFrame* frame); 69 virtual int AnalyzeReverseStream(AudioFrame* frame); 70 virtual int set_stream_delay_ms(int delay); 71 virtual int stream_delay_ms() const; 72 virtual int StartDebugRecording(const char filename[kMaxFilenameSize]); 73 virtual int StopDebugRecording(); 74 virtual EchoCancellation* echo_cancellation() const; 75 virtual EchoControlMobile* echo_control_mobile() const; 76 virtual GainControl* gain_control() const; 77 virtual HighPassFilter* high_pass_filter() const; 78 virtual LevelEstimator* level_estimator() const; 79 virtual NoiseSuppression* noise_suppression() const; 80 virtual VoiceDetection* voice_detection() const; 81 82 // Module methods. 83 virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id); 84 85 private: 86 bool stream_data_changed() const; 87 bool synthesis_needed(bool stream_data_changed) const; 88 bool analysis_needed(bool stream_data_changed) const; 89 90 int id_; 91 92 EchoCancellationImpl* echo_cancellation_; 93 EchoControlMobileImpl* echo_control_mobile_; 94 GainControlImpl* gain_control_; 95 HighPassFilterImpl* high_pass_filter_; 96 LevelEstimatorImpl* level_estimator_; 97 NoiseSuppressionImpl* noise_suppression_; 98 VoiceDetectionImpl* voice_detection_; 99 100 std::list<ProcessingComponent*> component_list_; 101 CriticalSectionWrapper* crit_; 102 AudioBuffer* render_audio_; 103 AudioBuffer* capture_audio_; 104 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP 105 // TODO(andrew): make this more graceful. Ideally we would split this stuff 106 // out into a separate class with an "enabled" and "disabled" implementation. 107 int WriteMessageToDebugFile(); 108 int WriteInitMessage(); 109 scoped_ptr<FileWrapper> debug_file_; 110 scoped_ptr<audioproc::Event> event_msg_; // Protobuf message. 111 std::string event_str_; // Memory for protobuf serialization. 112 #endif 113 114 int sample_rate_hz_; 115 int split_sample_rate_hz_; 116 int samples_per_channel_; 117 int stream_delay_ms_; 118 bool was_stream_delay_set_; 119 120 int num_reverse_channels_; 121 int num_input_channels_; 122 int num_output_channels_; 123 }; 124 } // namespace webrtc 125 126 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_ 127