Use MonoView for deinterleaved channels in AudioFrameView
Allow skipping the deinterleaving steps in PushResampler
before resampling when deinterleaved buffers already exist.
Bug: chromium:335805780
Change-Id: I2080ce2624636cb743beef78f6f08887db01120f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352202
Reviewed-by: Per Åhgren <peah@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42438}
diff --git a/api/audio/audio_view.h b/api/audio/audio_view.h
index c877ee4..e75769b 100644
--- a/api/audio/audio_view.h
+++ b/api/audio/audio_view.h
@@ -243,7 +243,7 @@
// "Incompatible view types");
RTC_DCHECK_EQ(NumChannels(destination), NumChannels(source));
RTC_DCHECK_EQ(SamplesPerChannel(destination), SamplesPerChannel(source));
- RTC_DCHECK_GE(destination.data().size(), source.data().size());
+ RTC_DCHECK_GE(destination.size(), source.size());
memcpy(&destination[0], &source[0],
source.size() * sizeof(typename S::value_type));
}
diff --git a/common_audio/resampler/include/push_resampler.h b/common_audio/resampler/include/push_resampler.h
index 0ed463b..4a23d46 100644
--- a/common_audio/resampler/include/push_resampler.h
+++ b/common_audio/resampler/include/push_resampler.h
@@ -37,8 +37,12 @@
// Returns the total number of samples provided in destination (e.g. 32 kHz,
// 2 channel audio gives 640 samples).
int Resample(InterleavedView<const T> src, InterleavedView<T> dst);
+ // For when a deinterleaved/mono channel already exists and we can skip the
+ // deinterleaved operation.
+ int Resample(MonoView<const T> src, MonoView<T> dst);
private:
+ // Buffers used for when a deinterleaving step is necessary.
std::unique_ptr<T[]> source_;
std::unique_ptr<T[]> destination_;
DeinterleavedView<T> source_view_;
diff --git a/common_audio/resampler/push_resampler.cc b/common_audio/resampler/push_resampler.cc
index 5d732dd..1cbf421 100644
--- a/common_audio/resampler/push_resampler.cc
+++ b/common_audio/resampler/push_resampler.cc
@@ -95,6 +95,20 @@
return static_cast<int>(dst.size());
}
+template <typename T>
+int PushResampler<T>::Resample(MonoView<const T> src, MonoView<T> dst) {
+ RTC_DCHECK_EQ(resamplers_.size(), 1);
+ RTC_DCHECK_EQ(SamplesPerChannel(src), SamplesPerChannel(source_view_));
+ RTC_DCHECK_EQ(SamplesPerChannel(dst), SamplesPerChannel(destination_view_));
+
+ if (SamplesPerChannel(src) == SamplesPerChannel(dst)) {
+ CopySamples(dst, src);
+ return static_cast<int>(src.size());
+ }
+
+ return resamplers_[0]->Resample(src, dst);
+}
+
// Explictly generate required instantiations.
template class PushResampler<int16_t>;
template class PushResampler<float>;
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 9764917..2f32a4a 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -269,7 +269,7 @@
rtc_source_set("audio_frame_view") {
sources = [ "include/audio_frame_view.h" ]
- deps = [ "../../api:array_view" ]
+ deps = [ "../../api/audio:audio_frame_api" ]
}
if (rtc_enable_protobuf) {
diff --git a/modules/audio_processing/agc2/vad_wrapper.cc b/modules/audio_processing/agc2/vad_wrapper.cc
index 331fd36..d4fa32a 100644
--- a/modules/audio_processing/agc2/vad_wrapper.cc
+++ b/modules/audio_processing/agc2/vad_wrapper.cc
@@ -102,18 +102,11 @@
vad_->Reset();
time_to_vad_reset_ = vad_reset_period_frames_;
}
+
// Resample the first channel of `frame`.
RTC_DCHECK_EQ(frame.samples_per_channel(), frame_size_);
-
- // TODO: b/335805780 - channel() should return a MonoView<> which there
- // should be a Resample() implementation for. There's no need to
- // "deinterleave" a mono buffer, which is what Resample() currently does,
- // so here we should be able to directly resample the channel buffer.
- auto channel = frame.channel(0);
- InterleavedView<const float> src(channel.data(), channel.size(), 1);
- InterleavedView<float> dst(resampled_buffer_.data(), resampled_buffer_.size(),
- 1);
- resampler_.Resample(src, dst);
+ MonoView<float> dst(resampled_buffer_.data(), resampled_buffer_.size());
+ resampler_.Resample(frame.channel(0), dst);
return vad_->Analyze(resampled_buffer_);
}
diff --git a/modules/audio_processing/include/audio_frame_view.h b/modules/audio_processing/include/audio_frame_view.h
index 164784a..1c717e9 100644
--- a/modules/audio_processing/include/audio_frame_view.h
+++ b/modules/audio_processing/include/audio_frame_view.h
@@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
-#include "api/array_view.h"
+#include "api/audio/audio_view.h"
namespace webrtc {
@@ -44,16 +44,16 @@
int samples_per_channel() const { return channel_size_; }
- rtc::ArrayView<T> channel(int idx) {
+ MonoView<T> channel(int idx) {
RTC_DCHECK_LE(0, idx);
RTC_DCHECK_LE(idx, num_channels_);
- return rtc::ArrayView<T>(audio_samples_[idx], channel_size_);
+ return MonoView<T>(audio_samples_[idx], channel_size_);
}
- rtc::ArrayView<const T> channel(int idx) const {
+ MonoView<const T> channel(int idx) const {
RTC_DCHECK_LE(0, idx);
RTC_DCHECK_LE(idx, num_channels_);
- return rtc::ArrayView<const T>(audio_samples_[idx], channel_size_);
+ return MonoView<const T>(audio_samples_[idx], channel_size_);
}
T* const* data() { return audio_samples_; }
