Moving LappedTransform, Blocker and AudioRingBuffer.

LappedTransform is only used in BandwidthAdaptationTest and therefore it
should not be anymore a visible target under common_audio.
This CL moves LappedTransform and other two classes it depends on (and which
are not used elsewhere) to modules/audio_coding/codecs/opus/test.

Bug: webrtc:9577, webrtc:5298
Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a
Reviewed-on: https://webrtc-review.googlesource.com/96440
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24509}
This commit is contained in:
Alessio Bazzica
2018-08-31 10:41:37 +02:00
committed by Commit Bot
parent 8a3c166fff
commit d4161a3c9d
14 changed files with 78 additions and 27 deletions

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@ -15,16 +15,10 @@ rtc_static_library("common_audio") {
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",
"include/audio_util.h",
"lapped_transform.cc",
"lapped_transform.h",
"real_fourier.cc",
"real_fourier.h",
"real_fourier_ooura.cc",
@ -349,12 +343,9 @@ if (rtc_include_tests) {
sources = [
"audio_converter_unittest.cc",
"audio_ring_buffer_unittest.cc",
"audio_util_unittest.cc",
"blocker_unittest.cc",
"channel_buffer_unittest.cc",
"fir_filter_unittest.cc",
"lapped_transform_unittest.cc",
"real_fourier_unittest.cc",
"resampler/push_resampler_unittest.cc",
"resampler/push_sinc_resampler_unittest.cc",

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@ -14,6 +14,8 @@
#ifndef COMMON_AUDIO_RING_BUFFER_H_
#define COMMON_AUDIO_RING_BUFFER_H_
// TODO(alessiob): Used by AEC, AECm and AudioRingBuffer. Remove when possible.
#ifdef __cplusplus
extern "C" {
#endif

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@ -2093,6 +2093,8 @@ if (rtc_include_tests) {
"../../test:rtp_test_utils",
"../../test:test_common",
"../../test:test_support",
"codecs/opus/test",
"codecs/opus/test:test_unittest",
"//testing/gtest",
"//third_party/abseil-cpp/absl/memory",
]

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@ -11,8 +11,8 @@
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "common_audio/include/audio_util.h"
#include "common_audio/lapped_transform.h"
#include "common_audio/window_generator.h"
#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "test/field_trial.h"
#include "test/gtest.h"

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@ -0,0 +1,55 @@
# Copyright (c) 2018 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.
import("../../../../../webrtc.gni")
visibility = [
":*",
"../../../:*",
]
if (rtc_include_tests) {
rtc_static_library("test") {
testonly = true
sources = [
"audio_ring_buffer.cc",
"audio_ring_buffer.h",
"blocker.cc",
"blocker.h",
"lapped_transform.cc",
"lapped_transform.h",
]
deps = [
"../../../../../common_audio:common_audio",
"../../../../../common_audio:common_audio_c",
"../../../../../rtc_base:checks",
"../../../../../rtc_base/memory:aligned_array",
]
}
rtc_source_set("test_unittest") {
testonly = true
sources = [
"audio_ring_buffer_unittest.cc",
"blocker_unittest.cc",
"lapped_transform_unittest.cc",
]
deps = [
":test",
"../../../../../common_audio:common_audio",
"../../../../../common_audio:common_audio_c",
"../../../../../rtc_base:rtc_base_approved",
"../../../../../test:test_support",
"//testing/gtest",
]
}
}

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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "common_audio/audio_ring_buffer.h"
#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
#include "common_audio/ring_buffer.h"
#include "rtc_base/checks.h"

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@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef COMMON_AUDIO_AUDIO_RING_BUFFER_H_
#define COMMON_AUDIO_AUDIO_RING_BUFFER_H_
#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
#include <stddef.h>
@ -54,4 +54,4 @@ class AudioRingBuffer final {
} // namespace webrtc
#endif // COMMON_AUDIO_AUDIO_RING_BUFFER_H_
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_

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@ -10,7 +10,7 @@
#include <memory>
#include "common_audio/audio_ring_buffer.h"
#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
#include "common_audio/channel_buffer.h"
#include "test/gtest.h"

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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "common_audio/blocker.h"
#include "modules/audio_coding/codecs/opus/test/blocker.h"
#include <string.h>

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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef COMMON_AUDIO_BLOCKER_H_
#define COMMON_AUDIO_BLOCKER_H_
#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
#include <memory>
#include "common_audio/audio_ring_buffer.h"
#include "common_audio/channel_buffer.h"
#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
namespace webrtc {
@ -124,4 +124,4 @@ class Blocker {
} // namespace webrtc
#endif // COMMON_AUDIO_BLOCKER_H_
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_

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@ -10,7 +10,7 @@
#include <memory>
#include "common_audio/blocker.h"
#include "modules/audio_coding/codecs/opus/test/blocker.h"
#include "rtc_base/arraysize.h"
#include "test/gtest.h"

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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "common_audio/lapped_transform.h"
#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
#include <algorithm>
#include <cstdlib>

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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef COMMON_AUDIO_LAPPED_TRANSFORM_H_
#define COMMON_AUDIO_LAPPED_TRANSFORM_H_
#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
#include <complex>
#include <memory>
#include "common_audio/blocker.h"
#include "common_audio/real_fourier.h"
#include "modules/audio_coding/codecs/opus/test/blocker.h"
#include "rtc_base/memory/aligned_array.h"
namespace webrtc {
@ -119,6 +119,7 @@ class LappedTransform {
Callback* const block_processor_;
Blocker blocker_;
// TODO(alessiob): Replace RealFourier with a different FFT library.
std::unique_ptr<RealFourier> fft_;
const size_t cplx_length_;
AlignedArray<float> real_buf_;
@ -128,4 +129,4 @@ class LappedTransform {
} // namespace webrtc
#endif // COMMON_AUDIO_LAPPED_TRANSFORM_H_
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_

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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "common_audio/lapped_transform.h"
#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
#include <algorithm>
#include <cmath>