Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 44 insertions, 0 deletions

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<int64_t> absolute_capture_timestamp_ms),
+              (override));
+};
+
+}  // namespace webrtc
+
+#endif  // API_TEST_MOCK_AUDIO_SINK_H_