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

Issue 1673263002: Update path for audioproc_debug proto output. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix GN build and move the test proto as well Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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_AUDIO_FILE_PROCESSOR_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
13 13
14 #include <algorithm> 14 #include <algorithm>
15 #include <limits> 15 #include <limits>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/scoped_ptr.h" 18 #include "webrtc/base/scoped_ptr.h"
19 #include "webrtc/common_audio/channel_buffer.h" 19 #include "webrtc/common_audio/channel_buffer.h"
20 #include "webrtc/common_audio/wav_file.h" 20 #include "webrtc/common_audio/wav_file.h"
21 #include "webrtc/modules/audio_processing/include/audio_processing.h" 21 #include "webrtc/modules/audio_processing/include/audio_processing.h"
22 #include "webrtc/modules/audio_processing/test/test_utils.h" 22 #include "webrtc/modules/audio_processing/test/test_utils.h"
23 #include "webrtc/system_wrappers/include/tick_util.h" 23 #include "webrtc/system_wrappers/include/tick_util.h"
24 24
25 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD 25 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
26 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" 26 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
27 #else 27 #else
28 #include "webrtc/audio_processing/debug.pb.h" 28 #include "webrtc/modules/audio_processing/debug.pb.h"
29 #endif 29 #endif
30 30
31 namespace webrtc { 31 namespace webrtc {
32 32
33 // Holds a few statistics about a series of TickIntervals. 33 // Holds a few statistics about a series of TickIntervals.
34 struct TickIntervalStats { 34 struct TickIntervalStats {
35 TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {} 35 TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
36 TickInterval sum; 36 TickInterval sum;
37 TickInterval max; 37 TickInterval max;
38 TickInterval min; 38 TickInterval min;
(...skipping 91 matching lines...) Expand 10 before | Expand all | Expand 10 after
130 ChannelBuffer<float> out_buf_; 130 ChannelBuffer<float> out_buf_;
131 StreamConfig input_config_; 131 StreamConfig input_config_;
132 StreamConfig reverse_config_; 132 StreamConfig reverse_config_;
133 const StreamConfig output_config_; 133 const StreamConfig output_config_;
134 ChannelBufferWavWriter buffer_writer_; 134 ChannelBufferWavWriter buffer_writer_;
135 }; 135 };
136 136
137 } // namespace webrtc 137 } // namespace webrtc
138 138
139 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_ 139 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
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