diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/test/call_config_utils.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/call_config_utils.cc')
-rw-r--r-- | third_party/libwebrtc/test/call_config_utils.cc | 110 |
1 files changed, 110 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/call_config_utils.cc b/third_party/libwebrtc/test/call_config_utils.cc new file mode 100644 index 0000000000..d93c902bbe --- /dev/null +++ b/third_party/libwebrtc/test/call_config_utils.cc @@ -0,0 +1,110 @@ +/* + * 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 VideoReceiveStreamInterface::Config +// back into a valid object. This will not initialize the decoders or the +// renderer. +VideoReceiveStreamInterface::Config ParseVideoReceiveStreamJsonConfig( + webrtc::Transport* transport, + const Json::Value& json) { + auto receive_config = VideoReceiveStreamInterface::Config(transport); + for (const auto& decoder_json : json["decoders"]) { + VideoReceiveStreamInterface::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.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.lntf.enabled = json["rtp"]["lntf"]["enabled"].asInt64(); + 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(); + } + return receive_config; +} + +Json::Value GenerateVideoReceiveStreamJsonConfig( + const VideoReceiveStreamInterface::Config& config) { + Json::Value root_json; + + root_json["decoders"] = Json::Value(Json::arrayValue); + for (const auto& decoder : config.decoders) { + Json::Value decoder_json; + decoder_json["payload_type"] = decoder.payload_type; + decoder_json["payload_name"] = decoder.video_format.name; + decoder_json["codec_params"] = Json::Value(Json::arrayValue); + for (const auto& codec_param_entry : decoder.video_format.parameters) { + Json::Value codec_param_json; + codec_param_json[codec_param_entry.first] = codec_param_entry.second; + decoder_json["codec_params"].append(codec_param_json); + } + root_json["decoders"].append(decoder_json); + } + + Json::Value rtp_json; + rtp_json["remote_ssrc"] = config.rtp.remote_ssrc; + rtp_json["local_ssrc"] = config.rtp.local_ssrc; + rtp_json["rtcp_mode"] = config.rtp.rtcp_mode == RtcpMode::kCompound + ? "RtcpMode::kCompound" + : "RtcpMode::kReducedSize"; + rtp_json["lntf"]["enabled"] = config.rtp.lntf.enabled; + rtp_json["nack"]["rtp_history_ms"] = config.rtp.nack.rtp_history_ms; + rtp_json["ulpfec_payload_type"] = config.rtp.ulpfec_payload_type; + rtp_json["red_payload_type"] = config.rtp.red_payload_type; + rtp_json["rtx_ssrc"] = config.rtp.rtx_ssrc; + rtp_json["rtx_payload_types"] = Json::Value(Json::arrayValue); + + for (auto& kv : config.rtp.rtx_associated_payload_types) { + Json::Value val; + val[std::to_string(kv.first)] = kv.second; + rtp_json["rtx_payload_types"].append(val); + } + + root_json["rtp"] = rtp_json; + + root_json["render_delay_ms"] = config.render_delay_ms; + + return root_json; +} + +} // namespace test. +} // namespace webrtc. |