Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ )
Reason for revert: This CL caused a google3 breakage due to dependencies in Google3. I will fix that, and reland. Original issue's description: > Moved ring-buffer related files from common_audio to audio_processing > > BUG=webrtc:5724 > NOPRESUBMIT=true > > Committed: https://crrev.com/711ccc8d96490f58cc3d7fd9207c19d4d881d4dc > Cr-Commit-Position: refs/heads/master@{#12227} TBR=ivoc@webrtc.org,henrik.lundin@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5724 Review URL: https://codereview.webrtc.org/1856323002 Cr-Commit-Position: refs/heads/master@{#12232}
This commit is contained in:
peah
@ -21,7 +21,11 @@ source_set("common_audio") {
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sources = [
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"audio_converter.cc",
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"audio_converter.h",
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"audio_ring_buffer.cc",
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"audio_ring_buffer.h",
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"audio_util.cc",
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"blocker.cc",
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"blocker.h",
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"channel_buffer.cc",
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"channel_buffer.h",
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"fft4g.c",
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@ -31,6 +35,8 @@ source_set("common_audio") {
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"fir_filter_neon.h",
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"fir_filter_sse.h",
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"include/audio_util.h",
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"lapped_transform.cc",
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"lapped_transform.h",
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"real_fourier.cc",
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"real_fourier.h",
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"real_fourier_ooura.cc",
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@ -43,6 +49,8 @@ source_set("common_audio") {
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"resampler/resampler.cc",
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"resampler/sinc_resampler.cc",
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"resampler/sinc_resampler.h",
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"ring_buffer.c",
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"ring_buffer.h",
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"signal_processing/auto_corr_to_refl_coef.c",
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"signal_processing/auto_correlation.c",
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"signal_processing/complex_fft_tables.h",
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@ -8,10 +8,10 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
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#include "webrtc/common_audio/audio_ring_buffer.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
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#include "webrtc/common_audio/ring_buffer.h"
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// This is a simple multi-channel wrapper over the ring_buffer.h C interface.
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@ -7,8 +7,8 @@
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_AUDIO_RING_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_AUDIO_RING_BUFFER_H_
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#ifndef WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
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#define WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
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#include <stddef.h>
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#include <vector>
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@ -52,4 +52,4 @@ class AudioRingBuffer final {
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_AUDIO_RING_BUFFER_H_
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#endif // WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
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@ -9,9 +9,8 @@
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*/
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#include <memory>
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#include <tuple>
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#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
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#include "webrtc/common_audio/audio_ring_buffer.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/common_audio/channel_buffer.h"
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@ -8,7 +8,7 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_processing/utility/blocker.h"
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#include "webrtc/common_audio/blocker.h"
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#include <string.h>
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@ -8,13 +8,13 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
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#ifndef WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
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#define WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
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#include <memory>
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#include "webrtc/common_audio/audio_ring_buffer.h"
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#include "webrtc/common_audio/channel_buffer.h"
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#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
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namespace webrtc {
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@ -121,4 +121,4 @@ class Blocker {
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
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#endif // WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
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@ -10,7 +10,7 @@
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#include <memory>
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#include "webrtc/modules/audio_processing/utility/blocker.h"
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#include "webrtc/common_audio/blocker.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/arraysize.h"
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@ -31,7 +31,11 @@
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'sources': [
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'audio_converter.cc',
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'audio_converter.h',
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'audio_ring_buffer.cc',
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'audio_ring_buffer.h',
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'audio_util.cc',
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'blocker.cc',
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'blocker.h',
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'channel_buffer.cc',
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'channel_buffer.h',
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'fft4g.c',
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@ -41,6 +45,8 @@
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'fir_filter_neon.h',
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'fir_filter_sse.h',
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'include/audio_util.h',
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'lapped_transform.cc',
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'lapped_transform.h',
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'real_fourier.cc',
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'real_fourier.h',
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'real_fourier_ooura.cc',
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@ -53,6 +59,8 @@
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'resampler/resampler.cc',
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'resampler/sinc_resampler.cc',
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'resampler/sinc_resampler.h',
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'ring_buffer.c',
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'ring_buffer.h',
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'signal_processing/include/real_fft.h',
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'signal_processing/include/signal_processing_library.h',
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'signal_processing/include/spl_inl.h',
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@ -232,14 +240,18 @@
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],
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'sources': [
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'audio_converter_unittest.cc',
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'audio_ring_buffer_unittest.cc',
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'audio_util_unittest.cc',
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'blocker_unittest.cc',
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'fir_filter_unittest.cc',
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'lapped_transform_unittest.cc',
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'real_fourier_unittest.cc',
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'resampler/resampler_unittest.cc',
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'resampler/push_resampler_unittest.cc',
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'resampler/push_sinc_resampler_unittest.cc',
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'resampler/sinusoidal_linear_chirp_source.cc',
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'resampler/sinusoidal_linear_chirp_source.h',
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'ring_buffer_unittest.cc',
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'signal_processing/real_fft_unittest.cc',
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'signal_processing/signal_processing_unittest.cc',
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'sparse_fir_filter_unittest.cc',
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@ -8,7 +8,7 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
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#include "webrtc/common_audio/lapped_transform.h"
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#include <algorithm>
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#include <cstdlib>
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@ -8,14 +8,14 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
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#ifndef WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
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#define WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
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#include <complex>
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#include <memory>
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#include "webrtc/common_audio/blocker.h"
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#include "webrtc/common_audio/real_fourier.h"
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#include "webrtc/modules/audio_processing/utility/blocker.h"
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#include "webrtc/system_wrappers/include/aligned_array.h"
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namespace webrtc {
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@ -121,4 +121,5 @@ class LappedTransform {
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
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#endif // WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
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@ -8,7 +8,7 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
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#include "webrtc/common_audio/lapped_transform.h"
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#include <algorithm>
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#include <cmath>
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@ -11,7 +11,7 @@
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// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
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// otherwise specified, functions return 0 on success and -1 on error.
