blob: 8b40360ae61504f4902ed7b57a0a29aa950b074a [file] [log] [blame]
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +00001/*
2 * Copyright (c) 2012 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// This sub-API supports the following functionalities:
12//
13// - Noise Suppression (NS).
14// - Automatic Gain Control (AGC).
15// - Echo Control (EC).
16// - Receiving side VAD, NS and AGC.
17// - Measurements of instantaneous speech, noise and echo levels.
18// - Generation of AP debug recordings.
19// - Detection of keyboard typing which can disrupt a voice conversation.
20//
21// Usage example, omitting error checking:
22//
23// using namespace webrtc;
24// VoiceEngine* voe = VoiceEngine::Create();
25// VoEBase* base = VoEBase::GetInterface();
26// VoEAudioProcessing* ap = VoEAudioProcessing::GetInterface(voe);
27// base->Init();
28// ap->SetEcStatus(true, kAgcAdaptiveAnalog);
29// ...
30// base->Terminate();
31// base->Release();
32// ap->Release();
33// VoiceEngine::Delete(voe);
34//
35#ifndef WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H
36#define WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H
37
henrikg@webrtc.org7b722642013-12-06 16:05:17 +000038#include <stdio.h>
39
pbos@webrtc.org471ae722013-05-21 13:52:32 +000040#include "webrtc/common_types.h"
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +000041
42namespace webrtc {
43
44class VoiceEngine;
45
46// VoERxVadCallback
47class WEBRTC_DLLEXPORT VoERxVadCallback
48{
49public:
50 virtual void OnRxVad(int channel, int vadDecision) = 0;
51
52protected:
53 virtual ~VoERxVadCallback() {}
54};
55
56// VoEAudioProcessing
57class WEBRTC_DLLEXPORT VoEAudioProcessing
58{
59public:
60 // Factory for the VoEAudioProcessing sub-API. Increases an internal
61 // reference counter if successful. Returns NULL if the API is not
62 // supported or if construction fails.
63 static VoEAudioProcessing* GetInterface(VoiceEngine* voiceEngine);
64
65 // Releases the VoEAudioProcessing sub-API and decreases an internal
66 // reference counter. Returns the new reference count. This value should
67 // be zero for all sub-API:s before the VoiceEngine object can be safely
68 // deleted.
69 virtual int Release() = 0;
70
71 // Sets Noise Suppression (NS) status and mode.
72 // The NS reduces noise in the microphone signal.
73 virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged) = 0;
74
75 // Gets the NS status and mode.
76 virtual int GetNsStatus(bool& enabled, NsModes& mode) = 0;
77
78 // Sets the Automatic Gain Control (AGC) status and mode.
79 // The AGC adjusts the microphone signal to an appropriate level.
80 virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged) = 0;
81
82 // Gets the AGC status and mode.
83 virtual int GetAgcStatus(bool& enabled, AgcModes& mode) = 0;
84
85 // Sets the AGC configuration.
86 // Should only be used in situations where the working environment
87 // is well known.
pbos@webrtc.orgca7a9a22013-05-14 08:31:39 +000088 virtual int SetAgcConfig(AgcConfig config) = 0;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +000089
90 // Gets the AGC configuration.
91 virtual int GetAgcConfig(AgcConfig& config) = 0;
92
93 // Sets the Echo Control (EC) status and mode.
94 // The EC mitigates acoustic echo where a user can hear their own
95 // speech repeated back due to an acoustic coupling between the
96 // speaker and the microphone at the remote end.
97 virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged) = 0;
98
99 // Gets the EC status and mode.
100 virtual int GetEcStatus(bool& enabled, EcModes& mode) = 0;
101
102 // Enables the compensation of clock drift between the capture and render
103 // streams by the echo canceller (i.e. only using EcMode==kEcAec). It will
104 // only be enabled if supported on the current platform; otherwise an error
105 // will be returned. Check if the platform is supported by calling
106 // |DriftCompensationSupported()|.
107 virtual int EnableDriftCompensation(bool enable) = 0;
108 virtual bool DriftCompensationEnabled() = 0;
109 static bool DriftCompensationSupported();
110
111 // Sets a delay |offset| in ms to add to the system delay reported by the
112 // OS, which is used by the AEC to synchronize far- and near-end streams.
113 // In some cases a system may introduce a delay which goes unreported by the
114 // OS, but which is known to the user. This method can be used to compensate
115 // for the unreported delay.
116 virtual void SetDelayOffsetMs(int offset) = 0;
117 virtual int DelayOffsetMs() = 0;
118
119 // Modifies settings for the AEC designed for mobile devices (AECM).
120 virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
121 bool enableCNG = true) = 0;
122
123 // Gets settings for the AECM.
124 virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG) = 0;
125
126 // Enables a high pass filter on the capture signal. This removes DC bias
127 // and low-frequency noise. Recommended to be enabled.
128 virtual int EnableHighPassFilter(bool enable) = 0;
129 virtual bool IsHighPassFilterEnabled() = 0;
130
131 // Sets status and mode of the receiving-side (Rx) NS.
