blob: 405948d8e09549037ed7a69c33fece00e9eabaf8 [file] [log] [blame]
/*
* Copyright (c) 2014 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 <stdio.h>
#include <fstream>
#include <map>
#include <memory>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "api/field_trials.h"
#include "api/media_types.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "api/units/timestamp.h"
#include "api/video/video_codec_type.h"
#include "api/video_codecs/video_decoder.h"
#include "call/call.h"
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "media/engine/internal_decoder_factory.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "modules/video_coding/utility/ivf_file_writer.h"
#include "rtc_base/checks.h"
#include "rtc_base/string_to_number.h"
#include "rtc_base/strings/json.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/sleep.h"
#include "test/call_config_utils.h"
#include "test/call_test.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"
#include "test/gtest.h"
#include "test/null_transport.h"
#include "test/rtp_file_reader.h"
#include "test/run_loop.h"
#include "test/run_test.h"
#include "test/test_video_capturer.h"
#include "test/testsupport/frame_writer.h"
#include "test/time_controller/simulated_time_controller.h"
#include "test/video_renderer.h"
// Flag for payload type.
ABSL_FLAG(int,
media_payload_type,
webrtc::test::CallTest::kPayloadTypeVP8,
"Media payload type");
// Flag for RED payload type.
ABSL_FLAG(int,
red_payload_type,
webrtc::test::CallTest::kRedPayloadType,
"RED payload type");
// Flag for ULPFEC payload type.
ABSL_FLAG(int,
ulpfec_payload_type,
webrtc::test::CallTest::kUlpfecPayloadType,
"ULPFEC payload type");
// Flag for FLEXFEC payload type.
ABSL_FLAG(int,
flexfec_payload_type,
webrtc::test::CallTest::kFlexfecPayloadType,
"FLEXFEC payload type");
ABSL_FLAG(int,
media_payload_type_rtx,
webrtc::test::CallTest::kSendRtxPayloadType,
"Media over RTX payload type");
ABSL_FLAG(int,
red_payload_type_rtx,
webrtc::test::CallTest::kRtxRedPayloadType,
"RED over RTX payload type");
// Flag for SSRC and RTX SSRC.
ABSL_FLAG(uint32_t,
ssrc,
webrtc::test::CallTest::kVideoSendSsrcs[0],
"Incoming SSRC");
ABSL_FLAG(uint32_t,
ssrc_rtx,
webrtc::test::CallTest::kSendRtxSsrcs[0],
"Incoming RTX SSRC");
ABSL_FLAG(uint32_t,
ssrc_flexfec,
webrtc::test::CallTest::kFlexfecSendSsrc,
"Incoming FLEXFEC SSRC");
// Flag for abs-send-time id.
ABSL_FLAG(int, abs_send_time_id, -1, "RTP extension ID for abs-send-time");
// Flag for transmission-offset id.
ABSL_FLAG(int,
transmission_offset_id,
-1,
"RTP extension ID for transmission-offset");
// Flag for rtpdump input file.
ABSL_FLAG(std::string, input_file, "", "input file");
ABSL_FLAG(std::string, config_file, "", "config file");
// Flag for raw output files.
ABSL_FLAG(std::string,
out_base,
"",
"Basename (excluding .jpg) for raw output");
ABSL_FLAG(std::string,
decoder_bitstream_filename,
"",
"Decoder bitstream output file");
ABSL_FLAG(std::string, decoder_ivf_filename, "", "Decoder ivf output file");
// Flag for video codec.
ABSL_FLAG(std::string, codec, "VP8", "Video codec");
// Flags for rtp start and stop timestamp.