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#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
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#include "webrtc/common_audio/ring_buffer.h"
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#include <stddef.h> // size_t
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#include <stdlib.h>
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@ -11,8 +11,8 @@
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// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
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// otherwise specified, functions return 0 on success and -1 on error.
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
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#ifndef WEBRTC_COMMON_AUDIO_RING_BUFFER_H_
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#define WEBRTC_COMMON_AUDIO_RING_BUFFER_H_
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#ifdef __cplusplus
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extern "C" {
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@ -63,4 +63,4 @@ size_t WebRtc_available_write(const RingBuffer* handle);
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}
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#endif
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
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#endif // WEBRTC_COMMON_AUDIO_RING_BUFFER_H_
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@ -8,7 +8,7 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
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#include "webrtc/common_audio/ring_buffer.h"
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#include <stdlib.h>
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#include <time.h>
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@ -108,21 +108,13 @@ source_set("audio_processing") {
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"transient/wpd_tree.h",
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"typing_detection.cc",
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"typing_detection.h",
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"utility/audio_ring_buffer.cc",
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"utility/audio_ring_buffer.h",
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"utility/block_mean_calculator.cc",
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"utility/block_mean_calculator.h",
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"utility/blocker.cc",
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"utility/blocker.h",
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"utility/delay_estimator.c",
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"utility/delay_estimator.h",
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"utility/delay_estimator_internal.h",
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"utility/delay_estimator_wrapper.c",
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"utility/delay_estimator_wrapper.h",
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"utility/lapped_transform.cc",
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"utility/lapped_transform.h",
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"utility/ring_buffer.c",
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"utility/ring_buffer.h",
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"vad/common.h",
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"vad/gmm.cc",
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"vad/gmm.h",
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@ -24,6 +24,9 @@
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#include <stdlib.h>
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#include <string.h>
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extern "C" {
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#include "webrtc/common_audio/ring_buffer.h"
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}
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/modules/audio_processing/aec/aec_common.h"
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#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
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@ -33,7 +36,6 @@ extern "C" {
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#include "webrtc/modules/audio_processing/logging/aec_logging.h"
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extern "C" {
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#include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
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#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
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}
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#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
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#include "webrtc/typedefs.h"
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@ -11,14 +11,13 @@
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
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extern "C" {
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#include "webrtc/common_audio/ring_buffer.h"
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}
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#include "webrtc/common_audio/wav_file.h"
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#include "webrtc/modules/audio_processing/aec/aec_common.h"
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#include "webrtc/modules/audio_processing/aec/aec_core.h"
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#include "webrtc/modules/audio_processing/utility/block_mean_calculator.h"
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extern "C" {
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#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
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}
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#include "webrtc/typedefs.h"
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namespace webrtc {
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@ -21,14 +21,12 @@
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#include <string.h>
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extern "C" {
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#include "webrtc/common_audio/ring_buffer.h"
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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}
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#include "webrtc/modules/audio_processing/aec/aec_core.h"
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#include "webrtc/modules/audio_processing/aec/aec_resampler.h"
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#include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
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extern "C" {
