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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /media/ffvpx/libavutil/samplefmt.c
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
Diffstat (limited to 'media/ffvpx/libavutil/samplefmt.c')
-rw-r--r--media/ffvpx/libavutil/samplefmt.c252
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;
+}