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Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will be used to store audio timestaps in future changes. This is a part of the A/V sync metric metric feature for mobile. The metric have already launched for web clients. Bug: webrtc:13609 Change-Id: I0031843476ff1b573b262308fca52d587fae30b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Minyue Li <minyue@google.com> Commit-Queue: Olov Brändström <brandstrom@google.com> Cr-Commit-Position: refs/heads/main@{#35851}
81 lines
2.5 KiB
C++
81 lines
2.5 KiB
C++
/*
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* Copyright (c) 2016 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 MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
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#define MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
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#include "modules/audio_device/include/audio_device_defines.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockAudioTransport : public AudioTransport {
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public:
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MockAudioTransport() {}
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~MockAudioTransport() {}
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MOCK_METHOD(int32_t,
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RecordedDataIsAvailable,
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(const void* audioSamples,
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size_t nSamples,
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size_t nBytesPerSample,
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size_t nChannels,
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uint32_t samplesPerSec,
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uint32_t totalDelayMS,
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int32_t clockDrift,
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uint32_t currentMicLevel,
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bool keyPressed,
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uint32_t& newMicLevel),
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(override));
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MOCK_METHOD(int32_t,
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RecordedDataIsAvailable,
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(const void* audioSamples,
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size_t nSamples,
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size_t nBytesPerSample,
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size_t nChannels,
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uint32_t samplesPerSec,
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uint32_t totalDelayMS,
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int32_t clockDrift,
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uint32_t currentMicLevel,
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bool keyPressed,
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uint32_t& newMicLevel,
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int64_t estimated_capture_time_ns),
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(override));
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MOCK_METHOD(int32_t,
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NeedMorePlayData,
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(size_t nSamples,
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size_t nBytesPerSample,
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size_t nChannels,
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uint32_t samplesPerSec,
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void* audioSamples,
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size_t& nSamplesOut,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms),
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(override));
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MOCK_METHOD(void,
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PullRenderData,
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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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void* audio_data,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms),
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(override));
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
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