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Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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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286 * presence of extension number i from kExtensionTypes / kExtensionNames. 286 * presence of extension number i from kExtensionTypes / kExtensionNames.
287 * The least significant bit extension_bitvector has number 0. 287 * The least significant bit extension_bitvector has number 0.
288 */ 288 */
289 size_t GenerateRtpPacket(uint32_t extensions_bitvector, 289 size_t GenerateRtpPacket(uint32_t extensions_bitvector,
290 uint32_t csrcs_count, 290 uint32_t csrcs_count,
291 uint8_t* packet, 291 uint8_t* packet,
292 size_t packet_size) { 292 size_t packet_size) {
293 RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions); 293 RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
294 Clock* clock = Clock::GetRealTimeClock(); 294 Clock* clock = Clock::GetRealTimeClock();
295 295
296 RTPSender rtp_sender(0, // int32_t id 296 RTPSender rtp_sender(false, // bool audio
297 false, // bool audio
298 clock, // Clock* clock 297 clock, // Clock* clock
299 nullptr, // Transport* 298 nullptr, // Transport*
300 nullptr, // RtpAudioFeedback* 299 nullptr, // RtpAudioFeedback*
301 nullptr, // PacedSender* 300 nullptr, // PacedSender*
302 nullptr, // PacketRouter* 301 nullptr, // PacketRouter*
303 nullptr, // SendTimeObserver* 302 nullptr, // SendTimeObserver*
304 nullptr, // BitrateStatisticsObserver* 303 nullptr, // BitrateStatisticsObserver*
305 nullptr, // FrameCountObserver* 304 nullptr, // FrameCountObserver*
306 nullptr); // SendSideDelayObserver* 305 nullptr); // SendSideDelayObserver*
307 306
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538 extensions, // Bit vector choosing extensions 537 extensions, // Bit vector choosing extensions
539 csrcs_count, // Number of contributing sources 538 csrcs_count, // Number of contributing sources
540 rand()); 539 rand());
541 } 540 }
542 } 541 }
543 } 542 }
544 543
545 } // namespace webrtc 544 } // namespace webrtc
546 545
547 #endif // ENABLE_RTC_EVENT_LOG 546 #endif // ENABLE_RTC_EVENT_LOG
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