3e6db2321c
This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
69 lines
2.0 KiB
C++
69 lines
2.0 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/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/include/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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