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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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82 | 82 |
83 // Starts AEC dump using an existing file. A maximum file size in bytes can be | 83 // Starts AEC dump using an existing file. A maximum file size in bytes can be |
84 // specified. When the maximum file size is reached, logging is stopped and | 84 // specified. When the maximum file size is reached, logging is stopped and |
85 // the file is closed. If max_size_bytes is set to <= 0, no limit will be | 85 // the file is closed. If max_size_bytes is set to <= 0, no limit will be |
86 // used. | 86 // used. |
87 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); | 87 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); |
88 | 88 |
89 // Stops AEC dump. | 89 // Stops AEC dump. |
90 void StopAecDump(); | 90 void StopAecDump(); |
91 | 91 |
| 92 const webrtc::AudioProcessing::Config& GetApmConfigForTest() const { |
| 93 return apm_config_; |
| 94 } |
| 95 |
92 private: | 96 private: |
93 // Every option that is "set" will be applied. Every option not "set" will be | 97 // Every option that is "set" will be applied. Every option not "set" will be |
94 // ignored. This allows us to selectively turn on and off different options | 98 // ignored. This allows us to selectively turn on and off different options |
95 // easily at any time. | 99 // easily at any time. |
96 bool ApplyOptions(const AudioOptions& options); | 100 bool ApplyOptions(const AudioOptions& options); |
97 void SetDefaultDevices(); | 101 void SetDefaultDevices(); |
98 | 102 |
99 // webrtc::TraceCallback: | 103 // webrtc::TraceCallback: |
100 void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 104 void Print(webrtc::TraceLevel level, const char* trace, int length) override; |
101 | 105 |
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275 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 279 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
276 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 280 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
277 | 281 |
278 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; | 282 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; |
279 | 283 |
280 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 284 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
281 }; | 285 }; |
282 } // namespace cricket | 286 } // namespace cricket |
283 | 287 |
284 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 288 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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