webrtc/api/call/audio_sink.h
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

53 lines
1.5 KiB
C++

/*
* Copyright (c) 2015 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_CALL_AUDIO_SINK_H_
#define API_CALL_AUDIO_SINK_H_
#if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
// Avoid conflict with format_macros.h.
#define __STDC_FORMAT_MACROS
#endif
#include <inttypes.h>
#include <stddef.h>
namespace webrtc {
// Represents a simple push audio sink.
class AudioSinkInterface {
public:
virtual ~AudioSinkInterface() {}
struct Data {
Data(const int16_t* data,
size_t samples_per_channel,
int sample_rate,
size_t channels,
uint32_t timestamp)
: data(data),
samples_per_channel(samples_per_channel),
sample_rate(sample_rate),
channels(channels),
timestamp(timestamp) {}
const int16_t* data; // The actual 16bit audio data.
size_t samples_per_channel; // Number of frames in the buffer.
int sample_rate; // Sample rate in Hz.
size_t channels; // Number of channels in the audio data.
uint32_t timestamp; // The RTP timestamp of the first sample.
};
virtual void OnData(const Data& audio) = 0;
};
} // namespace webrtc
#endif // API_CALL_AUDIO_SINK_H_