ABSL_FLAG(uint32_t,
start_timestamp,
0,
"RTP start timestamp, packets with smaller timestamp will be ignored "
"(no wraparound)");
ABSL_FLAG(uint32_t,
stop_timestamp,
4294967295,
"RTP stop timestamp, packets with larger timestamp will be ignored "
"(no wraparound)");
// Flags for render window width and height
ABSL_FLAG(uint32_t, render_width, 640, "Width of render window");
ABSL_FLAG(uint32_t, render_height, 480, "Height of render window");
ABSL_FLAG(
std::string,
force_fieldtrials,
"",
"Field trials control experimental feature code which can be forced. "
"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
" will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
"trials are separated by \"/\"");
ABSL_FLAG(bool, simulated_time, false, "Run in simulated time");
ABSL_FLAG(bool, disable_preview, false, "Disable decoded video preview.");
ABSL_FLAG(bool, disable_decoding, false, "Disable video decoding.");
ABSL_FLAG(int,
extend_run_time_duration,
0,
"Extends the run time of the receiving client after the last RTP "
"packet has been delivered. Typically useful to let the last few "
"frames be decoded and rendered. Duration given in seconds.");
namespace {
bool ValidatePayloadType(int32_t payload_type) {
return payload_type > 0 && payload_type <= 127;
}
bool ValidateOptionalPayloadType(int32_t payload_type) {
return payload_type == -1 || ValidatePayloadType(payload_type);
}
bool ValidateRtpHeaderExtensionId(int32_t extension_id) {
return extension_id >= -1 && extension_id < 15;
}
bool ValidateInputFilenameNotEmpty(const std::string& string) {
return !string.empty();
}
} // namespace
namespace webrtc {
namespace {
const uint32_t kReceiverLocalSsrc = 0x123456;
class NullRenderer : public rtc::VideoSinkInterface<VideoFrame> {
public:
void OnFrame(const VideoFrame& frame) override {}
};
class FileRenderPassthrough : public rtc::VideoSinkInterface<VideoFrame> {
public:
FileRenderPassthrough(const std::string& basename,
rtc::VideoSinkInterface<VideoFrame>* renderer)
: basename_(basename), renderer_(renderer), file_(nullptr), count_(0) {}
~FileRenderPassthrough() override {
if (file_)
fclose(file_);
}
private:
void OnFrame(const VideoFrame& video_frame) override {
if (renderer_)
renderer_->OnFrame(video_frame);
if (basename_.empty())
return;
std::stringstream filename;
filename << basename_ << count_++ << "_" << video_frame.timestamp()
<< ".jpg";
test::JpegFrameWriter frame_writer(filename.str());
RTC_CHECK(frame_writer.WriteFrame(video_frame, 100));
}
const std::string basename_;
rtc::VideoSinkInterface<VideoFrame>* const renderer_;
FILE* file_;
size_t count_;
};
class DecoderBitstreamFileWriter : public test::FakeDecoder {
public:
explicit DecoderBitstreamFileWriter(const char* filename)
: file_(fopen(filename, "wb")) {
RTC_DCHECK(file_);
}
~DecoderBitstreamFileWriter() override { fclose(file_); }
int32_t Decode(const EncodedImage& encoded_frame,
bool /* missing_frames */,
int64_t /* render_time_ms */) override {
if (fwrite(encoded_frame.data(), 1, encoded_frame.size(), file_) <
encoded_frame.size()) {
RTC_LOG_ERR(LS_ERROR) << "fwrite of encoded frame failed.";
return WEBRTC_VIDEO_CODEC_ERROR;
}
return WEBRTC_VIDEO_CODEC_OK;
}
private:
FILE* file_;
};
class DecoderIvfFileWriter : public test::FakeDecoder {
public:
explicit DecoderIvfFileWriter(const char* filename, const std::string& codec)
: file_writer_(
IvfFileWriter::Wrap(FileWrapper::OpenWriteOnly(filename), 0)) {
RTC_DCHECK(file_writer_.get());
if (codec == "VP8") {
video_codec_type_ = VideoCodecType::kVideoCodecVP8;
} else if (codec == "VP9") {
video_codec_type_ = VideoCodecType::kVideoCodecVP9;
} else if (codec == "H264") {
video_codec_type_ = VideoCodecType::kVideoCodecH264;
} else if (codec == "AV1") {
video_codec_type_ = VideoCodecType::kVideoCodecAV1;
} else {
RTC_LOG(LS_ERROR) << "Unsupported video codec " << codec;
RTC_DCHECK_NOTREACHED();
}
}
~DecoderIvfFileWriter() override { file_writer_->Close(); }
int32_t Decode(const EncodedImage& encoded_frame,
bool /* missing_frames */,
int64_t render_time_ms) override {
if (!file_writer_->WriteFrame(encoded_frame, video_codec_type_)) {
return WEBRTC_VIDEO_CODEC_ERROR;
}
return WEBRTC_VIDEO_CODEC_OK;
}
private:
std::unique_ptr<IvfFileWriter> file_writer_;
VideoCodecType video_codec_type_;
};
// Holds all the shared memory structures required for a receive stream. This
// structure is used to prevent members being deallocated before the replay
// has been finished.
