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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
henrike@webrtc.orgf2aafe42014-04-29 17:54:17 +000013#include <assert.h>
turaj@webrtc.org7959e162013-09-12 18:30:26 +000014#include <string.h>
15
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020016#include "common_audio/resampler/include/resampler.h"
17#include "rtc_base/logging.h"
turaj@webrtc.org7959e162013-09-12 18:30:26 +000018
19namespace webrtc {
turaj@webrtc.org6d5d2482013-10-06 04:47:28 +000020namespace acm2 {
21
Yves Gerey665174f2018-06-19 15:03:05 +020022ACMResampler::ACMResampler() {}
turaj@webrtc.org7959e162013-09-12 18:30:26 +000023
Yves Gerey665174f2018-06-19 15:03:05 +020024ACMResampler::~ACMResampler() {}
turaj@webrtc.org7959e162013-09-12 18:30:26 +000025
26int ACMResampler::Resample10Msec(const int16_t* in_audio,
27 int in_freq_hz,
28 int out_freq_hz,
Peter Kasting69558702016-01-12 16:26:35 -080029 size_t num_audio_channels,
Peter Kastingdce40cf2015-08-24 14:52:23 -070030 size_t out_capacity_samples,
turaj@webrtc.org7959e162013-09-12 18:30:26 +000031 int16_t* out_audio) {
Peter Kasting69558702016-01-12 16:26:35 -080032 size_t in_length = in_freq_hz * num_audio_channels / 100;
turaj@webrtc.org7959e162013-09-12 18:30:26 +000033 if (in_freq_hz == out_freq_hz) {
henrik.lundin@webrtc.org439a4c42014-04-24 19:05:33 +000034 if (out_capacity_samples < in_length) {
35 assert(false);
36 return -1;
37 }
andrew@webrtc.org40ee3d02014-04-03 21:56:01 +000038 memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
Peter Kastingdce40cf2015-08-24 14:52:23 -070039 return static_cast<int>(in_length / num_audio_channels);
turaj@webrtc.org7959e162013-09-12 18:30:26 +000040 }
41
andrew@webrtc.org40ee3d02014-04-03 21:56:01 +000042 if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
43 num_audio_channels) != 0) {
Mirko Bonadei675513b2017-11-09 11:09:25 +010044 RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
45 << out_freq_hz << ", " << num_audio_channels
46 << ") failed.";
turaj@webrtc.org7959e162013-09-12 18:30:26 +000047 return -1;
48 }
49
pkasting25702cb2016-01-08 13:50:27 -080050 int out_length =
henrik.lundin@webrtc.org439a4c42014-04-24 19:05:33 +000051 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
andrew@webrtc.org40ee3d02014-04-03 21:56:01 +000052 if (out_length == -1) {
Mirko Bonadei675513b2017-11-09 11:09:25 +010053 RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
54 << out_audio << ", " << out_capacity_samples
55 << ") failed.";
turaj@webrtc.org7959e162013-09-12 18:30:26 +000056 return -1;
57 }
58
Peter Kasting69558702016-01-12 16:26:35 -080059 return static_cast<int>(out_length / num_audio_channels);
turaj@webrtc.org7959e162013-09-12 18:30:26 +000060}
61
turaj@webrtc.org6d5d2482013-10-06 04:47:28 +000062} // namespace acm2
turaj@webrtc.org7959e162013-09-12 18:30:26 +000063} // namespace webrtc