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81a3d95332
Bug: chromium:335805780 Change-Id: I26825941076e78573de268f6e2da7215ee1ea762 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355740 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42544}
54 lines
1.7 KiB
C++
54 lines
1.7 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
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#define MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
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#include <memory>
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#include <vector>
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#include "api/array_view.h"
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#include "api/audio/audio_frame.h"
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#include "modules/audio_processing/agc2/limiter.h"
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namespace webrtc {
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class ApmDataDumper;
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class FrameCombiner {
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public:
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explicit FrameCombiner(bool use_limiter);
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~FrameCombiner();
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// Combine several frames into one. Assumes sample_rate,
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// samples_per_channel of the input frames match the parameters. The
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// parameters 'number_of_channels' and 'sample_rate' are needed
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// because 'mix_list' can be empty. The parameter
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// 'number_of_streams' is used for determining whether to pass the
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// data through a limiter.
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void Combine(rtc::ArrayView<AudioFrame* const> mix_list,
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size_t number_of_channels,
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int sample_rate,
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size_t number_of_streams,
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AudioFrame* audio_frame_for_mixing);
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// Stereo, 48 kHz, 10 ms.
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static constexpr size_t kMaximumNumberOfChannels = 8;
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static constexpr size_t kMaximumChannelSize = 48 * 10;
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private:
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std::unique_ptr<ApmDataDumper> data_dumper_;
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Limiter limiter_;
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const bool use_limiter_;
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std::array<float, kMaximumChannelSize * kMaximumNumberOfChannels>
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mixing_buffer_ = {};
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
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