| /* |
| * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "test/call_config_utils.h" |
| |
| #include <string> |
| #include <vector> |
| |
| namespace webrtc { |
| namespace test { |
| |
| // Deserializes a JSON representation of the VideoReceiveStream::Config back |
| // into a valid object. This will not initialize the decoders or the renderer. |
| VideoReceiveStream::Config ParseVideoReceiveStreamJsonConfig( |
| webrtc::Transport* transport, |
| const Json::Value& json) { |
| auto receive_config = VideoReceiveStream::Config(transport); |
| for (const auto decoder_json : json["decoders"]) { |
| VideoReceiveStream::Decoder decoder; |
| decoder.video_format = |
| SdpVideoFormat(decoder_json["payload_name"].asString()); |
| decoder.payload_type = decoder_json["payload_type"].asInt64(); |
| for (const auto& params_json : decoder_json["codec_params"]) { |
| std::vector<std::string> members = params_json.getMemberNames(); |
| RTC_CHECK_EQ(members.size(), 1); |
| decoder.video_format.parameters[members[0]] = |
| params_json[members[0]].asString(); |
| } |
| receive_config.decoders.push_back(decoder); |
| } |
| receive_config.render_delay_ms = json["render_delay_ms"].asInt64(); |
| receive_config.target_delay_ms = json["target_delay_ms"].asInt64(); |
| receive_config.rtp.remote_ssrc = json["rtp"]["remote_ssrc"].asInt64(); |
| receive_config.rtp.local_ssrc = json["rtp"]["local_ssrc"].asInt64(); |
| receive_config.rtp.rtcp_mode = |
| json["rtp"]["rtcp_mode"].asString() == "RtcpMode::kCompound" |
| ? RtcpMode::kCompound |
| : RtcpMode::kReducedSize; |
| receive_config.rtp.remb = json["rtp"]["remb"].asBool(); |
| receive_config.rtp.transport_cc = json["rtp"]["transport_cc"].asBool(); |
| receive_config.rtp.nack.rtp_history_ms = |
| json["rtp"]["nack"]["rtp_history_ms"].asInt64(); |
| receive_config.rtp.ulpfec_payload_type = |
| json["rtp"]["ulpfec_payload_type"].asInt64(); |
| receive_config.rtp.red_payload_type = |
| json["rtp"]["red_payload_type"].asInt64(); |
| receive_config.rtp.rtx_ssrc = json["rtp"]["rtx_ssrc"].asInt64(); |
| |
| for (const auto& pl_json : json["rtp"]["rtx_payload_types"]) { |
| std::vector<std::string> members = pl_json.getMemberNames(); |
| RTC_CHECK_EQ(members.size(), 1); |
| Json::Value rtx_payload_type = pl_json[members[0]]; |
| receive_config.rtp.rtx_associated_payload_types[std::stoi(members[0])] = |
| rtx_payload_type.asInt64(); |
| } |
| for (const auto& ext_json : json["rtp"]["extensions"]) { |
| receive_config.rtp.extensions.emplace_back(ext_json["uri"].asString(), |
| ext_json["id"].asInt64(), |
| ext_json["encrypt"].asBool()); |
| } |
| return receive_config; |
| } |
| |
| } // namespace test. |
| } // namespace webrtc. |