73 lines
3.0 KiB
C++
73 lines
3.0 KiB
C++
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/*
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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 VideoReceiveStream::Config back
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// into a valid object. This will not initialize the decoders or the renderer.
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VideoReceiveStream::Config ParseVideoReceiveStreamJsonConfig(
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webrtc::Transport* transport,
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const Json::Value& json) {
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auto receive_config = VideoReceiveStream::Config(transport);
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for (const auto decoder_json : json["decoders"]) {
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VideoReceiveStream::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.target_delay_ms = json["target_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.remb = json["rtp"]["remb"].asBool();
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receive_config.rtp.transport_cc = json["rtp"]["transport_cc"].asBool();
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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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for (const auto& ext_json : json["rtp"]["extensions"]) {
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receive_config.rtp.extensions.emplace_back(ext_json["uri"].asString(),
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ext_json["id"].asInt64(),
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ext_json["encrypt"].asBool());
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}
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return receive_config;
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}
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} // namespace test.
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} // namespace webrtc.
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