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| 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 #include <utility> | 14 #include <utility> |
| 15 | 15 |
| 16 #include "webrtc/api/test/mock_audio_mixer.h" | 16 #include "webrtc/api/test/mock_audio_mixer.h" |
| 17 #include "webrtc/base/ptr_util.h" | |
| 18 #include "webrtc/call/audio_state.h" | 17 #include "webrtc/call/audio_state.h" |
| 19 #include "webrtc/call/call.h" | 18 #include "webrtc/call/call.h" |
| 20 #include "webrtc/call/fake_rtp_transport_controller_send.h" | 19 #include "webrtc/call/fake_rtp_transport_controller_send.h" |
| 21 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 20 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| 22 #include "webrtc/modules/audio_device/include/mock_audio_device.h" | 21 #include "webrtc/modules/audio_device/include/mock_audio_device.h" |
| 23 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" | 22 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
| 24 #include "webrtc/modules/congestion_controller/include/mock/mock_send_side_conge
stion_controller.h" | 23 #include "webrtc/modules/congestion_controller/include/mock/mock_send_side_conge
stion_controller.h" |
| 25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 24 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| 25 #include "webrtc/rtc_base/ptr_util.h" |
| 26 #include "webrtc/test/gtest.h" | 26 #include "webrtc/test/gtest.h" |
| 27 #include "webrtc/test/mock_audio_decoder_factory.h" | 27 #include "webrtc/test/mock_audio_decoder_factory.h" |
| 28 #include "webrtc/test/mock_transport.h" | 28 #include "webrtc/test/mock_transport.h" |
| 29 #include "webrtc/test/mock_voice_engine.h" | 29 #include "webrtc/test/mock_voice_engine.h" |
| 30 | 30 |
| 31 namespace { | 31 namespace { |
| 32 | 32 |
| 33 struct CallHelper { | 33 struct CallHelper { |
| 34 explicit CallHelper( | 34 explicit CallHelper( |
| 35 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) | 35 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) |
| (...skipping 670 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 706 mask.min_bitrate_bps = rtc::Optional<int>(2000); | 706 mask.min_bitrate_bps = rtc::Optional<int>(2000); |
| 707 EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, -1, 1000)); | 707 EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, -1, 1000)); |
| 708 call->SetBitrateConfigMask(mask); | 708 call->SetBitrateConfigMask(mask); |
| 709 | 709 |
| 710 // Set min to 3000; the clamped value stays the same so nothing happens. | 710 // Set min to 3000; the clamped value stays the same so nothing happens. |
| 711 mask.min_bitrate_bps = rtc::Optional<int>(3000); | 711 mask.min_bitrate_bps = rtc::Optional<int>(3000); |
| 712 call->SetBitrateConfigMask(mask); | 712 call->SetBitrateConfigMask(mask); |
| 713 } | 713 } |
| 714 | 714 |
| 715 } // namespace webrtc | 715 } // namespace webrtc |
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