2c9306ed50
Before this change, the APM limiter used in FrameCombiner (a sub-component of AudioMixer) only gets to process the data when the number of non-muted streams is >1. If this number varies between <=1 and >1, the limiter's view of the data will have gaps during the periods with <= 1 active stream. This leads to discontinuities in the applied gain. These discontinuities cause clicks in the output audio. This change activates APM limiter processing based on the number of audio streams, independently of their mutedness status. BUG=chromium:695993 Review-Url: https://codereview.webrtc.org/2776113002 Cr-Commit-Position: refs/heads/master@{#17442}
46 lines
1.5 KiB
C++
46 lines
1.5 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 WEBRTC_MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
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#define WEBRTC_MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
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#include <memory>
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#include <vector>
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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class FrameCombiner {
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public:
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explicit FrameCombiner(bool use_apm_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) const;
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private:
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const bool use_apm_limiter_;
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std::unique_ptr<AudioProcessing> limiter_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
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