mirror of https://github.com/FFmpeg/FFmpeg.git
James Almer
7 years ago
parent
24a8603a8e
commit
8f483108b5
5 changed files with 0 additions and 860 deletions
@ -1,439 +0,0 @@ |
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/*
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* samplerate conversion for both audio and video |
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* Copyright (c) 2000 Fabrice Bellard |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/**
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* @file |
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* samplerate conversion for both audio and video |
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*/ |
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#include <string.h> |
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#include "avcodec.h" |
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#include "audioconvert.h" |
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#include "libavutil/opt.h" |
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#include "libavutil/mem.h" |
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#include "libavutil/samplefmt.h" |
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#if FF_API_AVCODEC_RESAMPLE |
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FF_DISABLE_DEPRECATION_WARNINGS |
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#define MAX_CHANNELS 8 |
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struct AVResampleContext; |
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static const char *context_to_name(void *ptr) |
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{ |
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return "audioresample"; |
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} |
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static const AVOption options[] = {{NULL}}; |
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static const AVClass audioresample_context_class = { |
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"ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT |
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}; |
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struct ReSampleContext { |
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struct AVResampleContext *resample_context; |
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short *temp[MAX_CHANNELS]; |
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int temp_len; |
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float ratio; |
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/* channel convert */ |
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int input_channels, output_channels, filter_channels; |
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AVAudioConvert *convert_ctx[2]; |
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enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
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unsigned sample_size[2]; ///< size of one sample in sample_fmt
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short *buffer[2]; ///< buffers used for conversion to S16
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unsigned buffer_size[2]; ///< sizes of allocated buffers
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}; |
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/* n1: number of samples */ |
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static void stereo_to_mono(short *output, short *input, int n1) |
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{ |
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short *p, *q; |
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int n = n1; |
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p = input; |
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q = output; |
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while (n >= 4) { |
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q[0] = (p[0] + p[1]) >> 1; |
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q[1] = (p[2] + p[3]) >> 1; |
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q[2] = (p[4] + p[5]) >> 1; |
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q[3] = (p[6] + p[7]) >> 1; |
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q += 4; |
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p += 8; |
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n -= 4; |
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} |
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while (n > 0) { |
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q[0] = (p[0] + p[1]) >> 1; |
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q++; |
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p += 2; |
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n--; |
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} |
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} |
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/* n1: number of samples */ |
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static void mono_to_stereo(short *output, short *input, int n1) |
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{ |
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short *p, *q; |
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int n = n1; |
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int v; |
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p = input; |
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q = output; |
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while (n >= 4) { |
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v = p[0]; q[0] = v; q[1] = v; |
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v = p[1]; q[2] = v; q[3] = v; |
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v = p[2]; q[4] = v; q[5] = v; |
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v = p[3]; q[6] = v; q[7] = v; |
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q += 8; |
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p += 4; |
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n -= 4; |
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} |
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while (n > 0) { |
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v = p[0]; q[0] = v; q[1] = v; |
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q += 2; |
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p += 1; |
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n--; |
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} |
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} |
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/*
