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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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297 * presence of extension number i from kExtensionTypes / kExtensionNames. | 297 * presence of extension number i from kExtensionTypes / kExtensionNames. |
298 * The least significant bit extension_bitvector has number 0. | 298 * The least significant bit extension_bitvector has number 0. |
299 */ | 299 */ |
300 size_t GenerateRtpPacket(uint32_t extensions_bitvector, | 300 size_t GenerateRtpPacket(uint32_t extensions_bitvector, |
301 uint32_t csrcs_count, | 301 uint32_t csrcs_count, |
302 uint8_t* packet, | 302 uint8_t* packet, |
303 size_t packet_size, | 303 size_t packet_size, |
304 Random* prng) { | 304 Random* prng) { |
305 RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions); | 305 RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions); |
306 Clock* clock = Clock::GetRealTimeClock(); | 306 Clock* clock = Clock::GetRealTimeClock(); |
307 | 307 SSRCDatabase ssrc_database; |
308 RTPSender rtp_sender(false, // bool audio | 308 RTPSender rtp_sender(false, // bool audio |
309 clock, // Clock* clock | 309 clock, // Clock* clock |
310 nullptr, // Transport* | 310 nullptr, // Transport* |
311 nullptr, // RtpAudioFeedback* | 311 nullptr, // RtpAudioFeedback* |
312 nullptr, // PacedSender* | 312 nullptr, // PacedSender* |
313 nullptr, // PacketRouter* | 313 nullptr, // PacketRouter* |
314 nullptr, // SendTimeObserver* | 314 nullptr, // SendTimeObserver* |
315 nullptr, // BitrateStatisticsObserver* | 315 nullptr, // BitrateStatisticsObserver* |
316 nullptr, // FrameCountObserver* | 316 nullptr, // FrameCountObserver* |
317 nullptr, // SendSideDelayObserver* | 317 nullptr, // SendSideDelayObserver* |
318 nullptr); // RtcEventLog* | 318 nullptr, // RtcEventLog* |
| 319 &ssrc_database); |
319 | 320 |
320 std::vector<uint32_t> csrcs; | 321 std::vector<uint32_t> csrcs; |
321 for (unsigned i = 0; i < csrcs_count; i++) { | 322 for (unsigned i = 0; i < csrcs_count; i++) { |
322 csrcs.push_back(prng->Rand<uint32_t>()); | 323 csrcs.push_back(prng->Rand<uint32_t>()); |
323 } | 324 } |
324 rtp_sender.SetCsrcs(csrcs); | 325 rtp_sender.SetCsrcs(csrcs); |
325 rtp_sender.SetSSRC(prng->Rand<uint32_t>()); | 326 rtp_sender.SetSSRC(prng->Rand<uint32_t>()); |
326 rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true); | 327 rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true); |
327 rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>()); | 328 rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>()); |
328 | 329 |
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685 // Enable all header extensions | 686 // Enable all header extensions |
686 uint32_t extensions = (1u << kNumExtensions) - 1; | 687 uint32_t extensions = (1u << kNumExtensions) - 1; |
687 uint32_t csrcs_count = 2; | 688 uint32_t csrcs_count = 2; |
688 DropOldEvents(extensions, csrcs_count, 141421356); | 689 DropOldEvents(extensions, csrcs_count, 141421356); |
689 DropOldEvents(extensions, csrcs_count, 173205080); | 690 DropOldEvents(extensions, csrcs_count, 173205080); |
690 } | 691 } |
691 | 692 |
692 } // namespace webrtc | 693 } // namespace webrtc |
693 | 694 |
694 #endif // ENABLE_RTC_EVENT_LOG | 695 #endif // ENABLE_RTC_EVENT_LOG |
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