Added JSON generator for VideoReceiveStream::Config objects.
This change adds a new way for test code to serialize the important information
from a VideoReceiveStream::Config so that it can be stored as configuration data
for WebRTC fuzzers. This code isn't included in the object itself as it is only
going to be used to generate new configurations for the fuzzer each time a new
error_correction or video format is added to WebRTC.
Bug: webrtc:10117
Change-Id: I9b6fb8e0345890ab16f6d319d91e4e316d1f2888
Reviewed-on: https://webrtc-review.googlesource.com/c/116920
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26255}diff --git a/test/call_config_utils.cc b/test/call_config_utils.cc
index 48d4849..671220b 100644
--- a/test/call_config_utils.cc
+++ b/test/call_config_utils.cc
@@ -68,5 +68,59 @@
return receive_config;
}
+Json::Value GenerateVideoReceiveStreamJsonConfig(
+ const VideoReceiveStream::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["remb"] = config.rtp.remb;
+ rtp_json["transport_cc"] = config.rtp.transport_cc;
+ 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;
+ root_json["target_delay_ms"] = config.target_delay_ms;
+
+ return root_json;
+}
+
} // namespace test.
} // namespace webrtc.