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5.1 to stereo input: [fl, fr, c, lfe, rl, rr] |
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- Left = front_left + rear_gain * rear_left + center_gain * center |
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- Right = front_right + rear_gain * rear_right + center_gain * center |
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Where rear_gain is usually around 0.5-1.0 and |
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center_gain is almost always 0.7 (-3 dB) |
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*/ |
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static void surround_to_stereo(short **output, short *input, int channels, int samples) |
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{ |
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int i; |
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short l, r; |
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for (i = 0; i < samples; i++) { |
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int fl,fr,c,rl,rr; |
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fl = input[0]; |
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fr = input[1]; |
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c = input[2]; |
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// lfe = input[3];
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rl = input[4]; |
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rr = input[5]; |
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l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c)); |
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r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c)); |
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/* output l & r. */ |
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*output[0]++ = l; |
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*output[1]++ = r; |
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/* increment input. */ |
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input += channels; |
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} |
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} |
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static void deinterleave(short **output, short *input, int channels, int samples) |
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{ |
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int i, j; |
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for (i = 0; i < samples; i++) { |
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for (j = 0; j < channels; j++) { |
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*output[j]++ = *input++; |
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} |
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} |
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} |
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static void interleave(short *output, short **input, int channels, int samples) |
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{ |
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int i, j; |
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for (i = 0; i < samples; i++) { |
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for (j = 0; j < channels; j++) { |
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*output++ = *input[j]++; |
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} |
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} |
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} |
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static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) |
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{ |
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int i; |
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short l, r; |
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for (i = 0; i < n; i++) { |
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l = *input1++; |
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r = *input2++; |
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*output++ = l; /* left */ |
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*output++ = (l / 2) + (r / 2); /* center */ |
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*output++ = r; /* right */ |
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*output++ = 0; /* left surround */ |
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*output++ = 0; /* right surroud */ |
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*output++ = 0; /* low freq */ |
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} |
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} |
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#define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \ |
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ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0 |
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static const uint8_t supported_resampling[MAX_CHANNELS] = { |
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// output ch: 1 2 3 4 5 6 7 8
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SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
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SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
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SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
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SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
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SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
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SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
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SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
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SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
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}; |
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ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, |
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int output_rate, int input_rate, |
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enum AVSampleFormat sample_fmt_out, |
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enum AVSampleFormat sample_fmt_in, |
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int filter_length, int log2_phase_count, |
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int linear, double cutoff) |
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{ |
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ReSampleContext *s; |
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if (input_channels > MAX_CHANNELS) { |
