/* * Copyright (c) 2012 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 #include #include #include "absl/flags/flag.h" #include "absl/flags/parse.h" #include "modules/audio_coding/neteq/tools/packet.h" #include "modules/audio_coding/neteq/tools/rtp_file_source.h" ABSL_FLAG(int, red, 117, "RTP payload type for RED"); ABSL_FLAG(int, audio_level, -1, "Extension ID for audio level (RFC 6464); " "-1 not to print audio level"); ABSL_FLAG(int, abs_send_time, -1, "Extension ID for absolute sender time; " "-1 not to print absolute send time"); int main(int argc, char* argv[]) { std::vector args = absl::ParseCommandLine(argc, argv); std::string usage = "Tool for parsing an RTP dump file to text output.\n" "Example usage:\n" "./rtp_analyze input.rtp output.txt\n\n" "Output is sent to stdout if no output file is given. " "Note that this tool can read files with or without payloads.\n"; if (args.size() != 2 && args.size() != 3) { printf("%s", usage.c_str()); return 1; } RTC_CHECK(absl::GetFlag(FLAGS_red) >= 0 && absl::GetFlag(FLAGS_red) <= 127); // Payload type RTC_CHECK(absl::GetFlag(FLAGS_audio_level) == -1 || // Default (absl::GetFlag(FLAGS_audio_level) > 0 && absl::GetFlag(FLAGS_audio_level) <= 255)); // Extension ID RTC_CHECK(absl::GetFlag(FLAGS_abs_send_time) == -1 || // Default (absl::GetFlag(FLAGS_abs_send_time) > 0 && absl::GetFlag(FLAGS_abs_send_time) <= 255)); // Extension ID printf("Input file: %s\n", args[1]); std::unique_ptr file_source( webrtc::test::RtpFileSource::Create(args[1])); RTC_DCHECK(file_source.get()); // Set RTP extension IDs. bool print_audio_level = false; if (absl::GetFlag(FLAGS_audio_level) != -1) { print_audio_level = true; file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, absl::GetFlag(FLAGS_audio_level)); } bool print_abs_send_time = false; if (absl::GetFlag(FLAGS_abs_send_time) != -1) { print_abs_send_time = true; file_source->RegisterRtpHeaderExtension( webrtc::kRtpExtensionAbsoluteSendTime, absl::GetFlag(FLAGS_abs_send_time)); } FILE* out_file; if (args.size() == 3) { out_file = fopen(args[2], "wt"); if (!out_file) { printf("Cannot open output file %s\n", args[2]); return -1; } printf("Output file: %s\n\n", args[2]); } else { out_file = stdout; } // Print file header. fprintf(out_file, "SeqNo TimeStamp SendTime Size PT M SSRC"); if (print_audio_level) { fprintf(out_file, " AuLvl (V)"); } if (print_abs_send_time) { fprintf(out_file, " AbsSendTime"); } fprintf(out_file, "\n"); uint32_t max_abs_send_time = 0; int cycles = -1; std::unique_ptr packet; while (true) { packet = file_source->NextPacket(); if (!packet.get()) { // End of file reached. break; } // Write packet data to file. Use virtual_packet_length_bytes so that the // correct packet sizes are printed also for RTP header-only dumps. fprintf(out_file, "%5u %10u %10u %5i %5i %2i %#08X", packet->header().sequenceNumber, packet->header().timestamp, static_cast(packet->time_ms()), static_cast(packet->virtual_packet_length_bytes()), packet->header().payloadType, packet->header().markerBit, packet->header().ssrc); if (print_audio_level && packet->header().extension.hasAudioLevel) { fprintf(out_file, " %5u (%1i)", packet->header().extension.audioLevel, packet->header().extension.voiceActivity); } if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) { if (cycles == -1) { // Initialize. max_abs_send_time = packet->header().extension.absoluteSendTime; cycles = 0; } // Abs sender time is 24 bit 6.18 fixed point. Shift by 8 to normalize to // 32 bits (unsigned). Calculate the difference between this packet's // send time and the maximum observed. Cast to signed 32-bit to get the // desired wrap-around behavior. if (static_cast( (packet->header().extension.absoluteSendTime << 8) - (max_abs_send_time << 8)) >= 0) { // The difference is non-negative, meaning that this packet is newer // than the previously observed maximum absolute send time. if (packet->header().extension.absoluteSendTime < max_abs_send_time) { // Wrap detected. cycles++; } max_abs_send_time = packet->header().extension.absoluteSendTime; } // Abs sender time is 24 bit 6.18 fixed point. Divide by 2^18 to convert // to floating point representation. double send_time_seconds = static_cast(packet->header().extension.absoluteSendTime) / 262144 + 64.0 * cycles; fprintf(out_file, " %11f", send_time_seconds); } fprintf(out_file, "\n"); if (packet->header().payloadType == absl::GetFlag(FLAGS_red)) { std::list red_headers; packet->ExtractRedHeaders(&red_headers); while (!red_headers.empty()) { webrtc::RTPHeader* red = red_headers.front(); RTC_DCHECK(red); fprintf(out_file, "* %5u %10u %10u %5i\n", red->sequenceNumber, red->timestamp, static_cast(packet->time_ms()), red->payloadType); red_headers.pop_front(); delete red; } } } fclose(out_file); return 0; }