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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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