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Remove gflags dependency for event_log_visualizer and activity_metric

Browse Source 
This is the first step towards getting rid of gflags as a dependency.
It has been causing us extra work for a long time since it's not present
in Chromium as one example.

BUG=webrtc:7644
NOTRY=True

Review-Url: https://codereview.webrtc.org/2874403003
Cr-Commit-Position: refs/heads/master@{#18157}
This commit is contained in:
kjellander
2017-05-16 00:01:23 -07:00
committed by Commit bot
parent 09f0561675
commit 4fa5be451f
3 changed files with 72 additions and 59 deletions
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4
webrtc/tools/BUILD.gn
View File

@ -244,8 +244,8 @@ if (rtc_include_tests) {
defines = [ "ENABLE_RTC_EVENT_LOG" ]
deps = [
":event_log_visualizer_utils",
"../base:rtc_base_approved",
"../test:field_trial",
"//third_party/gflags",
]
}
}
@ -262,13 +262,13 @@ if (rtc_include_tests) {
}
deps = [
"../base:rtc_base_approved",
"../modules:module_api",
"../modules/audio_processing",
"../system_wrappers:metrics_default",
"../test:test_support",
"//build/win:default_exe_manifest",
"//testing/gtest",
"//third_party/gflags",
]
}

53
webrtc/tools/agc/activity_metric.cc
View File

@ -16,7 +16,7 @@
#include <algorithm>
#include <memory>
#include "gflags/gflags.h"
#include "webrtc/base/flags.h"
#include "webrtc/modules/audio_processing/agc/agc.h"
#include "webrtc/modules/audio_processing/agc/loudness_histogram.h"
#include "webrtc/modules/audio_processing/agc/utility.h"
@ -28,7 +28,7 @@
#include "webrtc/test/gtest.h"
static const int kAgcAnalWindowSamples = 100;
static const double kDefaultActivityThreshold = 0.3;
static const float kDefaultActivityThreshold = 0.3f;
DEFINE_bool(standalone_vad, true, "enable stand-alone VAD");
DEFINE_string(true_vad, "", "name of a file containing true VAD in 'int'"
@ -44,8 +44,9 @@ DEFINE_string(result, "", "name of a file to write the results. The results"
" will be appended to the end of the file. This is optional.");
DEFINE_string(audio_content, "", "name of a file where audio content is written"
" to, in double format.");
DEFINE_double(activity_threshold, kDefaultActivityThreshold,
DEFINE_float(activity_threshold, kDefaultActivityThreshold,
"Activity threshold");
DEFINE_bool(help, false, "prints this message");
namespace webrtc {
@ -105,13 +106,13 @@ class AgcStat {
AudioFeatures features;
audio_processing_->ExtractFeatures(
frame.data_, frame.samples_per_channel_, &features);
if (FLAGS_standalone_vad) {
if (FLAG_standalone_vad) {
standalone_vad_->AddAudio(frame.data_,
frame.samples_per_channel_);
}
if (features.num_frames > 0) {
double p[kMaxNumFrames] = {0.5, 0.5, 0.5, 0.5};
if (FLAGS_standalone_vad) {
if (FLAG_standalone_vad) {
standalone_vad_->GetActivity(p, kMaxNumFrames);
}
// TODO(turajs) combining and limiting are used in the source files as
@ -175,20 +176,20 @@ void void_main(int argc, char* argv[]) {
}
FILE* true_vad_fid = NULL;
ASSERT_GT(FLAGS_true_vad.size(), 0u) << "Specify the file containing true "
ASSERT_GT(strlen(FLAG_true_vad), 0u) << "Specify the file containing true "
"VADs using --true_vad flag.";
true_vad_fid = fopen(FLAGS_true_vad.c_str(), "rb");
