webrtc_m130/webrtc/api/call/audio_sink.h
zhihuang 0acebe2238 Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ )
Reason for revert:
Re-land the original CL because the reverting it didn't fix the problem.

Original issue's description:
> Revert of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2873803002/ )
>
> Reason for revert:
> Reverted because it possibly breaks the internal project.
>
> Original issue's description:
> > Make AudioSinkInterface::Data hold a const pointer to the audio buffer.
> >
> > This is in preparation for https://codereview.webrtc.org/2750783004/, where
> > requiring a non-const pointer for AudioSinkInterface would force an unmuting
> > and zeroing of every frame.
> >
> > BUG=webrtc:7343
> >
> > Review-Url: https://codereview.webrtc.org/2873803002
> > Cr-Commit-Position: refs/heads/master@{#18107}
> > Committed: 38605965bd
>
> TBR=solenberg@webrtc.org,henrik.lundin@webrtc.org,yujo@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7343
>
> Review-Url: https://codereview.webrtc.org/2877013002
> Cr-Commit-Position: refs/heads/master@{#18112}
> Committed: c904634823

TBR=solenberg@webrtc.org,henrik.lundin@webrtc.org,yujo@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7343

Review-Url: https://codereview.webrtc.org/2880663003
Cr-Commit-Position: refs/heads/master@{#18113}
2017-05-12 05:07:37 +00:00

54 lines
1.6 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 WEBRTC_API_CALL_AUDIO_SINK_H_
#define WEBRTC_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 // WEBRTC_API_CALL_AUDIO_SINK_H_