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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.h

Issue 1917193008: Adding getParameters/setParameters APIs to RtpReceiver. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: objc compile errors Created 4 years, 7 months ago
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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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142 const AudioOptions& options, 142 const AudioOptions& options,
143 webrtc::Call* call); 143 webrtc::Call* call);
144 ~WebRtcVoiceMediaChannel() override; 144 ~WebRtcVoiceMediaChannel() override;
145 145
146 const AudioOptions& options() const { return options_; } 146 const AudioOptions& options() const { return options_; }
147 147
148 rtc::DiffServCodePoint PreferredDscp() const override; 148 rtc::DiffServCodePoint PreferredDscp() const override;
149 149
150 bool SetSendParameters(const AudioSendParameters& params) override; 150 bool SetSendParameters(const AudioSendParameters& params) override;
151 bool SetRecvParameters(const AudioRecvParameters& params) override; 151 bool SetRecvParameters(const AudioRecvParameters& params) override;
152 webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const override; 152 webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
153 bool SetRtpParameters(uint32_t ssrc, 153 bool SetRtpSendParameters(uint32_t ssrc,
154 const webrtc::RtpParameters& parameters) override; 154 const webrtc::RtpParameters& parameters) override;
155 webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
156 bool SetRtpReceiveParameters(
157 uint32_t ssrc,
158 const webrtc::RtpParameters& parameters) override;
155 159
156 bool SetPlayout(bool playout) override; 160 bool SetPlayout(bool playout) override;
157 bool PausePlayout(); 161 bool PausePlayout();
158 bool ResumePlayout(); 162 bool ResumePlayout();
159 void SetSend(bool send) override; 163 void SetSend(bool send) override;
160 bool SetAudioSend(uint32_t ssrc, 164 bool SetAudioSend(uint32_t ssrc,
161 bool enable, 165 bool enable,
162 const AudioOptions* options, 166 const AudioOptions* options,
163 AudioSource* source) override; 167 AudioSource* source) override;
164 bool AddSendStream(const StreamParams& sp) override; 168 bool AddSendStream(const StreamParams& sp) override;
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223 int GetLastEngineError() { return engine()->GetLastEngineError(); } 227 int GetLastEngineError() { return engine()->GetLastEngineError(); }
224 int GetOutputLevel(int channel); 228 int GetOutputLevel(int channel);
225 bool SetPlayout(int channel, bool playout); 229 bool SetPlayout(int channel, bool playout);
226 bool ChangePlayout(bool playout); 230 bool ChangePlayout(bool playout);
227 int CreateVoEChannel(); 231 int CreateVoEChannel();
228 bool DeleteVoEChannel(int channel); 232 bool DeleteVoEChannel(int channel);
229 bool IsDefaultRecvStream(uint32_t ssrc) { 233 bool IsDefaultRecvStream(uint32_t ssrc) {
230 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); 234 return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
231 } 235 }
232 bool SetMaxSendBitrate(int bps); 236 bool SetMaxSendBitrate(int bps);
233 bool SetChannelParameters(int channel, 237 bool SetChannelSendParameters(int channel,
234 const webrtc::RtpParameters& parameters); 238 const webrtc::RtpParameters& parameters);
235 bool SetMaxSendBitrate(int channel, int bps); 239 bool SetMaxSendBitrate(int channel, int bps);
236 bool HasSendCodec() const { 240 bool HasSendCodec() const {
237 return send_codec_spec_.codec_inst.pltype != -1; 241 return send_codec_spec_.codec_inst.pltype != -1;
238 } 242 }
239 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); 243 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
240 void SetupRecording(); 244 void SetupRecording();
241 245
242 rtc::ThreadChecker worker_thread_checker_; 246 rtc::ThreadChecker worker_thread_checker_;
243 247
244 WebRtcVoiceEngine* const engine_ = nullptr; 248 WebRtcVoiceEngine* const engine_ = nullptr;
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287 int cng_payload_type = -1; 291 int cng_payload_type = -1;
288 int cng_plfreq = -1; 292 int cng_plfreq = -1;
289 webrtc::CodecInst codec_inst; 293 webrtc::CodecInst codec_inst;
290 } send_codec_spec_; 294 } send_codec_spec_;
291 295
292 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 296 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
293 }; 297 };
294 } // namespace cricket 298 } // namespace cricket
295 299
296 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 300 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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