Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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.