Reformat the WebRTC code base
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
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@ -20,10 +20,14 @@
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// Define command line flags.
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DEFINE_int(red, 117, "RTP payload type for RED");
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DEFINE_int(audio_level, -1, "Extension ID for audio level (RFC 6464); "
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"-1 not to print audio level");
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DEFINE_int(abs_send_time, -1, "Extension ID for absolute sender time; "
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"-1 not to print absolute send time");
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DEFINE_int(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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DEFINE_int(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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DEFINE_bool(help, false, "Print this message");
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int main(int argc, char* argv[]) {
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@ -37,8 +41,8 @@ int main(int argc, char* argv[]) {
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program_name + " 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 (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
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FLAG_help || (argc != 2 && argc != 3)) {
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if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
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(argc != 2 && argc != 3)) {
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printf("%s", usage.c_str());
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if (FLAG_help) {
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rtc::FlagList::Print(nullptr, false);
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@ -47,10 +51,11 @@ int main(int argc, char* argv[]) {
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return 1;
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}
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RTC_CHECK(FLAG_red >= 0 && FLAG_red <= 127); // Payload type
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RTC_CHECK(FLAG_audio_level == -1 || // Default
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(FLAG_audio_level > 0 && FLAG_audio_level <= 255)); // Extension ID
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RTC_CHECK(FLAG_abs_send_time == -1 || // Default
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RTC_CHECK(FLAG_red >= 0 && FLAG_red <= 127); // Payload type
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RTC_CHECK(FLAG_audio_level == -1 || // Default
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(FLAG_audio_level > 0 && FLAG_audio_level <= 255)); // Extension ID
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RTC_CHECK(
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FLAG_abs_send_time == -1 || // Default
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(FLAG_abs_send_time > 0 && FLAG_abs_send_time <= 255)); // Extension ID
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printf("Input file: %s\n", argv[1]);
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@ -104,19 +109,14 @@ int main(int argc, char* argv[]) {
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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,
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"%5u %10u %10u %5i %5i %2i %#08X",
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packet->header().sequenceNumber,
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packet->header().timestamp,
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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,
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packet->header().markerBit,
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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,
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" %5u (%1i)",
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packet->header().extension.audioLevel,
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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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@ -156,11 +156,8 @@ int main(int argc, char* argv[]) {
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while (!red_headers.empty()) {
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webrtc::RTPHeader* red = red_headers.front();
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assert(red);
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fprintf(out_file,
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"* %5u %10u %10u %5i\n",
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red->sequenceNumber,
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red->timestamp,
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static_cast<unsigned int>(packet->time_ms()),
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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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