webrtc_m130/api/test/mock_audio_sink.h
Dor Hen 1921fa5ea1 Apply include-cleaner to api/test/[^/]*
e.g all files in the api/test folder not including subdirectories

Bug: webrtc:42226242
Change-Id: I18d74a18f8feec41eb252faa9acfffd1d6f45ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42773}
2024-08-13 15:28:34 +00:00

48 lines
1.3 KiB
C++

/*
* 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 <cstddef>
#include <cstdint>
#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<int64_t> absolute_capture_timestamp_ms),
(override));
};
} // namespace webrtc
#endif // API_TEST_MOCK_AUDIO_SINK_H_