true_vad_fid = fopen(FLAG_true_vad, "rb");
ASSERT_TRUE(true_vad_fid != NULL) << "Cannot open the active list " <<
FLAGS_true_vad;
FLAG_true_vad;
FILE* results_fid = NULL;
if (FLAGS_result.size() > 0) {
if (strlen(FLAG_result) > 0) {
// True if this is the first time writing to this function and we add a
// header to the beginning of the file.
bool write_header;
// Open in the read mode. If it fails, the file doesn't exist and has to
// write a header for it. Otherwise no need to write a header.
results_fid = fopen(FLAGS_result.c_str(), "r");
results_fid = fopen(FLAG_result, "r");
if (results_fid == NULL) {
write_header = true;
} else {
@ -196,9 +197,9 @@ void void_main(int argc, char* argv[]) {
write_header = false;
}
// Open in append mode.
results_fid = fopen(FLAGS_result.c_str(), "a");
results_fid = fopen(FLAG_result, "a");
ASSERT_TRUE(results_fid != NULL) << "Cannot open the file, " <<
FLAGS_result << ", to write the results.";
FLAG_result << ", to write the results.";
// Write the header if required.
if (write_header) {
fprintf(results_fid, "%% Total Active, Misdetection, "
@ -208,19 +209,19 @@ void void_main(int argc, char* argv[]) {
}
FILE* video_vad_fid = NULL;
if (FLAGS_video_vad.size() > 0) {
video_vad_fid = fopen(FLAGS_video_vad.c_str(), "rb");
if (strlen(FLAG_video_vad) > 0) {
video_vad_fid = fopen(FLAG_video_vad, "rb");
ASSERT_TRUE(video_vad_fid != NULL) << "Cannot open the file, " <<
FLAGS_video_vad << " to read video-based VAD decisions.\n";
FLAG_video_vad << " to read video-based VAD decisions.\n";
}
// AgsStat will be the owner of this file and will close it at its
// destructor.
FILE* audio_content_fid = NULL;
if (FLAGS_audio_content.size() > 0) {
audio_content_fid = fopen(FLAGS_audio_content.c_str(), "wb");
if (strlen(FLAG_audio_content) > 0) {
audio_content_fid = fopen(FLAG_audio_content, "wb");
ASSERT_TRUE(audio_content_fid != NULL) << "Cannot open file, " <<
FLAGS_audio_content << " to write audio-content.\n";
FLAG_audio_content << " to write audio-content.\n";
agc_stat.set_audio_content_file(audio_content_fid);
}
@ -231,7 +232,7 @@ void void_main(int argc, char* argv[]) {
const size_t kSamplesToRead = frame.num_channels_ *
frame.samples_per_channel_;
agc_stat.SetActivityThreshold(FLAGS_activity_threshold);
agc_stat.SetActivityThreshold(FLAG_activity_threshold);
int ret_val = 0;
int num_frames = 0;
@ -369,17 +370,25 @@ void void_main(int argc, char* argv[]) {
} // namespace webrtc
int main(int argc, char* argv[]) {
char kUsage[] =
if (argc == 1) {
// Print usage information.
std::cout <<
"\nCompute the number of misdetected and false-positive frames. Not\n"
" that for each frame of audio (10 ms) there should be one true\n"
" activity. If any video-based activity is given, there should also be\n"
" one probability per frame.\n"
"Run with --help for more details on available flags.\n"
"\nUsage:\n\n"
"activity_metric input_pcm [options]\n"
"where 'input_pcm' is the input audio sampled at 16 kHz in 16 bits "
"format.\n\n";
google::SetUsageMessage(kUsage);
google::ParseCommandLineFlags(&argc, &argv, true);
return 0;
}
rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
webrtc::void_main(argc, argv);
return 0;
}