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av_log(NULL, AV_LOG_ERROR, |
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"Resampling with input channels greater than %d is unsupported.\n", |
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MAX_CHANNELS); |
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return NULL; |
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} |
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if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) { |
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int i; |
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av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed " |
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"output channels for %d input channel%s", input_channels, |
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input_channels > 1 ? "s:" : ":"); |
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for (i = 0; i < MAX_CHANNELS; i++) |
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if (supported_resampling[input_channels-1] & (1<<i)) |
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av_log(NULL, AV_LOG_ERROR, " %d", i + 1); |
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av_log(NULL, AV_LOG_ERROR, "\n"); |
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return NULL; |
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} |
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s = av_mallocz(sizeof(ReSampleContext)); |
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if (!s) { |
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av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n"); |
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return NULL; |
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} |
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s->ratio = (float)output_rate / (float)input_rate; |
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s->input_channels = input_channels; |
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s->output_channels = output_channels; |
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s->filter_channels = s->input_channels; |
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if (s->output_channels < s->filter_channels) |
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s->filter_channels = s->output_channels; |
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s->sample_fmt[0] = sample_fmt_in; |
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s->sample_fmt[1] = sample_fmt_out; |
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s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]); |
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s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]); |
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if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) { |
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if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1, |
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s->sample_fmt[0], 1, NULL, 0))) { |
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av_log(s, AV_LOG_ERROR, |
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"Cannot convert %s sample format to s16 sample format\n", |
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av_get_sample_fmt_name(s->sample_fmt[0])); |
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av_free(s); |
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return NULL; |
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} |
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} |
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if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { |
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if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1, |
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AV_SAMPLE_FMT_S16, 1, NULL, 0))) { |
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av_log(s, AV_LOG_ERROR, |
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"Cannot convert s16 sample format to %s sample format\n", |
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av_get_sample_fmt_name(s->sample_fmt[1])); |
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av_audio_convert_free(s->convert_ctx[0]); |
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av_free(s); |
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return NULL; |
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} |
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} |
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s->resample_context = av_resample_init(output_rate, input_rate, |
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filter_length, log2_phase_count, |
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linear, cutoff); |
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*(const AVClass**)s->resample_context = &audioresample_context_class; |
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return s; |
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} |
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/* resample audio. 'nb_samples' is the number of input samples */ |
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/* XXX: optimize it ! */ |
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int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) |
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{ |
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int i, nb_samples1; |
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short *bufin[MAX_CHANNELS]; |
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short *bufout[MAX_CHANNELS]; |
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short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS]; |
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short *output_bak = NULL; |
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int lenout; |
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if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) { |
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int istride[1] = { s->sample_size[0] }; |
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int ostride[1] = { 2 }; |
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const void *ibuf[1] = { input }; |
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void *obuf[1]; |
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unsigned input_size = nb_samples * s->input_channels * 2; |
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if (!s->buffer_size[0] || s->buffer_size[0] < input_size) { |
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av_free(s->buffer[0]); |
