Revert "Simplification and refactoring of the AudioBuffer code"
This reverts commit 81c0cf287c8514cb1cd6f3baca484d668c6eb128.
Reason for revert: internal test failures
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28931}
diff --git a/modules/audio_processing/level_estimator_impl.cc b/modules/audio_processing/level_estimator_impl.cc
index e796095..8adbf19 100644
--- a/modules/audio_processing/level_estimator_impl.cc
+++ b/modules/audio_processing/level_estimator_impl.cc
@@ -32,15 +32,16 @@
rms_->Reset();
}
-void LevelEstimatorImpl::ProcessStream(const AudioBuffer& audio) {
+void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
+ RTC_DCHECK(audio);
rtc::CritScope cs(crit_);
if (!enabled_) {
return;
}
- for (size_t i = 0; i < audio.num_channels(); i++) {
- rms_->Analyze(rtc::ArrayView<const float>(audio.channels_const()[i],
- audio.num_frames()));
+ for (size_t i = 0; i < audio->num_channels(); i++) {
+ rms_->Analyze(rtc::ArrayView<const float>(audio->channels_const_f()[i],
+ audio->num_frames()));
}
}