Index: webrtc/modules/audio_processing/test/conversational_speech/convspeech_gen.cc |
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/convspeech_gen.cc b/webrtc/modules/audio_processing/test/conversational_speech/convspeech_gen.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..9a2456edfae3b152583adbd41fa3170e983a8abe |
--- /dev/null |
+++ b/webrtc/modules/audio_processing/test/conversational_speech/convspeech_gen.cc |
@@ -0,0 +1,59 @@ |
+/* |
+ * Copyright (c) 2017 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 <iostream> |
+ |
+#include "gflags/gflags.h" |
+#include "webrtc/base/logging.h" |
+#include "webrtc/modules/audio_processing/test/conversational_speech/settings.h" |
+ |
+namespace webrtc { |
+namespace test { |
+namespace { |
+ |
+const char kUsageDescription[] = |
+ "Usage: convspeech_gen\n" |
+ " -i <path/to/source/audiotracks>\n" |
+ " -t <path/to/timing_file.txt>\n" |
+ " -o <output/path>\n" |
+ "\n\n" |
+ "Command-line tool to generate multiple-end audio tracks to simulate " |
+ "conversational speech with two or more participants."; |
+ |
+DEFINE_string(i, "", "Directory containing the speech turn wav files"); |
+DEFINE_string(t, "", "Path to the timing text file"); |
+DEFINE_string(o, "", "Output wav files destination path"); |
+ |
+} // namespace |
+ |
+int main(int argc, char* argv[]) { |
+ google::SetUsageMessage(kUsageDescription); |
+ google::ParseCommandLineFlags(&argc, &argv, true); |
+ |
+ convspeechgen::Settings settings(FLAGS_i, FLAGS_t, FLAGS_o); |
+ if (!settings.ValidateAndReport()) { |
hlundin-webrtc
2017/03/13 12:28:19
I've been increasingly radical about error handlin
AleBzk
2017/03/13 14:14:41
Done.
|
+ exit(1); |
+ } |
+ |
+ // TODO(alessiob): remove line below once debugged. |
+ rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE); |
+ LOG(LS_VERBOSE) << "i = " << settings.AudiotracksPath(); |
+ LOG(LS_VERBOSE) << "t = " << settings.TimingFilePath(); |
+ LOG(LS_VERBOSE) << "o = " << settings.OutputPath(); |
+ |
+ return 0; |
+} |
+ |
+} // namespace test |
+} // namespace webrtc |
+ |
+int main(int argc, char* argv[]) { |
+ return webrtc::test::main(argc, argv); |
+} |