Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(7)

Side by Side Diff: webrtc/test/call_test.cc

Issue 2668523004: Move AudioDecoder and related stuff to the api/ directory (Closed)
Patch Set: more review fixes Created 3 years, 10 months 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) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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
11 #include "webrtc/test/call_test.h" 11 #include "webrtc/test/call_test.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 14
15 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
15 #include "webrtc/base/checks.h" 16 #include "webrtc/base/checks.h"
16 #include "webrtc/config.h" 17 #include "webrtc/config.h"
17 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
18 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" 18 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
19 #include "webrtc/test/testsupport/fileutils.h" 19 #include "webrtc/test/testsupport/fileutils.h"
20 #include "webrtc/voice_engine/include/voe_base.h" 20 #include "webrtc/voice_engine/include/voe_base.h"
21 21
22 namespace webrtc { 22 namespace webrtc {
23 namespace test { 23 namespace test {
24 24
25 namespace { 25 namespace {
26 const int kVideoRotationRtpExtensionId = 4; 26 const int kVideoRotationRtpExtensionId = 4;
27 } 27 }
(...skipping 467 matching lines...) Expand 10 before | Expand all | Expand 10 after
495 495
496 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { 496 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
497 } 497 }
498 498
499 bool EndToEndTest::ShouldCreateReceivers() const { 499 bool EndToEndTest::ShouldCreateReceivers() const {
500 return true; 500 return true;
501 } 501 }
502 502
503 } // namespace test 503 } // namespace test
504 } // namespace webrtc 504 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/pc/peerconnectioninterface_unittest.cc ('k') | webrtc/test/fuzzers/audio_decoder_fuzzer.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698