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| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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_DEVICE_ANDROID_AUDIO_COMMON_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ |
| 12 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ | 12 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ |
| 13 | 13 |
| 14 namespace webrtc { | 14 namespace webrtc { |
| 15 | 15 |
| 16 enum { | 16 const int kDefaultSampleRate = 44100; |
| 17 kDefaultSampleRate = 44100, | 17 const int kNumChannels = 1; |
| 18 kNumChannels = 1, | 18 // Number of bytes per audio frame. |
| 19 // Number of bytes per audio frame. | 19 // Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame] |
| 20 // Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame] | 20 const size_t kBytesPerFrame = kNumChannels * (16 / 8); |
| 21 kBytesPerFrame = kNumChannels * (16 / 8), | 21 // Delay estimates for the two different supported modes. These values are based |
| 22 // Delay estimates for the two different supported modes. These values | 22 // on real-time round-trip delay estimates on a large set of devices and they |
| 23 // are based on real-time round-trip delay estimates on a large set of | 23 // are lower bounds since the filter length is 128 ms, so the AEC works for |
| 24 // devices and they are lower bounds since the filter length is 128 ms, | 24 // delays in the range [50, ~170] ms and [150, ~270] ms. Note that, in most |
| 25 // so the AEC works for delays in the range [50, ~170] ms and [150, ~270] ms. | 25 // cases, the lowest delay estimate will not be utilized since devices that |
| 26 // Note that, in most cases, the lowest delay estimate will not be utilized | 26 // support low-latency output audio often supports HW AEC as well. |
| 27 // since devices that support low-latency output audio often supports | 27 const int kLowLatencyModeDelayEstimateInMilliseconds = 50; |
| 28 // HW AEC as well. | 28 const int kHighLatencyModeDelayEstimateInMilliseconds = 150; |
| 29 kLowLatencyModeDelayEstimateInMilliseconds = 50, | |
| 30 kHighLatencyModeDelayEstimateInMilliseconds = 150, | |
| 31 }; | |
| 32 | 29 |
| 33 } // namespace webrtc | 30 } // namespace webrtc |
| 34 | 31 |
| 35 #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ | 32 #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ |
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