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/voice_detection_impl.cc b/modules/audio_processing/voice_detection_impl.cc
index 80b633c..3b0eb7c 100644
--- a/modules/audio_processing/voice_detection_impl.cc
+++ b/modules/audio_processing/voice_detection_impl.cc
@@ -63,16 +63,17 @@
std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> mixed_low_pass_data;
rtc::ArrayView<const int16_t> mixed_low_pass(mixed_low_pass_data.data(),
audio->num_frames_per_band());
- if (audio->num_channels() == 1) {
- FloatS16ToS16(audio->split_bands_const(0)[kBand0To8kHz],
+ if (audio->num_proc_channels() == 1) {
+ FloatS16ToS16(audio->split_bands_const_f(0)[kBand0To8kHz],
audio->num_frames_per_band(), mixed_low_pass_data.data());
} else {
const int num_channels = static_cast<int>(audio->num_channels());
for (size_t i = 0; i < audio->num_frames_per_band(); ++i) {
int32_t value =
- FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[0][i]);
+ FloatS16ToS16(audio->split_channels_const_f(kBand0To8kHz)[0][i]);
for (int j = 1; j < num_channels; ++j) {
- value += FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[j][i]);
+ value +=
+ FloatS16ToS16(audio->split_channels_const_f(kBand0To8kHz)[j][i]);
}
mixed_low_pass_data[i] = value / num_channels;
}