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123a0ed604
This reverts commit e87ec28b80.
Reason for revert: Breaks upstream.
Original change's description:
> Add checks for api/test mocks to make sure they're complete
>
> Also unifies the mock inheritance if they inherited from a ref counted
> interface:
> - it should only inherit from the interface
> - it should use make_ref_counted
>
> Bug: webrtc:14594
> Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38602}
Bug: webrtc:14594
Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38604}
44 lines
1.3 KiB
C++
44 lines
1.3 KiB
C++
/*
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* Copyright 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_TEST_MOCK_AUDIO_SINK_H_
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#define API_TEST_MOCK_AUDIO_SINK_H_
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#include "absl/types/optional.h"
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#include "api/media_stream_interface.h"
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#include "test/gmock.h"
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namespace webrtc {
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class MockAudioSink final : public webrtc::AudioTrackSinkInterface {
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public:
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MOCK_METHOD(void,
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OnData,
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(const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames),
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(override));
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MOCK_METHOD(void,
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OnData,
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(const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames,
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absl::optional<int64_t> absolute_capture_timestamp_ms),
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(override));
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};
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} // namespace webrtc
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#endif // API_TEST_MOCK_AUDIO_SINK_H_
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