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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 /* | 11 /* |
12 * Contains the API functions for the AEC. | 12 * Contains the API functions for the AEC. |
13 */ | 13 */ |
14 #include "webrtc/modules/audio_processing/aec/echo_cancellation.h" | 14 #include "webrtc/modules/audio_processing/aec/echo_cancellation.h" |
15 | 15 |
16 #include <math.h> | 16 #include <math.h> |
17 #include <stdlib.h> | 17 #include <stdlib.h> |
18 #include <string.h> | 18 #include <string.h> |
19 | 19 |
20 extern "C" { | 20 extern "C" { |
21 #include "webrtc/common_audio/ring_buffer.h" | 21 #include "webrtc/common_audio/ring_buffer.h" |
22 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" | 22 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" |
23 } | 23 } |
24 #include "webrtc/modules/audio_processing/aec/aec_core.h" | 24 #include "webrtc/modules/audio_processing/aec/aec_core.h" |
25 #include "webrtc/modules/audio_processing/aec/aec_resampler.h" | 25 #include "webrtc/modules/audio_processing/aec/aec_resampler.h" |
26 #include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h" | |
27 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" | 26 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" |
28 #include "webrtc/typedefs.h" | 27 #include "webrtc/typedefs.h" |
29 | 28 |
30 namespace webrtc { | 29 namespace webrtc { |
31 | 30 |
32 // Measured delays [ms] | 31 // Measured delays [ms] |
33 // Device Chrome GTP | 32 // Device Chrome GTP |
34 // MacBook Air 10 | 33 // MacBook Air 10 |
35 // MacBook Retina 10 100 | 34 // MacBook Retina 10 100 |
36 // MacPro 30? | 35 // MacPro 30? |
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847 } else { | 846 } else { |
848 self->timeForDelayChange = 0; | 847 self->timeForDelayChange = 0; |
849 } | 848 } |
850 self->lastDelayDiff = delay_difference; | 849 self->lastDelayDiff = delay_difference; |
851 | 850 |
852 if (self->timeForDelayChange > 25) { | 851 if (self->timeForDelayChange > 25) { |
853 self->knownDelay = WEBRTC_SPL_MAX((int)self->filtDelay - 256, 0); | 852 self->knownDelay = WEBRTC_SPL_MAX((int)self->filtDelay - 256, 0); |
854 } | 853 } |
855 } | 854 } |
856 } // namespace webrtc | 855 } // namespace webrtc |
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