Revert of Moved ring-buffer related files from common_audio to audio_processing" (patchset #2 id:20001 of https://codereview.webrtc.org/1858123003/ )
Reason for revert:
Because of down-stream dependencies, this CL needs to be reverted.
The dependencies will be resolved and then the CL will be relanded.
Original issue's description:
> Revert "Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ )"
>
> This reverts commit c54aad6ae07fe2a44a65be403386bd7d7d865e5b.
>
> BUG=webrtc:5724
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/8864fe5e08f8d8711612526dee9a812adfcd3be1
> Cr-Commit-Position: refs/heads/master@{#12247}
TBR=henrik.lundin@webrtc.org,ivoc@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/1855393004
Cr-Commit-Position: refs/heads/master@{#12248}
diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn
index 0103655..b01b318 100644
--- a/webrtc/common_audio/BUILD.gn
+++ b/webrtc/common_audio/BUILD.gn
@@ -21,7 +21,11 @@
sources = [
"audio_converter.cc",
"audio_converter.h",
+ "audio_ring_buffer.cc",
+ "audio_ring_buffer.h",
"audio_util.cc",
+ "blocker.cc",
+ "blocker.h",
"channel_buffer.cc",
"channel_buffer.h",
"fft4g.c",
@@ -31,6 +35,8 @@
"fir_filter_neon.h",
"fir_filter_sse.h",
"include/audio_util.h",
+ "lapped_transform.cc",
+ "lapped_transform.h",
"real_fourier.cc",
"real_fourier.h",
"real_fourier_ooura.cc",
@@ -43,6 +49,8 @@
"resampler/resampler.cc",
"resampler/sinc_resampler.cc",
"resampler/sinc_resampler.h",
+ "ring_buffer.c",
+ "ring_buffer.h",
"signal_processing/auto_corr_to_refl_coef.c",
"signal_processing/auto_correlation.c",
"signal_processing/complex_fft_tables.h",
diff --git a/webrtc/common_audio/DEPS b/webrtc/common_audio/DEPS
index 2b59d23..7df03ea 100644
--- a/webrtc/common_audio/DEPS
+++ b/webrtc/common_audio/DEPS
@@ -1,6 +1,5 @@
include_rules = [
"+dl/sp/api", # For openmax_dl.
"+webrtc/base",
- "+webrtc/modules/audio_processing",
"+webrtc/system_wrappers",
]
diff --git a/webrtc/modules/audio_processing/utility/audio_ring_buffer.cc b/webrtc/common_audio/audio_ring_buffer.cc
similarity index 93%
rename from webrtc/modules/audio_processing/utility/audio_ring_buffer.cc
rename to webrtc/common_audio/audio_ring_buffer.cc
index 73f578f..a29e53a 100644
--- a/webrtc/modules/audio_processing/utility/audio_ring_buffer.cc
+++ b/webrtc/common_audio/audio_ring_buffer.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
+#include "webrtc/common_audio/audio_ring_buffer.h"
#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
+#include "webrtc/common_audio/ring_buffer.h"
// This is a simple multi-channel wrapper over the ring_buffer.h C interface.
diff --git a/webrtc/common_audio/audio_ring_buffer.h b/webrtc/common_audio/audio_ring_buffer.h
index 820d127..ae825a3 100644
--- a/webrtc/common_audio/audio_ring_buffer.h
+++ b/webrtc/common_audio/audio_ring_buffer.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -10,7 +10,46 @@
#ifndef WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
#define WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
-// TODO(peah): Remove as soon as all downstream dependencies are resolved.
-#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
+#include <stddef.h>
+#include <vector>
+
+struct RingBuffer;
+
+namespace webrtc {
+
+// A ring buffer tailored for float deinterleaved audio. Any operation that
+// cannot be performed as requested will cause a crash (e.g. insufficient data
+// in the buffer to fulfill a read request.)
+class AudioRingBuffer final {
+ public:
+ // Specify the number of channels and maximum number of frames the buffer will
+ // contain.
+ AudioRingBuffer(size_t channels, size_t max_frames);
+ ~AudioRingBuffer();
+
+ // Copies |data| to the buffer and advances the write pointer. |channels| must
+ // be the same as at creation time.
