| /* |
| * 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 <map> |
| #include <memory> |
| #include <sstream> |
| |
| #include "webrtc/api/video_codecs/video_decoder.h" |
| #include "webrtc/call/call.h" |
| #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
| #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/rtc_base/flags.h" |
| #include "webrtc/rtc_base/string_to_number.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/system_wrappers/include/sleep.h" |
| #include "webrtc/test/call_test.h" |
| #include "webrtc/test/encoder_settings.h" |
| #include "webrtc/test/fake_decoder.h" |
| #include "webrtc/test/gtest.h" |
| #include "webrtc/test/null_transport.h" |
| #include "webrtc/test/rtp_file_reader.h" |
| #include "webrtc/test/run_loop.h" |
| #include "webrtc/test/run_test.h" |
| #include "webrtc/test/testsupport/frame_writer.h" |
| #include "webrtc/test/video_capturer.h" |
| #include "webrtc/test/video_renderer.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace { |
| |
| static bool ValidatePayloadType(int32_t payload_type) { |
| return payload_type > 0 && payload_type <= 127; |
| } |
| |
| static bool ValidateSsrc(const char* ssrc_string) { |
| return rtc::StringToNumber<uint32_t>(ssrc_string).has_value(); |
| } |
| |
| 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(); |
| } |
| |
| } // namespace |
| |
| namespace webrtc { |
| namespace flags { |
| |
| // TODO(pbos): Multiple receivers. |
| |
| // Flag for payload type. |
| DEFINE_int(payload_type, test::CallTest::kPayloadTypeVP8, "Payload type"); |
| static int PayloadType() { return static_cast<int>(FLAG_payload_type); } |
| |
| DEFINE_int(payload_type_rtx, |
| test::CallTest::kSendRtxPayloadType, |
| "RTX payload type"); |
| static int PayloadTypeRtx() { |
| return static_cast<int>(FLAG_payload_type_rtx); |
| } |
| |
| // Flag for SSRC. |
| const std::string& DefaultSsrc() { |
| static const std::string ssrc = std::to_string( |
| test::CallTest::kVideoSendSsrcs[0]); |
| return ssrc; |
| } |
| DEFINE_string(ssrc, DefaultSsrc().c_str(), "Incoming SSRC"); |
| static uint32_t Ssrc() { |
| return rtc::StringToNumber<uint32_t>(FLAG_ssrc).value(); |
| } |
| |
| const std::string& DefaultSsrcRtx() { |
| static const std::string ssrc_rtx = std::to_string( |
| test::CallTest::kSendRtxSsrcs[0]); |
| return ssrc_rtx; |
| } |
| DEFINE_string(ssrc_rtx, DefaultSsrcRtx().c_str(), "Incoming RTX SSRC"); |
| static uint32_t SsrcRtx() { |
| return rtc::StringToNumber<uint32_t>(FLAG_ssrc_rtx).value(); |
| } |
| |
| // Flag for RED payload type. |
| DEFINE_int(red_payload_type, -1, "RED payload type"); |
| static int RedPayloadType() { |
| return static_cast<int>(FLAG_red_payload_type); |
| } |
| |
| // Flag for ULPFEC payload type. |
| DEFINE_int(fec_payload_type, -1, "ULPFEC payload type"); |
| static int FecPayloadType() { |
| return static_cast<int>(FLAG_fec_payload_type); |
| } |
| |
| // Flag for abs-send-time id. |
| DEFINE_int(abs_send_time_id, -1, "RTP extension ID for abs-send-time"); |
| static int AbsSendTimeId() { return static_cast<int>(FLAG_abs_send_time_id); } |
| |
| // Flag for transmission-offset id. |
| DEFINE_int(transmission_offset_id, |
| -1, |
| "RTP extension ID for transmission-offset"); |
| static int TransmissionOffsetId() { |
| return static_cast<int>(FLAG_transmission_offset_id); |
| } |
| |
| // Flag for rtpdump input file. |
| DEFINE_string(input_file, "", "input file"); |
| static std::string InputFile() { |
| return static_cast<std::string>(FLAG_input_file); |
| } |
| |
| // Flag for raw output files. |
| DEFINE_string(out_base, "", "Basename (excluding .jpg) for raw output"); |
| static std::string OutBase() { |
| return static_cast<std::string>(FLAG_out_base); |
| } |
| |
| DEFINE_string(decoder_bitstream_filename, "", "Decoder bitstream output file"); |
| static std::string DecoderBitstreamFilename() { |
| return static_cast<std::string>(FLAG_decoder_bitstream_filename); |
| } |
| |
| // Flag for video codec. |
| DEFINE_string(codec, "VP8", "Video codec"); |
| static std::string Codec() { return static_cast<std::string>(FLAG_codec); } |
| |
| DEFINE_bool(help, false, "Print this message."); |
| } // namespace flags |
| |
| 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() { |
| 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 EncodedFrameObserver { |
| public: |
| explicit DecoderBitstreamFileWriter(const char* filename) |
| : file_(fopen(filename, "wb")) { |
| RTC_DCHECK(file_); |
| } |
| ~DecoderBitstreamFileWriter() { fclose(file_); } |
| |
| virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) { |
| fwrite(encoded_frame.data_, 1, encoded_frame.length_, file_); |
| } |
| |
| private: |
| FILE* file_; |
| }; |
| |
| void RtpReplay() { |
| std::unique_ptr<test::VideoRenderer> playback_video( |
| test::VideoRenderer::Create("Playback Video", 640, 480)); |
