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