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/echo_control_mobile_impl.cc b/modules/audio_processing/echo_control_mobile_impl.cc
index c8084ea..982287b 100644
--- a/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/modules/audio_processing/echo_control_mobile_impl.cc
@@ -56,6 +56,7 @@
return AudioProcessing::kUnspecifiedError;
}
}
+
} // namespace
struct EchoControlMobileImpl::StreamProperties {
@@ -131,7 +132,8 @@
size_t num_output_channels,
size_t num_channels,
std::vector<int16_t>* packed_buffer) {
- RTC_DCHECK_GE(160, audio->num_frames_per_band());
+ RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
+ audio->num_frames_per_band());
RTC_DCHECK_EQ(num_channels, audio->num_channels());
// The ordering convention must be followed to pass to the correct AECM.
@@ -139,12 +141,14 @@
int render_channel = 0;
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],
+ audio->num_frames_per_band(), data_to_buffer.data());
+
// Buffer the samples in the render queue.
packed_buffer->insert(
- packed_buffer->end(),
- audio->split_bands_const(render_channel)[kBand0To8kHz],
- (audio->split_bands_const(render_channel)[kBand0To8kHz] +
- audio->num_frames_per_band()));
+ packed_buffer->end(), data_to_buffer.data(),
+ data_to_buffer.data() + audio->num_frames_per_band());
render_channel = (render_channel + 1) % audio->num_channels();
}
}
@@ -174,7 +178,21 @@
RTC_DCHECK_LT(capture, low_pass_reference_.size());
const int16_t* noisy =
reference_copied_ ? low_pass_reference_[capture].data() : nullptr;
- const int16_t* clean = audio->split_bands_const(capture)[kBand0To8kHz];
+
+ RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
+ audio->num_frames_per_band());
+
+ 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],
+ audio->num_frames_per_band(), split_bands_data.data());
+ } else {
+ clean = nullptr;
+ split_bands = nullptr;
+ }
+
if (noisy == NULL) {
noisy = clean;
clean = NULL;
@@ -182,8 +200,13 @@
for (size_t render = 0; render < stream_properties_->num_reverse_channels;
++render) {
err = WebRtcAecm_Process(cancellers_[handle_index]->state(), noisy, clean,
- audio->split_bands(capture)[kBand0To8kHz],
- audio->num_frames_per_band(), stream_delay_ms);
+ split_bands, audio->num_frames_per_band(),
+ stream_delay_ms);
+
+ if (split_bands) {
+ S16ToFloatS16(split_bands, audio->num_frames_per_band(),
+ audio->split_bands_f(capture)[kBand0To8kHz]);
+ }
if (err != AudioProcessing::kNoError) {
return MapError(err);
@@ -192,9 +215,9 @@
++handle_index;
}
for (size_t band = 1u; band < audio->num_bands(); ++band) {
- memset(audio->split_bands(capture)[band], 0,
+ memset(audio->split_bands_f(capture)[band], 0,
audio->num_frames_per_band() *
- sizeof(audio->split_bands(capture)[band][0]));
+ sizeof(audio->split_bands_f(capture)[band][0]));
}
}
return AudioProcessing::kNoError;
@@ -204,9 +227,9 @@
RTC_DCHECK_LE(audio->num_channels(), low_pass_reference_.size());
reference_copied_ = true;
for (size_t capture = 0; capture < audio->num_channels(); ++capture) {
- memcpy(low_pass_reference_[capture].data(),
- audio->split_bands_const(capture)[kBand0To8kHz],
- audio->num_frames_per_band() * sizeof(int16_t));
+ FloatS16ToS16(audio->split_bands_const_f(capture)[kBand0To8kHz],
+ audio->num_frames_per_band(),
+ low_pass_reference_[capture].data());
}
}