Add m-esr52 at 52.6.0

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;
+}