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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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30 #include "webrtc/media/engine/webrtcvoe.h" | 30 #include "webrtc/media/engine/webrtcvoe.h" |
31 #include "webrtc/pc/channel.h" | 31 #include "webrtc/pc/channel.h" |
32 | 32 |
33 namespace cricket { | 33 namespace cricket { |
34 | 34 |
35 class AudioDeviceModule; | 35 class AudioDeviceModule; |
36 class AudioSource; | 36 class AudioSource; |
37 class VoEWrapper; | 37 class VoEWrapper; |
38 class WebRtcVoiceMediaChannel; | 38 class WebRtcVoiceMediaChannel; |
39 | 39 |
40 struct SendCodecSpec { | |
41 SendCodecSpec() { | |
42 webrtc::CodecInst empty_inst = {0}; | |
43 codec_inst = empty_inst; | |
44 codec_inst.pltype = -1; | |
45 } | |
46 bool operator==(const SendCodecSpec& rhs) const; | |
47 bool operator!=(const SendCodecSpec& rhs) const; | |
48 | |
49 bool nack_enabled = false; | |
50 bool transport_cc_enabled = false; | |
51 bool enable_codec_fec = false; | |
52 bool enable_opus_dtx = false; | |
53 int opus_max_playback_rate = 0; | |
54 int red_payload_type = -1; | |
55 int cng_payload_type = -1; | |
56 int cng_plfreq = -1; | |
57 webrtc::CodecInst codec_inst; | |
58 }; | |
59 | |
60 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine. | 40 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine. |
61 // It uses the WebRtc VoiceEngine library for audio handling. | 41 // It uses the WebRtc VoiceEngine library for audio handling. |
62 class WebRtcVoiceEngine final : public webrtc::TraceCallback { | 42 class WebRtcVoiceEngine final : public webrtc::TraceCallback { |
63 friend class WebRtcVoiceMediaChannel; | 43 friend class WebRtcVoiceMediaChannel; |
64 public: | 44 public: |
65 // Exposed for the WVoE/MC unit test. | 45 // Exposed for the WVoE/MC unit test. |
66 static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out); | 46 static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out); |
67 | 47 |
68 WebRtcVoiceEngine( | 48 WebRtcVoiceEngine( |
69 webrtc::AudioDeviceModule* adm, | 49 webrtc::AudioDeviceModule* adm, |
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230 return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions()); | 210 return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions()); |
231 } | 211 } |
232 | 212 |
233 int GetReceiveChannelId(uint32_t ssrc) const; | 213 int GetReceiveChannelId(uint32_t ssrc) const; |
234 int GetSendChannelId(uint32_t ssrc) const; | 214 int GetSendChannelId(uint32_t ssrc) const; |
235 | 215 |
236 private: | 216 private: |
237 bool SetOptions(const AudioOptions& options); | 217 bool SetOptions(const AudioOptions& options); |
238 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); | 218 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); |
239 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); | 219 bool SetSendCodecs(const std::vector<AudioCodec>& codecs); |
240 bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters); | |
241 bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec); | |
242 bool SetLocalSource(uint32_t ssrc, AudioSource* source); | 220 bool SetLocalSource(uint32_t ssrc, AudioSource* source); |
243 bool MuteStream(uint32_t ssrc, bool mute); | 221 bool MuteStream(uint32_t ssrc, bool mute); |
244 | 222 |
245 WebRtcVoiceEngine* engine() { return engine_; } | 223 WebRtcVoiceEngine* engine() { return engine_; } |
246 int GetLastEngineError() { return engine()->GetLastEngineError(); } | 224 int GetLastEngineError() { return engine()->GetLastEngineError(); } |
247 int GetOutputLevel(int channel); | 225 int GetOutputLevel(int channel); |
248 int CreateVoEChannel(); | 226 int CreateVoEChannel(); |
249 bool DeleteVoEChannel(int channel); | 227 bool DeleteVoEChannel(int channel); |
250 bool IsDefaultRecvStream(uint32_t ssrc) { | 228 bool IsDefaultRecvStream(uint32_t ssrc) { |
251 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); | 229 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); |
252 } | 230 } |
253 bool SetMaxSendBitrate(int bps); | 231 bool SetMaxSendBitrate(int bps); |
254 bool SetChannelSendParameters(int channel, | |
255 const webrtc::RtpParameters& parameters); | |
256 bool SetMaxSendBitrate(int channel, int bps); | |
257 bool HasSendCodec() const { | |
258 return send_codec_spec_.codec_inst.pltype != -1; | |
259 } | |
260 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); | 232 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); |
261 void SetupRecording(); | 233 void SetupRecording(); |
262 | 234 |
263 rtc::ThreadChecker worker_thread_checker_; | 235 rtc::ThreadChecker worker_thread_checker_; |
264 | 236 |
265 WebRtcVoiceEngine* const engine_ = nullptr; | 237 WebRtcVoiceEngine* const engine_ = nullptr; |
266 std::vector<AudioCodec> send_codecs_; | 238 std::vector<AudioCodec> send_codecs_; |
267 std::vector<AudioCodec> recv_codecs_; | 239 std::vector<AudioCodec> recv_codecs_; |
268 int max_send_bitrate_bps_ = 0; | 240 int max_send_bitrate_bps_ = 0; |
269 AudioOptions options_; | 241 AudioOptions options_; |
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286 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; | 258 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; |
287 | 259 |
288 class WebRtcAudioSendStream; | 260 class WebRtcAudioSendStream; |
289 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; | 261 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; |
290 std::vector<webrtc::RtpExtension> send_rtp_extensions_; | 262 std::vector<webrtc::RtpExtension> send_rtp_extensions_; |
291 | 263 |
292 class WebRtcAudioReceiveStream; | 264 class WebRtcAudioReceiveStream; |
293 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 265 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
294 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 266 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
295 | 267 |
296 SendCodecSpec send_codec_spec_; | 268 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; |
297 | 269 |
298 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 270 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
299 }; | 271 }; |
300 } // namespace cricket | 272 } // namespace cricket |
301 | 273 |
302 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 274 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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