blob: 3c9e22e63e94e45bac5fcc26667e6feb31d9eb07 [file] [log] [blame]
/*
* 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 "rtc_tools/rtp_generator/rtp_generator.h"
#include <algorithm>
#include <utility>
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/video_encoder_config.h"
#include "media/base/media_constants.h"
#include "rtc_base/strings/json.h"
#include "rtc_base/system/file_wrapper.h"
#include "rtc_base/thread.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace {
// Payload types.
constexpr int kPayloadTypeVp8 = 125;
constexpr int kPayloadTypeVp9 = 124;
constexpr int kPayloadTypeH264 = 123;
constexpr int kFakeVideoSendPayloadType = 122;
// Defaults
constexpr int kDefaultSsrc = 1337;
constexpr int kMaxConfigBufferSize = 8192;
// Utility function to validate a correct codec type has been passed in.
bool IsValidCodecType(const std::string& codec_name) {
return cricket::kVp8CodecName == codec_name ||
cricket::kVp9CodecName == codec_name ||
cricket::kH264CodecName == codec_name;
}
// Utility function to return some base payload type for a codec_name.
int GetDefaultTypeForPayloadName(const std::string& codec_name) {
if (cricket::kVp8CodecName == codec_name) {
return kPayloadTypeVp8;
}
if (cricket::kVp9CodecName == codec_name) {
return kPayloadTypeVp9;
}
if (cricket::kH264CodecName == codec_name) {
return kPayloadTypeH264;
}
return kFakeVideoSendPayloadType;
}
// Creates a single VideoSendStream configuration.
absl::optional<RtpGeneratorOptions::VideoSendStreamConfig>
ParseVideoSendStreamConfig(const Json::Value& json) {
RtpGeneratorOptions::VideoSendStreamConfig config;
// Parse video source settings.
if (!rtc::GetIntFromJsonObject(json, "duration_ms", &config.duration_ms)) {
RTC_LOG(LS_WARNING) << "duration_ms not specified using default: "
<< config.duration_ms;
}
if (!rtc::GetIntFromJsonObject(json, "video_width", &config.video_width)) {
RTC_LOG(LS_WARNING) << "video_width not specified using default: "
<< config.video_width;
}
if (!rtc::GetIntFromJsonObject(json, "video_height", &config.video_height)) {
RTC_LOG(LS_WARNING) << "video_height not specified using default: "
<< config.video_height;
}
if (!rtc::GetIntFromJsonObject(json, "video_fps", &config.video_fps)) {
RTC_LOG(LS_WARNING) << "video_fps not specified using default: "
<< config.video_fps;
}
if (!rtc::GetIntFromJsonObject(json, "num_squares", &config.num_squares)) {
RTC_LOG(LS_WARNING) << "num_squares not specified using default: "
<< config.num_squares;
}
// Parse RTP settings for this configuration.
config.rtp.ssrcs.push_back(kDefaultSsrc);
Json::Value rtp_json;
if (!rtc::GetValueFromJsonObject(json, "rtp", &rtp_json)) {
RTC_LOG(LS_ERROR) << "video_streams must have an rtp section";
return absl::nullopt;
}
if (!rtc::GetStringFromJsonObject(rtp_json, "payload_name",
&config.rtp.payload_name)) {
RTC_LOG(LS_ERROR) << "rtp.payload_name must be specified";
return absl::nullopt;
}
if (!IsValidCodecType(config.rtp.payload_name)) {
RTC_LOG(LS_ERROR) << "rtp.payload_name must be VP8,VP9 or H264";
return absl::nullopt;
}
config.rtp.payload_type =
GetDefaultTypeForPayloadName(config.rtp.payload_name);
if (!rtc::GetIntFromJsonObject(rtp_json, "payload_type",
&config.rtp.payload_type)) {
RTC_LOG(LS_WARNING)
<< "rtp.payload_type not specified using default for codec type"
<< config.rtp.payload_type;
}
return config;
}
} // namespace
absl::optional<RtpGeneratorOptions> ParseRtpGeneratorOptionsFromFile(
const std::string& options_file) {
if (!test::FileExists(options_file)) {
RTC_LOG(LS_ERROR) << " configuration file does not exist";
return absl::nullopt;
}
// Read the configuration file from disk.