+ void Write(const float* const* data, size_t channels, size_t frames);
+
+ // Copies from the buffer to |data| and advances the read pointer. |channels|
+ // must be the same as at creation time.
+ void Read(float* const* data, size_t channels, size_t frames);
+
+ size_t ReadFramesAvailable() const;
+ size_t WriteFramesAvailable() const;
+
+ // Moves the read position. The forward version advances the read pointer
+ // towards the write pointer and the backward verison withdraws the read
+ // pointer away from the write pointer (i.e. flushing and stuffing the buffer
+ // respectively.)
+ void MoveReadPositionForward(size_t frames);
+ void MoveReadPositionBackward(size_t frames);
+
+ private:
+ // TODO(kwiberg): Use std::vector<std::unique_ptr<RingBuffer>> instead.
+ std::vector<RingBuffer*> buffers_;
+};
+
+} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_AUDIO_RING_BUFFER_H_
diff --git a/webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc b/webrtc/common_audio/audio_ring_buffer_unittest.cc
similarity index 97%
rename from webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc
rename to webrtc/common_audio/audio_ring_buffer_unittest.cc
index c2d5e7a..c5c38de 100644
--- a/webrtc/modules/audio_processing/utility/audio_ring_buffer_unittest.cc
+++ b/webrtc/common_audio/audio_ring_buffer_unittest.cc
@@ -9,9 +9,8 @@
*/
#include <memory>
-#include <tuple>
-#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
+#include "webrtc/common_audio/audio_ring_buffer.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_audio/channel_buffer.h"
diff --git a/webrtc/modules/audio_processing/utility/blocker.cc b/webrtc/common_audio/blocker.cc
similarity index 98%
rename from webrtc/modules/audio_processing/utility/blocker.cc
rename to webrtc/common_audio/blocker.cc
index a3661cc..13432f2 100644
--- a/webrtc/modules/audio_processing/utility/blocker.cc
+++ b/webrtc/common_audio/blocker.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/utility/blocker.h"
+#include "webrtc/common_audio/blocker.h"
#include <string.h>
diff --git a/webrtc/common_audio/blocker.h b/webrtc/common_audio/blocker.h
index 2ce2e93..edf81d3 100644
--- a/webrtc/common_audio/blocker.h
+++ b/webrtc/common_audio/blocker.h
@@ -1,17 +1,124 @@
- /*
- * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
-#ifndef WEBRTC_COMMON_AUDIO_BLOCKER_H_
-#define WEBRTC_COMMON_AUDIO_BLOCKER_H_
+#ifndef WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
+#define WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
-// TODO(peah): Remove as soon as all downstream dependencies are resolved.
-#include "webrtc/modules/audio_processing/utility/blocker.h"
+#include <memory>
-#endif // WEBRTC_COMMON_AUDIO_BLOCKER_H_
+#include "webrtc/common_audio/audio_ring_buffer.h"
+#include "webrtc/common_audio/channel_buffer.h"
+
+namespace webrtc {
+
+// The callback function to process audio in the time domain. Input has already
+// been windowed, and output will be windowed. The number of input channels
+// must be >= the number of output channels.
+class BlockerCallback {
+ public:
+ virtual ~BlockerCallback() {}
+
+ virtual void ProcessBlock(const float* const* input,
+ size_t num_frames,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ float* const* output) = 0;
+};
+
+// The main purpose of Blocker is to abstract away the fact that often we
+// receive a different number of audio frames than our transform takes. For
+// example, most FFTs work best when the fft-size is a power of 2, but suppose
+// we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames
+// of audio, which is not a power of 2. Blocker allows us to specify the
+// transform and all other necessary processing via the Process() callback
+// function without any constraints on the transform-size
+// (read: |block_size_|) or received-audio-size (read: |chunk_size_|).
+// We handle this for the multichannel audio case, allowing for different
+// numbers of input and output channels (for example, beamforming takes 2 or
+// more input channels and returns 1 output channel). Audio signals are
+// represented as deinterleaved floats in the range [-1, 1].
+//
+// Blocker is responsible for:
+// - blocking audio while handling potential discontinuities on the edges
+// of chunks
+// - windowing blocks before sending them to Process()
+// - windowing processed blocks, and overlap-adding them together before
+// sending back a processed chunk
+//
+// To use blocker:
+// 1. Impelment a BlockerCallback object |bc|.
