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| Rev | Author | Line No. | Line |
|---|---|---|---|
| 6147 | serge | 1 | /* |
| 2 | * audio resampling with soxr |
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| 3 | * Copyright (c) 2012 Rob Sykes |
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| 4 | * |
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| 5 | * This file is part of FFmpeg. |
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| 6 | * |
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| 7 | * FFmpeg is free software; you can redistribute it and/or |
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| 8 | * modify it under the terms of the GNU Lesser General Public |
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| 9 | * License as published by the Free Software Foundation; either |
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| 10 | * version 2.1 of the License, or (at your option) any later version. |
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| 11 | * |
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| 12 | * FFmpeg is distributed in the hope that it will be useful, |
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| 13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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| 14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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| 15 | * Lesser General Public License for more details. |
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| 16 | * |
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| 17 | * You should have received a copy of the GNU Lesser General Public |
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| 18 | * License along with FFmpeg; if not, write to the Free Software |
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| 19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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| 20 | */ |
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| 21 | |||
| 22 | /** |
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| 23 | * @file |
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| 24 | * audio resampling with soxr |
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| 25 | */ |
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| 26 | |||
| 27 | #include "libavutil/log.h" |
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| 28 | #include "swresample_internal.h" |
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| 29 | |||
| 30 | #include |
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| 31 | |||
| 32 | static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, |
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| 33 | double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby){ |
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| 34 | soxr_error_t error; |
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| 35 | |||
| 36 | soxr_datatype_t type = |
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| 37 | format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S : |
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| 38 | format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I : |
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| 39 | format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S : |
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| 40 | format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I : |
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| 41 | format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S : |
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| 42 | format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I : |
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| 43 | format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S : |
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| 44 | format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1; |
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| 45 | |||
| 46 | soxr_io_spec_t io_spec = soxr_io_spec(type, type); |
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| 47 | |||
| 48 | soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby); |
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| 49 | q_spec.precision = linear? 0 : precision; |
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| 50 | #if !defined SOXR_VERSION /* Deprecated @ March 2013: */ |
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| 51 | q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc; |
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| 52 | #else |
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| 53 | q_spec.passband_end = cutoff? FFMAX(FFMIN(cutoff,.995),.8) : q_spec.passband_end; |
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| 54 | #endif |
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| 55 | |||
| 56 | soxr_delete((soxr_t)c); |
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| 57 | c = (struct ResampleContext *) |
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| 58 | soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0); |
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| 59 | if (!c) |
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| 60 | av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error); |
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| 61 | return c; |
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| 62 | } |
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| 63 | |||
| 64 | static void destroy(struct ResampleContext * *c){ |
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| 65 | soxr_delete((soxr_t)*c); |
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| 66 | *c = NULL; |
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| 67 | } |
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| 68 | |||
| 69 | static int flush(struct SwrContext *s){ |
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| 70 | s->delayed_samples_fixup = soxr_delay((soxr_t)s->resample); |
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| 71 | |||
| 72 | soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL); |
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| 73 | |||
| 74 | { |
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| 75 | float f; |
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| 76 | size_t idone, odone; |
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| 77 | soxr_process((soxr_t)s->resample, &f, 0, &idone, &f, 0, &odone); |
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| 78 | s->delayed_samples_fixup -= soxr_delay((soxr_t)s->resample); |
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| 79 | } |
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| 80 | |||
| 81 | return 0; |
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| 82 | } |
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| 83 | |||
| 84 | static int process( |
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| 85 | struct ResampleContext * c, AudioData *dst, int dst_size, |
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| 86 | AudioData *src, int src_size, int *consumed){ |
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| 87 | size_t idone, odone; |
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| 88 | soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count)); |
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| 89 | if (!error) |
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| 90 | error = soxr_process((soxr_t)c, src->ch, (size_t)src_size, |
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| 91 | &idone, dst->ch, (size_t)dst_size, &odone); |
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| 92 | else |
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| 93 | idone = 0; |
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| 94 | |||
| 95 | *consumed = (int)idone; |
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| 96 | return error? -1 : odone; |
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| 97 | } |
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| 98 | |||
| 99 | static int64_t get_delay(struct SwrContext *s, int64_t base){ |
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| 100 | double delayed_samples = soxr_delay((soxr_t)s->resample); |
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| 101 | double delay_s; |
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| 102 | |||
| 103 | if (s->flushed) |
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| 104 | delayed_samples += s->delayed_samples_fixup; |
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| 105 | |||
| 106 | delay_s = delayed_samples / s->out_sample_rate; |
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| 107 | |||
| 108 | return (int64_t)(delay_s * base + .5); |
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| 109 | } |
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| 110 | |||
| 111 | static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src, |
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| 112 | int in_count, int *out_idx, int *out_sz){ |
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| 113 | return 0; |
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| 114 | } |
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| 115 | |||
| 116 | static int64_t get_out_samples(struct SwrContext *s, int in_samples){ |
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| 117 | double out_samples = (double)s->out_sample_rate / s->in_sample_rate * in_samples; |
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| 118 | double delayed_samples = soxr_delay((soxr_t)s->resample); |
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| 119 | |||
| 120 | if (s->flushed) |
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| 121 | delayed_samples += s->delayed_samples_fixup; |
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| 122 | |||
| 123 | return (int64_t)(out_samples + delayed_samples + 1 + .5); |
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| 124 | } |
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| 125 | |||
| 126 | struct Resampler const swri_soxr_resampler={ |
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| 127 | create, destroy, process, flush, NULL /* set_compensation */, get_delay, |
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| 128 | invert_initial_buffer, get_out_samples |
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| 129 | }; |
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| 130 |