blob: 3f35a5525d690840e0d5c1da0bc2bdc2f199b2a1 [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/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/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/source/rtp_packet.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/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");
namespace {
static bool ValidatePayloadType(int32_t payload_type) {
return payload_type > 0 && payload_type <= 127;
}
static bool ValidateOptionalPayloadType(int32_t payload_type) {
return payload_type == -1 || ValidatePayloadType(payload_type);
}
static bool ValidateRtpHeaderExtensionId(int32_t extension_id) {
return extension_id >= -1 && extension_id < 15;
}
bool ValidateInputFilenameNotEmpty(const std::string& string) {
return !string.empty();
}
static int MediaPayloadType() {
return absl::GetFlag(FLAGS_media_payload_type);
}
static int RedPayloadType() {
return absl::GetFlag(FLAGS_red_payload_type);
}
static int UlpfecPayloadType() {
return absl::GetFlag(FLAGS_ulpfec_payload_type);
}
static int FlexfecPayloadType() {
return absl::GetFlag(FLAGS_flexfec_payload_type);
}
static int MediaPayloadTypeRtx() {
return absl::GetFlag(FLAGS_media_payload_type_rtx);
}
static int RedPayloadTypeRtx() {
return absl::GetFlag(FLAGS_red_payload_type_rtx);
}
static uint32_t Ssrc() {
return absl::GetFlag(FLAGS_ssrc);
}
static uint32_t SsrcRtx() {
return absl::GetFlag(FLAGS_ssrc_rtx);
}
static uint32_t SsrcFlexfec() {
return absl::GetFlag(FLAGS_ssrc_flexfec);
}
static int AbsSendTimeId() {
return absl::GetFlag(FLAGS_abs_send_time_id);
}
static int TransmissionOffsetId() {
return absl::GetFlag(FLAGS_transmission_offset_id);
}
static std::string InputFile() {
return absl::GetFlag(FLAGS_input_file);
}
static std::string ConfigFile() {
return absl::GetFlag(FLAGS_config_file);
}
static std::string OutBase() {
return absl::GetFlag(FLAGS_out_base);
}
static std::string DecoderBitstreamFilename() {
return absl::GetFlag(FLAGS_decoder_bitstream_filename);
}
static std::string IVFFilename() {
return absl::GetFlag(FLAGS_decoder_ivf_filename);
}
static std::string Codec() {
return absl::GetFlag(FLAGS_codec);
}
static uint32_t RenderWidth() {
return absl::GetFlag(FLAGS_render_width);
}
static uint32_t RenderHeight() {
return absl::GetFlag(FLAGS_render_height);
}
} // namespace
namespace webrtc {
static const uint32_t kReceiverLocalSsrc = 0x123456;
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_;
};
// 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:
// Replay a rtp dump with an optional json configuration.
static void Replay(const std::string& replay_config_path,
const std::string& rtp_dump_path) {
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
webrtc::CreateDefaultTaskQueueFactory();
auto worker_thread = task_queue_factory->CreateTaskQueue(
"worker_thread", TaskQueueFactory::Priority::NORMAL);
rtc::Event sync_event(/*manual_reset=*/false,
/*initially_signalled=*/false);
webrtc::RtcEventLogNull event_log;
Call::Config call_config(&event_log);
call_config.task_queue_factory = task_queue_factory.get();
call_config.trials = new FieldTrialBasedConfig();
std::unique_ptr<Call> call;
std::unique_ptr<StreamState> stream_state;
// Creation of the streams must happen inside a task queue because it is
// resued as a worker thread.
worker_thread->PostTask([&]() {
call.reset(Call::Create(call_config));
// Attempt to load the configuration
if (replay_config_path.empty()) {
stream_state = ConfigureFromFlags(rtp_dump_path, call.get());
} else {
stream_state = ConfigureFromFile(replay_config_path, call.get());
}
if (stream_state == nullptr) {
return;
}
// 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();
}
sync_event.Set();
});
// Attempt to create an RtpReader from the input file.
std::unique_ptr<test::RtpFileReader> rtp_reader =
CreateRtpReader(rtp_dump_path);
// Wait for streams creation.
sync_event.Wait(/*give_up_after_ms=*/10000);
if (stream_state == nullptr || rtp_reader == nullptr) {
return;
}
ReplayPackets(call.get(), rtp_reader.get(), worker_thread.get());
// Destruction of streams and the call must happen on the same thread as
// their creation.
