Move VideoStreamReceiver JSON configuration parser to test source_set.
This change moves the configuration parser that converts a JSON representation
of the VideoStreamReceiver::Config structure into a native object into the test
directory so that it can be shared with the new corpus_generator utility that is
being built. This rtc_source_set will have an additional utility function added
in a subsequent CL that will allow the generation of a VideoStreamSender::Config
from a given VideoStreamReceiver::Config and visa versa.
Bug: webrtc:10117
Change-Id: I3035826f799f8d1fcdeaa76997391f030c855a5c
Reviewed-on: https://webrtc-review.googlesource.com/c/116880
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26252}
diff --git a/test/call_config_utils.cc b/test/call_config_utils.cc
new file mode 100644
index 0000000..48d4849
--- /dev/null
+++ b/test/call_config_utils.cc
@@ -0,0 +1,72 @@
+/*
+ * 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.