Make path to wav file for jitter buffer simulation in event_log_visualizer configurable.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2996953002
Cr-Commit-Position: refs/heads/master@{#19364}
diff --git a/webrtc/rtc_tools/event_log_visualizer/main.cc b/webrtc/rtc_tools/event_log_visualizer/main.cc
index 12b55e6..0aafb2c 100644
--- a/webrtc/rtc_tools/event_log_visualizer/main.cc
+++ b/webrtc/rtc_tools/event_log_visualizer/main.cc
@@ -111,6 +111,9 @@
"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_string(wav_filename,
+ "",
+ "Path to wav file used for simulation of jitter buffer");
DEFINE_bool(help, false, "prints this message");
DEFINE_bool(show_detector_state,
@@ -255,11 +258,15 @@
analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_jitter_buffer) {
- analyzer.CreateAudioJitterBufferGraph(
- webrtc::test::ResourcePath(
- "audio_processing/conversational_speech/EN_script2_F_sp2_B1",
- "wav"),
- 48000, collection->AppendNewPlot());
+ std::string wav_path;
+ if (FLAG_wav_filename[0] != '\0') {
+ wav_path = FLAG_wav_filename;
+ } else {
+ wav_path = webrtc::test::ResourcePath(
+ "audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
+ }
+ analyzer.CreateAudioJitterBufferGraph(wav_path, 48000,
+ collection->AppendNewPlot());
}
collection->Draw();