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2c4b7f98ba
Upstream commit: https://webrtc.googlesource.com/src/+/217b384c1be7e692086464bf3503e7e4f35d1abf Remove rtp header extension from config of Call audio and video receivers These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived. Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d Bug: webrtc:7135 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480 Owners-Override: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39236}
62 lines
2.6 KiB
C++
62 lines
2.6 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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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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}
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} // namespace test
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} // namespace webrtc
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