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  1. /*
  2.  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
  3.  *
  4.  * This file is part of FFmpeg.
  5.  *
  6.  * FFmpeg is free software; you can redistribute it and/or
  7.  * modify it under the terms of the GNU Lesser General Public
  8.  * License as published by the Free Software Foundation; either
  9.  * version 2.1 of the License, or (at your option) any later version.
  10.  *
  11.  * FFmpeg is distributed in the hope that it will be useful,
  12.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  14.  * Lesser General Public License for more details.
  15.  *
  16.  * You should have received a copy of the GNU Lesser General Public
  17.  * License along with FFmpeg; if not, write to the Free Software
  18.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  19.  */
  20.  
  21. #include <stdint.h>
  22. #include <string.h>
  23.  
  24. #include "libavutil/mem.h"
  25. #include "audio_data.h"
  26.  
  27. static const AVClass audio_data_class = {
  28.     .class_name = "AudioData",
  29.     .item_name  = av_default_item_name,
  30.     .version    = LIBAVUTIL_VERSION_INT,
  31. };
  32.  
  33. /*
  34.  * Calculate alignment for data pointers.
  35.  */
  36. static void calc_ptr_alignment(AudioData *a)
  37. {
  38.     int p;
  39.     int min_align = 128;
  40.  
  41.     for (p = 0; p < a->planes; p++) {
  42.         int cur_align = 128;
  43.         while ((intptr_t)a->data[p] % cur_align)
  44.             cur_align >>= 1;
  45.         if (cur_align < min_align)
  46.             min_align = cur_align;
  47.     }
  48.     a->ptr_align = min_align;
  49. }
  50.  
  51. int ff_audio_data_set_channels(AudioData *a, int channels)
  52. {
  53.     if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS ||
  54.         channels > a->allocated_channels)
  55.         return AVERROR(EINVAL);
  56.  
  57.     a->channels  = channels;
  58.     a->planes    = a->is_planar ? channels : 1;
  59.  
  60.     calc_ptr_alignment(a);
  61.  
  62.     return 0;
  63. }
  64.  
  65. int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels,
  66.                        int nb_samples, enum AVSampleFormat sample_fmt,
  67.                        int read_only, const char *name)
  68. {
  69.     int p;
  70.  
  71.     memset(a, 0, sizeof(*a));
  72.     a->class = &audio_data_class;
  73.  
  74.     if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) {
  75.         av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels);
  76.         return AVERROR(EINVAL);
  77.     }
  78.  
  79.     a->sample_size = av_get_bytes_per_sample(sample_fmt);
  80.     if (!a->sample_size) {
  81.         av_log(a, AV_LOG_ERROR, "invalid sample format\n");
  82.         return AVERROR(EINVAL);
  83.     }
  84.     a->is_planar = av_sample_fmt_is_planar(sample_fmt);
  85.     a->planes    = a->is_planar ? channels : 1;
  86.     a->stride    = a->sample_size * (a->is_planar ? 1 : channels);
  87.  
  88.     for (p = 0; p < (a->is_planar ? channels : 1); p++) {
  89.         if (!src[p]) {
  90.             av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p);
  91.             return AVERROR(EINVAL);
  92.         }
  93.         a->data[p] = src[p];
  94.     }
  95.     a->allocated_samples  = nb_samples * !read_only;
  96.     a->nb_samples         = nb_samples;
  97.     a->sample_fmt         = sample_fmt;
  98.     a->channels           = channels;
  99.     a->allocated_channels = channels;
  100.     a->read_only          = read_only;
  101.     a->allow_realloc      = 0;
  102.     a->name               = name ? name : "{no name}";
  103.  
  104.     calc_ptr_alignment(a);
  105.     a->samples_align = plane_size / a->stride;
  106.  
  107.     return 0;
  108. }
  109.  
  110. AudioData *ff_audio_data_alloc(int channels, int nb_samples,
  111.                                enum AVSampleFormat sample_fmt, const char *name)
  112. {
  113.     AudioData *a;
  114.     int ret;
  115.  
  116.     if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS)
  117.         return NULL;
  118.  
  119.     a = av_mallocz(sizeof(*a));
  120.     if (!a)
  121.         return NULL;
  122.  
  123.     a->sample_size = av_get_bytes_per_sample(sample_fmt);
  124.     if (!a->sample_size) {
  125.         av_free(a);
  126.         return NULL;
  127.     }
  128.     a->is_planar = av_sample_fmt_is_planar(sample_fmt);
  129.     a->planes    = a->is_planar ? channels : 1;
  130.     a->stride    = a->sample_size * (a->is_planar ? 1 : channels);
  131.  