+// 2. Instantiate a Blocker object |b|, passing in |bc|.
+// 3. As you receive audio, call b.ProcessChunk() to get processed audio.
+//
+// A small amount of delay is added to the first received chunk to deal with
+// the difference in chunk/block sizes. This delay is <= chunk_size.
+//
+// Ownership of window is retained by the caller. That is, Blocker makes a
+// copy of window and does not attempt to delete it.
+class Blocker {
+ public:
+ Blocker(size_t chunk_size,
+ size_t block_size,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ const float* window,
+ size_t shift_amount,
+ BlockerCallback* callback);
+
+ void ProcessChunk(const float* const* input,
+ size_t chunk_size,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ float* const* output);
+
+ private:
+ const size_t chunk_size_;
+ const size_t block_size_;
+ const size_t num_input_channels_;
+ const size_t num_output_channels_;
+
+ // The number of frames of delay to add at the beginning of the first chunk.
+ const size_t initial_delay_;
+
+ // The frame index into the input buffer where the first block should be read
+ // from. This is necessary because shift_amount_ is not necessarily a
+ // multiple of chunk_size_, so blocks won't line up at the start of the
+ // buffer.
+ size_t frame_offset_;
+
+ // Since blocks nearly always overlap, there are certain blocks that require
+ // frames from the end of one chunk and the beginning of the next chunk. The
+ // input and output buffers are responsible for saving those frames between
+ // calls to ProcessChunk().
+ //
+ // Both contain |initial delay| + |chunk_size| frames. The input is a fairly
+ // standard FIFO, but due to the overlap-add it's harder to use an
+ // AudioRingBuffer for the output.
+ AudioRingBuffer input_buffer_;
+ ChannelBuffer<float> output_buffer_;
+
+ // Space for the input block (can't wrap because of windowing).
+ ChannelBuffer<float> input_block_;
+
+ // Space for the output block (can't wrap because of overlap/add).
+ ChannelBuffer<float> output_block_;
+
+ std::unique_ptr<float[]> window_;
+
+ // The amount of frames between the start of contiguous blocks. For example,
+ // |shift_amount_| = |block_size_| / 2 for a Hann window.
+ size_t shift_amount_;
+
+ BlockerCallback* callback_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_INTERNAL_BEAMFORMER_BLOCKER_H_
diff --git a/webrtc/modules/audio_processing/utility/blocker_unittest.cc b/webrtc/common_audio/blocker_unittest.cc
similarity index 99%
rename from webrtc/modules/audio_processing/utility/blocker_unittest.cc
rename to webrtc/common_audio/blocker_unittest.cc
index 3f53f68..eea3e25 100644
--- a/webrtc/modules/audio_processing/utility/blocker_unittest.cc
+++ b/webrtc/common_audio/blocker_unittest.cc
@@ -10,7 +10,7 @@
#include <memory>
-#include "webrtc/modules/audio_processing/utility/blocker.h"
+#include "webrtc/common_audio/blocker.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/arraysize.h"
diff --git a/webrtc/common_audio/common_audio.gyp b/webrtc/common_audio/common_audio.gyp
index 4a6377a..57d9f1c 100644
--- a/webrtc/common_audio/common_audio.gyp
+++ b/webrtc/common_audio/common_audio.gyp
@@ -31,8 +31,10 @@
'sources': [
'audio_converter.cc',
'audio_converter.h',
+ 'audio_ring_buffer.cc',
'audio_ring_buffer.h',
'audio_util.cc',
+ 'blocker.cc',
'blocker.h',
'channel_buffer.cc',
'channel_buffer.h',
@@ -43,6 +45,7 @@
'fir_filter_neon.h',
'fir_filter_sse.h',
'include/audio_util.h',
+ 'lapped_transform.cc',
'lapped_transform.h',
'real_fourier.cc',
'real_fourier.h',
@@ -56,6 +59,8 @@
'resampler/resampler.cc',
'resampler/sinc_resampler.cc',
'resampler/sinc_resampler.h',
+ 'ring_buffer.c',
+ 'ring_buffer.h',
'signal_processing/include/real_fft.h',
'signal_processing/include/signal_processing_library.h',
'signal_processing/include/spl_inl.h',
@@ -235,14 +240,18 @@
],
'sources': [
'audio_converter_unittest.cc',
+ 'audio_ring_buffer_unittest.cc',
'audio_util_unittest.cc',
+ 'blocker_unittest.cc',
'fir_filter_unittest.cc',
+ 'lapped_transform_unittest.cc',