struct StreamState {
test::NullTransport transport;
std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks;
std::vector<VideoReceiveStreamInterface*> receive_streams;
std::vector<FlexfecReceiveStream*> flexfec_streams;
std::unique_ptr<VideoDecoderFactory> decoder_factory;
};
// Loads multiple configurations from the provided configuration file.
std::unique_ptr<StreamState> ConfigureFromFile(const std::string& config_path,
Call* call) {
auto stream_state = std::make_unique<StreamState>();
// Parse the configuration file.
std::ifstream config_file(config_path);
std::stringstream raw_json_buffer;
raw_json_buffer << config_file.rdbuf();
std::string raw_json = raw_json_buffer.str();
Json::CharReaderBuilder builder;
Json::Value json_configs;
std::string error_message;
std::unique_ptr<Json::CharReader> json_reader(builder.newCharReader());
if (!json_reader->parse(raw_json.data(), raw_json.data() + raw_json.size(),
&json_configs, &error_message)) {
fprintf(stderr, "Error parsing JSON config\n");
fprintf(stderr, "%s\n", error_message.c_str());
return nullptr;
}
if (absl::GetFlag(FLAGS_disable_decoding)) {
stream_state->decoder_factory =
std::make_unique<test::FunctionVideoDecoderFactory>(
[]() { return std::make_unique<test::FakeDecoder>(); });
} else {
stream_state->decoder_factory = std::make_unique<InternalDecoderFactory>();
}
size_t config_count = 0;
for (const auto& json : json_configs) {
// Create the configuration and parse the JSON into the config.
auto receive_config =
ParseVideoReceiveStreamJsonConfig(&(stream_state->transport), json);
// Instantiate the underlying decoder.
for (auto& decoder : receive_config.decoders) {
decoder = test::CreateMatchingDecoder(decoder.payload_type,
decoder.video_format.name);
}
// Create a window for this config.
std::stringstream window_title;
window_title << "Playback Video (" << config_count++ << ")";
if (absl::GetFlag(FLAGS_disable_preview)) {
stream_state->sinks.emplace_back(std::make_unique<NullRenderer>());
} else {
stream_state->sinks.emplace_back(test::VideoRenderer::Create(
window_title.str().c_str(), absl::GetFlag(FLAGS_render_width),
absl::GetFlag(FLAGS_render_height)));
}
// 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)));
}
return stream_state;
}
// Loads the base configuration from flags passed in on the commandline.
std::unique_ptr<StreamState> ConfigureFromFlags(
const std::string& rtp_dump_path,
Call* call) {
auto stream_state = std::make_unique<StreamState>();
// Create the video renderers. We must add both to the stream state to keep
// them from deallocating.
std::stringstream window_title;
window_title << "Playback Video (" << rtp_dump_path << ")";
std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> playback_video;
if (absl::GetFlag(FLAGS_disable_preview)) {
playback_video = std::make_unique<NullRenderer>();
} else {
playback_video.reset(test::VideoRenderer::Create(
window_title.str().c_str(), absl::GetFlag(FLAGS_render_width),
absl::GetFlag(FLAGS_render_height)));
}
auto file_passthrough = std::make_unique<FileRenderPassthrough>(
absl::GetFlag(FLAGS_out_base), playback_video.get());
stream_state->sinks.push_back(std::move(playback_video));
stream_state->sinks.push_back(std::move(file_passthrough));
// Setup the configuration from the flags.