74
webrtc/tools/event_log_visualizer/main.cc
View File

@ -10,7 +10,7 @@
#include <iostream>
#include "gflags/gflags.h"
#include "webrtc/base/flags.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/tools/event_log_visualizer/analyzer.h"
@ -84,6 +84,7 @@ DEFINE_string(
"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
" will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
"trials are separated by \"/\"");
DEFINE_bool(help, false, "prints this message");
int main(int argc, char* argv[]) {
std::string program_name = argv[0];
@ -92,16 +93,19 @@ int main(int argc, char* argv[]) {
"Example usage:\n" +
program_name + " <logfile> | python\n" + "Run " + program_name +
" --help for a list of command line options\n";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (argc != 2) {
// Print usage information.
std::cout << google::ProgramUsage();
rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
webrtc::test::InitFieldTrialsFromString(FLAGS_force_fieldtrials);
if (argc != 2) {
// Print usage information.
std::cout << usage;
return 0;
}
webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
std::string filename = argv[1];
@ -118,15 +122,15 @@ int main(int argc, char* argv[]) {
std::unique_ptr<webrtc::plotting::PlotCollection> collection(
new webrtc::plotting::PythonPlotCollection());
if (FLAGS_plot_all || FLAGS_plot_packets) {
if (FLAGS_incoming) {
if (FLAG_plot_all || FLAG_plot_packets) {
if (FLAG_incoming) {
analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
collection->AppendNewPlot());
analyzer.CreateAccumulatedPacketsGraph(
webrtc::PacketDirection::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAGS_outgoing) {
if (FLAG_outgoing) {
analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
collection->AppendNewPlot());
analyzer.CreateAccumulatedPacketsGraph(
@ -135,95 +139,95 @@ int main(int argc, char* argv[]) {
}
}
if (FLAGS_plot_all || FLAGS_plot_audio_playout) {
if (FLAG_plot_all || FLAG_plot_audio_playout) {
analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_plot_audio_level) {
if (FLAG_plot_all || FLAG_plot_audio_level) {
analyzer.CreateAudioLevelGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_plot_sequence_number) {
if (FLAGS_incoming) {
if (FLAG_plot_all || FLAG_plot_sequence_number) {
if (FLAG_incoming) {
analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
}
}
if (FLAGS_plot_all || FLAGS_plot_delay_change) {
if (FLAGS_incoming) {
if (FLAG_plot_all || FLAG_plot_delay_change) {
if (FLAG_incoming) {
analyzer.CreateDelayChangeGraph(collection->AppendNewPlot());
}
}
if (FLAGS_plot_all || FLAGS_plot_accumulated_delay_change) {
if (FLAGS_incoming) {
if (FLAG_plot_all || FLAG_plot_accumulated_delay_change) {
if (FLAG_incoming) {
analyzer.CreateAccumulatedDelayChangeGraph(collection->AppendNewPlot());
}
}
if (FLAGS_plot_all || FLAGS_plot_fraction_loss) {
if (FLAG_plot_all || FLAG_plot_fraction_loss) {
analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_plot_total_bitrate) {
if (FLAGS_incoming) {
if (FLAG_plot_all || FLAG_plot_total_bitrate) {
if (FLAG_incoming) {
analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAGS_outgoing) {
if (FLAG_outgoing) {
analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
collection->AppendNewPlot());
}
}
if (FLAGS_plot_all || FLAGS_plot_stream_bitrate) {
if (FLAGS_incoming) {
if (FLAG_plot_all || FLAG_plot_stream_bitrate) {
if (FLAG_incoming) {
analyzer.CreateStreamBitrateGraph(
webrtc::PacketDirection::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAGS_outgoing) {
if (FLAG_outgoing) {
analyzer.CreateStreamBitrateGraph(
webrtc::PacketDirection::kOutgoingPacket,
collection->AppendNewPlot());
}
}
if (FLAGS_plot_all || FLAGS_plot_bwe) {
if (FLAG_plot_all || FLAG_plot_bwe) {
analyzer.CreateBweSimulationGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_plot_network_delay_feedback) {
if (FLAG_plot_all || FLAG_plot_network_delay_feedback) {
analyzer.CreateNetworkDelayFeedbackGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_plot_timestamps) {
if (FLAG_plot_all || FLAG_plot_timestamps) {
analyzer.CreateTimestampGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_audio_encoder_bitrate_bps) {
if (FLAG_plot_all || FLAG_audio_encoder_bitrate_bps) {
analyzer.CreateAudioEncoderTargetBitrateGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_audio_encoder_frame_length_ms) {
if (FLAG_plot_all || FLAG_audio_encoder_frame_length_ms) {
analyzer.CreateAudioEncoderFrameLengthGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_audio_encoder_uplink_packet_loss_fraction) {
if (FLAG_plot_all || FLAG_audio_encoder_uplink_packet_loss_fraction) {
analyzer.CreateAudioEncoderUplinkPacketLossFractionGraph(
collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_audio_encoder_fec) {
if (FLAG_plot_all || FLAG_audio_encoder_fec) {
analyzer.CreateAudioEncoderEnableFecGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_audio_encoder_dtx) {
if (FLAG_plot_all || FLAG_audio_encoder_dtx) {
analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot());
}
if (FLAGS_plot_all || FLAGS_audio_encoder_num_channels) {
if (FLAG_plot_all || FLAG_audio_encoder_num_channels) {
analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
}