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#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
|
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}
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#include "webrtc/typedefs.h"
|
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|
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namespace webrtc {
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|
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@ -11,10 +11,10 @@
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
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#include "webrtc/modules/audio_processing/aec/aec_core.h"
|
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extern "C" {
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#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
|
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#include "webrtc/common_audio/ring_buffer.h"
|
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}
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#include "webrtc/modules/audio_processing/aec/aec_core.h"
|
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|
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namespace webrtc {
|
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|
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|
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@ -14,10 +14,10 @@
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#include <stddef.h>
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#include <stdlib.h>
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#include "webrtc/common_audio/ring_buffer.h"
|
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#include "webrtc/common_audio/signal_processing/include/real_fft.h"
|
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#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
|
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#include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
|
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#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
|
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#include "webrtc/system_wrappers/include/compile_assert_c.h"
|
||||
#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
|
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#include "webrtc/typedefs.h"
|
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|
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@ -13,9 +13,9 @@
|
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
|
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|
||||
#include "webrtc/common_audio/ring_buffer.h"
|
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
||||
#include "webrtc/modules/audio_processing/aecm/aecm_defines.h"
|
||||
#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
|
||||
#include "webrtc/typedefs.h"
|
||||
|
||||
#ifdef _MSC_VER // visual c++
|
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|
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@ -14,10 +14,10 @@
|
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#include <stddef.h>
|
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#include <stdlib.h>
|
||||
|
||||
#include "webrtc/common_audio/ring_buffer.h"
|
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#include "webrtc/common_audio/signal_processing/include/real_fft.h"
|
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#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
|
||||
#include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
|
||||
#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
|
||||
#include "webrtc/system_wrappers/include/compile_assert_c.h"
|
||||
#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
|
||||
#include "webrtc/typedefs.h"
|
||||
@ -768,3 +768,4 @@ static void ComfortNoise(AecmCore* aecm,
|
||||
out[i].imag = WebRtcSpl_AddSatW16(out[i].imag, uImag[i]);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
@ -15,9 +15,9 @@
|
||||
#endif
|
||||
#include <stdlib.h>
|
||||
|
||||
#include "webrtc/common_audio/ring_buffer.h"
|
||||
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
||||
#include "webrtc/modules/audio_processing/aecm/aecm_core.h"
|
||||
#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
|
||||
|
||||
#define BUF_SIZE_FRAMES 50 // buffer size (frames)
|
||||
// Maximum length of resampled signal. Must be an integer multiple of frames
|
||||
|
||||
@ -118,21 +118,13 @@
|
||||
'transient/wpd_tree.h',
|
||||
'typing_detection.cc',
|
||||
'typing_detection.h',
|
||||
'utility/audio_ring_buffer.cc',
|
||||
'utility/audio_ring_buffer.h',
|
||||
'utility/block_mean_calculator.cc',
|
||||
'utility/block_mean_calculator.h',
|
||||
'utility/blocker.cc',
|
||||
'utility/blocker.h',
|
||||
'utility/delay_estimator.c',
|
||||
'utility/delay_estimator.h',
|
||||
'utility/delay_estimator_internal.h',
|
||||
'utility/delay_estimator_wrapper.c',
|
||||
'utility/delay_estimator_wrapper.h',
|
||||
'utility/lapped_transform.cc',
|
||||
'utility/lapped_transform.h',
|
||||
'utility/ring_buffer.c',
|
||||
'utility/ring_buffer.h',
|
||||
'vad/common.h',
|
||||
'vad/gmm.cc',
|
||||
'vad/gmm.h',
|
||||
|
||||
@ -19,10 +19,11 @@
|
||||
#include <memory>
|
||||
#include <vector>
|
||||
|
||||
#include "webrtc/common_audio/lapped_transform.h"
|
||||
#include "webrtc/common_audio/channel_buffer.h"
|
||||
#include "webrtc/modules/audio_processing/beamformer/beamformer.h"
|
||||
#include "webrtc/modules/audio_processing/beamformer/complex_matrix.h"
|
||||
#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
// Enhances sound sources coming directly in front of a uniform linear array
|
||||
|
||||
@ -16,10 +16,10 @@
|
||||
#include <vector>
|
||||
|
||||
#include "webrtc/base/swap_queue.h"
|
||||
#include "webrtc/common_audio/lapped_transform.h"
|
||||
#include "webrtc/common_audio/channel_buffer.h"
|
||||
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
|
||||
#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
|
||||
#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
|
||||
#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
@ -258,12 +258,8 @@
|
||||
'audio_processing/transient/transient_suppressor_unittest.cc',
|
||||
'audio_processing/transient/wpd_node_unittest.cc',
|
||||
'audio_processing/transient/wpd_tree_unittest.cc',
|
||||
'audio_processing/utility/audio_ring_buffer_unittest.cc',
|
||||
'audio_processing/utility/block_mean_calculator_unittest.cc',
|
||||
'audio_processing/utility/blocker_unittest.cc',
|
||||
'audio_processing/utility/delay_estimator_unittest.cc',
|
||||
'audio_processing/utility/lapped_transform_unittest.cc',
|
||||
'audio_processing/utility/ring_buffer_unittest.cc',
|
||||
'audio_processing/vad/gmm_unittest.cc',
|
||||
'audio_processing/vad/pitch_based_vad_unittest.cc',
|
||||
'audio_processing/vad/pitch_internal_unittest.cc',
|
||||
|
||||
Reference in New Issue
Block a user