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s->buffer_size[0] = input_size; |
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s->buffer[0] = av_malloc(s->buffer_size[0]); |
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if (!s->buffer[0]) { |
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av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); |
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return 0; |
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} |
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} |
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obuf[0] = s->buffer[0]; |
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if (av_audio_convert(s->convert_ctx[0], obuf, ostride, |
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ibuf, istride, nb_samples * s->input_channels) < 0) { |
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av_log(s->resample_context, AV_LOG_ERROR, |
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"Audio sample format conversion failed\n"); |
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return 0; |
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} |
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input = s->buffer[0]; |
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} |
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lenout= 2*s->output_channels*nb_samples * s->ratio + 16; |
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if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { |
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int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) * |
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s->output_channels; |
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output_bak = output; |
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if (!s->buffer_size[1] || s->buffer_size[1] < out_size) { |
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av_free(s->buffer[1]); |
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s->buffer_size[1] = out_size; |
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s->buffer[1] = av_malloc(s->buffer_size[1]); |
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if (!s->buffer[1]) { |
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av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); |
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return 0; |
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} |
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} |
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output = s->buffer[1]; |
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} |
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/* XXX: move those malloc to resample init code */ |
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for (i = 0; i < s->filter_channels; i++) { |
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bufin[i] = av_malloc_array((nb_samples + s->temp_len), sizeof(short)); |
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bufout[i] = av_malloc_array(lenout, sizeof(short)); |
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if (!bufin[i] || !bufout[i]) { |
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av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); |
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nb_samples1 = 0; |
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goto fail; |
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} |
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memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); |
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buftmp2[i] = bufin[i] + s->temp_len; |
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} |
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if (s->input_channels == 2 && s->output_channels == 1) { |
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buftmp3[0] = output; |
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stereo_to_mono(buftmp2[0], input, nb_samples); |
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} else if (s->output_channels >= 2 && s->input_channels == 1) { |
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buftmp3[0] = bufout[0]; |
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memcpy(buftmp2[0], input, nb_samples * sizeof(short)); |
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} else if (s->input_channels == 6 && s->output_channels ==2) { |
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buftmp3[0] = bufout[0]; |
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buftmp3[1] = bufout[1]; |
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surround_to_stereo(buftmp2, input, s->input_channels, nb_samples); |
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} else if (s->output_channels >= s->input_channels && s->input_channels >= 2) { |
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for (i = 0; i < s->input_channels; i++) { |
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buftmp3[i] = bufout[i]; |
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} |
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deinterleave(buftmp2, input, s->input_channels, nb_samples); |
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} else { |
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buftmp3[0] = output; |
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memcpy(buftmp2[0], input, nb_samples * sizeof(short)); |
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} |
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nb_samples += s->temp_len; |
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/* resample each channel */ |
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nb_samples1 = 0; /* avoid warning */ |
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for (i = 0; i < s->filter_channels; i++) { |
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int consumed; |
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int is_last = i + 1 == s->filter_channels; |
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nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], |
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&consumed, nb_samples, lenout, is_last); |
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s->temp_len = nb_samples - consumed; |
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s->temp[i] = av_realloc_array(s->temp[i], s->temp_len, sizeof(short)); |
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memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short)); |
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} |
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if (s->output_channels == 2 && s->input_channels == 1) { |
