summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/test/call_config_utils.cc
diff options
context:
space:
mode:
authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/test/call_config_utils.cc
parentInitial commit. (diff)
downloadfirefox-esr-upstream.tar.xz
firefox-esr-upstream.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
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.cc123
1 files changed, 123 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..da3d76c689
--- /dev/null
+++ b/third_party/libwebrtc/test/call_config_utils.cc
@@ -0,0 +1,123 @@
+/*
+ * 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();
+ }
+ 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;
+}
+
+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);
+ }
+
+ rtp_json["extensions"] = Json::Value(Json::arrayValue);
+ for (auto& ext : config.rtp.extensions) {
+ Json::Value ext_json;
+ ext_json["uri"] = ext.uri;
+ ext_json["id"] = ext.id;
+ ext_json["encrypt"] = ext.encrypt;
+ rtp_json["extensions"].append(ext_json);
+ }
+ root_json["rtp"] = rtp_json;
+
+ root_json["render_delay_ms"] = config.render_delay_ms;
+
+ return root_json;
+}
+
+} // namespace test.
+} // namespace webrtc.