1 /* 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ 12 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ 13 14 namespace webrtc { 15 16 const int kDefaultSampleRate = 44100; 17 const int kNumChannels = 1; 18 // Number of bytes per audio frame. 19 // Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame] 20 const size_t kBytesPerFrame = kNumChannels * (16 / 8); 21 // Delay estimates for the two different supported modes. These values are based 22 // on real-time round-trip delay estimates on a large set of devices and they 23 // are lower bounds since the filter length is 128 ms, so the AEC works for 24 // delays in the range [50, ~170] ms and [150, ~270] ms. Note that, in most 25 // cases, the lowest delay estimate will not be utilized since devices that 26 // support low-latency output audio often supports HW AEC as well. 27 const int kLowLatencyModeDelayEstimateInMilliseconds = 50; 28 const int kHighLatencyModeDelayEstimateInMilliseconds = 150; 29 30 } // namespace webrtc 31 32 #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_ 33
