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      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