'real_fourier_unittest.cc',
'resampler/resampler_unittest.cc',
'resampler/push_resampler_unittest.cc',
'resampler/push_sinc_resampler_unittest.cc',
'resampler/sinusoidal_linear_chirp_source.cc',
'resampler/sinusoidal_linear_chirp_source.h',
+ 'ring_buffer_unittest.cc',
'signal_processing/real_fft_unittest.cc',
'signal_processing/signal_processing_unittest.cc',
'sparse_fir_filter_unittest.cc',
diff --git a/webrtc/modules/audio_processing/utility/lapped_transform.cc b/webrtc/common_audio/lapped_transform.cc
similarity index 97%
rename from webrtc/modules/audio_processing/utility/lapped_transform.cc
rename to webrtc/common_audio/lapped_transform.cc
index cb5496d..0edf586 100644
--- a/webrtc/modules/audio_processing/utility/lapped_transform.cc
+++ b/webrtc/common_audio/lapped_transform.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
+#include "webrtc/common_audio/lapped_transform.h"
#include <algorithm>
#include <cstdlib>
diff --git a/webrtc/common_audio/lapped_transform.h b/webrtc/common_audio/lapped_transform.h
index 18b7768..8327359 100644
--- a/webrtc/common_audio/lapped_transform.h
+++ b/webrtc/common_audio/lapped_transform.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -11,7 +11,115 @@
#ifndef WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
#define WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
-// TODO(peah): Remove as soon as all downstream dependencies are resolved.
-#include "webrtc/modules/audio_processing/utility/lapped_transform.h
+#include <complex>
+#include <memory>
+
+#include "webrtc/common_audio/blocker.h"
+#include "webrtc/common_audio/real_fourier.h"
+#include "webrtc/system_wrappers/include/aligned_array.h"
+
+namespace webrtc {
+
+// Helper class for audio processing modules which operate on frequency domain
+// input derived from the windowed time domain audio stream.
+//
+// The input audio chunk is sliced into possibly overlapping blocks, multiplied
+// by a window and transformed with an FFT implementation. The transformed data
+// is supplied to the given callback for processing. The processed output is
+// then inverse transformed into the time domain and spliced back into a chunk
+// which constitutes the final output of this processing module.
+class LappedTransform {
+ public:
+ class Callback {
+ public:
+ virtual ~Callback() {}
+
+ virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
+ size_t num_in_channels, size_t frames,
+ size_t num_out_channels,
+ std::complex<float>* const* out_block) = 0;
+ };
+
+ // Construct a transform instance. |chunk_length| is the number of samples in
+ // each channel. |window| defines the window, owned by the caller (a copy is
+ // made internally); |window| should have length equal to |block_length|.
+ // |block_length| defines the length of a block, in samples.
+ // |shift_amount| is in samples. |callback| is the caller-owned audio
+ // processing function called for each block of the input chunk.
+ LappedTransform(size_t num_in_channels,
+ size_t num_out_channels,
+ size_t chunk_length,
+ const float* window,
+ size_t block_length,
+ size_t shift_amount,
+ Callback* callback);
+ ~LappedTransform() {}
+
+ // Main audio processing helper method. Internally slices |in_chunk| into
+ // blocks, transforms them to frequency domain, calls the callback for each
+ // block and returns a de-blocked time domain chunk of audio through
+ // |out_chunk|. Both buffers are caller-owned.
+ void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
+
+ // Get the chunk length.
+ //
+ // The chunk length is the number of samples per channel that must be passed
+ // to ProcessChunk via the parameter in_chunk.
+ //
+ // Returns the same chunk_length passed to the LappedTransform constructor.
+ size_t chunk_length() const { return chunk_length_; }
+
+ // Get the number of input channels.
+ //
+ // This is the number of arrays that must be passed to ProcessChunk via
+ // in_chunk.
+ //
+ // Returns the same num_in_channels passed to the LappedTransform constructor.
