| /* |
| * 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/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/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); |
| } |
| |
| if (absl::GetFlag(FLAGS_transmission_offset_id) != -1) { |
| receive_config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, |
| absl::GetFlag(FLAGS_transmission_offset_id))); |
| } |
| if (absl::GetFlag(FLAGS_abs_send_time_id) != -1) { |
| receive_config.rtp.extensions.push_back(RtpExtension( |
| RtpExtension::kAbsSendTimeUri, absl::GetFlag(FLAGS_abs_send_time_id))); |
| } |
| 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() { |
| 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 = 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; |
| PacketReceiver::DeliveryStatus result = PacketReceiver::DELIVERY_OK; |
| worker_thread_->PostTask([&]() { |
| MediaType media_type = |
| IsRtcpPacket(packet_buffer) ? MediaType::ANY : MediaType::VIDEO; |
| result = call_->Receiver()->DeliverPacket(media_type, |
| std::move(packet_buffer), |
| /* packet_time_us */ -1); |
| event.Set(); |
| }); |
| event.Wait(/*give_up_after=*/TimeDelta::Seconds(10)); |
| |
| 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; |
| } |
| } |
| } |
| // 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; |
| } |
