Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(346)

Side by Side Diff: webrtc/audio/conversion.h

Issue 1414743004: Implement AudioSendStream::GetStats(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: workaround for android build issue Created 5 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/audio/audio_send_stream_unittest.cc ('k') | webrtc/audio_send_stream.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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_AUDIO_CONVERSION_H_ 11 #ifndef WEBRTC_AUDIO_CONVERSION_H_
12 #define WEBRTC_AUDIO_CONVERSION_H_ 12 #define WEBRTC_AUDIO_CONVERSION_H_
13 13
14 namespace webrtc { 14 namespace webrtc {
15 15
16 // Convert fixed point number with 8 bit fractional part, to floating point.
17 inline float Q8ToFloat(uint32_t v) {
18 return static_cast<float>(v) / (1 << 8);
19 }
20
16 // Convert fixed point number with 14 bit fractional part, to floating point. 21 // Convert fixed point number with 14 bit fractional part, to floating point.
17 inline float Q14ToFloat(uint16_t v) { 22 inline float Q14ToFloat(uint32_t v) {
18 return static_cast<float>(v) / (1 << 14); 23 return static_cast<float>(v) / (1 << 14);
19 } 24 }
20 } // namespace webrtc 25 } // namespace webrtc
21 26
22 #endif // WEBRTC_AUDIO_CONVERSION_H_ 27 #endif // WEBRTC_AUDIO_CONVERSION_H_
OLDNEW
« no previous file with comments | « webrtc/audio/audio_send_stream_unittest.cc ('k') | webrtc/audio_send_stream.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698