| FileRenderPassthrough file_passthrough(flags::OutBase(), |
| playback_video.get()); |
| |
| webrtc::RtcEventLogNullImpl event_log; |
| std::unique_ptr<Call> call(Call::Create(Call::Config(&event_log))); |
| |
| test::NullTransport transport; |
| VideoReceiveStream::Config receive_config(&transport); |
| receive_config.rtp.remote_ssrc = flags::Ssrc(); |
| receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
| receive_config.rtp.rtx_ssrc = flags::SsrcRtx(); |
| receive_config.rtp.rtx_associated_payload_types[flags::PayloadTypeRtx()] = |
| flags::PayloadType(); |
| receive_config.rtp.ulpfec.ulpfec_payload_type = flags::FecPayloadType(); |
| receive_config.rtp.ulpfec.red_payload_type = flags::RedPayloadType(); |
| receive_config.rtp.nack.rtp_history_ms = 1000; |
| if (flags::TransmissionOffsetId() != -1) { |
| receive_config.rtp.extensions.push_back(RtpExtension( |
| RtpExtension::kTimestampOffsetUri, flags::TransmissionOffsetId())); |
| } |
| if (flags::AbsSendTimeId() != -1) { |
| receive_config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTimeUri, flags::AbsSendTimeId())); |
| } |
| receive_config.renderer = &file_passthrough; |
| |
| VideoSendStream::Config::EncoderSettings encoder_settings; |
| encoder_settings.payload_name = flags::Codec(); |
| encoder_settings.payload_type = flags::PayloadType(); |
| VideoReceiveStream::Decoder decoder; |
| std::unique_ptr<DecoderBitstreamFileWriter> bitstream_writer; |
| if (!flags::DecoderBitstreamFilename().empty()) { |
| bitstream_writer.reset(new DecoderBitstreamFileWriter( |
| flags::DecoderBitstreamFilename().c_str())); |
| receive_config.pre_decode_callback = bitstream_writer.get(); |
| } |
| decoder = test::CreateMatchingDecoder(encoder_settings); |
| if (!flags::DecoderBitstreamFilename().empty()) { |
| // Replace with a null decoder if we're writing the bitstream to a file |
| // instead. |
| delete decoder.decoder; |
| decoder.decoder = new test::FakeNullDecoder(); |
| } |
| receive_config.decoders.push_back(decoder); |
| |
| VideoReceiveStream* receive_stream = |
| call->CreateVideoReceiveStream(std::move(receive_config)); |
| |
| std::unique_ptr<test::RtpFileReader> rtp_reader(test::RtpFileReader::Create( |
| test::RtpFileReader::kRtpDump, flags::InputFile())); |
| if (!rtp_reader) { |
| rtp_reader.reset(test::RtpFileReader::Create(test::RtpFileReader::kPcap, |
| flags::InputFile())); |
| 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, flags::InputFile())); |
| if (!rtp_reader) { |
| fprintf(stderr, |
| "Unable to open input file with any supported format\n"); |
| return; |
| } |
| } |
| } |
| receive_stream->Start(); |
| |
| uint32_t last_time_ms = 0; |
| int num_packets = 0; |
| std::map<uint32_t, int> unknown_packets; |
| while (true) { |
| test::RtpPacket packet; |
| if (!rtp_reader->NextPacket(&packet)) |
| break; |
| ++num_packets; |
| switch (call->Receiver()->DeliverPacket( |
| webrtc::MediaType::VIDEO, packet.data, packet.length, PacketTime())) { |
| case PacketReceiver::DELIVERY_OK: |
| break; |
| case PacketReceiver::DELIVERY_UNKNOWN_SSRC: { |
| RTPHeader header; |
| std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); |
| parser->Parse(packet.data, packet.length, &header); |
| 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"); |
| RTPHeader header; |
| std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); |
| parser->Parse(packet.data, packet.length, &header); |
| 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; |
| } |
| } |
| if (last_time_ms != 0 && last_time_ms != packet.time_ms) { |
| SleepMs(packet.time_ms - last_time_ms); |
| } |
| last_time_ms = packet.time_ms; |
| } |
| 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); |
| } |
| |
| call->DestroyVideoReceiveStream(receive_stream); |
| |
| delete decoder.decoder; |
| } |
| } // namespace webrtc |
| |
| int main(int argc, char* argv[]) { |
| ::testing::InitGoogleTest(&argc, argv); |
| if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) { |
| return 1; |
| } |
| if (webrtc::flags::FLAG_help) { |
| rtc::FlagList::Print(nullptr, false); |
| return 0; |
| } |
| |
| RTC_CHECK(ValidatePayloadType(webrtc::flags::FLAG_payload_type)); |
| RTC_CHECK(ValidatePayloadType(webrtc::flags::FLAG_payload_type_rtx)); |
| RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc)); |
| RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc_rtx)); |
| RTC_CHECK(ValidateOptionalPayloadType(webrtc::flags::FLAG_red_payload_type)); |
| RTC_CHECK(ValidateOptionalPayloadType(webrtc::flags::FLAG_fec_payload_type)); |
| RTC_CHECK(ValidateRtpHeaderExtensionId(webrtc::flags::FLAG_abs_send_time_id)); |
| RTC_CHECK(ValidateRtpHeaderExtensionId( |
| webrtc::flags::FLAG_transmission_offset_id)); |
| RTC_CHECK(ValidateInputFilenameNotEmpty(webrtc::flags::FLAG_input_file)); |
| |
| webrtc::test::RunTest(webrtc::RtpReplay); |
| return 0; |
| } |