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

Issue 1885473004: Adding codecs to the RtpParameters returned by an RtpSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Responding to review comments. Created 4 years, 8 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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232 const webrtc::RtpParameters& parameters); 232 const webrtc::RtpParameters& parameters);
233 bool SetSendBitrate(int channel, int bps); 233 bool SetSendBitrate(int channel, int bps);
234 bool HasSendCodec() const { 234 bool HasSendCodec() const {
235 return send_codec_spec_.codec_inst.pltype != -1; 235 return send_codec_spec_.codec_inst.pltype != -1;
236 } 236 }
237 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); 237 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
238 238
239 rtc::ThreadChecker worker_thread_checker_; 239 rtc::ThreadChecker worker_thread_checker_;
240 240
241 WebRtcVoiceEngine* const engine_ = nullptr; 241 WebRtcVoiceEngine* const engine_ = nullptr;
242 std::vector<AudioCodec> send_codecs_;
242 std::vector<AudioCodec> recv_codecs_; 243 std::vector<AudioCodec> recv_codecs_;
243 int send_bitrate_bps_ = 0; 244 int send_bitrate_bps_ = 0;
244 AudioOptions options_; 245 AudioOptions options_;
245 rtc::Optional<int> dtmf_payload_type_; 246 rtc::Optional<int> dtmf_payload_type_;
246 bool desired_playout_ = false; 247 bool desired_playout_ = false;
247 bool recv_transport_cc_enabled_ = false; 248 bool recv_transport_cc_enabled_ = false;
248 bool playout_ = false; 249 bool playout_ = false;
249 bool send_ = false; 250 bool send_ = false;
250 webrtc::Call* const call_ = nullptr; 251 webrtc::Call* const call_ = nullptr;
251 252
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283 int cng_payload_type = -1; 284 int cng_payload_type = -1;
284 int cng_plfreq = -1; 285 int cng_plfreq = -1;
285 webrtc::CodecInst codec_inst; 286 webrtc::CodecInst codec_inst;
286 } send_codec_spec_; 287 } send_codec_spec_;
287 288
288 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 289 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
289 }; 290 };
290 } // namespace cricket 291 } // namespace cricket
291 292
292 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 293 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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