132 // The Rx NS reduces noise in the received signal for the specified
133 // |channel|. Intended for advanced usage only.
134 virtual int SetRxNsStatus(int channel,
135 bool enable,
136 NsModes mode = kNsUnchanged) = 0;
137
138 // Gets status and mode of the receiving-side NS.
139 virtual int GetRxNsStatus(int channel,
140 bool& enabled,
141 NsModes& mode) = 0;
142
143 // Sets status and mode of the receiving-side (Rx) AGC.
144 // The Rx AGC adjusts the received signal to an appropriate level
145 // for the specified |channel|. Intended for advanced usage only.
146 virtual int SetRxAgcStatus(int channel,
147 bool enable,
148 AgcModes mode = kAgcUnchanged) = 0;
149
150 // Gets status and mode of the receiving-side AGC.
151 virtual int GetRxAgcStatus(int channel,
152 bool& enabled,
153 AgcModes& mode) = 0;
154
155 // Modifies the AGC configuration on the receiving side for the
156 // specified |channel|.
pbos@webrtc.orgca7a9a22013-05-14 08:31:39 +0000157 virtual int SetRxAgcConfig(int channel, AgcConfig config) = 0;
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000158
159 // Gets the AGC configuration on the receiving side.
160 virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;
161
162 // Registers a VoERxVadCallback |observer| instance and enables Rx VAD
163 // notifications for the specified |channel|.
164 virtual int RegisterRxVadObserver(int channel,
165 VoERxVadCallback &observer) = 0;
166
167 // Deregisters the VoERxVadCallback |observer| and disables Rx VAD
168 // notifications for the specified |channel|.
169 virtual int DeRegisterRxVadObserver(int channel) = 0;
170
171 // Gets the VAD/DTX activity for the specified |channel|.
172 // The returned value is 1 if frames of audio contains speech
173 // and 0 if silence. The output is always 1 if VAD is disabled.
174 virtual int VoiceActivityIndicator(int channel) = 0;
175
176 // Enables or disables the possibility to retrieve echo metrics and delay
177 // logging values during an active call. The metrics are only supported in
178 // AEC.
179 virtual int SetEcMetricsStatus(bool enable) = 0;
180
181 // Gets the current EC metric status.
182 virtual int GetEcMetricsStatus(bool& enabled) = 0;
183
184 // Gets the instantaneous echo level metrics.
185 virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP) = 0;
186
187 // Gets the EC internal |delay_median| and |delay_std| in ms between
188 // near-end and far-end. The values are calculated over the time period
189 // since the last GetEcDelayMetrics() call.
190 virtual int GetEcDelayMetrics(int& delay_median, int& delay_std) = 0;
191
192 // Enables recording of Audio Processing (AP) debugging information.
193 // The file can later be used for off-line analysis of the AP performance.
194 virtual int StartDebugRecording(const char* fileNameUTF8) = 0;
195
henrikg@webrtc.org7b722642013-12-06 16:05:17 +0000196 // Same as above but sets and uses an existing file handle. Takes ownership
197 // of |file_handle| and passes it on to the audio processing module.
198 virtual int StartDebugRecording(FILE* file_handle) = 0;
199
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000200 // Disables recording of AP debugging information.
201 virtual int StopDebugRecording() = 0;
202
203 // Enables or disables detection of disturbing keyboard typing.
204 // An error notification will be given as a callback upon detection.
205 virtual int SetTypingDetectionStatus(bool enable) = 0;
206
207 // Gets the current typing detection status.
208 virtual int GetTypingDetectionStatus(bool& enabled) = 0;
209
210 // Reports the lower of:
211 // * Time in seconds since the last typing event.
212 // * Time in seconds since the typing detection was enabled.
213 // Returns error if typing detection is disabled.
214 virtual int TimeSinceLastTyping(int &seconds) = 0;
215
216 // Optional setting of typing detection parameters
217 // Parameter with value == 0 will be ignored
218 // and left with default config.
219 // TODO(niklase) Remove default argument as soon as libJingle is updated!
220 virtual int SetTypingDetectionParameters(int timeWindow,
221 int costPerTyping,
222 int reportingThreshold,
223 int penaltyDecay,
224 int typeEventDelay = 0) = 0;
225
226 // Swaps the capture-side left and right audio channels when enabled. It
227 // only has an effect when using a stereo send codec. The setting is
228 // persistent; it will be applied whenever a stereo send codec is enabled.
229 //
230 // The swap is applied only to the captured audio, and not mixed files. The
231 // swap will appear in file recordings and when accessing audio through the
232 // external media interface.
233 virtual void EnableStereoChannelSwapping(bool enable) = 0;
234 virtual bool IsStereoChannelSwappingEnabled() = 0;
235
236protected:
237 VoEAudioProcessing() {}
238 virtual ~VoEAudioProcessing() {}
239};
240
pbos@webrtc.org3b89e102013-07-03 15:12:26 +0000241} // namespace webrtc
andrew@webrtc.orgb015cbe2012-10-22 18:19:23 +0000242
243#endif // WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H