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Side by Side Diff: webrtc/modules/audio_device/android/audio_common.h

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Resync Created 5 years, 4 months ago
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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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