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Remove bwe_rtp_play and add rtp_to_text to the build.

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This CL also switches rtp_to_text to ABSL_FLAG.

Bug: webrtc:10616
Change-Id: I6a2ce921e4c622a9fe08e7de724b8c7ed06f3597
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144630
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28502}
This commit is contained in:
Mirko Bonadei 2019-07-08 11:52:24 +02:00 committed by Commit Bot
parent ab0b9d4236
commit 79e4c92d70
4 changed files with 42 additions and 147 deletions
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1
BUILD.gn
View File

@ -52,6 +52,7 @@ if (!build_with_chromium) {
"modules:modules_unittests",
"modules/audio_coding:audio_coding_tests",
"modules/audio_processing:audio_processing_tests",
"modules/remote_bitrate_estimator:rtp_to_text",
"modules/rtp_rtcp:test_packet_masks_metrics",
"modules/video_capture:video_capture_internal_impl",
"pc:peerconnection_unittests",

16
modules/remote_bitrate_estimator/BUILD.gn
View File

@ -75,6 +75,22 @@ if (!build_with_chromium) {
"../../rtc_base:rtc_base_approved",
"../../test:rtp_test_utils",
"../rtp_rtcp",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
]
}
rtc_executable("rtp_to_text") {
testonly = true
sources = [
"tools/rtp_to_text.cc",
]
deps = [
":bwe_rtp",
"../../modules/rtp_rtcp",
"../../rtc_base:macromagic",
"../../rtc_base:stringutils",
"../../test:rtp_test_utils",
]
}
}

61
modules/remote_bitrate_estimator/tools/bwe_rtp.cc
View File

@ -16,40 +16,39 @@
#include <sstream>
#include <string>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "rtc_base/flags.h"
#include "test/rtp_file_reader.h"
namespace flags {
WEBRTC_DEFINE_string(
extension_type,
"abs",
"Extension type, either abs for absolute send time or tsoffset "
"for timestamp offset.");
ABSL_FLAG(std::string,
extension_type,
"abs",
"Extension type, either abs for absolute send time or tsoffset "
"for timestamp offset.");
std::string ExtensionType() {
return static_cast<std::string>(FLAG_extension_type);
return absl::GetFlag(FLAGS_extension_type);
}
WEBRTC_DEFINE_int(extension_id, 3, "Extension id.");
ABSL_FLAG(int, extension_id, 3, "Extension id.");
int ExtensionId() {
return static_cast<int>(FLAG_extension_id);
return absl::GetFlag(FLAGS_extension_id);
}
WEBRTC_DEFINE_string(input_file, "", "Input file.");
ABSL_FLAG(std::string, input_file, "", "Input file.");
std::string InputFile() {
return static_cast<std::string>(FLAG_input_file);
return absl::GetFlag(FLAGS_input_file);
}
WEBRTC_DEFINE_string(
ssrc_filter,
"",
"Comma-separated list of SSRCs in hexadecimal which are to be "
"used as input to the BWE (only applicable to pcap files).");
ABSL_FLAG(std::string,
ssrc_filter,
"",
"Comma-separated list of SSRCs in hexadecimal which are to be "
"used as input to the BWE (only applicable to pcap files).");
std::set<uint32_t> SsrcFilter() {
std::string ssrc_filter_string = static_cast<std::string>(FLAG_ssrc_filter);
std::string ssrc_filter_string = absl::GetFlag(FLAGS_ssrc_filter);
if (ssrc_filter_string.empty())
return std::set<uint32_t>();
std::stringstream ss;
@ -66,9 +65,6 @@ std::set<uint32_t> SsrcFilter() {
return ssrcs;
}
WEBRTC_DEFINE_bool(help, false, "Print this message.");
} // namespace flags
bool ParseArgsAndSetupEstimator(int argc,
char** argv,
webrtc::Clock* clock,
@ -77,16 +73,10 @@ bool ParseArgsAndSetupEstimator(int argc,
webrtc::RtpHeaderParser** parser,
webrtc::RemoteBitrateEstimator** estimator,
std::string* estimator_used) {
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
return 1;
}
if (flags::FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
std::string filename = flags::InputFile();
absl::ParseCommandLine(argc, argv);
std::string filename = InputFile();
std::set<uint32_t> ssrc_filter = flags::SsrcFilter();
std::set<uint32_t> ssrc_filter = SsrcFilter();
fprintf(stderr, "Filter on SSRC: ");
for (auto& s : ssrc_filter) {
fprintf(stderr, "0x%08x, ", s);
@ -95,8 +85,7 @@ bool ParseArgsAndSetupEstimator(int argc,
if (filename.substr(filename.find_last_of('.')) == ".pcap") {
fprintf(stderr, "Opening as pcap\n");
*rtp_reader = webrtc::test::RtpFileReader::Create(
webrtc::test::RtpFileReader::kPcap, filename.c_str(),
flags::SsrcFilter());
webrtc::test::RtpFileReader::kPcap, filename.c_str(), SsrcFilter());
} else {
fprintf(stderr, "Opening as rtp\n");
*rtp_reader = webrtc::test::RtpFileReader::Create(
@ -109,10 +98,10 @@ bool ParseArgsAndSetupEstimator(int argc,
fprintf(stderr, "Input file: %s\n\n", filename.c_str());
webrtc::RTPExtensionType extension = webrtc::kRtpExtensionAbsoluteSendTime;
if (flags::ExtensionType() == "tsoffset") {
if (ExtensionType() == "tsoffset") {
extension = webrtc::kRtpExtensionTransmissionTimeOffset;
fprintf(stderr, "Extension: toffset\n");
} else if (flags::ExtensionType() == "abs") {
} else if (ExtensionType() == "abs") {
fprintf(stderr, "Extension: abs\n");
} else {
fprintf(stderr, "Unknown extension type\n");
@ -121,7 +110,7 @@ bool ParseArgsAndSetupEstimator(int argc,
// Setup the RTP header parser and the bitrate estimator.
*parser = webrtc::RtpHeaderParser::Create();
(*parser)->RegisterRtpHeaderExtension(extension, flags::ExtensionId());
(*parser)->RegisterRtpHeaderExtension(extension, ExtensionId());
if (estimator) {
switch (extension) {
case webrtc::kRtpExtensionAbsoluteSendTime: {