+ size_t num_in_channels() const { return num_in_channels_; }
+
+ // Get the number of output channels.
+ //
+ // This is the number of arrays that must be passed to ProcessChunk via
+ // out_chunk.
+ //
+ // Returns the same num_out_channels passed to the LappedTransform
+ // constructor.
+ size_t num_out_channels() const { return num_out_channels_; }
+
+ private:
+ // Internal middleware callback, given to the blocker. Transforms each block
+ // and hands it over to the processing method given at construction time.
+ class BlockThunk : public BlockerCallback {
+ public:
+ explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
+
+ virtual void ProcessBlock(const float* const* input,
+ size_t num_frames,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ float* const* output);
+
+ private:
+ LappedTransform* const parent_;
+ } blocker_callback_;
+
+ const size_t num_in_channels_;
+ const size_t num_out_channels_;
+
+ const size_t block_length_;
+ const size_t chunk_length_;
+
+ Callback* const block_processor_;
+ Blocker blocker_;
+
+ std::unique_ptr<RealFourier> fft_;
+ const size_t cplx_length_;
+ AlignedArray<float> real_buf_;
+ AlignedArray<std::complex<float> > cplx_pre_;
+ AlignedArray<std::complex<float> > cplx_post_;
+};
+
+} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
+
diff --git a/webrtc/modules/audio_processing/utility/lapped_transform_unittest.cc b/webrtc/common_audio/lapped_transform_unittest.cc
similarity index 98%
rename from webrtc/modules/audio_processing/utility/lapped_transform_unittest.cc
rename to webrtc/common_audio/lapped_transform_unittest.cc
index f6c8ebd..a78488e 100644
--- a/webrtc/modules/audio_processing/utility/lapped_transform_unittest.cc
+++ b/webrtc/common_audio/lapped_transform_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/utility/lapped_transform.h"
+#include "webrtc/common_audio/lapped_transform.h"
#include <algorithm>
#include <cmath>
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer.c b/webrtc/common_audio/ring_buffer.c
similarity index 98%
rename from webrtc/modules/audio_processing/utility/ring_buffer.c
rename to webrtc/common_audio/ring_buffer.c
index c71bac1..60fb5df 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer.c
+++ b/webrtc/common_audio/ring_buffer.c
@@ -11,7 +11,7 @@
// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
// otherwise specified, functions return 0 on success and -1 on error.
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
+#include "webrtc/common_audio/ring_buffer.h"
#include <stddef.h> // size_t
#include <stdlib.h>
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer.h b/webrtc/common_audio/ring_buffer.h
similarity index 92%
rename from webrtc/modules/audio_processing/utility/ring_buffer.h
rename to webrtc/common_audio/ring_buffer.h
index a46c262..4125c48 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer.h
+++ b/webrtc/common_audio/ring_buffer.h
@@ -11,8 +11,8 @@
// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
// otherwise specified, functions return 0 on success and -1 on error.
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
+#ifndef WEBRTC_COMMON_AUDIO_RING_BUFFER_H_
+#define WEBRTC_COMMON_AUDIO_RING_BUFFER_H_
#ifdef __cplusplus
extern "C" {
@@ -63,4 +63,4 @@
}
#endif
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
+#endif // WEBRTC_COMMON_AUDIO_RING_BUFFER_H_
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc b/webrtc/common_audio/ring_buffer_unittest.cc
similarity index 98%
rename from webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc
rename to webrtc/common_audio/ring_buffer_unittest.cc
index 7972657..92c470a 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer_unittest.cc
+++ b/webrtc/common_audio/ring_buffer_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
+#include "webrtc/common_audio/ring_buffer.h"
#include <stdlib.h>
#include <time.h>
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index bf7bdc5..22c904d 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -108,21 +108,13 @@
"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",
diff --git a/webrtc/modules/audio_processing/aec/aec_core.cc b/webrtc/modules/audio_processing/aec/aec_core.cc
index 4fe635d..e23a793 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.cc
+++ b/webrtc/modules/audio_processing/aec/aec_core.cc
@@ -24,6 +24,9 @@
#include <stdlib.h>
#include <string.h>
+extern "C" {
+#include "webrtc/common_audio/ring_buffer.h"
+}
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/aec/aec_common.h"
#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
@@ -33,7 +36,6 @@
#include "webrtc/modules/audio_processing/logging/aec_logging.h"
extern "C" {
#include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
}
#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/audio_processing/aec/aec_core_internal.h b/webrtc/modules/audio_processing/aec/aec_core_internal.h
index 1f4f99b..ea5889f 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_internal.h
+++ b/webrtc/modules/audio_processing/aec/aec_core_internal.h
