webrtc/audio/remix_resample.h
Fredrik Solenberg bbf21a3fd6 Remove dependencies on modules:module_api from AudioProcessing.
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.

Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:05:27 +00:00

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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_