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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: Adding nil check and removing unneeded methods. 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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141 const AudioOptions& options, 141 const AudioOptions& options,
142 webrtc::Call* call); 142 webrtc::Call* call);
143 ~WebRtcVoiceMediaChannel() override; 143 ~WebRtcVoiceMediaChannel() override;
144 144
145 const AudioOptions& options() const { return options_; } 145 const AudioOptions& options() const { return options_; }
146 146
147 rtc::DiffServCodePoint PreferredDscp() const override; 147 rtc::DiffServCodePoint PreferredDscp() const override;
148 148
149 bool SetSendParameters(const AudioSendParameters& params) override; 149 bool SetSendParameters(const AudioSendParameters& params) override;
150 bool SetRecvParameters(const AudioRecvParameters& params) override; 150 bool SetRecvParameters(const AudioRecvParameters& params) override;
151 webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const override; 151 webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
152 bool SetRtpParameters(uint32_t ssrc, 152 bool SetRtpSendParameters(uint32_t ssrc,
153 const webrtc::RtpParameters& parameters) override; 153 const webrtc::RtpParameters& parameters) override;
154 webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
155 bool SetRtpReceiveParameters(
156 uint32_t ssrc,
157 const webrtc::RtpParameters& parameters) override;
154 158
155 bool SetPlayout(bool playout) override; 159 bool SetPlayout(bool playout) override;
156 bool PausePlayout(); 160 bool PausePlayout();
157 bool ResumePlayout(); 161 bool ResumePlayout();
158 void SetSend(bool send) override; 162 void SetSend(bool send) override;
159 bool SetAudioSend(uint32_t ssrc, 163 bool SetAudioSend(uint32_t ssrc,
160 bool enable, 164 bool enable,
161 const AudioOptions* options, 165 const AudioOptions* options,
162 AudioSource* source) override; 166 AudioSource* source) override;
163 bool AddSendStream(const StreamParams& sp) override; 167 bool AddSendStream(const StreamParams& sp) override;
(...skipping 58 matching lines...) Expand 10 before | Expand all | Expand 10 after
222 int GetLastEngineError() { return engine()->GetLastEngineError(); } 226 int GetLastEngineError() { return engine()->GetLastEngineError(); }
223 int GetOutputLevel(int channel); 227 int GetOutputLevel(int channel);
224 bool SetPlayout(int channel, bool playout); 228 bool SetPlayout(int channel, bool playout);
225 bool ChangePlayout(bool playout); 229 bool ChangePlayout(bool playout);
226 int CreateVoEChannel(); 230 int CreateVoEChannel();
227 bool DeleteVoEChannel(int channel); 231 bool DeleteVoEChannel(int channel);
228 bool IsDefaultRecvStream(uint32_t ssrc) { 232 bool IsDefaultRecvStream(uint32_t ssrc) {
229 return default_recv_ssrc_ == static_cast<int64_t>(ssrc); 233 return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
230 } 234 }
231 bool SetMaxSendBitrate(int bps); 235 bool SetMaxSendBitrate(int bps);
232 bool SetChannelParameters(int channel, 236 bool SetChannelSendParameters(int channel,
233 const webrtc::RtpParameters& parameters); 237 const webrtc::RtpParameters& parameters);
234 bool SetMaxSendBitrate(int channel, int bps); 238 bool SetMaxSendBitrate(int channel, int bps);
235 bool HasSendCodec() const { 239 bool HasSendCodec() const {
236 return send_codec_spec_.codec_inst.pltype != -1; 240 return send_codec_spec_.codec_inst.pltype != -1;
237 } 241 }
238 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); 242 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
239 void SetupRecording(); 243 void SetupRecording();
240 244
241 rtc::ThreadChecker worker_thread_checker_; 245 rtc::ThreadChecker worker_thread_checker_;
242 246
243 WebRtcVoiceEngine* const engine_ = nullptr; 247 WebRtcVoiceEngine* const engine_ = nullptr;
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286 int cng_payload_type = -1; 290 int cng_payload_type = -1;
287 int cng_plfreq = -1; 291 int cng_plfreq = -1;
288 webrtc::CodecInst codec_inst; 292 webrtc::CodecInst codec_inst;
289 } send_codec_spec_; 293 } send_codec_spec_;
290 294
291 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 295 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
292 }; 296 };
293 } // namespace cricket 297 } // namespace cricket
294 298
295 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 299 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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