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We want to know how the AudioMixer is used and how FixedGainController behaves. The WebRTC.Audio.Agc2.FixedDigitalGainCurveRegion.* metrics measures how often the input level hits different regions of the Fixed Gain Controller gain curve (when the limiter is enabled). They also measure how long the metrics stay in different regions. They are related to WebRTC.Audio.ApmCaptureOutputLevelPeakRms, but the new metrics measure the level before any processing done in APM. The AudioMixer mixes incoming audio streams. Their number should be mostly constant, and often some of them could be muted. The metrics WebRTC.Audio.AudioMixer.NumIncomingStreams, WebRTC.Audio.AudioMixer.NumIncomingActiveStreams log the number of incoming stream and how many are not muted. We currently don't have any stats related to that. The metric WebRTC.Audio.AudioMixer.MixingRate logs the rate selected for mixing. The rate can sometimes be inferred from WebRTC.Audio.Encoder.CodecType. But that metric measures encoding and not decoding, and codecs don't always map to rates. See also accompanying Chromium CL https://chromium-review.googlesource.com/c/chromium/src/+/939473 Bug: webrtc:8925 Change-Id: Ib1405877fc1b39e5d2f0ceccba04434813f20b0d Reviewed-on: https://webrtc-review.googlesource.com/57740 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22443}
60 lines
2.0 KiB
C++
60 lines
2.0 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 "modules/audio_processing/agc2/fixed_gain_controller.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/include/module_common_types.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(LimiterType limiter_type);
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explicit FrameCombiner(bool use_limiter);
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~FrameCombiner();
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void SetLimiterType(LimiterType limiter_type);
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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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LimiterType limiter_type_;
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std::unique_ptr<AudioProcessing> apm_agc_limiter_;
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std::unique_ptr<ApmDataDumper> data_dumper_;
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FixedGainController apm_agc2_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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