9158df2aa4
Turns out it was needed. TBR=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15849004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6493 4adac7df-926f-26a2-2b94-8c16560cd09d
47 lines
1.4 KiB
C++
47 lines
1.4 KiB
C++
/*
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* Copyright (c) 2014 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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namespace test {
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// Interface class for an object receiving raw output audio from test
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// applications.
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class AudioSink {
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public:
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AudioSink() {}
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virtual ~AudioSink() {}
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// Writes |num_samples| from |audio| to the AudioSink. Returns true if
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// successful, otherwise false.
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virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
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// Writes |audio_frame| to the AudioSink. Returns true if successful,
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// otherwise false.
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bool WriteAudioFrame(const AudioFrame& audio_frame) {
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return WriteArray(
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audio_frame.data_,
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audio_frame.samples_per_channel_ * audio_frame.num_channels_);
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}
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private:
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DISALLOW_COPY_AND_ASSIGN(AudioSink);
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
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