blob: 7b7e240b3dd710aa592ade9acd7601c6e6d1137d [file] [log] [blame]
* 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.
#include <memory>
#include <vector>
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
class FrameCombiner {
explicit FrameCombiner(bool use_apm_limiter);
// 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;
const bool use_apm_limiter_;
std::unique_ptr<AudioProcessing> limiter_;
} // namespace webrtc