9253240305
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}
65 lines
2.7 KiB
C++
65 lines
2.7 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 "test/call_config_utils.h"
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#include "call/video_receive_stream.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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TEST(CallConfigUtils, MarshalUnmarshalProcessSameObject) {
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VideoReceiveStreamInterface::Config recv_config(nullptr);
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VideoReceiveStreamInterface::Decoder decoder;
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decoder.payload_type = 10;
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decoder.video_format.name = "test";
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decoder.video_format.parameters["99"] = "b";
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recv_config.decoders.push_back(decoder);
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recv_config.render_delay_ms = 10;
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recv_config.rtp.remote_ssrc = 100;
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recv_config.rtp.local_ssrc = 101;
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recv_config.rtp.rtcp_mode = RtcpMode::kCompound;
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recv_config.rtp.lntf.enabled = false;
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recv_config.rtp.nack.rtp_history_ms = 150;
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recv_config.rtp.red_payload_type = 50;
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recv_config.rtp.rtx_ssrc = 1000;
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recv_config.rtp.rtx_associated_payload_types[10] = 10;
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recv_config.rtp.extensions.emplace_back("uri", 128, true);
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VideoReceiveStreamInterface::Config unmarshaled_config =
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ParseVideoReceiveStreamJsonConfig(
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nullptr, GenerateVideoReceiveStreamJsonConfig(recv_config));
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EXPECT_EQ(recv_config.decoders[0].payload_type,
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unmarshaled_config.decoders[0].payload_type);
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EXPECT_EQ(recv_config.decoders[0].video_format.name,
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unmarshaled_config.decoders[0].video_format.name);
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EXPECT_EQ(recv_config.decoders[0].video_format.parameters,
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unmarshaled_config.decoders[0].video_format.parameters);
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EXPECT_EQ(recv_config.render_delay_ms, unmarshaled_config.render_delay_ms);
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EXPECT_EQ(recv_config.rtp.remote_ssrc, unmarshaled_config.rtp.remote_ssrc);
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EXPECT_EQ(recv_config.rtp.local_ssrc, unmarshaled_config.rtp.local_ssrc);
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EXPECT_EQ(recv_config.rtp.rtcp_mode, unmarshaled_config.rtp.rtcp_mode);
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EXPECT_EQ(recv_config.rtp.lntf.enabled, unmarshaled_config.rtp.lntf.enabled);
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EXPECT_EQ(recv_config.rtp.nack.rtp_history_ms,
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unmarshaled_config.rtp.nack.rtp_history_ms);
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EXPECT_EQ(recv_config.rtp.red_payload_type,
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unmarshaled_config.rtp.red_payload_type);
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EXPECT_EQ(recv_config.rtp.rtx_ssrc, unmarshaled_config.rtp.rtx_ssrc);
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EXPECT_EQ(recv_config.rtp.rtx_associated_payload_types,
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unmarshaled_config.rtp.rtx_associated_payload_types);
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EXPECT_EQ(recv_config.rtp.extensions, recv_config.rtp.extensions);
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}
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} // namespace test
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} // namespace webrtc
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