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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
66 lines
2.4 KiB
C
66 lines
2.4 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 "common_audio/signal_processing/include/signal_processing_library.h"
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#include "rtc_base/checks.h"
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#include "rtc_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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// Negative overflow is permitted here, because this is
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// auto-regressive filters, and the state for each batch run is
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// stored in the "negative" positions of the output vector.
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rtc_MsanCheckInitialized(&data_in[(ptrdiff_t) i - (ptrdiff_t) j],
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sizeof(data_in[0]), 1);
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// out_s32 is in Q12 domain.
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out_s32 += coefficients[j] * data_in[(ptrdiff_t) i - (ptrdiff_t) j];
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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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