Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(33)

Side by Side Diff: webrtc/modules/audio_processing/test/simulator_buffers.h

Issue 2964773002: Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" (Closed)
Patch Set: Created 3 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
13 13
14 #include <memory> 14 #include <memory>
15 #include <vector> 15 #include <vector>
16 16
17 #include "webrtc/base/random.h"
17 #include "webrtc/modules/audio_processing/audio_buffer.h" 18 #include "webrtc/modules/audio_processing/audio_buffer.h"
18 #include "webrtc/modules/audio_processing/include/audio_processing.h" 19 #include "webrtc/modules/audio_processing/include/audio_processing.h"
19 #include "webrtc/rtc_base/random.h"
20 20
21 namespace webrtc { 21 namespace webrtc {
22 namespace test { 22 namespace test {
23 23
24 struct SimulatorBuffers { 24 struct SimulatorBuffers {
25 SimulatorBuffers(int render_input_sample_rate_hz, 25 SimulatorBuffers(int render_input_sample_rate_hz,
26 int capture_input_sample_rate_hz, 26 int capture_input_sample_rate_hz,
27 int render_output_sample_rate_hz, 27 int render_output_sample_rate_hz,
28 int capture_output_sample_rate_hz, 28 int capture_output_sample_rate_hz,
29 size_t num_render_input_channels, 29 size_t num_render_input_channels,
(...skipping 27 matching lines...) Expand all
57 std::vector<float*> render_output; 57 std::vector<float*> render_output;
58 std::vector<float> render_output_samples; 58 std::vector<float> render_output_samples;
59 std::vector<float*> capture_output; 59 std::vector<float*> capture_output;
60 std::vector<float> capture_output_samples; 60 std::vector<float> capture_output_samples;
61 }; 61 };
62 62
63 } // namespace test 63 } // namespace test
64 } // namespace webrtc 64 } // namespace webrtc
65 65
66 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ 66 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698