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Consistent 30% improvement in audio mixer running time.

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(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}
This commit is contained in:
aleloi 2017-02-01 03:43:31 -08:00 committed by Commit bot
parent 35fc2aa82f
commit 4637b6afca
1 changed files with 3 additions and 0 deletions
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3
webrtc/modules/audio_mixer/audio_frame_manipulator.cc
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@ -29,6 +29,9 @@ void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
RTC_DCHECK(audio_frame); RTC_DCHECK(audio_frame);
RTC_DCHECK_GE(start_gain, 0.0f); RTC_DCHECK_GE(start_gain, 0.0f);
RTC_DCHECK_GE(target_gain, 0.0f); RTC_DCHECK_GE(target_gain, 0.0f);
if (start_gain == target_gain) {
return;
}
size_t samples = audio_frame->samples_per_channel_; size_t samples = audio_frame->samples_per_channel_;
RTC_DCHECK_LT(0, samples); RTC_DCHECK_LT(0, samples);