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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
commit | 40a355a42d4a9444dc753c04c6608dade2f06a23 (patch) | |
tree | 871fc667d2de662f171103ce5ec067014ef85e61 /third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h | |
parent | Adding upstream version 124.0.1. (diff) | |
download | firefox-upstream/125.0.1.tar.xz firefox-upstream/125.0.1.zip |
Adding upstream version 125.0.1.upstream/125.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h | 32 |
1 files changed, 32 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h new file mode 100644 index 0000000000..1b2080cd94 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h @@ -0,0 +1,32 @@ +/* + * Copyright 2023 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_ +#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_ + +#include "modules/audio_coding/neteq/packet_arrival_history.h" +#include "test/gmock.h" + +namespace webrtc { + +class MockPacketArrivalHistory : public PacketArrivalHistory { + public: + MockPacketArrivalHistory() : PacketArrivalHistory(0) {} + + MOCK_METHOD(int, + GetDelayMs, + (uint32_t rtp_timestamp, int64_t time_ms), + (const override)); + MOCK_METHOD(int, GetMaxDelayMs, (), (const override)); +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_ |