Removing all external access to the integer sample data in AudioBuffer
This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.
The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.
Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 584111c..32668fa 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -169,29 +169,11 @@
}
}
-const int16_t* const* AudioBuffer::channels_const() const {
- return data_->ibuf_const()->channels();
-}
-
-int16_t* const* AudioBuffer::channels() {
- return data_->ibuf()->channels();
-}
-
-const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
- return split_data_.get() ? split_data_->ibuf_const()->bands(channel)
- : data_->ibuf_const()->bands(channel);
-}
-
-int16_t* const* AudioBuffer::split_bands(size_t channel) {
- return split_data_.get() ? split_data_->ibuf()->bands(channel)
- : data_->ibuf()->bands(channel);
-}
-
-const int16_t* const* AudioBuffer::split_channels_const(Band band) const {
+const float* const* AudioBuffer::split_channels_const_f(Band band) const {
if (split_data_.get()) {
- return split_data_->ibuf_const()->channels(band);
+ return split_data_->fbuf_const()->channels(band);
} else {
- return band == kBand0To8kHz ? data_->ibuf_const()->channels() : nullptr;
+ return band == kBand0To8kHz ? data_->fbuf_const()->channels() : nullptr;
}
}
@@ -308,4 +290,29 @@
splitting_filter_->Synthesis(split_data_.get(), data_.get());
}
+void AudioBuffer::CopySplitChannelDataTo(size_t channel,
+ int16_t* const* split_band_data) {
+ for (size_t k = 0; k < num_bands(); ++k) {
+ const float* band_data = split_bands_f(channel)[k];
+ RTC_DCHECK(split_band_data[k]);
+ RTC_DCHECK(band_data);
+ for (size_t i = 0; i < num_frames_per_band(); ++i) {
+ split_band_data[k][i] = FloatS16ToS16(band_data[i]);
+ }
+ }
+}
+
+void AudioBuffer::CopySplitChannelDataFrom(
+ size_t channel,
+ const int16_t* const* split_band_data) {
+ for (size_t k = 0; k < num_bands(); ++k) {
+ float* band_data = split_bands_f(channel)[k];
+ RTC_DCHECK(split_band_data[k]);
+ RTC_DCHECK(band_data);
+ for (size_t i = 0; i < num_frames_per_band(); ++i) {
+ band_data[i] = split_band_data[k][i];
+ }
+ }
+}
+
} // namespace webrtc