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mono_to_stereo(output, buftmp3[0], nb_samples1); |
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} else if (s->output_channels == 6 && s->input_channels == 2) { |
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ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); |
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} else if ((s->output_channels == s->input_channels && s->input_channels >= 2) || |
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(s->output_channels == 2 && s->input_channels == 6)) { |
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interleave(output, buftmp3, s->output_channels, nb_samples1); |
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} |
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if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { |
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int istride[1] = { 2 }; |
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int ostride[1] = { s->sample_size[1] }; |
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const void *ibuf[1] = { output }; |
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void *obuf[1] = { output_bak }; |
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if (av_audio_convert(s->convert_ctx[1], obuf, ostride, |
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ibuf, istride, nb_samples1 * s->output_channels) < 0) { |
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av_log(s->resample_context, AV_LOG_ERROR, |
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"Audio sample format conversion failed\n"); |
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return 0; |
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} |
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} |
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fail: |
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for (i = 0; i < s->filter_channels; i++) { |
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av_free(bufin[i]); |
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av_free(bufout[i]); |
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} |
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return nb_samples1; |
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} |
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void audio_resample_close(ReSampleContext *s) |
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{ |
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int i; |
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av_resample_close(s->resample_context); |
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for (i = 0; i < s->filter_channels; i++) |
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av_freep(&s->temp[i]); |
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av_freep(&s->buffer[0]); |
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av_freep(&s->buffer[1]); |
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av_audio_convert_free(s->convert_ctx[0]); |
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av_audio_convert_free(s->convert_ctx[1]); |
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av_free(s); |
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} |
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FF_ENABLE_DEPRECATION_WARNINGS |
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#endif |
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@ -1,319 +0,0 @@ |
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/*
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* audio resampling |
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/**
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* @file |
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* audio resampling |
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* @author Michael Niedermayer <michaelni@gmx.at> |
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*/ |
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#include "libavutil/avassert.h" |
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#include "avcodec.h" |
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#include "libavutil/common.h" |
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#if FF_API_AVCODEC_RESAMPLE |
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#ifndef CONFIG_RESAMPLE_HP |
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#define FILTER_SHIFT 15 |
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typedef int16_t FELEM; |
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typedef int32_t FELEM2; |
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typedef int64_t FELEML; |
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#define FELEM_MAX INT16_MAX |
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#define FELEM_MIN INT16_MIN |
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#define WINDOW_TYPE 9 |
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#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE) |
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#define FILTER_SHIFT 30 |
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#define FELEM int32_t |
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#define FELEM2 int64_t |
||||
#define FELEML int64_t |
||||
#define FELEM_MAX INT32_MAX |
||||
#define FELEM_MIN INT32_MIN |
||||
#define WINDOW_TYPE 12 |
||||
#else |
||||
#define FILTER_SHIFT 0 |
||||
|
||||
typedef double FELEM; |
||||
typedef double FELEM2; |
||||
typedef double FELEML; |
||||
#define WINDOW_TYPE 24 |
||||
#endif |
||||
|
||||
|
||||
typedef struct AVResampleContext{ |
||||
const AVClass *av_class; |
||||
FELEM *filter_bank; |
||||
int filter_length; |
||||
int ideal_dst_incr; |
||||
int dst_incr; |
||||
int index; |
||||
int frac; |
||||
int src_incr; |
||||
int compensation_distance; |
||||
int phase_shift; |
||||
int phase_mask; |
||||
int linear; |
||||
}AVResampleContext; |
||||
|
||||
/**
|
||||
* 0th order modified bessel function of the first kind. |
||||
*/ |
||||
static double bessel(double x){ |
||||
double v=1; |
||||
double lastv=0; |
||||
double t=1; |
||||
int i; |
||||
|
||||
x= x*x/4; |
||||
for(i=1; v != lastv; i++){ |
||||
lastv=v; |
||||
t *= x/(i*i); |
||||
v += t; |
||||
} |
||||
return v; |
||||
} |
||||
|
||||
/**
|
||||
* Build a polyphase filterbank. |
||||
* @param factor resampling factor |
||||
* @param scale wanted sum of coefficients for each filter |
||||
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16 |
||||
* @return 0 on success, negative on error |
||||
*/ |
||||
static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ |
||||
int ph, i; |
||||
double x, y, w; |
||||
double *tab = av_malloc_array(tap_count, sizeof(*tab)); |
||||
const int center= (tap_count-1)/2; |