VideoReceiveStreamInterface::Config receive_config(
&(stream_state->transport));
receive_config.rtp.remote_ssrc = absl::GetFlag(FLAGS_ssrc);
receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
receive_config.rtp.rtx_ssrc = absl::GetFlag(FLAGS_ssrc_rtx);
receive_config.rtp.rtx_associated_payload_types[absl::GetFlag(
FLAGS_media_payload_type_rtx)] = absl::GetFlag(FLAGS_media_payload_type);
receive_config.rtp
.rtx_associated_payload_types[absl::GetFlag(FLAGS_red_payload_type_rtx)] =
absl::GetFlag(FLAGS_red_payload_type);
receive_config.rtp.ulpfec_payload_type =
absl::GetFlag(FLAGS_ulpfec_payload_type);
receive_config.rtp.red_payload_type = absl::GetFlag(FLAGS_red_payload_type);
receive_config.rtp.nack.rtp_history_ms = 1000;
if (absl::GetFlag(FLAGS_flexfec_payload_type) != -1) {
receive_config.rtp.protected_by_flexfec = true;
FlexfecReceiveStream::Config flexfec_config(&(stream_state->transport));
flexfec_config.payload_type = absl::GetFlag(FLAGS_flexfec_payload_type);
flexfec_config.protected_media_ssrcs.push_back(absl::GetFlag(FLAGS_ssrc));
flexfec_config.rtp.remote_ssrc = absl::GetFlag(FLAGS_ssrc_flexfec);
FlexfecReceiveStream* flexfec_stream =
call->CreateFlexfecReceiveStream(flexfec_config);
receive_config.rtp.packet_sink_ = flexfec_stream;
stream_state->flexfec_streams.push_back(flexfec_stream);
}
receive_config.renderer = stream_state->sinks.back().get();
// Setup the receiving stream
VideoReceiveStreamInterface::Decoder decoder;
decoder = test::CreateMatchingDecoder(absl::GetFlag(FLAGS_media_payload_type),
absl::GetFlag(FLAGS_codec));
if (!absl::GetFlag(FLAGS_decoder_bitstream_filename).empty()) {
// Replace decoder with file writer if we're writing the bitstream to a
// file instead.
stream_state->decoder_factory =
std::make_unique<test::FunctionVideoDecoderFactory>([]() {
return std::make_unique<DecoderBitstreamFileWriter>(
absl::GetFlag(FLAGS_decoder_bitstream_filename).c_str());
});
} else if (!absl::GetFlag(FLAGS_decoder_ivf_filename).empty()) {
// Replace decoder with file writer if we're writing the ivf to a
// file instead.
stream_state->decoder_factory =
std::make_unique<test::FunctionVideoDecoderFactory>([]() {
return std::make_unique<DecoderIvfFileWriter>(
absl::GetFlag(FLAGS_decoder_ivf_filename).c_str(),
absl::GetFlag(FLAGS_codec));
});
} else if (absl::GetFlag(FLAGS_disable_decoding)) {
stream_state->decoder_factory =
std::make_unique<test::FunctionVideoDecoderFactory>(
[]() { return std::make_unique<test::FakeDecoder>(); });
} else {
stream_state->decoder_factory = std::make_unique<InternalDecoderFactory>();
}
receive_config.decoder_factory = stream_state->decoder_factory.get();
receive_config.decoders.push_back(decoder);
stream_state->receive_streams.emplace_back(
call->CreateVideoReceiveStream(std::move(receive_config)));
return stream_state;
}
std::unique_ptr<test::RtpFileReader> CreateRtpReader(
const std::string& rtp_dump_path) {
std::unique_ptr<test::RtpFileReader> rtp_reader(test::RtpFileReader::Create(
test::RtpFileReader::kRtpDump, rtp_dump_path));
if (!rtp_reader) {
rtp_reader.reset(
test::RtpFileReader::Create(test::RtpFileReader::kPcap, rtp_dump_path));
if (!rtp_reader) {
fprintf(stderr,
"Couldn't open input file as either a rtpdump or .pcap. Note "
"that .pcapng is not supported.\nTrying to interpret the file as "
"length/packet interleaved.\n");
rtp_reader.reset(test::RtpFileReader::Create(
test::RtpFileReader::kLengthPacketInterleaved, rtp_dump_path));
if (!rtp_reader) {
fprintf(stderr,
"Unable to open input file with any supported format\n");
return nullptr;
}
}
}
return rtp_reader;
}
// The RtpReplayer is responsible for parsing the configuration provided by
// the user, setting up the windows, receive streams and decoders and then
// replaying the provided RTP dump.
class RtpReplayer final {
public:
RtpReplayer(absl::string_view replay_config_path,
absl::string_view rtp_dump_path,
std::unique_ptr<FieldTrialsView> field_trials,
bool simulated_time)
: replay_config_path_(replay_config_path),
rtp_dump_path_(rtp_dump_path),
field_trials_(std::move(field_trials)),
rtp_reader_(CreateRtpReader(rtp_dump_path_)) {
TaskQueueFactory* task_queue_factory;
if (simulated_time) {
time_sim_ = std::make_unique<GlobalSimulatedTimeController>(
Timestamp::Millis(1 << 30));
task_queue_factory = time_sim_->GetTaskQueueFactory();
} else {
task_queue_factory_ = CreateDefaultTaskQueueFactory(field_trials_.get()),
task_queue_factory = task_queue_factory_.get();
}
worker_thread_ =
std::make_unique<rtc::TaskQueue>(task_queue_factory->CreateTaskQueue(
"worker_thread", TaskQueueFactory::Priority::NORMAL));
rtc::Event event;
worker_thread_->PostTask([&]() {
Call::Config call_config(&event_log_);
call_config.trials = field_trials_.get();
call_config.task_queue_factory = task_queue_factory;
call_.reset(Call::Create(call_config));
// Creation of the streams must happen inside a task queue because it is
// resued as a worker thread.