  132.     a->class              = &audio_data_class;
  133.     a->sample_fmt         = sample_fmt;
  134.     a->channels           = channels;
  135.     a->allocated_channels = channels;
  136.     a->read_only          = 0;
  137.     a->allow_realloc      = 1;
  138.     a->name               = name ? name : "{no name}";
  139.  
  140.     if (nb_samples > 0) {
  141.         ret = ff_audio_data_realloc(a, nb_samples);
  142.         if (ret < 0) {
  143.             av_free(a);
  144.             return NULL;
  145.         }
  146.         return a;
  147.     } else {
  148.         calc_ptr_alignment(a);
  149.         return a;
  150.     }
  151. }
  152.  
  153. int ff_audio_data_realloc(AudioData *a, int nb_samples)
  154. {
  155.     int ret, new_buf_size, plane_size, p;
  156.  
  157.     /* check if buffer is already large enough */
  158.     if (a->allocated_samples >= nb_samples)
  159.         return 0;
  160.  
  161.     /* validate that the output is not read-only and realloc is allowed */
  162.     if (a->read_only || !a->allow_realloc)
  163.         return AVERROR(EINVAL);
  164.  
  165.     new_buf_size = av_samples_get_buffer_size(&plane_size,
  166.                                               a->allocated_channels, nb_samples,
  167.                                               a->sample_fmt, 0);
  168.     if (new_buf_size < 0)
  169.         return new_buf_size;
  170.  
  171.     /* if there is already data in the buffer and the sample format is planar,
  172.        allocate a new buffer and copy the data, otherwise just realloc the
  173.        internal buffer and set new data pointers */
  174.     if (a->nb_samples > 0 && a->is_planar) {
  175.         uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL };
  176.  
  177.         ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels,
  178.                                nb_samples, a->sample_fmt, 0);
  179.         if (ret < 0)
  180.             return ret;
  181.  
  182.         for (p = 0; p < a->planes; p++)
  183.             memcpy(new_data[p], a->data[p], a->nb_samples * a->stride);
  184.  
  185.         av_freep(&a->buffer);
  186.         memcpy(a->data, new_data, sizeof(new_data));
  187.         a->buffer = a->data[0];
  188.     } else {
  189.         av_freep(&a->buffer);
  190.         a->buffer = av_malloc(new_buf_size);
  191.         if (!a->buffer)
  192.             return AVERROR(ENOMEM);
  193.         ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer,
  194.                                      a->allocated_channels, nb_samples,
  195.                                      a->sample_fmt, 0);
  196.         if (ret < 0)
  197.             return ret;
  198.     }
  199.     a->buffer_size       = new_buf_size;
  200.     a->allocated_samples = nb_samples;
  201.  
  202.     calc_ptr_alignment(a);
  203.     a->samples_align = plane_size / a->stride;
  204.  
  205.     return 0;
  206. }
  207.  
  208. void ff_audio_data_free(AudioData **a)
  209. {
  210.     if (!*a)
  211.         return;
  212.     av_free((*a)->buffer);
  213.     av_freep(a);
  214. }
  215.  
  216. int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map)
  217. {
  218.     int ret, p;
  219.  
  220.     /* validate input/output compatibility */
  221.     if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels)
  222.         return AVERROR(EINVAL);
  223.  
  224.     if (map && !src->is_planar) {
  225.         av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n");
  226.         return AVERROR(EINVAL);
  227.     }
  228.  
  229.     /* if the input is empty, just empty the output */
  230.     if (!src->nb_samples) {
  231.         dst->nb_samples = 0;
  232.         return 0;
  233.     }
  234.  
  235.     /* reallocate output if necessary */
  236.     ret = ff_audio_data_realloc(dst, src->nb_samples);
  237.     if (ret < 0)
  238.         return ret;
  239.  
  240.     /* copy data */
  241.     if (map) {
  242.         if (map->do_remap) {
  243.             for (p = 0; p < src->planes; p++) {
  244.                 if (map->channel_map[p] >= 0)
  245.                     memcpy(dst->data[p], src->data[map->channel_map[p]],
  246.                            src->nb_samples * src->stride);
  247.             }
  248.         }
  249.         if (map->do_copy || map->do_zero) {
  250.             for (p = 0; p < src->planes; p++) {
  251.                 if (map->channel_copy[p])
  252.                     memcpy(dst->data[p], dst->data[map->channel_copy[p]],
  253.                            src->nb_samples * src->stride);
  254.                 else if (map->channel_zero[p])
  255.                     av_samples_set_silence(&dst->data[p], 0, src->nb_samples,
  256.                                            1, dst->sample_fmt);
  257.             }
  258.         }
  259.     } else {
  260.         for (p = 0; p < src->planes; p++)
  261.             memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
  262.     }
  263.  
