OLD | NEW |
1 /* | 1 /* |
2 * Copyright (c) 2008 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2008 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 <memory> | 11 #include <memory> |
12 | 12 |
13 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h" | 13 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h" |
14 #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h" | 14 #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h" |
15 #include "webrtc/base/arraysize.h" | |
16 #include "webrtc/base/byteorder.h" | |
17 #include "webrtc/base/safe_conversions.h" | |
18 #include "webrtc/base/scoped_ref_ptr.h" | |
19 #include "webrtc/call/call.h" | 15 #include "webrtc/call/call.h" |
20 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 16 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
21 #include "webrtc/media/base/fakemediaengine.h" | 17 #include "webrtc/media/base/fakemediaengine.h" |
22 #include "webrtc/media/base/fakenetworkinterface.h" | 18 #include "webrtc/media/base/fakenetworkinterface.h" |
23 #include "webrtc/media/base/fakertp.h" | 19 #include "webrtc/media/base/fakertp.h" |
24 #include "webrtc/media/base/mediaconstants.h" | 20 #include "webrtc/media/base/mediaconstants.h" |
25 #include "webrtc/media/engine/fakewebrtccall.h" | 21 #include "webrtc/media/engine/fakewebrtccall.h" |
26 #include "webrtc/media/engine/fakewebrtcvoiceengine.h" | 22 #include "webrtc/media/engine/fakewebrtcvoiceengine.h" |
27 #include "webrtc/media/engine/webrtcvoiceengine.h" | 23 #include "webrtc/media/engine/webrtcvoiceengine.h" |
28 #include "webrtc/modules/audio_device/include/mock_audio_device.h" | 24 #include "webrtc/modules/audio_device/include/mock_audio_device.h" |
29 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" | 25 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" |
30 #include "webrtc/pc/channel.h" | 26 #include "webrtc/pc/channel.h" |
| 27 #include "webrtc/rtc_base/arraysize.h" |
| 28 #include "webrtc/rtc_base/byteorder.h" |
| 29 #include "webrtc/rtc_base/safe_conversions.h" |
| 30 #include "webrtc/rtc_base/scoped_ref_ptr.h" |
31 #include "webrtc/test/field_trial.h" | 31 #include "webrtc/test/field_trial.h" |
32 #include "webrtc/test/gtest.h" | 32 #include "webrtc/test/gtest.h" |
33 #include "webrtc/test/mock_audio_decoder_factory.h" | 33 #include "webrtc/test/mock_audio_decoder_factory.h" |
34 #include "webrtc/test/mock_audio_encoder_factory.h" | 34 #include "webrtc/test/mock_audio_encoder_factory.h" |
35 #include "webrtc/voice_engine/transmit_mixer.h" | 35 #include "webrtc/voice_engine/transmit_mixer.h" |
36 | 36 |
37 using testing::ContainerEq; | 37 using testing::ContainerEq; |
38 using testing::Return; | 38 using testing::Return; |
39 using testing::StrictMock; | 39 using testing::StrictMock; |
40 | 40 |
(...skipping 3441 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
3482 // Without this cast, the comparison turned unsigned and, thus, failed for -1. | 3482 // Without this cast, the comparison turned unsigned and, thus, failed for -1. |
3483 const int num_specs = static_cast<int>(specs.size()); | 3483 const int num_specs = static_cast<int>(specs.size()); |
3484 EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs); | 3484 EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs); |
3485 EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs); | 3485 EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs); |
3486 EXPECT_EQ(find_codec({"cn", 32000, 1}), -1); | 3486 EXPECT_EQ(find_codec({"cn", 32000, 1}), -1); |
3487 EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs); | 3487 EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs); |
3488 EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs); | 3488 EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs); |
3489 EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs); | 3489 EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs); |
3490 EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs); | 3490 EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs); |
3491 } | 3491 } |
OLD | NEW |