webrtc/modules/audio_mixer/frame_combiner.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

45 lines
1.5 KiB
C++

/*
* Copyright (c) 2017 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 MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
#define MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
#include <memory>
#include <vector>
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/include/module_common_types.h"
namespace webrtc {
class FrameCombiner {
public:
explicit FrameCombiner(bool use_apm_limiter);
~FrameCombiner();
// Combine several frames into one. Assumes sample_rate,
// samples_per_channel of the input frames match the parameters. The
// parameters 'number_of_channels' and 'sample_rate' are needed
// because 'mix_list' can be empty. The parameter
// 'number_of_streams' is used for determining whether to pass the
// data through a limiter.
void Combine(const std::vector<AudioFrame*>& mix_list,
size_t number_of_channels,
int sample_rate,
size_t number_of_streams,
AudioFrame* audio_frame_for_mixing) const;
private:
const bool use_apm_limiter_;
std::unique_ptr<AudioProcessing> limiter_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_MIXER_FRAME_COMBINER_H_