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adf35a359e
Extends the mocks for rtpreceiver rtpsender and videotrack. This change allows the external HangoutsKit client to remove its own mocks of rtc types. Bug: none Change-Id: I8ba1752fe7633f9e0bba264a1279f74cc1368a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282900 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jack Smith <jackdsmith@google.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38782}
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 : 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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