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Move VideoStreamReceiver JSON configuration parser to test source_set.

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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}
This commit is contained in:
Benjamin Wright
2019-01-11 10:48:42 -08:00
committed by Commit Bot
parent 4895b45703
commit 8efafdf84b
5 changed files with 123 additions and 63 deletions
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72
test/call_config_utils.cc Normal file
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/*
* 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.