Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(397)

Side by Side Diff: webrtc/modules/audio_processing/test/process_test.cc

Issue 1409943002: Add aecdump support to audioproc_f. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 5 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 618 matching lines...) Expand 10 before | Expand all | Expand 10 after
629 printf(" Primary rates: %d Hz (in), %d Hz (out)\n", 629 printf(" Primary rates: %d Hz (in), %d Hz (out)\n",
630 msg.sample_rate(), output_sample_rate); 630 msg.sample_rate(), output_sample_rate);
631 printf(" Primary channels: %d (in), %d (out)\n", 631 printf(" Primary channels: %d (in), %d (out)\n",
632 msg.num_input_channels(), 632 msg.num_input_channels(),
633 msg.num_output_channels()); 633 msg.num_output_channels());
634 printf(" Reverse rate: %d\n", reverse_sample_rate); 634 printf(" Reverse rate: %d\n", reverse_sample_rate);
635 printf(" Reverse channels: %d\n", msg.num_reverse_channels()); 635 printf(" Reverse channels: %d\n", msg.num_reverse_channels());
636 } 636 }
637 637
638 if (!raw_output) { 638 if (!raw_output) {
639 // The WAV file needs to be reset every time, because it cant change 639 // The WAV file needs to be reset every time, because it can't change
640 // it's sample rate or number of channels. 640 // its sample rate or number of channels.
641 output_wav_file.reset(new WavWriter(out_filename + ".wav", 641 output_wav_file.reset(new WavWriter(out_filename + ".wav",
642 output_sample_rate, 642 output_sample_rate,
643 msg.num_output_channels())); 643 msg.num_output_channels()));
644 } 644 }
645 645
646 } else if (event_msg.type() == Event::REVERSE_STREAM) { 646 } else if (event_msg.type() == Event::REVERSE_STREAM) {
647 ASSERT_TRUE(event_msg.has_reverse_stream()); 647 ASSERT_TRUE(event_msg.has_reverse_stream());
648 ReverseStream msg = event_msg.reverse_stream(); 648 ReverseStream msg = event_msg.reverse_stream();
649 reverse_count++; 649 reverse_count++;
650 650
(...skipping 488 matching lines...) Expand 10 before | Expand all | Expand 10 after
1139 } // namespace 1139 } // namespace
1140 } // namespace webrtc 1140 } // namespace webrtc
1141 1141
1142 int main(int argc, char* argv[]) { 1142 int main(int argc, char* argv[]) {
1143 webrtc::void_main(argc, argv); 1143 webrtc::void_main(argc, argv);
1144 1144
1145 // Optional, but removes memory leak noise from Valgrind. 1145 // Optional, but removes memory leak noise from Valgrind.
1146 google::protobuf::ShutdownProtobufLibrary(); 1146 google::protobuf::ShutdownProtobufLibrary();
1147 return 0; 1147 return 0;
1148 } 1148 }
OLDNEW
« no previous file with comments | « webrtc/modules/audio_processing/test/audioproc_float.cc ('k') | webrtc/modules/audio_processing/test/test_utils.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698