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1 // Copyright 2014 The Chromium Authors. All rights reserved. | 1 // Copyright 2014 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "media/cast/net/cast_transport_sender_impl.h" | 5 #include "media/cast/net/cast_transport_sender_impl.h" |
6 | 6 |
7 #include <stddef.h> | 7 #include <stddef.h> |
8 #include <algorithm> | 8 #include <algorithm> |
9 #include <string> | 9 #include <string> |
10 #include <utility> | 10 #include <utility> |
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367 } else if (video_sender_ && video_sender_->ssrc() == ssrc) { | 367 } else if (video_sender_ && video_sender_->ssrc() == ssrc) { |
368 dedup_info.resend_interval = video_rtcp_session_->current_round_trip_time(); | 368 dedup_info.resend_interval = video_rtcp_session_->current_round_trip_time(); |
369 | 369 |
370 // Only use audio stream to dedup if there is one. | 370 // Only use audio stream to dedup if there is one. |
371 if (audio_sender_) { | 371 if (audio_sender_) { |
372 dedup_info.last_byte_acked_for_audio = last_byte_acked_for_audio_; | 372 dedup_info.last_byte_acked_for_audio = last_byte_acked_for_audio_; |
373 } | 373 } |
374 } | 374 } |
375 | 375 |
376 if (!cast_message.missing_frames_and_packets.empty()) { | 376 if (!cast_message.missing_frames_and_packets.empty()) { |
377 // This call does two things. | 377 VLOG(2) << "feedback_count: " << cast_message.feedback_count; |
378 // 1. Specifies that retransmissions for packets not listed in the set are | 378 // This call specifies a deduplication window. For video this would be the |
379 // cancelled. | 379 // most recent RTT. For audio there is no deduplication. |
380 // 2. Specifies a deduplication window. For video this would be the most | 380 ResendPackets(ssrc, cast_message.missing_frames_and_packets, false, |
miu
2016/03/29 19:14:14
Looks like all possible callers of ResendPackets w
xjz
2016/03/29 19:33:49
As discussed, this is not applicable any more.
| |
381 // recent RTT. For audio there is no deduplication. | |
382 ResendPackets(ssrc, cast_message.missing_frames_and_packets, true, | |
383 dedup_info); | 381 dedup_info); |
384 } | 382 } |
385 | 383 |
386 if (!cast_message.received_later_frames.empty()) { | 384 if (!cast_message.received_later_frames.empty()) { |
387 // Cancel resending frames that were received by the RTP receiver. | 385 // Cancel resending frames that were received by the RTP receiver. |
388 CancelSendingFrames(ssrc, cast_message.received_later_frames); | 386 CancelSendingFrames(ssrc, cast_message.received_later_frames); |
389 } | 387 } |
390 } | 388 } |
391 | 389 |
392 void CastTransportSenderImpl::AddValidRtpReceiver(uint32_t rtp_sender_ssrc, | 390 void CastTransportSenderImpl::AddValidRtpReceiver(uint32_t rtp_sender_ssrc, |
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486 "calling CastTransportSenderImpl::SendRtcpFromRtpReceiver."; | 484 "calling CastTransportSenderImpl::SendRtcpFromRtpReceiver."; |
487 return; | 485 return; |
488 } | 486 } |
489 pacer_.SendRtcpPacket(rtcp_builder_at_rtp_receiver_->local_ssrc(), | 487 pacer_.SendRtcpPacket(rtcp_builder_at_rtp_receiver_->local_ssrc(), |
490 rtcp_builder_at_rtp_receiver_->Finish()); | 488 rtcp_builder_at_rtp_receiver_->Finish()); |
491 rtcp_builder_at_rtp_receiver_.reset(); | 489 rtcp_builder_at_rtp_receiver_.reset(); |
492 } | 490 } |
493 | 491 |
494 } // namespace cast | 492 } // namespace cast |
495 } // namespace media | 493 } // namespace media |
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