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();
sync_event.Set();
});
sync_event.Wait(/*give_up_after_ms=*/10000);
}
private:
// 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.
static 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;
}
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++ << ")";
stream_state->sinks.emplace_back(test::VideoRenderer::Create(
window_title.str().c_str(), RenderWidth(), RenderHeight()));
// 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.
static 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<test::VideoRenderer> playback_video(
test::VideoRenderer::Create(window_title.str().c_str(), RenderWidth(),
RenderHeight()));
auto file_passthrough = std::make_unique<FileRenderPassthrough>(
OutBase(), 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 = Ssrc();
receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
receive_config.rtp.rtx_ssrc = SsrcRtx();
receive_config.rtp.rtx_associated_payload_types[MediaPayloadTypeRtx()] =
MediaPayloadType();
receive_config.rtp.rtx_associated_payload_types[RedPayloadTypeRtx()] =
RedPayloadType();
receive_config.rtp.ulpfec_payload_type = UlpfecPayloadType();
receive_config.rtp.red_payload_type = RedPayloadType();
receive_config.rtp.nack.rtp_history_ms = 1000;
if (FlexfecPayloadType() != -1) {
receive_config.rtp.protected_by_flexfec = true;
webrtc::FlexfecReceiveStream::Config flexfec_config(
&(stream_state->transport));
flexfec_config.payload_type = FlexfecPayloadType();
flexfec_config.protected_media_ssrcs.push_back(Ssrc());
flexfec_config.rtp.remote_ssrc = SsrcFlexfec();
FlexfecReceiveStream* flexfec_stream =
call->CreateFlexfecReceiveStream(flexfec_config);
receive_config.rtp.packet_sink_ = flexfec_stream;
stream_state->flexfec_streams.push_back(flexfec_stream);
}
if (TransmissionOffsetId() != -1) {
receive_config.rtp.extensions.push_back(RtpExtension(
RtpExtension::kTimestampOffsetUri, TransmissionOffsetId()));
}
if (AbsSendTimeId() != -1) {
receive_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, AbsSendTimeId()));
}
receive_config.renderer = stream_state->sinks.back().get();
// Setup the receiving stream
VideoReceiveStreamInterface::Decoder decoder;
decoder = test::CreateMatchingDecoder(MediaPayloadType(), Codec());
if (!DecoderBitstreamFilename().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>(
DecoderBitstreamFilename().c_str());
});
} else if (!IVFFilename().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>(IVFFilename().c_str(),
Codec());
});
} 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;
}
static 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;
}
static void ReplayPackets(Call* call,
test::RtpFileReader* rtp_reader,
TaskQueueBase* worker_thread) {
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);
while (true) {
int64_t now_ms = rtc::TimeMillis();
if (replay_start_ms == -1) {
replay_start_ms = now_ms;
}
test::RtpPacket packet;
if (!rtp_reader->NextPacket(&packet)) {
break;
}
rtc::CopyOnWriteBuffer packet_buffer(packet.data, packet.length);
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;
if (deliver_in_ms > 0) {
SleepMs(deliver_in_ms);
}
++num_packets;
PacketReceiver::DeliveryStatus result = PacketReceiver::DELIVERY_OK;
worker_thread->PostTask([&]() {
result = call->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
std::move(packet_buffer),
/* packet_time_us */ -1);
event.Set();
});
event.Wait(/*give_up_after_ms=*/10000);
switch (result) {
case PacketReceiver::DELIVERY_OK:
break;
case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
if (unknown_packets[header.Ssrc()] == 0)
fprintf(stderr, "Unknown SSRC: %u!\n", header.Ssrc());
++unknown_packets[header.Ssrc()];
break;
}
case PacketReceiver::DELIVERY_PACKET_ERROR: {
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;
}
}
}
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);
}
}
}; // class RtpReplayer
void RtpReplay() {
RtpReplayer::Replay(ConfigFile(), InputFile());
}
} // 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::ThreadManager::Instance()->WrapCurrentThread();
webrtc::test::RunTest(webrtc::RtpReplay);
return 0;
}