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authorMichael Niedermayer <michaelni@gmx.at>2011-11-24 02:08:21 +0100
committerMichael Niedermayer <michaelni@gmx.at>2011-11-24 03:32:24 +0100
commit8e576d58306df95d6373dd0ca2c1f21f1afaeca9 (patch)
tree5f7b9c8783b342e80e32b58b94ded819eb414b3c /libavutil/samplefmt.c
parent7ffa9ea05aa951b6b13e615f1bd3b8280f758561 (diff)
parentbbb46f3ec7128d8a624f2aa5b4f99ec44c0b9567 (diff)
downloadffmpeg-8e576d58306df95d6373dd0ca2c1f21f1afaeca9.tar.gz
Merge remote-tracking branch 'qatar/master'
* qatar/master: libavutil: add utility functions to simplify allocation of audio buffers. libavutil: add planar sample formats and av_sample_fmt_is_planar() avconv: fix segfault at EOF with delayed pictures pcmdec: remove unneeded resetting of samples pointer avconv: remove a now unused parameter from output_packet(). avconv: formatting fixes in output_packet() avconv: declare some variables in blocks where they are used avconv: use the same behavior when decoding audio/video/subs bethsoftvideo: return proper consumed size for palette packets. cdg: skip packets that don't contain a cdg command. crcenc: add flags avconv: use vsync 0 for AVFMT_NOTIMESTAMPS formats. tiffenc: add a private option for selecting compression algorithm md5enc: add flags ARM: remove needless .text/.align directives Conflicts: doc/APIchanges libavcodec/tiffenc.c libavutil/avutil.h libavutil/samplefmt.c libavutil/samplefmt.h tests/ref/fate/bethsoft-vid tests/ref/fate/cdgraphics tests/ref/fate/film-cvid-pcm-stereo-8bit tests/ref/fate/mpeg2-field-enc tests/ref/fate/nuv tests/ref/fate/tiertex-seq tests/ref/fate/tscc-32bit tests/ref/fate/vmnc-32bit Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavutil/samplefmt.c')
-rw-r--r--libavutil/samplefmt.c103
1 files changed, 66 insertions, 37 deletions
diff --git a/libavutil/samplefmt.c b/libavutil/samplefmt.c
index bdbbf0a37d..fb9c46cab2 100644
--- a/libavutil/samplefmt.c
+++ b/libavutil/samplefmt.c
@@ -25,15 +25,21 @@
typedef struct SampleFmtInfo {
char name[4];
int bits;
+ int planar;
} 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 },
- [AV_SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 },
- [AV_SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 },
- [AV_SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 },
- [AV_SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64 },
+ [AV_SAMPLE_FMT_U8] = { .name = "u8", .bits = 8, .planar = 0 },
+ [AV_SAMPLE_FMT_S16] = { .name = "s16", .bits = 16, .planar = 0 },
+ [AV_SAMPLE_FMT_S32] = { .name = "s32", .bits = 32, .planar = 0 },
+ [AV_SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32, .planar = 0 },
+ [AV_SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64, .planar = 0 },
+ [AV_SAMPLE_FMT_U8P] = { .name = "u8p", .bits = 8, .planar = 1 },
+ [AV_SAMPLE_FMT_S16P] = { .name = "s16p", .bits = 16, .planar = 1 },
+ [AV_SAMPLE_FMT_S32P] = { .name = "s32p", .bits = 32, .planar = 1 },
+ [AV_SAMPLE_FMT_FLTP] = { .name = "fltp", .bits = 32, .planar = 1 },
+ [AV_SAMPLE_FMT_DBLP] = { .name = "dblp", .bits = 64, .planar = 1 },
};
const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
@@ -80,51 +86,74 @@ int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt)
}
#endif
-int av_samples_fill_arrays(uint8_t *pointers[8], int linesizes[8],
- uint8_t *buf, int nb_channels, int nb_samples,
- enum AVSampleFormat sample_fmt, int planar, int align)
+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 i, linesize;
+ int line_size;
int sample_size = av_get_bytes_per_sample(sample_fmt);
+ int planar = av_sample_fmt_is_planar(sample_fmt);
- if (nb_channels * (uint64_t)nb_samples * sample_size >= INT_MAX - align*(uint64_t)nb_channels)
+ /* validate parameter ranges */
+ if (!sample_size || nb_samples <= 0 || nb_channels <= 0)
return AVERROR(EINVAL);
- linesize = planar ? FFALIGN(nb_samples*sample_size, align) :
- FFALIGN(nb_samples*sample_size*nb_channels, align);
-
- if (pointers) {
- pointers[0] = buf;
- for (i = 1; planar && i < nb_channels; i++) {
- pointers[i] = pointers[i-1] + linesize;
- }
- memset(&pointers[i], 0, (8-i) * sizeof(pointers[0]));
- }
- if (linesizes) {
- linesizes[0] = linesize;
- for (i = 1; planar && i < nb_channels; i++)
- linesizes[i] = linesizes[0];
- memset(&linesizes[i], 0, (8-i) * sizeof(linesizes[0]));
- }
+ /* 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 ? linesize * nb_channels : linesize;
+ return planar ? line_size * nb_channels : line_size;
}
-int av_samples_alloc(uint8_t *pointers[8], int linesizes[8],
- int nb_channels, int nb_samples,
- enum AVSampleFormat sample_fmt, int planar,
- int align)
+int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
+ uint8_t *buf, int nb_channels, int nb_samples,
+ enum AVSampleFormat sample_fmt, int align)
+{
+ int ch, planar, buf_size;
+
+ planar = av_sample_fmt_is_planar(sample_fmt);
+ buf_size = av_samples_get_buffer_size(linesize, nb_channels, nb_samples,
+ sample_fmt, align);
+ if (buf_size < 0)
+ return buf_size;
+
+ audio_data[0] = buf;
+ for (ch = 1; planar && ch < nb_channels; ch++)
+ audio_data[ch] = audio_data[ch-1] + *linesize;
+
+ return 0;
+}
+
+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_fill_arrays(NULL, NULL,
- NULL, nb_channels, nb_samples,
- sample_fmt, planar, align);
+ int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,
+ sample_fmt, align);
+ if (size < 0)
+ return size;
buf = av_mallocz(size);
if (!buf)
return AVERROR(ENOMEM);
- return av_samples_fill_arrays(pointers, linesizes,
- buf, nb_channels, nb_samples,
- sample_fmt, planar, align);
+ size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,
+ nb_samples, sample_fmt, align);
+ if (size < 0) {
+ av_free(buf);
+ return size;
+ }
+ return 0;
}