@@ -11,14 +11,13 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_
+extern "C" {
+#include "webrtc/common_audio/ring_buffer.h"
+}
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/aec/aec_common.h"
#include "webrtc/modules/audio_processing/aec/aec_core.h"
#include "webrtc/modules/audio_processing/utility/block_mean_calculator.h"
-extern "C" {
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
-}
-
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.cc b/webrtc/modules/audio_processing/aec/echo_cancellation.cc
index 493068d..32496ca 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.cc
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.cc
@@ -21,14 +21,12 @@
#include <string.h>
extern "C" {
+#include "webrtc/common_audio/ring_buffer.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
}
#include "webrtc/modules/audio_processing/aec/aec_core.h"
#include "webrtc/modules/audio_processing/aec/aec_resampler.h"
#include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
-extern "C" {
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
-}
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
index 537ab5d..b4a6fd8 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
@@ -11,10 +11,10 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
-#include "webrtc/modules/audio_processing/aec/aec_core.h"
extern "C" {
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
+#include "webrtc/common_audio/ring_buffer.h"
}
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c
index 084feb8..6bf1cf7 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.c
@@ -14,10 +14,10 @@
#include <stddef.h>
#include <stdlib.h>
+#include "webrtc/common_audio/ring_buffer.h"
#include "webrtc/common_audio/signal_processing/include/real_fft.h"
#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"
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.h b/webrtc/modules/audio_processing/aecm/aecm_core.h
index 9b70f47..b52bb62 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.h
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.h
@@ -13,9 +13,9 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_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_defines.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
#include "webrtc/typedefs.h"
#ifdef _MSC_VER // visual c++
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_c.c b/webrtc/modules/audio_processing/aecm/aecm_core_c.c
index ea5f2bc..3a8fafa 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_c.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_c.c
@@ -14,10 +14,10 @@
#include <stddef.h>
#include <stdlib.h>
+#include "webrtc/common_audio/ring_buffer.h"
#include "webrtc/common_audio/signal_processing/include/real_fft.h"
#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 @@
out[i].imag = WebRtcSpl_AddSatW16(out[i].imag, uImag[i]);
}
}
+
diff --git a/webrtc/modules/audio_processing/aecm/echo_control_mobile.c b/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
index d58ee3e..91e6f0e 100644
--- a/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
+++ b/webrtc/modules/audio_processing/aecm/echo_control_mobile.c
@@ -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
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index 75da59f..264f3e5 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -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',
diff --git a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
index c1f9188..b8953b0 100644
--- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
+++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.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
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
index 728e4cf..1413212 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
@@ -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 {
diff --git a/webrtc/modules/audio_processing/utility/audio_ring_buffer.h b/webrtc/modules/audio_processing/utility/audio_ring_buffer.h
deleted file mode 100644
index 8f97587..0000000
--- a/webrtc/modules/audio_processing/utility/audio_ring_buffer.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_AUDIO_RING_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_AUDIO_RING_BUFFER_H_
-
-#include <stddef.h>
-#include <vector>
-
-struct RingBuffer;
-
-namespace webrtc {
-
-// A ring buffer tailored for float deinterleaved audio. Any operation that
-// cannot be performed as requested will cause a crash (e.g. insufficient data
-// in the buffer to fulfill a read request.)
-class AudioRingBuffer final {
- public:
- // Specify the number of channels and maximum number of frames the buffer will
- // contain.
- AudioRingBuffer(size_t channels, size_t max_frames);
- ~AudioRingBuffer();
-
- // Copies |data| to the buffer and advances the write pointer. |channels| must
- // be the same as at creation time.
- void Write(const float* const* data, size_t channels, size_t frames);
-
- // Copies from the buffer to |data| and advances the read pointer. |channels|
- // must be the same as at creation time.
