/*
* 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>
#include "api/call/transport.h"
#include "api/rtp_headers.h"
#include "api/video_codecs/sdp_video_format.h"
#include "call/video_receive_stream.h"
#include "rtc_base/checks.h"
#include "rtc_base/strings/json.h"
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(
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.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["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.
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