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turaj@webrtc.org7959e162013-09-12 18:30:26 +00001/*
2 * Copyright (c) 2012 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
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#include "modules/audio_coding/acm2/acm_resampler.h"
turaj@webrtc.org7959e162013-09-12 18:30:26 +000012
13#include <string.h>
14
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020015#include "rtc_base/logging.h"
turaj@webrtc.org7959e162013-09-12 18:30:26 +000016
17namespace webrtc {
turaj@webrtc.org6d5d2482013-10-06 04:47:28 +000018namespace acm2 {
19
Yves Gerey665174f2018-06-19 15:03:05 +020020ACMResampler::ACMResampler() {}
turaj@webrtc.org7959e162013-09-12 18:30:26 +000021
Yves Gerey665174f2018-06-19 15:03:05 +020022ACMResampler::~ACMResampler() {}
turaj@webrtc.org7959e162013-09-12 18:30:26 +000023
24int ACMResampler::Resample10Msec(const int16_t* in_audio,
25 int in_freq_hz,
26 int out_freq_hz,
Peter Kasting69558702016-01-12 16:26:35 -080027 size_t num_audio_channels,
Peter Kastingdce40cf2015-08-24 14:52:23 -070028 size_t out_capacity_samples,
turaj@webrtc.org7959e162013-09-12 18:30:26 +000029 int16_t* out_audio) {
Peter Kasting69558702016-01-12 16:26:35 -080030 size_t in_length = in_freq_hz * num_audio_channels / 100;
turaj@webrtc.org7959e162013-09-12 18:30:26 +000031 if (in_freq_hz == out_freq_hz) {
henrik.lundin@webrtc.org439a4c42014-04-24 19:05:33 +000032 if (out_capacity_samples < in_length) {
Artem Titovd3251962021-11-15 16:57:07 +010033 RTC_DCHECK_NOTREACHED();
henrik.lundin@webrtc.org439a4c42014-04-24 19:05:33 +000034 return -1;
35 }
andrew@webrtc.org40ee3d02014-04-03 21:56:01 +000036 memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
Peter Kastingdce40cf2015-08-24 14:52:23 -070037 return static_cast<int>(in_length / num_audio_channels);
turaj@webrtc.org7959e162013-09-12 18:30:26 +000038 }
39
andrew@webrtc.org40ee3d02014-04-03 21:56:01 +000040 if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
41 num_audio_channels) != 0) {
Mirko Bonadei675513b2017-11-09 11:09:25 +010042 RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
43 << out_freq_hz << ", " << num_audio_channels
44 << ") failed.";
turaj@webrtc.org7959e162013-09-12 18:30:26 +000045 return -1;
46 }
47
pkasting25702cb2016-01-08 13:50:27 -080048 int out_length =
henrik.lundin@webrtc.org439a4c42014-04-24 19:05:33 +000049 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
andrew@webrtc.org40ee3d02014-04-03 21:56:01 +000050 if (out_length == -1) {
Mirko Bonadei675513b2017-11-09 11:09:25 +010051 RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
52 << out_audio << ", " << out_capacity_samples
53 << ") failed.";
turaj@webrtc.org7959e162013-09-12 18:30:26 +000054 return -1;
55 }
56
Peter Kasting69558702016-01-12 16:26:35 -080057 return static_cast<int>(out_length / num_audio_channels);
turaj@webrtc.org7959e162013-09-12 18:30:26 +000058}
59
turaj@webrtc.org6d5d2482013-10-06 04:47:28 +000060} // namespace acm2
turaj@webrtc.org7959e162013-09-12 18:30:26 +000061} // namespace webrtc