- void Read(float* const* data, size_t channels, size_t frames);
-
- size_t ReadFramesAvailable() const;
- size_t WriteFramesAvailable() const;
-
- // Moves the read position. The forward version advances the read pointer
- // towards the write pointer and the backward verison withdraws the read
- // pointer away from the write pointer (i.e. flushing and stuffing the buffer
- // respectively.)
- void MoveReadPositionForward(size_t frames);
- void MoveReadPositionBackward(size_t frames);
-
- private:
- // TODO(kwiberg): Use std::vector<std::unique_ptr<RingBuffer>> instead.
- std::vector<RingBuffer*> buffers_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_AUDIO_RING_BUFFER_H_
diff --git a/webrtc/modules/audio_processing/utility/blocker.h b/webrtc/modules/audio_processing/utility/blocker.h
deleted file mode 100644
index 7d9bf66..0000000
--- a/webrtc/modules/audio_processing/utility/blocker.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
-
-#include <memory>
-
-#include "webrtc/common_audio/channel_buffer.h"
-#include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h"
-
-namespace webrtc {
-
-// The callback function to process audio in the time domain. Input has already
-// been windowed, and output will be windowed. The number of input channels
-// must be >= the number of output channels.
-class BlockerCallback {
- public:
- virtual ~BlockerCallback() {}
-
- virtual void ProcessBlock(const float* const* input,
- size_t num_frames,
- size_t num_input_channels,
- size_t num_output_channels,
- float* const* output) = 0;
-};
-
-// The main purpose of Blocker is to abstract away the fact that often we
-// receive a different number of audio frames than our transform takes. For
-// example, most FFTs work best when the fft-size is a power of 2, but suppose
-// we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames
-// of audio, which is not a power of 2. Blocker allows us to specify the
-// transform and all other necessary processing via the Process() callback
-// function without any constraints on the transform-size
-// (read: |block_size_|) or received-audio-size (read: |chunk_size_|).
-// We handle this for the multichannel audio case, allowing for different
-// numbers of input and output channels (for example, beamforming takes 2 or
-// more input channels and returns 1 output channel). Audio signals are
-// represented as deinterleaved floats in the range [-1, 1].
-//
-// Blocker is responsible for:
-// - blocking audio while handling potential discontinuities on the edges
-// of chunks
-// - windowing blocks before sending them to Process()
-// - windowing processed blocks, and overlap-adding them together before
-// sending back a processed chunk
-//
-// To use blocker:
-// 1. Impelment a BlockerCallback object |bc|.
-// 2. Instantiate a Blocker object |b|, passing in |bc|.
-// 3. As you receive audio, call b.ProcessChunk() to get processed audio.
-//
-// A small amount of delay is added to the first received chunk to deal with
-// the difference in chunk/block sizes. This delay is <= chunk_size.
-//
-// Ownership of window is retained by the caller. That is, Blocker makes a
-// copy of window and does not attempt to delete it.
-class Blocker {
- public:
- Blocker(size_t chunk_size,
- size_t block_size,
- size_t num_input_channels,
- size_t num_output_channels,
- const float* window,
- size_t shift_amount,
- BlockerCallback* callback);
-
- void ProcessChunk(const float* const* input,
- size_t chunk_size,
- size_t num_input_channels,
- size_t num_output_channels,
- float* const* output);
-
- private:
- const size_t chunk_size_;
- const size_t block_size_;
- const size_t num_input_channels_;
- const size_t num_output_channels_;
-
- // The number of frames of delay to add at the beginning of the first chunk.
- const size_t initial_delay_;
-
- // The frame index into the input buffer where the first block should be read
- // from. This is necessary because shift_amount_ is not necessarily a
- // multiple of chunk_size_, so blocks won't line up at the start of the
- // buffer.
- size_t frame_offset_;
-
- // Since blocks nearly always overlap, there are certain blocks that require
- // frames from the end of one chunk and the beginning of the next chunk. The
- // input and output buffers are responsible for saving those frames between
- // calls to ProcessChunk().
- //
- // Both contain |initial delay| + |chunk_size| frames. The input is a fairly
- // standard FIFO, but due to the overlap-add it's harder to use an
- // AudioRingBuffer for the output.
- AudioRingBuffer input_buffer_;
- ChannelBuffer<float> output_buffer_;
-
- // Space for the input block (can't wrap because of windowing).