FileWrapper config_file = FileWrapper::OpenReadOnly(options_file);
std::vector<char> raw_json_buffer(kMaxConfigBufferSize, 0);
size_t bytes_read =
config_file.Read(raw_json_buffer.data(), raw_json_buffer.size() - 1);
if (bytes_read == 0) {
RTC_LOG(LS_ERROR) << "Unable to read the configuration file.";
return absl::nullopt;
}
// Parse the file as JSON
Json::Reader json_reader;
Json::Value json;
if (!json_reader.parse(raw_json_buffer.data(), json)) {
RTC_LOG(LS_ERROR) << "Unable to parse the corpus config json file";
return absl::nullopt;
}
RtpGeneratorOptions gen_options;
for (const auto& video_stream_json : json["video_streams"]) {
absl::optional<RtpGeneratorOptions::VideoSendStreamConfig>
video_stream_config = ParseVideoSendStreamConfig(video_stream_json);
if (!video_stream_config.has_value()) {
RTC_LOG(LS_ERROR) << "Unable to parse the corpus config json file";
return absl::nullopt;
}
gen_options.video_streams.push_back(*video_stream_config);
}
return gen_options;
}
RtpGenerator::RtpGenerator(const RtpGeneratorOptions& options)
: options_(options),
video_encoder_factory_(CreateBuiltinVideoEncoderFactory()),
video_decoder_factory_(CreateBuiltinVideoDecoderFactory()),
video_bitrate_allocator_factory_(
CreateBuiltinVideoBitrateAllocatorFactory()),
event_log_(webrtc::RtcEventLog::CreateNull()),
call_(Call::Create(CallConfig(event_log_.get()))) {
constexpr int kMinBitrateBps = 30000; // 30 Kbps
constexpr int kMaxBitrateBps = 2500000; // 2.5 Mbps
int stream_count = 0;
for (const auto& send_config : options.video_streams) {
webrtc::VideoSendStream::Config video_config(this);
video_config.encoder_settings.encoder_factory =
video_encoder_factory_.get();
video_config.encoder_settings.bitrate_allocator_factory =
video_bitrate_allocator_factory_.get();
video_config.rtp = send_config.rtp;
// Update some required to be unique values.
stream_count++;
video_config.rtp.mid = "mid-" + std::to_string(stream_count);
video_config.track_id = "track-" + std::to_string(stream_count);
// Configure the video encoder configuration.
VideoEncoderConfig encoder_config;
encoder_config.content_type =
VideoEncoderConfig::ContentType::kRealtimeVideo;
encoder_config.codec_type =
PayloadStringToCodecType(video_config.rtp.payload_name);
if (video_config.rtp.payload_name == cricket::kVp8CodecName) {
VideoCodecVP8 settings = VideoEncoder::GetDefaultVp8Settings();
encoder_config.encoder_specific_settings = new rtc::RefCountedObject<
VideoEncoderConfig::Vp8EncoderSpecificSettings>(settings);
} else if (video_config.rtp.payload_name == cricket::kVp9CodecName) {
VideoCodecVP9 settings = VideoEncoder::GetDefaultVp9Settings();
encoder_config.encoder_specific_settings = new rtc::RefCountedObject<
VideoEncoderConfig::Vp9EncoderSpecificSettings>(settings);
} else if (video_config.rtp.payload_name == cricket::kH264CodecName) {
VideoCodecH264 settings = VideoEncoder::GetDefaultH264Settings();
encoder_config.encoder_specific_settings = new rtc::RefCountedObject<
VideoEncoderConfig::H264EncoderSpecificSettings>(settings);
}
encoder_config.video_format.name = video_config.rtp.payload_name;
encoder_config.min_transmit_bitrate_bps = 0;
encoder_config.max_bitrate_bps = kMaxBitrateBps;
encoder_config.content_type =
VideoEncoderConfig::ContentType::kRealtimeVideo;
// Configure the simulcast layers.
