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https://github.com/mollyim/webrtc.git
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217b384c1b
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}
110 lines
4.3 KiB
C++
110 lines
4.3 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 <string>
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#include <vector>
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namespace webrtc {
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namespace test {
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// Deserializes a JSON representation of the VideoReceiveStreamInterface::Config
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// back into a valid object. This will not initialize the decoders or the
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// renderer.
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VideoReceiveStreamInterface::Config ParseVideoReceiveStreamJsonConfig(
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webrtc::Transport* transport,
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const Json::Value& json) {
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auto receive_config = VideoReceiveStreamInterface::Config(transport);
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for (const auto& decoder_json : json["decoders"]) {
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VideoReceiveStreamInterface::Decoder decoder;
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decoder.video_format =
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SdpVideoFormat(decoder_json["payload_name"].asString());
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decoder.payload_type = decoder_json["payload_type"].asInt64();
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for (const auto& params_json : decoder_json["codec_params"]) {
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std::vector<std::string> members = params_json.getMemberNames();
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RTC_CHECK_EQ(members.size(), 1);
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decoder.video_format.parameters[members[0]] =
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params_json[members[0]].asString();
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}
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receive_config.decoders.push_back(decoder);
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}
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receive_config.render_delay_ms = json["render_delay_ms"].asInt64();
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receive_config.rtp.remote_ssrc = json["rtp"]["remote_ssrc"].asInt64();
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receive_config.rtp.local_ssrc = json["rtp"]["local_ssrc"].asInt64();
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receive_config.rtp.rtcp_mode =
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json["rtp"]["rtcp_mode"].asString() == "RtcpMode::kCompound"
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? RtcpMode::kCompound
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: RtcpMode::kReducedSize;
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receive_config.rtp.lntf.enabled = json["rtp"]["lntf"]["enabled"].asInt64();
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receive_config.rtp.nack.rtp_history_ms =
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json["rtp"]["nack"]["rtp_history_ms"].asInt64();
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receive_config.rtp.ulpfec_payload_type =
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json["rtp"]["ulpfec_payload_type"].asInt64();
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receive_config.rtp.red_payload_type =
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json["rtp"]["red_payload_type"].asInt64();
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receive_config.rtp.rtx_ssrc = json["rtp"]["rtx_ssrc"].asInt64();
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for (const auto& pl_json : json["rtp"]["rtx_payload_types"]) {
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std::vector<std::string> members = pl_json.getMemberNames();
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RTC_CHECK_EQ(members.size(), 1);
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Json::Value rtx_payload_type = pl_json[members[0]];
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receive_config.rtp.rtx_associated_payload_types[std::stoi(members[0])] =
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rtx_payload_type.asInt64();
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}
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return receive_config;
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}
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Json::Value GenerateVideoReceiveStreamJsonConfig(
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const VideoReceiveStreamInterface::Config& config) {
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Json::Value root_json;
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root_json["decoders"] = Json::Value(Json::arrayValue);
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for (const auto& decoder : config.decoders) {
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Json::Value decoder_json;
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decoder_json["payload_type"] = decoder.payload_type;
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decoder_json["payload_name"] = decoder.video_format.name;
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decoder_json["codec_params"] = Json::Value(Json::arrayValue);
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for (const auto& codec_param_entry : decoder.video_format.parameters) {
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Json::Value codec_param_json;
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codec_param_json[codec_param_entry.first] = codec_param_entry.second;
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decoder_json["codec_params"].append(codec_param_json);
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}
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root_json["decoders"].append(decoder_json);
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}
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Json::Value rtp_json;
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rtp_json["remote_ssrc"] = config.rtp.remote_ssrc;
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rtp_json["local_ssrc"] = config.rtp.local_ssrc;
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rtp_json["rtcp_mode"] = config.rtp.rtcp_mode == RtcpMode::kCompound
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? "RtcpMode::kCompound"
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: "RtcpMode::kReducedSize";
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rtp_json["lntf"]["enabled"] = config.rtp.lntf.enabled;
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rtp_json["nack"]["rtp_history_ms"] = config.rtp.nack.rtp_history_ms;
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rtp_json["ulpfec_payload_type"] = config.rtp.ulpfec_payload_type;
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rtp_json["red_payload_type"] = config.rtp.red_payload_type;
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rtp_json["rtx_ssrc"] = config.rtp.rtx_ssrc;
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rtp_json["rtx_payload_types"] = Json::Value(Json::arrayValue);
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for (auto& kv : config.rtp.rtx_associated_payload_types) {
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Json::Value val;
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val[std::to_string(kv.first)] = kv.second;
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rtp_json["rtx_payload_types"].append(val);
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}
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root_json["rtp"] = rtp_json;
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root_json["render_delay_ms"] = config.render_delay_ms;
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return root_json;
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}
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} // namespace test.
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} // namespace webrtc.
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