Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(285)

Side by Side Diff: webrtc/call/call_unittest.cc

Issue 2798063004: Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ (Closed)
Patch Set: Created 3 years, 8 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
« no previous file with comments | « webrtc/call/BUILD.gn ('k') | webrtc/media/BUILD.gn » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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 <list> 11 #include <list>
12 #include <map> 12 #include <map>
13 #include <memory> 13 #include <memory>
14 14
15 #include "webrtc/call/audio_state.h" 15 #include "webrtc/call/audio_state.h"
16 #include "webrtc/call/call.h" 16 #include "webrtc/call/call.h"
17 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" 17 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
18 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
19 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" 18 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
20 #include "webrtc/test/gtest.h" 19 #include "webrtc/test/gtest.h"
20 #include "webrtc/test/mock_audio_decoder_factory.h"
21 #include "webrtc/test/mock_transport.h" 21 #include "webrtc/test/mock_transport.h"
22 #include "webrtc/test/mock_voice_engine.h" 22 #include "webrtc/test/mock_voice_engine.h"
23 23
24 namespace { 24 namespace {
25 25
26 struct CallHelper { 26 struct CallHelper {
27 explicit CallHelper( 27 explicit CallHelper(
28 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) 28 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
29 : voice_engine_(decoder_factory) { 29 : voice_engine_(decoder_factory) {
30 webrtc::AudioState::Config audio_state_config; 30 webrtc::AudioState::Config audio_state_config;
(...skipping 268 matching lines...) Expand 10 before | Expand all | Expand 10 after
299 stream = call->CreateFlexfecReceiveStream(config); 299 stream = call->CreateFlexfecReceiveStream(config);
300 EXPECT_NE(stream, nullptr); 300 EXPECT_NE(stream, nullptr);
301 streams.push_back(stream); 301 streams.push_back(stream);
302 302
303 for (auto s : streams) { 303 for (auto s : streams) {
304 call->DestroyFlexfecReceiveStream(s); 304 call->DestroyFlexfecReceiveStream(s);
305 } 305 }
306 } 306 }
307 307
308 } // namespace webrtc 308 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/call/BUILD.gn ('k') | webrtc/media/BUILD.gn » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698