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675513b96a
This CL has been generated with the following script: for m in PLOG \ LOG_TAG \ LOG_GLEM \ LOG_GLE_EX \ LOG_GLE \ LAST_SYSTEM_ERROR \ LOG_ERRNO_EX \ LOG_ERRNO \ LOG_ERR_EX \ LOG_ERR \ LOG_V \ LOG_F \ LOG_T_F \ LOG_E \ LOG_T \ LOG_CHECK_LEVEL_V \ LOG_CHECK_LEVEL \ LOG do git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g" done git checkout rtc_base/logging.h git cl format Bug: webrtc:8452 Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600 Reviewed-on: https://webrtc-review.googlesource.com/21325 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20617}
65 lines
2.1 KiB
C++
65 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 "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 "common_audio/resampler/include/resampler.h"
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#include "rtc_base/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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size_t 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 = in_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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RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
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<< out_freq_hz << ", " << num_audio_channels
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<< ") failed.";
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return -1;
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}
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int 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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RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
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<< out_audio << ", " << out_capacity_samples
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<< ") failed.";
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return -1;
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}
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return static_cast<int>(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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