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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Issue 2177523002: Fix issues with RestartingSendStreamPreservesRtpStatesWithRtx (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 4 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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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411 uint8_t dtmfbuffer[IP_PACKET_SIZE]; 411 uint8_t dtmfbuffer[IP_PACKET_SIZE];
412 uint8_t sendCount = 1; 412 uint8_t sendCount = 1;
413 int32_t retVal = 0; 413 int32_t retVal = 0;
414 414
415 if (ended) { 415 if (ended) {
416 // resend last packet in an event 3 times 416 // resend last packet in an event 3 times
417 sendCount = 3; 417 sendCount = 3;
418 } 418 }
419 do { 419 do {
420 // Send DTMF data 420 // Send DTMF data
421 rtp_sender_->BuildRtpHeader(dtmfbuffer, dtmf_payload_type, marker_bit, 421 int32_t header_length = rtp_sender_->BuildRtpHeader(
422 dtmf_timestamp, clock_->TimeInMilliseconds()); 422 dtmfbuffer, dtmf_payload_type, marker_bit, dtmf_timestamp,
423 clock_->TimeInMilliseconds());
424 if (header_length <= 0)
425 return -1;
423 426
424 // reset CSRC and X bit 427 // reset CSRC and X bit
425 dtmfbuffer[0] &= 0xe0; 428 dtmfbuffer[0] &= 0xe0;
426 429
427 // Create DTMF data 430 // Create DTMF data
428 /* From RFC 2833: 431 /* From RFC 2833:
429 432
430 0 1 2 3 433 0 1 2 3
431 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 434 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
432 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 435 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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450 "timestamp", dtmf_timestamp, "seqnum", rtp_sender_->SequenceNumber()); 453 "timestamp", dtmf_timestamp, "seqnum", rtp_sender_->SequenceNumber());
451 retVal = rtp_sender_->SendToNetwork(dtmfbuffer, 4, 12, rtc::TimeMillis(), 454 retVal = rtp_sender_->SendToNetwork(dtmfbuffer, 4, 12, rtc::TimeMillis(),
452 kAllowRetransmission, 455 kAllowRetransmission,
453 RtpPacketSender::kHighPriority); 456 RtpPacketSender::kHighPriority);
454 sendCount--; 457 sendCount--;
455 } while (sendCount > 0 && retVal == 0); 458 } while (sendCount > 0 && retVal == 0);
456 459
457 return retVal; 460 return retVal;
458 } 461 }
459 } // namespace webrtc 462 } // namespace webrtc
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