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https://github.com/danog/libtgvoip.git
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333c4a1101
Added simple audio resampler Replaced prebuilt static libs with their sources & added that to all project files (closes #5)
61 lines
2.2 KiB
C
61 lines
2.2 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/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/sanitizer.h"
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// TODO(Bjornv): Change the function parameter order to WebRTC code style.
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// C version of WebRtcSpl_DownsampleFast() for generic platforms.
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int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
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size_t data_in_length,
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int16_t* data_out,
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size_t data_out_length,
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const int16_t* __restrict coefficients,
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size_t coefficients_length,
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int factor,
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size_t delay) {
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int16_t* const original_data_out = data_out;
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size_t i = 0;
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size_t j = 0;
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int32_t out_s32 = 0;
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size_t endpos = delay + factor * (data_out_length - 1) + 1;
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// Return error if any of the running conditions doesn't meet.
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if (data_out_length == 0 || coefficients_length == 0
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|| data_in_length < endpos) {
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return -1;
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}
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rtc_MsanCheckInitialized(coefficients, sizeof(coefficients[0]),
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coefficients_length);
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for (i = delay; i < endpos; i += factor) {
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out_s32 = 2048; // Round value, 0.5 in Q12.
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for (j = 0; j < coefficients_length; j++) {
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rtc_MsanCheckInitialized(&data_in[i - j], sizeof(data_in[0]), 1);
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out_s32 += coefficients[j] * data_in[i - j]; // Q12.
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}
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out_s32 >>= 12; // Q0.
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// Saturate and store the output.
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*data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
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}
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RTC_DCHECK_EQ(original_data_out + data_out_length, data_out);
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rtc_MsanCheckInitialized(original_data_out, sizeof(original_data_out[0]),
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data_out_length);
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return 0;
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}
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