encoder_config.number_of_streams = video_config.rtp.ssrcs.size();
encoder_config.bitrate_priority = 1.0;
encoder_config.simulcast_layers.resize(encoder_config.number_of_streams);
for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
encoder_config.simulcast_layers[i].active = true;
encoder_config.simulcast_layers[i].min_bitrate_bps = kMinBitrateBps;
encoder_config.simulcast_layers[i].max_bitrate_bps = kMaxBitrateBps;
encoder_config.simulcast_layers[i].max_framerate = send_config.video_fps;
}
encoder_config.video_stream_factory =
new rtc::RefCountedObject<cricket::EncoderStreamFactory>(
video_config.rtp.payload_name, /*max qp*/ 56, /*screencast*/ false,
/*screenshare enabled*/ false);
// Setup the fake video stream for this.
std::unique_ptr<test::FrameGeneratorCapturer> frame_generator(
test::FrameGeneratorCapturer::Create(
send_config.video_width, send_config.video_height, absl::nullopt,
absl::nullopt, send_config.video_fps, Clock::GetRealTimeClock()));
frame_generator->Init();
VideoSendStream* video_send_stream = call_->CreateVideoSendStream(
std::move(video_config), std::move(encoder_config));
video_send_stream->SetSource(
frame_generator.get(),
webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
// Store these objects so we can destropy them at the end.
frame_generators_.push_back(std::move(frame_generator));
video_send_streams_.push_back(video_send_stream);
}
}
RtpGenerator::~RtpGenerator() {
for (VideoSendStream* send_stream : video_send_streams_) {
call_->DestroyVideoSendStream(send_stream);
}
}
void RtpGenerator::GenerateRtpDump(const std::string& rtp_dump_path) {
rtp_dump_writer_.reset(test::RtpFileWriter::Create(
test::RtpFileWriter::kRtpDump, rtp_dump_path));
call_->SignalChannelNetworkState(webrtc::MediaType::VIDEO,
webrtc::kNetworkUp);
for (VideoSendStream* send_stream : video_send_streams_) {
send_stream->Start();
}
// Spinlock until all the durations end.
WaitUntilAllVideoStreamsFinish();
call_->SignalChannelNetworkState(webrtc::MediaType::VIDEO,
webrtc::kNetworkDown);
}
bool RtpGenerator::SendRtp(const uint8_t* packet,
size_t length,
const webrtc::PacketOptions& options) {
test::RtpPacket rtp_packet = DataToRtpPacket(packet, length);
rtp_dump_writer_->WritePacket(&rtp_packet);
return true;
}
bool RtpGenerator::SendRtcp(const uint8_t* packet, size_t length) {
test::RtpPacket rtcp_packet = DataToRtpPacket(packet, length);
rtp_dump_writer_->WritePacket(&rtcp_packet);
return true;
}
int RtpGenerator::GetMaxDuration() const {
int max_end_ms = 0;
for (const auto& video_stream : options_.video_streams) {
max_end_ms = std::max(video_stream.duration_ms, max_end_ms);
}
return max_end_ms;
}
void RtpGenerator::WaitUntilAllVideoStreamsFinish() {
// Find the maximum duration required by the streams.
start_ms_ = Clock::GetRealTimeClock()->TimeInMilliseconds();
int64_t max_end_ms = start_ms_ + GetMaxDuration();
int64_t current_time = 0;
do {
int64_t min_wait_time = 0;
current_time = Clock::GetRealTimeClock()->TimeInMilliseconds();
// Stop any streams that are no longer active.
for (size_t i = 0; i < options_.video_streams.size(); ++i) {
const int64_t end_ms = start_ms_ + options_.video_streams[i].duration_ms;
if (current_time > end_ms) {
video_send_streams_[i]->Stop();
} else {
min_wait_time = std::min(min_wait_time, end_ms - current_time);
}
}
rtc::Thread::Current()->SleepMs(min_wait_time);
} while (current_time < max_end_ms);
}
test::RtpPacket RtpGenerator::DataToRtpPacket(const uint8_t* packet,
size_t packet_len) {
webrtc::test::RtpPacket rtp_packet;
memcpy(rtp_packet.data, packet, packet_len);
rtp_packet.length = packet_len;
rtp_packet.original_length = packet_len;
rtp_packet.time_ms =
webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_;
return rtp_packet;
}
} // namespace webrtc