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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/fuzzers/utils/rtp_replayer.h"
#include <algorithm>
#include <memory>
#include <string>
#include <utility>
#include "absl/memory/memory.h"
#include "api/environment/environment_factory.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/strings/json.h"
#include "system_wrappers/include/clock.h"
#include "test/call_config_utils.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"
#include "test/rtp_file_reader.h"
#include "test/run_loop.h"
namespace webrtc {
namespace test {
void RtpReplayer::Replay(const std::string& replay_config_filepath,
const uint8_t* rtp_dump_data,
size_t rtp_dump_size) {
auto stream_state = std::make_unique<StreamState>();
std::vector<VideoReceiveStreamInterface::Config> receive_stream_configs =
ReadConfigFromFile(replay_config_filepath, &(stream_state->transport));
return Replay(std::move(stream_state), std::move(receive_stream_configs),
rtp_dump_data, rtp_dump_size);
}
void RtpReplayer::Replay(
std::unique_ptr<StreamState> stream_state,
std::vector<VideoReceiveStreamInterface::Config> receive_stream_configs,
const uint8_t* rtp_dump_data,
size_t rtp_dump_size) {
RunLoop loop;
ScopedBaseFakeClock fake_clock;
// Work around: webrtc calls webrtc::Random(clock.TimeInMicroseconds())
// everywhere and Random expects non-zero seed. Let's set the clock non-zero
// to make them happy.
fake_clock.SetTime(Timestamp::Millis(1));
// Attempt to create an RtpReader from the input file.
auto rtp_reader = CreateRtpReader(rtp_dump_data, rtp_dump_size);
if (rtp_reader == nullptr) {
RTC_LOG(LS_ERROR) << "Failed to create the rtp_reader";
return;
}
RtpHeaderExtensionMap extensions(/*extmap_allow_mixed=*/true);
// Skip i = 0 since it maps to kRtpExtensionNone.
for (int i = 1; i < kRtpExtensionNumberOfExtensions; i++) {
RTPExtensionType extension_type = static_cast<RTPExtensionType>(i);
// Extensions are registered with an ID, which you signal to the
// peer so they know what to expect. This code only cares about
// parsing so the value of the ID isn't relevant.
extensions.RegisterByType(i, extension_type);
}
// Setup the video streams based on the configuration.
CallConfig call_config(CreateEnvironment());
std::unique_ptr<Call> call(Call::Create(std::move(call_config)));
SetupVideoStreams(&receive_stream_configs, stream_state.get(), call.get());
// Start replaying the provided stream now that it has been configured.
for (const auto& receive_stream : stream_state->receive_streams) {
receive_stream->Start();
}
ReplayPackets(&fake_clock, call.get(), rtp_reader.get(), extensions);
for (const auto& receive_stream : stream_state->receive_streams) {
call->DestroyVideoReceiveStream(receive_stream);
}
}
std::vector<VideoReceiveStreamInterface::Config>
RtpReplayer::ReadConfigFromFile(const std::string& replay_config,
Transport* transport) {
Json::CharReaderBuilder factory;
std::unique_ptr<Json::CharReader> json_reader =
absl::WrapUnique(factory.newCharReader());
Json::Value json_configs;
Json::String errors;
if (!json_reader->parse(replay_config.data(),
replay_config.data() + replay_config.length(),
&json_configs, &errors)) {
RTC_LOG(LS_ERROR)
<< "Error parsing JSON replay configuration for the fuzzer: " << errors;
return {};
}
std::vector<VideoReceiveStreamInterface::Config> receive_stream_configs;
receive_stream_configs.reserve(json_configs.size());
for (const auto& json : json_configs) {
receive_stream_configs.push_back(
ParseVideoReceiveStreamJsonConfig(transport, json));
}
return receive_stream_configs;
}
void RtpReplayer::SetupVideoStreams(
std::vector<VideoReceiveStreamInterface::Config>* receive_stream_configs,
StreamState* stream_state,
Call* call) {
stream_state->decoder_factory = std::make_unique<InternalDecoderFactory>();
for (auto& receive_config : *receive_stream_configs) {
// Attach the decoder for the corresponding payload type in the config.
for (auto& decoder : receive_config.decoders) {
decoder = test::CreateMatchingDecoder(decoder.payload_type,
decoder.video_format.name);
}
// Create the window to display the rendered video.
stream_state->sinks.emplace_back(
test::VideoRenderer::Create("Fuzzing WebRTC Video Config", 640, 480));
// Create a receive stream for this config.
receive_config.renderer = stream_state->sinks.back().get();
receive_config.decoder_factory = stream_state->decoder_factory.get();
stream_state->receive_streams.emplace_back(
call->CreateVideoReceiveStream(std::move(receive_config)));
}
}
std::unique_ptr<test::RtpFileReader> RtpReplayer::CreateRtpReader(
const uint8_t* rtp_dump_data,
size_t rtp_dump_size) {
std::unique_ptr<test::RtpFileReader> rtp_reader(test::RtpFileReader::Create(
test::RtpFileReader::kRtpDump, rtp_dump_data, rtp_dump_size, {}));
if (!rtp_reader) {
RTC_LOG(LS_ERROR) << "Unable to open input file with any supported format";
return nullptr;
}
return rtp_reader;
}
void RtpReplayer::ReplayPackets(
FakeClock* clock,
Call* call,
test::RtpFileReader* rtp_reader,
const RtpPacketReceived::ExtensionManager& extensions) {
int64_t replay_start_ms = -1;
while (true) {
int64_t now_ms = TimeMillis();
if (replay_start_ms == -1) {
replay_start_ms = now_ms;
}
test::RtpPacket packet;
if (!rtp_reader->NextPacket(&packet)) {
break;
}
int64_t deliver_in_ms = replay_start_ms + packet.time_ms - now_ms;
if (deliver_in_ms > 0) {
// StatsCounter::ReportMetricToAggregatedCounter is O(elapsed time).
// Set an upper limit to prevent waste time.
clock->AdvanceTime(TimeDelta::Millis(
std::min(deliver_in_ms, static_cast<int64_t>(100))));
}
RtpPacketReceived received_packet(
&extensions, Timestamp::Micros(clock->TimeNanos() / 1000));
if (!received_packet.Parse(packet.data, packet.length)) {
RTC_LOG(LS_ERROR) << "Packet error, corrupt packets or incorrect setup?";
break;
}
// Set the clock rate - always 90K for video
received_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
call->Receiver()->DeliverRtpPacket(
MediaType::VIDEO, std::move(received_packet),
[&](const RtpPacketReceived& parsed_packet) {
RTC_LOG(LS_ERROR) << "Unknown SSRC: " << parsed_packet.Ssrc();
return false;
});
}
}
} // namespace test
} // namespace webrtc
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