- ChannelBuffer<float> input_block_;
-
- // Space for the output block (can't wrap because of overlap/add).
- ChannelBuffer<float> output_block_;
-
- std::unique_ptr<float[]> window_;
-
- // The amount of frames between the start of contiguous blocks. For example,
- // |shift_amount_| = |block_size_| / 2 for a Hann window.
- size_t shift_amount_;
-
- BlockerCallback* callback_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_
diff --git a/webrtc/modules/audio_processing/utility/lapped_transform.h b/webrtc/modules/audio_processing/utility/lapped_transform.h
deleted file mode 100644
index 1286ecf..0000000
--- a/webrtc/modules/audio_processing/utility/lapped_transform.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
-
-#include <complex>
-#include <memory>
-
-#include "webrtc/common_audio/real_fourier.h"
-#include "webrtc/modules/audio_processing/utility/blocker.h"
-#include "webrtc/system_wrappers/include/aligned_array.h"
-
-namespace webrtc {
-
-// Helper class for audio processing modules which operate on frequency domain
-// input derived from the windowed time domain audio stream.
-//
-// The input audio chunk is sliced into possibly overlapping blocks, multiplied
-// by a window and transformed with an FFT implementation. The transformed data
-// is supplied to the given callback for processing. The processed output is
-// then inverse transformed into the time domain and spliced back into a chunk
-// which constitutes the final output of this processing module.
-class LappedTransform {
- public:
- class Callback {
- public:
- virtual ~Callback() {}
-
- virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
- size_t num_in_channels, size_t frames,
- size_t num_out_channels,
- std::complex<float>* const* out_block) = 0;
- };
-
- // Construct a transform instance. |chunk_length| is the number of samples in
- // each channel. |window| defines the window, owned by the caller (a copy is
- // made internally); |window| should have length equal to |block_length|.
- // |block_length| defines the length of a block, in samples.
- // |shift_amount| is in samples. |callback| is the caller-owned audio
- // processing function called for each block of the input chunk.
- LappedTransform(size_t num_in_channels,
- size_t num_out_channels,
- size_t chunk_length,
- const float* window,
- size_t block_length,
- size_t shift_amount,
- Callback* callback);
- ~LappedTransform() {}
-
- // Main audio processing helper method. Internally slices |in_chunk| into
- // blocks, transforms them to frequency domain, calls the callback for each
- // block and returns a de-blocked time domain chunk of audio through
- // |out_chunk|. Both buffers are caller-owned.
- void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
-
- // Get the chunk length.
- //
- // The chunk length is the number of samples per channel that must be passed
- // to ProcessChunk via the parameter in_chunk.
- //
- // Returns the same chunk_length passed to the LappedTransform constructor.
- size_t chunk_length() const { return chunk_length_; }
-
- // Get the number of input channels.
- //
- // This is the number of arrays that must be passed to ProcessChunk via
- // in_chunk.
- //
- // Returns the same num_in_channels passed to the LappedTransform constructor.
- size_t num_in_channels() const { return num_in_channels_; }
-
- // Get the number of output channels.
- //
- // This is the number of arrays that must be passed to ProcessChunk via
- // out_chunk.
- //
- // Returns the same num_out_channels passed to the LappedTransform
- // constructor.
- size_t num_out_channels() const { return num_out_channels_; }
-
- private:
- // Internal middleware callback, given to the blocker. Transforms each block
- // and hands it over to the processing method given at construction time.
- class BlockThunk : public BlockerCallback {
- public:
- explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
-
- virtual void ProcessBlock(const float* const* input,
- size_t num_frames,
- size_t num_input_channels,
- size_t num_output_channels,
- float* const* output);
-
- private:
- LappedTransform* const parent_;
- } blocker_callback_;
-
- const size_t num_in_channels_;
- const size_t num_out_channels_;
-
- const size_t block_length_;
- const size_t chunk_length_;
-
- Callback* const block_processor_;
- Blocker blocker_;
-
- std::unique_ptr<RealFourier> fft_;
- const size_t cplx_length_;
- AlignedArray<float> real_buf_;
- AlignedArray<std::complex<float> > cplx_pre_;
- AlignedArray<std::complex<float> > cplx_post_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index eb886a9..d5df853 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -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',