if (replay_config_path_.empty()) {
stream_state_ = ConfigureFromFlags(rtp_dump_path_, call_.get());
} else {
stream_state_ = ConfigureFromFile(replay_config_path_, call_.get());
}
event.Set();
});
event.Wait(/*give_up_after=*/TimeDelta::Seconds(10));
RTC_CHECK(stream_state_);
RTC_CHECK(rtp_reader_);
}
~RtpReplayer() {
// Destruction of streams and the call must happen on the same thread as
// their creation.
rtc::Event event;
worker_thread_->PostTask([&]() {
for (const auto& receive_stream : stream_state_->receive_streams) {
call_->DestroyVideoReceiveStream(receive_stream);
}
for (const auto& flexfec_stream : stream_state_->flexfec_streams) {
call_->DestroyFlexfecReceiveStream(flexfec_stream);
}
call_.reset();
event.Set();
});
event.Wait(/*give_up_after=*/TimeDelta::Seconds(10));
}
void Run() {
rtc::Event event;
worker_thread_->PostTask([&]() {
// Start replaying the provided stream now that it has been configured.
// VideoReceiveStreams must be started on the same thread as they were
// created on.
for (const auto& receive_stream : stream_state_->receive_streams) {
receive_stream->Start();
}
event.Set();
});
event.Wait(/*give_up_after=*/TimeDelta::Seconds(10));
ReplayPackets();
}
private:
void ReplayPackets() {
enum class Result { kOk, kUnknownSsrc, kParsingFailed };
int64_t replay_start_ms = -1;
int num_packets = 0;
std::map<uint32_t, int> unknown_packets;
rtc::Event event(/*manual_reset=*/false, /*initially_signalled=*/false);
uint32_t start_timestamp = absl::GetFlag(FLAGS_start_timestamp);
uint32_t stop_timestamp = absl::GetFlag(FLAGS_stop_timestamp);
RtpHeaderExtensionMap extensions;
if (absl::GetFlag(FLAGS_transmission_offset_id) != -1) {
extensions.RegisterByUri(absl::GetFlag(FLAGS_transmission_offset_id),
RtpExtension::kTimestampOffsetUri);
}
if (absl::GetFlag(FLAGS_abs_send_time_id) != -1) {
extensions.RegisterByUri(absl::GetFlag(FLAGS_abs_send_time_id),
RtpExtension::kAbsSendTimeUri);
}
while (true) {
int64_t now_ms = CurrentTimeMs();
if (replay_start_ms == -1) {
replay_start_ms = now_ms;
}
test::RtpPacket packet;
if (!rtp_reader_->NextPacket(&packet)) {
break;
}
rtc::CopyOnWriteBuffer packet_buffer(
packet.original_length > 0 ? packet.original_length : packet.length);
memcpy(packet_buffer.MutableData(), packet.data, packet.length);
if (packet.length < packet.original_length) {
// Only the RTP header was recorded in the RTP dump, payload is not
// known and and padding length is not known, zero the payload and
// clear the padding bit.
memset(packet_buffer.MutableData() + packet.length, 0,
packet.original_length - packet.length);
packet_buffer.MutableData()[0] &= ~0x20;
}
RtpPacket header;
header.Parse(packet_buffer);
if (header.Timestamp() < start_timestamp ||
header.Timestamp() > stop_timestamp) {
continue;
}
int64_t deliver_in_ms = replay_start_ms + packet.time_ms - now_ms;
SleepOrAdvanceTime(deliver_in_ms);
++num_packets;
Result result = Result::kOk;
worker_thread_->PostTask([&]() {
if (IsRtcpPacket(packet_buffer)) {
call_->Receiver()->DeliverRtcpPacket(std::move(packet_buffer));
}
RtpPacketReceived received_packet(&extensions,
Timestamp::Millis(CurrentTimeMs()));
if (!received_packet.Parse(std::move(packet_buffer))) {
result = Result::kParsingFailed;
return;
}
call_->Receiver()->DeliverRtpPacket(
MediaType::VIDEO, received_packet,
[&result](const RtpPacketReceived& parsed_packet) -> bool {
result = Result::kUnknownSsrc;
// No point in trying to demux again.