111
modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
View File

@ -1,111 +0,0 @@
/*
* 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 <memory>
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/tools/bwe_rtp.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "rtc_base/format_macros.h"
#include "test/rtp_file_reader.h"
class Observer : public webrtc::RemoteBitrateObserver {
public:
explicit Observer(webrtc::Clock* clock) : clock_(clock) {}
// Called when a receive channel group has a new bitrate estimate for the
// incoming streams.
virtual void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
uint32_t bitrate) {
printf("[%u] Num SSRCs: %d, bitrate: %u\n",
static_cast<uint32_t>(clock_->TimeInMilliseconds()),
static_cast<int>(ssrcs.size()), bitrate);
}
virtual ~Observer() {}
private:
webrtc::Clock* clock_;
};
int main(int argc, char* argv[]) {
webrtc::test::RtpFileReader* reader;
webrtc::RemoteBitrateEstimator* estimator;
webrtc::RtpHeaderParser* parser;
std::string estimator_used;
webrtc::SimulatedClock clock(0);
Observer observer(&clock);
if (!ParseArgsAndSetupEstimator(argc, argv, &clock, &observer, &reader,
&parser, &estimator, &estimator_used)) {
return -1;
}
std::unique_ptr<webrtc::test::RtpFileReader> rtp_reader(reader);
std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
std::unique_ptr<webrtc::RemoteBitrateEstimator> rbe(estimator);
// Process the file.
int packet_counter = 0;
int64_t next_rtp_time_ms = 0;
int64_t first_rtp_time_ms = -1;
int abs_send_time_count = 0;
int ts_offset_count = 0;
webrtc::test::RtpPacket packet;
if (!rtp_reader->NextPacket(&packet)) {
printf("No RTP packet found\n");
return 0;
}
first_rtp_time_ms = packet.time_ms;
packet.time_ms = packet.time_ms - first_rtp_time_ms;
while (true) {
if (next_rtp_time_ms <= clock.TimeInMilliseconds()) {
if (!parser->IsRtcp(packet.data, packet.length)) {
webrtc::RTPHeader header;
parser->Parse(packet.data, packet.length, &header);
if (header.extension.hasAbsoluteSendTime)
++abs_send_time_count;
if (header.extension.hasTransmissionTimeOffset)
++ts_offset_count;
size_t packet_length = packet.length;
// Some RTP dumps only include the header, in which case packet.length
// is equal to the header length. In those cases packet.original_length
// usually contains the original packet length.
if (packet.original_length > 0) {
packet_length = packet.original_length;
}
rbe->IncomingPacket(clock.TimeInMilliseconds(),
packet_length - header.headerLength, header);
++packet_counter;
}
if (!rtp_reader->NextPacket(&packet)) {
break;
}
packet.time_ms = packet.time_ms - first_rtp_time_ms;
next_rtp_time_ms = packet.time_ms;
}
int64_t time_until_process_ms = rbe->TimeUntilNextProcess();
if (time_until_process_ms <= 0) {
rbe->Process();
}
int64_t time_until_next_event =
std::min(rbe->TimeUntilNextProcess(),
next_rtp_time_ms - clock.TimeInMilliseconds());
clock.AdvanceTimeMilliseconds(std::max<int64_t>(time_until_next_event, 0));
}
printf("Parsed %d packets\nTime passed: %" PRId64 " ms\n", packet_counter,
clock.TimeInMilliseconds());
printf("Estimator used: %s\n", estimator_used.c_str());
printf("Packets with absolute send time: %d\n", abs_send_time_count);
printf("Packets with timestamp offset: %d\n", ts_offset_count);
printf("Packets with no extension: %d\n",
packet_counter - ts_offset_count - abs_send_time_count);
return 0;
}