From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/api/test/mock_audio_sink.h | 44 ++++++++++++++++++++++++ 1 file changed, 44 insertions(+) create mode 100644 third_party/libwebrtc/api/test/mock_audio_sink.h (limited to 'third_party/libwebrtc/api/test/mock_audio_sink.h') diff --git a/third_party/libwebrtc/api/test/mock_audio_sink.h b/third_party/libwebrtc/api/test/mock_audio_sink.h new file mode 100644 index 0000000000..88f38a3c57 --- /dev/null +++ b/third_party/libwebrtc/api/test/mock_audio_sink.h @@ -0,0 +1,44 @@ +/* + * Copyright 2021 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 API_TEST_MOCK_AUDIO_SINK_H_ +#define API_TEST_MOCK_AUDIO_SINK_H_ + +#include "absl/types/optional.h" +#include "api/media_stream_interface.h" +#include "test/gmock.h" + +namespace webrtc { + +class MockAudioSink : public webrtc::AudioTrackSinkInterface { + public: + MOCK_METHOD(void, + OnData, + (const void* audio_data, + int bits_per_sample, + int sample_rate, + size_t number_of_channels, + size_t number_of_frames), + (override)); + + MOCK_METHOD(void, + OnData, + (const void* audio_data, + int bits_per_sample, + int sample_rate, + size_t number_of_channels, + size_t number_of_frames, + absl::optional absolute_capture_timestamp_ms), + (override)); +}; + +} // namespace webrtc + +#endif // API_TEST_MOCK_AUDIO_SINK_H_ -- cgit v1.2.3