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This CL completes the removal of assert() and relative headers from the codebase (excluded //examples/objc/AppRTCMobile/third_party/SocketRocket which is in a third_party sub-directory). Bug: webrtc:6779 Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34528}
166 lines
6 KiB
C++
166 lines
6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stdio.h>
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#include <memory>
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#include <vector>
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#include "absl/flags/flag.h"
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#include "absl/flags/parse.h"
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
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ABSL_FLAG(int, red, 117, "RTP payload type for RED");
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ABSL_FLAG(int,
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audio_level,
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-1,
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"Extension ID for audio level (RFC 6464); "
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"-1 not to print audio level");
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ABSL_FLAG(int,
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abs_send_time,
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-1,
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"Extension ID for absolute sender time; "
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"-1 not to print absolute send time");
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int main(int argc, char* argv[]) {
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std::vector<char*> args = absl::ParseCommandLine(argc, argv);
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std::string usage =
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"Tool for parsing an RTP dump file to text output.\n"
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"Example usage:\n"
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"./rtp_analyze input.rtp output.txt\n\n"
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"Output is sent to stdout if no output file is given. "
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"Note that this tool can read files with or without payloads.\n";
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if (args.size() != 2 && args.size() != 3) {
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printf("%s", usage.c_str());
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return 1;
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}
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RTC_CHECK(absl::GetFlag(FLAGS_red) >= 0 &&
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absl::GetFlag(FLAGS_red) <= 127); // Payload type
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RTC_CHECK(absl::GetFlag(FLAGS_audio_level) == -1 || // Default
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(absl::GetFlag(FLAGS_audio_level) > 0 &&
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absl::GetFlag(FLAGS_audio_level) <= 255)); // Extension ID
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RTC_CHECK(absl::GetFlag(FLAGS_abs_send_time) == -1 || // Default
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(absl::GetFlag(FLAGS_abs_send_time) > 0 &&
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absl::GetFlag(FLAGS_abs_send_time) <= 255)); // Extension ID
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printf("Input file: %s\n", args[1]);
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std::unique_ptr<webrtc::test::RtpFileSource> file_source(
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webrtc::test::RtpFileSource::Create(args[1]));
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RTC_DCHECK(file_source.get());
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// Set RTP extension IDs.
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bool print_audio_level = false;
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if (absl::GetFlag(FLAGS_audio_level) != -1) {
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print_audio_level = true;
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file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
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absl::GetFlag(FLAGS_audio_level));
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}
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bool print_abs_send_time = false;
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if (absl::GetFlag(FLAGS_abs_send_time) != -1) {
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print_abs_send_time = true;
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file_source->RegisterRtpHeaderExtension(
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webrtc::kRtpExtensionAbsoluteSendTime,
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absl::GetFlag(FLAGS_abs_send_time));
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}
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FILE* out_file;
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if (args.size() == 3) {
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out_file = fopen(args[2], "wt");
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if (!out_file) {
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printf("Cannot open output file %s\n", args[2]);
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return -1;
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}
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printf("Output file: %s\n\n", args[2]);
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} else {
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out_file = stdout;
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}
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// Print file header.
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fprintf(out_file, "SeqNo TimeStamp SendTime Size PT M SSRC");
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if (print_audio_level) {
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fprintf(out_file, " AuLvl (V)");
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}
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if (print_abs_send_time) {
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fprintf(out_file, " AbsSendTime");
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}
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fprintf(out_file, "\n");
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uint32_t max_abs_send_time = 0;
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int cycles = -1;
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std::unique_ptr<webrtc::test::Packet> packet;
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while (true) {
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packet = file_source->NextPacket();
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if (!packet.get()) {
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// End of file reached.
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break;
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}
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// Write packet data to file. Use virtual_packet_length_bytes so that the
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// correct packet sizes are printed also for RTP header-only dumps.
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fprintf(out_file, "%5u %10u %10u %5i %5i %2i %#08X",
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packet->header().sequenceNumber, packet->header().timestamp,
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static_cast<unsigned int>(packet->time_ms()),
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static_cast<int>(packet->virtual_packet_length_bytes()),
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packet->header().payloadType, packet->header().markerBit,
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packet->header().ssrc);
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if (print_audio_level && packet->header().extension.hasAudioLevel) {
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fprintf(out_file, " %5u (%1i)", packet->header().extension.audioLevel,
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packet->header().extension.voiceActivity);
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}
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if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
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if (cycles == -1) {
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// Initialize.
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max_abs_send_time = packet->header().extension.absoluteSendTime;
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cycles = 0;
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}
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// Abs sender time is 24 bit 6.18 fixed point. Shift by 8 to normalize to
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// 32 bits (unsigned). Calculate the difference between this packet's
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// send time and the maximum observed. Cast to signed 32-bit to get the
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// desired wrap-around behavior.
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if (static_cast<int32_t>(
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(packet->header().extension.absoluteSendTime << 8) -
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(max_abs_send_time << 8)) >= 0) {
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// The difference is non-negative, meaning that this packet is newer
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// than the previously observed maximum absolute send time.
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if (packet->header().extension.absoluteSendTime < max_abs_send_time) {
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// Wrap detected.
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cycles++;
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}
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max_abs_send_time = packet->header().extension.absoluteSendTime;
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}
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// Abs sender time is 24 bit 6.18 fixed point. Divide by 2^18 to convert
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// to floating point representation.
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double send_time_seconds =
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static_cast<double>(packet->header().extension.absoluteSendTime) /
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262144 +
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64.0 * cycles;
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fprintf(out_file, " %11f", send_time_seconds);
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}
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fprintf(out_file, "\n");
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if (packet->header().payloadType == absl::GetFlag(FLAGS_red)) {
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std::list<webrtc::RTPHeader*> red_headers;
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packet->ExtractRedHeaders(&red_headers);
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while (!red_headers.empty()) {
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webrtc::RTPHeader* red = red_headers.front();
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RTC_DCHECK(red);
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fprintf(out_file, "* %5u %10u %10u %5i\n", red->sequenceNumber,
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red->timestamp, static_cast<unsigned int>(packet->time_ms()),
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red->payloadType);
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red_headers.pop_front();
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delete red;
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}
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}
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}
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fclose(out_file);
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return 0;
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}
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