kjellander c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
2017-06-30 21:02:00 +00:00

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2.1 KiB
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/*
* Copyright (c) 2014 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/typedefs.h"
namespace webrtc {
namespace test {
// Interface class for an object receiving raw output audio from test
// applications.
class AudioSink {
public:
AudioSink() {}
virtual ~AudioSink() {}
// Writes |num_samples| from |audio| to the AudioSink. Returns true if
// successful, otherwise false.
virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
// Writes |audio_frame| to the AudioSink. Returns true if successful,
// otherwise false.
bool WriteAudioFrame(const AudioFrame& audio_frame) {
return WriteArray(
audio_frame.data(),
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
}
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioSink);
};
// Forks the output audio to two AudioSink objects.
class AudioSinkFork : public AudioSink {
public:
AudioSinkFork(AudioSink* left, AudioSink* right)
: left_sink_(left), right_sink_(right) {}
bool WriteArray(const int16_t* audio, size_t num_samples) override;
private:
AudioSink* left_sink_;
AudioSink* right_sink_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioSinkFork);
};
// An AudioSink implementation that does nothing.
class VoidAudioSink : public AudioSink {
public:
VoidAudioSink() = default;
bool WriteArray(const int16_t* audio, size_t num_samples) override;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(VoidAudioSink);
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_