dce40cf804
use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
69 lines
2.1 KiB
C++
69 lines
2.1 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
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#include <assert.h>
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#include <string.h>
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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namespace webrtc {
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namespace acm2 {
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ACMResampler::ACMResampler() {
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}
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ACMResampler::~ACMResampler() {
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}
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int ACMResampler::Resample10Msec(const int16_t* in_audio,
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int in_freq_hz,
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int out_freq_hz,
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int num_audio_channels,
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size_t out_capacity_samples,
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int16_t* out_audio) {
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size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
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int out_length = out_freq_hz * num_audio_channels / 100;
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if (in_freq_hz == out_freq_hz) {
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if (out_capacity_samples < in_length) {
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assert(false);
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return -1;
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}
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memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
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return static_cast<int>(in_length / num_audio_channels);
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}
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if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
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num_audio_channels) != 0) {
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LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
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num_audio_channels);
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return -1;
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}
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out_length =
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resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
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if (out_length == -1) {
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LOG_FERR4(LS_ERROR,
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Resample,
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in_audio,
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in_length,
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out_audio,
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out_capacity_samples);
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return -1;
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}
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return out_length / num_audio_channels;
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}
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} // namespace acm2
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} // namespace webrtc
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