This is a reland of commit 97ba853295578975a04fc504315cccd465f9f0bd This cl did not cause the regression in Chrome rolls https://chromium-review.googlesource.com/c/chromium/src/+/4132644?tab=checks. Real culprit reverted in https://webrtc-review.googlesource.com/c/src/+/290502. Original change's description: > Remove use of ReceiveStreamRtpConfig:transport_cc > > With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension > http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated. > I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored. > > > Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841 > > Bug: webrtc:14802 > Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38980} Bug: webrtc:14802 Change-Id: Ib98e61413161d462da60144942cdb0140e12bc42 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290503 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38997}
43 lines
1.4 KiB
C++
43 lines
1.4 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stddef.h>
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#include <stdint.h>
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#include <memory>
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#include "test/fuzzers/utils/rtp_replayer.h"
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namespace webrtc {
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void FuzzOneInput(const uint8_t* data, size_t size) {
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auto stream_state = std::make_unique<test::RtpReplayer::StreamState>();
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VideoReceiveStreamInterface::Config vp8_config(&(stream_state->transport));
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VideoReceiveStreamInterface::Decoder vp8_decoder;
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vp8_decoder.video_format = SdpVideoFormat("VP8");
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vp8_decoder.payload_type = 125;
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vp8_config.decoders.push_back(std::move(vp8_decoder));
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vp8_config.rtp.local_ssrc = 7731;
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vp8_config.rtp.remote_ssrc = 1337;
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vp8_config.rtp.rtx_ssrc = 100;
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vp8_config.rtp.nack.rtp_history_ms = 1000;
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vp8_config.rtp.lntf.enabled = true;
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std::vector<VideoReceiveStreamInterface::Config> replay_configs;
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replay_configs.push_back(std::move(vp8_config));
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test::RtpReplayer::Replay(std::move(stream_state), std::move(replay_configs),
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data, size);
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}
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} // namespace webrtc
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