niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1 | /* |
braveyao@webrtc.org | d713143 | 2012-03-29 10:39:44 +0000 | [diff] [blame] | 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 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 | |
Fredrik Solenberg | a8b7c7f | 2018-01-17 11:18:31 +0100 | [diff] [blame] | 11 | #ifndef AUDIO_REMIX_RESAMPLE_H_ |
| 12 | #define AUDIO_REMIX_RESAMPLE_H_ |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 13 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 14 | #include "common_audio/resampler/include/push_resampler.h" |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 15 | |
andrew@webrtc.org | 40ee3d0 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 16 | namespace webrtc { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 17 | |
andrew@webrtc.org | 40ee3d0 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 18 | class AudioFrame; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 19 | |
andrew@webrtc.org | 40ee3d0 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 20 | namespace voe { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 21 | |
Alejandro Luebs | cdfe20b | 2015-09-23 12:49:12 -0700 | [diff] [blame] | 22 | // Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame| |
| 23 | // to have its sample rate and channels members set to the desired values. |
| 24 | // Updates the |samples_per_channel_| member accordingly. |
| 25 | // |
| 26 | // This version has an AudioFrame |src_frame| as input and sets the output |
| 27 | // |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the |
| 28 | // input ones. |
andrew@webrtc.org | 40ee3d0 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 29 | void RemixAndResample(const AudioFrame& src_frame, |
andrew@webrtc.org | f5a33f1 | 2014-04-19 00:32:07 +0000 | [diff] [blame] | 30 | PushResampler<int16_t>* resampler, |
andrew@webrtc.org | 40ee3d0 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 31 | AudioFrame* dst_frame); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 32 | |
Alejandro Luebs | cdfe20b | 2015-09-23 12:49:12 -0700 | [diff] [blame] | 33 | // This version has a pointer to the samples |src_data| as input and receives |
| 34 | // |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as |
| 35 | // parameters. |
| 36 | void RemixAndResample(const int16_t* src_data, |
| 37 | size_t samples_per_channel, |
Peter Kasting | 6955870 | 2016-01-12 16:26:35 -0800 | [diff] [blame] | 38 | size_t num_channels, |
Alejandro Luebs | cdfe20b | 2015-09-23 12:49:12 -0700 | [diff] [blame] | 39 | int sample_rate_hz, |
| 40 | PushResampler<int16_t>* resampler, |
| 41 | AudioFrame* dst_frame); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 42 | |
pbos@webrtc.org | d900e8b | 2013-07-03 15:12:26 +0000 | [diff] [blame] | 43 | } // namespace voe |
pbos@webrtc.org | d900e8b | 2013-07-03 15:12:26 +0000 | [diff] [blame] | 44 | } // namespace webrtc |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 45 | |
Fredrik Solenberg | a8b7c7f | 2018-01-17 11:18:31 +0100 | [diff] [blame] | 46 | #endif // AUDIO_REMIX_RESAMPLE_H_ |