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Diffstat (limited to 'media/ffvpx/libavutil/samplefmt.c')
| -rw-r--r-- | media/ffvpx/libavutil/samplefmt.c | 252 |
1 files changed, 252 insertions, 0 deletions
diff --git a/media/ffvpx/libavutil/samplefmt.c b/media/ffvpx/libavutil/samplefmt.c new file mode 100644 index 000000000..c605b5ebd --- /dev/null +++ b/media/ffvpx/libavutil/samplefmt.c @@ -0,0 +1,252 @@ +/* + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "common.h" +#include "samplefmt.h" + +#include <stdio.h> +#include <stdlib.h> +#include <string.h> + +typedef struct SampleFmtInfo { + char name[8]; + int bits; + int planar; + enum AVSampleFormat altform; ///< planar<->packed alternative form +} SampleFmtInfo; + +/** this table gives more information about formats */ +static const SampleFmtInfo sample_fmt_info[AV_SAMPLE_FMT_NB] = { + [AV_SAMPLE_FMT_U8] = { .name = "u8", .bits = 8, .planar = 0, .altform = AV_SAMPLE_FMT_U8P }, + [AV_SAMPLE_FMT_S16] = { .name = "s16", .bits = 16, .planar = 0, .altform = AV_SAMPLE_FMT_S16P }, + [AV_SAMPLE_FMT_S32] = { .name = "s32", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_S32P }, + [AV_SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_FLTP }, + [AV_SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_DBLP }, + [AV_SAMPLE_FMT_U8P] = { .name = "u8p", .bits = 8, .planar = 1, .altform = AV_SAMPLE_FMT_U8 }, + [AV_SAMPLE_FMT_S16P] = { .name = "s16p", .bits = 16, .planar = 1, .altform = AV_SAMPLE_FMT_S16 }, + [AV_SAMPLE_FMT_S32P] = { .name = "s32p", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_S32 }, + [AV_SAMPLE_FMT_FLTP] = { .name = "fltp", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_FLT }, + [AV_SAMPLE_FMT_DBLP] = { .name = "dblp", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_DBL }, +}; + +const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt) +{ + if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB) + return NULL; + return sample_fmt_info[sample_fmt].name; +} + +enum AVSampleFormat av_get_sample_fmt(const char *name) +{ + int i; + + for (i = 0; i < AV_SAMPLE_FMT_NB; i++) + if (!strcmp(sample_fmt_info[i].name, name)) + return i; + return AV_SAMPLE_FMT_NONE; +} + +enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar) +{ + if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB) + return AV_SAMPLE_FMT_NONE; + if (sample_fmt_info[sample_fmt].planar == planar) + return sample_fmt; + return sample_fmt_info[sample_fmt].altform; +} + +enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt) +{ + if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB) + return AV_SAMPLE_FMT_NONE; + if (sample_fmt_info[sample_fmt].planar) + return sample_fmt_info[sample_fmt].altform; + return sample_fmt; +} + +enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt) +{ + if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB) + return AV_SAMPLE_FMT_NONE; + if (sample_fmt_info[sample_fmt].planar) + return sample_fmt; + return sample_fmt_info[sample_fmt].altform; +} + +char *av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt) +{ + /* print header */ + if (sample_fmt < 0) + snprintf(buf, buf_size, "name " " depth"); + else if (sample_fmt < AV_SAMPLE_FMT_NB) { + SampleFmtInfo info = sample_fmt_info[sample_fmt]; + snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits); + } + + return buf; +} + +int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt) +{ + return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ? + 0 : sample_fmt_info[sample_fmt].bits >> 3; +} + +int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt) +{ + if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB) + return 0; + return sample_fmt_info[sample_fmt].planar; +} + +int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, + enum AVSampleFormat sample_fmt, int align) +{ + int line_size; + int sample_size = av_get_bytes_per_sample(sample_fmt); + int planar = av_sample_fmt_is_planar(sample_fmt); + + /* validate parameter ranges */ + if (!sample_size || nb_samples <= 0 || nb_channels <= 0) + return AVERROR(EINVAL); + + /* auto-select alignment if not specified */ + if (!align) { + if (nb_samples > INT_MAX - 31) + return AVERROR(EINVAL); + align = 1; + nb_samples = FFALIGN(nb_samples, 32); + } + + /* check for integer overflow */ + if (nb_channels > INT_MAX / align || + (int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size) + return AVERROR(EINVAL); + + line_size = planar ? FFALIGN(nb_samples * sample_size, align) : + FFALIGN(nb_samples * sample_size * nb_channels, align); + if (linesize) + *linesize = line_size; + + return planar ? line_size * nb_channels : line_size; +} + +int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, + const uint8_t *buf, int nb_channels, int nb_samples, + enum AVSampleFormat sample_fmt, int align) +{ + int ch, planar, buf_size, line_size; + + planar = av_sample_fmt_is_planar(sample_fmt); + buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples, + sample_fmt, align); + if (buf_size < 0) + return buf_size; + + audio_data[0] = (uint8_t *)buf; + for (ch = 1; planar && ch < nb_channels; ch++) + audio_data[ch] = audio_data[ch-1] + line_size; + + if (linesize) + *linesize = line_size; + + return buf_size; +} + +int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, + int nb_samples, enum AVSampleFormat sample_fmt, int align) +{ + uint8_t *buf; + int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples, + sample_fmt, align); + if (size < 0) + return size; + + buf = av_malloc(size); + if (!buf) + return AVERROR(ENOMEM); + + size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels, + nb_samples, sample_fmt, align); + if (size < 0) { + av_free(buf); + return size; + } + + av_samples_set_silence(audio_data, 0, nb_samples, nb_channels, sample_fmt); + + return size; +} + +int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, + int nb_samples, enum AVSampleFormat sample_fmt, int align) +{ + int ret, nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1; + + *audio_data = av_calloc(nb_planes, sizeof(**audio_data)); + if (!*audio_data) + return AVERROR(ENOMEM); + ret = av_samples_alloc(*audio_data, linesize, nb_channels, + nb_samples, sample_fmt, align); + if (ret < 0) + av_freep(audio_data); + return ret; +} + +int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset, + int src_offset, int nb_samples, int nb_channels, + enum AVSampleFormat sample_fmt) +{ + int planar = av_sample_fmt_is_planar(sample_fmt); + int planes = planar ? nb_channels : 1; + int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels); + int data_size = nb_samples * block_align; + int i; + + dst_offset *= block_align; + src_offset *= block_align; + + if((dst[0] < src[0] ? src[0] - dst[0] : dst[0] - src[0]) >= data_size) { + for (i = 0; i < planes; i++) + memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size); + } else { + for (i = 0; i < planes; i++) + memmove(dst[i] + dst_offset, src[i] + src_offset, data_size); + } + + return 0; +} + +int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, + int nb_channels, enum AVSampleFormat sample_fmt) +{ + int planar = av_sample_fmt_is_planar(sample_fmt); + int planes = planar ? nb_channels : 1; + int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels); + int data_size = nb_samples * block_align; + int fill_char = (sample_fmt == AV_SAMPLE_FMT_U8 || + sample_fmt == AV_SAMPLE_FMT_U8P) ? 0x80 : 0x00; + int i; + + offset *= block_align; + + for (i = 0; i < planes; i++) + memset(audio_data[i] + offset, fill_char, data_size); + + return 0; +} |