  264.     dst->nb_samples = src->nb_samples;
  265.  
  266.     return 0;
  267. }
  268.  
  269. int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
  270.                           int src_offset, int nb_samples)
  271. {
  272.     int ret, p, dst_offset2, dst_move_size;
  273.  
  274.     /* validate input/output compatibility */
  275.     if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) {
  276.         av_log(src, AV_LOG_ERROR, "sample format mismatch\n");
  277.         return AVERROR(EINVAL);
  278.     }
  279.  
  280.     /* validate offsets are within the buffer bounds */
  281.     if (dst_offset < 0 || dst_offset > dst->nb_samples ||
  282.         src_offset < 0 || src_offset > src->nb_samples) {
  283.         av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n",
  284.                src_offset, dst_offset);
  285.         return AVERROR(EINVAL);
  286.     }
  287.  
  288.     /* check offsets and sizes to see if we can just do nothing and return */
  289.     if (nb_samples > src->nb_samples - src_offset)
  290.         nb_samples = src->nb_samples - src_offset;
  291.     if (nb_samples <= 0)
  292.         return 0;
  293.  
  294.     /* validate that the output is not read-only */
  295.     if (dst->read_only) {
  296.         av_log(dst, AV_LOG_ERROR, "dst is read-only\n");
  297.         return AVERROR(EINVAL);
  298.     }
  299.  
  300.     /* reallocate output if necessary */
  301.     ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples);
  302.     if (ret < 0) {
  303.         av_log(dst, AV_LOG_ERROR, "error reallocating dst\n");
  304.         return ret;
  305.     }
  306.  
  307.     dst_offset2   = dst_offset + nb_samples;
  308.     dst_move_size = dst->nb_samples - dst_offset;
  309.  
  310.     for (p = 0; p < src->planes; p++) {
  311.         if (dst_move_size > 0) {
  312.             memmove(dst->data[p] + dst_offset2 * dst->stride,
  313.                     dst->data[p] + dst_offset  * dst->stride,
  314.                     dst_move_size * dst->stride);
  315.         }
  316.         memcpy(dst->data[p] + dst_offset * dst->stride,
  317.                src->data[p] + src_offset * src->stride,
  318.                nb_samples * src->stride);
  319.     }
  320.     dst->nb_samples += nb_samples;
  321.  
  322.     return 0;
  323. }
  324.  
  325. void ff_audio_data_drain(AudioData *a, int nb_samples)
  326. {
  327.     if (a->nb_samples <= nb_samples) {
  328.         /* drain the whole buffer */
  329.         a->nb_samples = 0;
  330.     } else {
  331.         int p;
  332.         int move_offset = a->stride * nb_samples;
  333.         int move_size   = a->stride * (a->nb_samples - nb_samples);
  334.  
  335.         for (p = 0; p < a->planes; p++)
  336.             memmove(a->data[p], a->data[p] + move_offset, move_size);
  337.  
  338.         a->nb_samples -= nb_samples;
  339.     }
  340. }
  341.  
  342. int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
  343.                               int nb_samples)
  344. {
  345.     uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS];
  346.     int offset_size, p;
  347.  
  348.     if (offset >= a->nb_samples)
  349.         return 0;
  350.     offset_size = offset * a->stride;
  351.     for (p = 0; p < a->planes; p++)
  352.         offset_data[p] = a->data[p] + offset_size;
  353.  
  354.     return av_audio_fifo_write(af, (void **)offset_data, nb_samples);
  355. }
  356.  
  357. int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples)
  358. {
  359.     int ret;
  360.  
  361.     if (a->read_only)
  362.         return AVERROR(EINVAL);
  363.  
  364.     ret = ff_audio_data_realloc(a, nb_samples);
  365.     if (ret < 0)
  366.         return ret;
  367.  
  368.     ret = av_audio_fifo_read(af, (void **)a->data, nb_samples);
  369.     if (ret >= 0)
  370.         a->nb_samples = ret;
  371.     return ret;
  372. }
  373.