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8396e3498f
The FrameCombiner sub-module of the AudioMixer uses one of two limiters. One is an AudioProcessingModule with AGC1 enabled and configured as a limiter. The other is the limiter part of AGC2. This change removes the APM-AGC1 limiter. This requires small changes to FrameCombiner, AudioMixerImpl and tests. We also stop using the finch experiment flag. Bug: webrtc:8925 Change-Id: Id7b8349ec4720b6417b15eaf70ed1a850b6ddbed Reviewed-on: https://webrtc-review.googlesource.com/84620 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23727}
54 lines
1.8 KiB
C++
54 lines
1.8 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/audio/audio_frame.h"
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#include "modules/audio_processing/agc2/fixed_gain_controller.h"
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namespace webrtc {
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class ApmDataDumper;
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class FixedGainController;
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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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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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FixedGainController 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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