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}
diff --git a/modules/audio_processing/echo_control_mobile_impl.cc b/modules/audio_processing/echo_control_mobile_impl.cc
index 982287b..8057e33 100644
--- a/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/modules/audio_processing/echo_control_mobile_impl.cc
@@ -142,7 +142,7 @@
for (size_t i = 0; i < num_output_channels; i++) {
for (size_t j = 0; j < audio->num_channels(); j++) {
std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> data_to_buffer;
- FloatS16ToS16(audio->split_bands_const_f(render_channel)[kBand0To8kHz],
+ FloatS16ToS16(audio->split_bands_const(render_channel)[kBand0To8kHz],
audio->num_frames_per_band(), data_to_buffer.data());
// Buffer the samples in the render queue.
@@ -185,8 +185,8 @@
std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> split_bands_data;
int16_t* split_bands = split_bands_data.data();
const int16_t* clean = split_bands_data.data();
- if (audio->split_bands_f(capture)[kBand0To8kHz]) {
- FloatS16ToS16(audio->split_bands_f(capture)[kBand0To8kHz],
+ if (audio->split_bands(capture)[kBand0To8kHz]) {
+ FloatS16ToS16(audio->split_bands(capture)[kBand0To8kHz],
audio->num_frames_per_band(), split_bands_data.data());
} else {
clean = nullptr;
@@ -205,7 +205,7 @@
if (split_bands) {
S16ToFloatS16(split_bands, audio->num_frames_per_band(),
- audio->split_bands_f(capture)[kBand0To8kHz]);
+ audio->split_bands(capture)[kBand0To8kHz]);
}
if (err != AudioProcessing::kNoError) {
@@ -227,7 +227,7 @@
RTC_DCHECK_LE(audio->num_channels(), low_pass_reference_.size());
reference_copied_ = true;
for (size_t capture = 0; capture < audio->num_channels(); ++capture) {
- FloatS16ToS16(audio->split_bands_const_f(capture)[kBand0To8kHz],
+ FloatS16ToS16(audio->split_bands_const(capture)[kBand0To8kHz],
audio->num_frames_per_band(),
low_pass_reference_[capture].data());
}