return false;
});
event.Set();
});
event.Wait(/*give_up_after=*/TimeDelta::Seconds(10));
switch (result) {
case Result::kOk:
break;
case Result::kUnknownSsrc: {
if (unknown_packets[header.Ssrc()] == 0)
fprintf(stderr, "Unknown SSRC: %u!\n", header.Ssrc());
++unknown_packets[header.Ssrc()];
break;
}
case Result::kParsingFailed: {
fprintf(stderr,
"Packet error, corrupt packets or incorrect setup?\n");
fprintf(stderr, "Packet len=%zu pt=%u seq=%u ts=%u ssrc=0x%8x\n",
packet.length, header.PayloadType(), header.SequenceNumber(),
header.Timestamp(), header.Ssrc());
break;
}
}
}
// Note that even when `extend_run_time_duration` is zero
// `SleepOrAdvanceTime` should still be called in order to process the last
// delivered packet when running in simulated time.
SleepOrAdvanceTime(absl::GetFlag(FLAGS_extend_run_time_duration) * 1000);
fprintf(stderr, "num_packets: %d\n", num_packets);
for (std::map<uint32_t, int>::const_iterator it = unknown_packets.begin();
it != unknown_packets.end(); ++it) {
fprintf(stderr, "Packets for unknown ssrc '%u': %d\n", it->first,
it->second);
}
}
int64_t CurrentTimeMs() {
return time_sim_ ? time_sim_->GetClock()->TimeInMilliseconds()
: rtc::TimeMillis();
}
void SleepOrAdvanceTime(int64_t duration_ms) {
if (time_sim_) {
time_sim_->AdvanceTime(TimeDelta::Millis(duration_ms));
} else if (duration_ms > 0) {
SleepMs(duration_ms);
}
}
const std::string replay_config_path_;
const std::string rtp_dump_path_;
RtcEventLogNull event_log_;
std::unique_ptr<FieldTrialsView> field_trials_;
std::unique_ptr<GlobalSimulatedTimeController> time_sim_;
std::unique_ptr<TaskQueueFactory> task_queue_factory_;
std::unique_ptr<rtc::TaskQueue> worker_thread_;
std::unique_ptr<Call> call_;
std::unique_ptr<test::RtpFileReader> rtp_reader_;
std::unique_ptr<StreamState> stream_state_;
};
void RtpReplay() {
RtpReplayer replayer(
absl::GetFlag(FLAGS_config_file), absl::GetFlag(FLAGS_input_file),
std::make_unique<FieldTrials>(absl::GetFlag(FLAGS_force_fieldtrials)),
absl::GetFlag(FLAGS_simulated_time));
replayer.Run();
}
} // namespace
} // namespace webrtc
int main(int argc, char* argv[]) {
::testing::InitGoogleTest(&argc, argv);
absl::ParseCommandLine(argc, argv);
RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_media_payload_type)));
RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_media_payload_type_rtx)));
RTC_CHECK(ValidateOptionalPayloadType(absl::GetFlag(FLAGS_red_payload_type)));
RTC_CHECK(
ValidateOptionalPayloadType(absl::GetFlag(FLAGS_red_payload_type_rtx)));
RTC_CHECK(
ValidateOptionalPayloadType(absl::GetFlag(FLAGS_ulpfec_payload_type)));
RTC_CHECK(
ValidateOptionalPayloadType(absl::GetFlag(FLAGS_flexfec_payload_type)));
RTC_CHECK(
ValidateRtpHeaderExtensionId(absl::GetFlag(FLAGS_abs_send_time_id)));
RTC_CHECK(ValidateRtpHeaderExtensionId(
absl::GetFlag(FLAGS_transmission_offset_id)));
RTC_CHECK(ValidateInputFilenameNotEmpty(absl::GetFlag(FLAGS_input_file)));
RTC_CHECK_GE(absl::GetFlag(FLAGS_extend_run_time_duration), 0);
rtc::ThreadManager::Instance()->WrapCurrentThread();
webrtc::test::RunTest(webrtc::RtpReplay);
return 0;
}