mirror of
https://github.com/DrKLO/Telegram.git
synced 2024-12-23 15:00:50 +01:00
44 lines
1.6 KiB
C++
44 lines
1.6 KiB
C++
/*
|
|
* Copyright (c) 2012 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 AUDIO_REMIX_RESAMPLE_H_
|
|
#define AUDIO_REMIX_RESAMPLE_H_
|
|
|
|
#include "api/audio/audio_frame.h"
|
|
#include "common_audio/resampler/include/push_resampler.h"
|
|
|
|
namespace webrtc {
|
|
namespace voe {
|
|
|
|
// Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|
|
|
// to have its sample rate and channels members set to the desired values.
|
|
// Updates the |samples_per_channel_| member accordingly.
|
|
//
|
|
// This version has an AudioFrame |src_frame| as input and sets the output
|
|
// |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the
|
|
// input ones.
|
|
void RemixAndResample(const AudioFrame& src_frame,
|
|
PushResampler<int16_t>* resampler,
|
|
AudioFrame* dst_frame);
|
|
|
|
// This version has a pointer to the samples |src_data| as input and receives
|
|
// |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
|
|
// parameters.
|
|
void RemixAndResample(const int16_t* src_data,
|
|
size_t samples_per_channel,
|
|
size_t num_channels,
|
|
int sample_rate_hz,
|
|
PushResampler<int16_t>* resampler,
|
|
AudioFrame* dst_frame);
|
|
|
|
} // namespace voe
|
|
} // namespace webrtc
|
|
|
|
#endif // AUDIO_REMIX_RESAMPLE_H_
|