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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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47 bool operator!=(const SendCodecSpec& rhs) const; | 47 bool operator!=(const SendCodecSpec& rhs) const; |
48 | 48 |
49 bool nack_enabled = false; | 49 bool nack_enabled = false; |
50 bool transport_cc_enabled = false; | 50 bool transport_cc_enabled = false; |
51 bool enable_codec_fec = false; | 51 bool enable_codec_fec = false; |
52 bool enable_opus_dtx = false; | 52 bool enable_opus_dtx = false; |
53 int opus_max_playback_rate = 0; | 53 int opus_max_playback_rate = 0; |
54 int red_payload_type = -1; | 54 int red_payload_type = -1; |
55 int cng_payload_type = -1; | 55 int cng_payload_type = -1; |
56 int cng_plfreq = -1; | 56 int cng_plfreq = -1; |
| 57 // max_ptime_ms: maximal frame length in ms. |
| 58 int max_ptime_ms = kPreferredMaxPTime; |
| 59 // min_ptime_ms: minimal frame length in ms. |
| 60 int min_ptime_ms = kPreferredMinPTime; |
57 webrtc::CodecInst codec_inst; | 61 webrtc::CodecInst codec_inst; |
58 }; | 62 }; |
59 | 63 |
60 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine. | 64 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine. |
61 // It uses the WebRtc VoiceEngine library for audio handling. | 65 // It uses the WebRtc VoiceEngine library for audio handling. |
62 class WebRtcVoiceEngine final : public webrtc::TraceCallback { | 66 class WebRtcVoiceEngine final : public webrtc::TraceCallback { |
63 friend class WebRtcVoiceMediaChannel; | 67 friend class WebRtcVoiceMediaChannel; |
64 public: | 68 public: |
65 // Exposed for the WVoE/MC unit test. | 69 // Exposed for the WVoE/MC unit test. |
66 static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out); | 70 static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out); |
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230 return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions()); | 234 return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions()); |
231 } | 235 } |
232 | 236 |
233 int GetReceiveChannelId(uint32_t ssrc) const; | 237 int GetReceiveChannelId(uint32_t ssrc) const; |
234 int GetSendChannelId(uint32_t ssrc) const; | 238 int GetSendChannelId(uint32_t ssrc) const; |
235 | 239 |
236 private: | 240 private: |
237 bool SetOptions(const AudioOptions& options); | 241 bool SetOptions(const AudioOptions& options); |
238 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); | 242 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); |
239 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); | 243 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); |
240 bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters); | 244 bool SetSendCodecs(uint32_t ssrc, |
| 245 const webrtc::RtpParameters& rtp_parameters); |
241 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec); | 246 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec); |
242 bool SetLocalSource(uint32_t ssrc, AudioSource* source); | 247 bool SetLocalSource(uint32_t ssrc, AudioSource* source); |
243 bool MuteStream(uint32_t ssrc, bool mute); | 248 bool MuteStream(uint32_t ssrc, bool mute); |
244 | 249 |
245 WebRtcVoiceEngine* engine() { return engine_; } | 250 WebRtcVoiceEngine* engine() { return engine_; } |
246 int GetLastEngineError() { return engine()->GetLastEngineError(); } | 251 int GetLastEngineError() { return engine()->GetLastEngineError(); } |
247 int GetOutputLevel(int channel); | 252 int GetOutputLevel(int channel); |
248 void ChangePlayout(bool playout); | 253 void ChangePlayout(bool playout); |
249 int CreateVoEChannel(); | 254 int CreateVoEChannel(); |
250 bool DeleteVoEChannel(int channel); | 255 bool DeleteVoEChannel(int channel); |
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295 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 300 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
296 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 301 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
297 | 302 |
298 SendCodecSpec send_codec_spec_; | 303 SendCodecSpec send_codec_spec_; |
299 | 304 |
300 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 305 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
301 }; | 306 }; |
302 } // namespace cricket | 307 } // namespace cricket |
303 | 308 |
304 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 309 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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