62 lines
2.1 KiB
C
62 lines
2.1 KiB
C
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/*
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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/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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enum class LimiterType { kNoLimiter, kApmAgcLimiter, kApmAgc2Limiter };
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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(const std::vector<AudioFrame*>& 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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using MixingBuffer = std::array<std::array<float, kMaximumChannelSize>,
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kMaximumNumberOfChannels>;
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private:
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void LogMixingStats(const std::vector<AudioFrame*>& mix_list,
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int sample_rate,
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size_t number_of_streams) const;
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std::unique_ptr<ApmDataDumper> data_dumper_;
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std::unique_ptr<MixingBuffer> mixing_buffer_;
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Limiter limiter_;
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const bool use_limiter_;
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mutable int uma_logging_counter_ = 0;
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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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