4637b6afca
(Or, in less flattering terms, fixing a performance issue introduced a few months ago by me). In GN release mode (is_debug = false), the version of the mixer code before this CL generated code that multiplied each sample (tens of thousands/second for each input stream) with a floating point number. This number is almost always exactly 1.0f. The only situation when it's not 1 is when an audio steam is added or removed. For one input stream early return leads to a 30% improvement of audio mixing time profiled on x86-64 under a release build (is_debug = false, enable_profiling, enable_full_stack_frames_for_profiling) with 16kHz and no APM limiter. There can be up to 3 streams. BUG=chromium:687502 Review-Url: https://codereview.webrtc.org/2659423002 Cr-Commit-Position: refs/heads/master@{#16396}
60 lines
2.1 KiB
C++
60 lines
2.1 KiB
C++
/*
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* Copyright (c) 2012 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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#include "webrtc/audio/utility/audio_frame_operations.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
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uint32_t energy = 0;
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for (size_t position = 0; position < audio_frame.samples_per_channel_;
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position++) {
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// TODO(aleloi): This can overflow. Convert to floats.
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energy += audio_frame.data_[position] * audio_frame.data_[position];
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}
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return energy;
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}
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void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
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RTC_DCHECK(audio_frame);
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RTC_DCHECK_GE(start_gain, 0.0f);
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RTC_DCHECK_GE(target_gain, 0.0f);
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if (start_gain == target_gain) {
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return;
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}
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size_t samples = audio_frame->samples_per_channel_;
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RTC_DCHECK_LT(0, samples);
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float increment = (target_gain - start_gain) / samples;
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float gain = start_gain;
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for (size_t i = 0; i < samples; ++i) {
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// If the audio is interleaved of several channels, we want to
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// apply the same gain change to the ith sample of every channel.
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for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
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audio_frame->data_[audio_frame->num_channels_ * i + ch] *= gain;
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}
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gain += increment;
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}
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}
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void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
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RTC_DCHECK_GE(target_number_of_channels, 1);
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RTC_DCHECK_LE(target_number_of_channels, 2);
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if (frame->num_channels_ == 1 && target_number_of_channels == 2) {
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AudioFrameOperations::MonoToStereo(frame);
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} else if (frame->num_channels_ == 2 && target_number_of_channels == 1) {
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AudioFrameOperations::StereoToMono(frame);
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}
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}
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} // namespace webrtc
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