||||
|
||||
if (!tab) |
||||
return AVERROR(ENOMEM); |
||||
|
||||
/* if upsampling, only need to interpolate, no filter */ |
||||
if (factor > 1.0) |
||||
factor = 1.0; |
||||
|
||||
for(ph=0;ph<phase_count;ph++) { |
||||
double norm = 0; |
||||
for(i=0;i<tap_count;i++) { |
||||
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; |
||||
if (x == 0) y = 1.0; |
||||
else y = sin(x) / x; |
||||
switch(type){ |
||||
case 0:{ |
||||
const float d= -0.5; //first order derivative = -0.5
|
||||
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); |
||||
if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); |
||||
else y= d*(-4 + 8*x - 5*x*x + x*x*x); |
||||
break;} |
||||
case 1: |
||||
w = 2.0*x / (factor*tap_count) + M_PI; |
||||
y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); |
||||
break; |
||||
default: |
||||
w = 2.0*x / (factor*tap_count*M_PI); |
||||
y *= bessel(type*sqrt(FFMAX(1-w*w, 0))); |
||||
break; |
||||
} |
||||
|
||||
tab[i] = y; |
||||
norm += y; |
||||
} |
||||
|
||||
/* normalize so that an uniform color remains the same */ |
||||
for(i=0;i<tap_count;i++) { |
||||
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE |
||||
filter[ph * tap_count + i] = tab[i] / norm; |
||||
#else |
||||
filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX); |
||||
#endif |
||||
} |
||||
} |
||||
#if 0 |
||||
{ |
||||
#define LEN 1024 |
||||
int j,k; |
||||
double sine[LEN + tap_count]; |
||||
double filtered[LEN]; |
||||
double maxff=-2, minff=2, maxsf=-2, minsf=2; |
||||
for(i=0; i<LEN; i++){ |
||||
double ss=0, sf=0, ff=0; |
||||
for(j=0; j<LEN+tap_count; j++) |
||||
sine[j]= cos(i*j*M_PI/LEN); |
||||
for(j=0; j<LEN; j++){ |
||||
double sum=0; |
||||
ph=0; |
||||
for(k=0; k<tap_count; k++) |
||||
sum += filter[ph * tap_count + k] * sine[k+j]; |
||||
filtered[j]= sum / (1<<FILTER_SHIFT); |
||||
ss+= sine[j + center] * sine[j + center]; |
||||
ff+= filtered[j] * filtered[j]; |
||||
sf+= sine[j + center] * filtered[j]; |
||||
} |
||||
ss= sqrt(2*ss/LEN); |
||||
ff= sqrt(2*ff/LEN); |
||||
sf= 2*sf/LEN; |
||||
maxff= FFMAX(maxff, ff); |
||||
minff= FFMIN(minff, ff); |
||||
maxsf= FFMAX(maxsf, sf); |
||||
minsf= FFMIN(minsf, sf); |
||||
if(i%11==0){ |
||||
av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf); |
||||
minff=minsf= 2; |
||||
maxff=maxsf= -2; |
||||
} |
||||
} |
||||
} |
||||
#endif |
||||
|
||||
av_free(tab); |
||||
return 0; |
||||
} |
||||
|
||||
AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ |
||||
AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); |
||||
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); |
||||
int phase_count= 1<<phase_shift; |
||||
|
||||
if (!c) |
||||
return NULL; |
||||
|
||||
c->phase_shift= phase_shift; |
||||
c->phase_mask= phase_count-1; |
||||
c->linear= linear; |
||||
|
||||
c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); |
||||
c->filter_bank= av_mallocz_array(c->filter_length, (phase_count+1)*sizeof(FELEM)); |
||||
if (!c->filter_bank) |
||||
goto error; |
||||
if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE)) |
||||
goto error; |
||||
memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); |
||||
c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1]; |
||||
|
||||
if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2)) |
||||
goto error; |
||||
c->ideal_dst_incr= c->dst_incr; |
||||
|
||||
c->index= -phase_count*((c->filter_length-1)/2); |
||||
|
||||
return c; |
||||
error: |
||||
av_free(c->filter_bank); |
||||
av_free(c); |
||||
return NULL; |
||||
} |
||||
|
||||
void av_resample_close(AVResampleContext *c){ |
||||
av_freep(&c->filter_bank); |
||||
av_freep(&c); |
||||
} |
||||
|
||||
void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ |
||||
// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
|
||||
c->compensation_distance= compensation_distance; |
||||
c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; |
||||
} |
||||
|
||||
int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ |
||||
int dst_index, i; |
||||
int index= c->index; |
||||
int frac= c->frac; |
||||
int dst_incr_frac= c->dst_incr % c->src_incr; |
||||
int dst_incr= c->dst_incr / c->src_incr; |
||||
int compensation_distance= c->compensation_distance; |
||||
|
||||
if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ |
||||
int64_t index2= ((int64_t)index)<<32; |
||||
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; |
||||
dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); |
||||
|
||||
for(dst_index=0; dst_index < dst_size; dst_index++){ |
||||
dst[dst_index] = src[index2>>32]; |
||||
index2 += incr; |
||||
} |
||||
index += dst_index * dst_incr; |
||||
index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr; |
||||
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr; |
||||
}else{ |
||||
for(dst_index=0; dst_index < dst_size; dst_index++){ |
||||
FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); |
||||
int sample_index= index >> c->phase_shift; |
||||
FELEM2 val=0; |
||||
|
||||
if(sample_index < 0){ |
||||
for(i=0; i<c->filter_length; i++) |
||||
val += src[FFABS(sample_index + i) % src_size] * filter[i]; |
||||
}else if(sample_index + c->filter_length > src_size){ |
||||
break; |
||||
}else if(c->linear){ |
||||
FELEM2 v2=0; |
||||
for(i=0; i<c->filter_length; i++){ |
||||
val += src[sample_index + i] * (FELEM2)filter[i]; |
||||
v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; |
||||
} |
||||
val+=(v2-val)*(FELEML)frac / c->src_incr; |
||||
}else{ |
||||
for(i=0; i<c->filter_length; i++){ |
||||
val += src[sample_index + i] * (FELEM2)filter[i]; |
||||
} |
||||
} |
||||
|
||||
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE |
||||
dst[dst_index] = av_clip_int16(lrintf(val)); |
||||
#else |
||||
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; |
||||
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; |
||||
#endif |
||||
|
||||
frac += dst_incr_frac; |
||||
index += dst_incr; |
||||
if(frac >= c->src_incr){ |
||||
frac -= c->src_incr; |
||||
index++; |
||||
} |
||||
|
||||
if(dst_index + 1 == compensation_distance){ |
||||
compensation_distance= 0; |
||||
dst_incr_frac= c->ideal_dst_incr % c->src_incr; |
||||
dst_incr= c->ideal_dst_incr / c->src_incr; |
||||
} |
||||
} |
||||
} |
||||
*consumed= FFMAX(index, 0) >> c->phase_shift; |
||||
if(index>=0) index &= c->phase_mask; |
||||
|
||||
if(compensation_distance){ |
||||
compensation_distance -= dst_index; |
||||
av_assert2(compensation_distance > 0); |
||||
} |
||||
if(update_ctx){ |
||||
c->frac= frac; |
||||
c->index= index; |
||||
c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; |
||||
c->compensation_distance= compensation_distance; |
||||
} |
||||
|
||||
return dst_index; |
||||
} |
||||
|
||||
#endif |
||||
Loading…
Reference in new issue