falkTX
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FFmpeg
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Merge remote-tracking branch 'qatar/master' 

* qatar/master:
  configure: add functions for testing code fragments
  af_amix: avoid spurious EAGAIN.
  af_amix: return AVERROR(EAGAIN) when request_frame didn't produce output.
  af_amix: only consider negative return codes as errors.
  avconv: use only meaningful timestamps in start time check.
  avconv: fix the check for -ss as an output option.
  mss3: add forgotten 'static' qualifier for private table
  lavc: options: add planar names for request_sample_fmt
  flacdec: add planar output support
  flvdec: Treat all nellymoser versions as the same codec

Conflicts:
	ffmpeg.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
tags/n1.0
Michael Niedermayer 13 years ago
parent
a6da14ec3c 5adc829eb0
commit
46a47077ee
9 changed files with 121 additions and 57 deletions
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  1. +18
    -21
      configure
  2. +26
    -8
      libavcodec/flacdec.c
  3. +32
    -3
      libavcodec/flacdsp.c
  4. +1
    -1
      libavcodec/flacdsp.h
  5. +32
    -15
      libavcodec/flacdsp_template.c
  6. +1
    -1
      libavcodec/mss3.c
  7. +5
    -0
      libavcodec/options_table.h
  8. +6
    -4
      libavfilter/af_amix.c
  9. +0
    -4
      libavformat/flvdec.c

+ 18
- 21
configure View File

@@ -716,6 +716,20 @@ check_ld(){
check_cmd $ld $LDFLAGS $flags -o $TMPE $TMPO $libs $extralibs
}

check_code(){
log check_code "$@"
check=$1
headers=$2
code=$3
shift 3
{
for hdr in $headers; do
echo "#include <$hdr>"
done
echo "int main(void) { $code; return 0; }"
} | check_$check "$@"
}

check_cppflags(){
log check_cppflags "$@"
set -- $($filter_cppflags "$@")
@@ -912,15 +926,7 @@ check_type(){
type=$2
shift 2
disable_safe "$type"
incs=""
for hdr in $headers; do
incs="$incs
#include <$hdr>"
done
check_cc "$@" <<EOF && enable_safe "$type"
$incs
$type v;
EOF
check_code cc "$headers" "$type v" "$@" && enable_safe "$type"
}

check_struct(){
@@ -930,15 +936,8 @@ check_struct(){
member=$3
shift 3
disable_safe "${struct}_${member}"
incs=""
for hdr in $headers; do
incs="$incs
#include <$hdr>"
done
check_cc "$@" <<EOF && enable_safe "${struct}_${member}"
$incs
const void *p = &(($struct *)0)->$member;
EOF
check_code cc "$headers" "const void *p = &(($struct *)0)->$member" "$@" &&
enable_safe "${struct}_${member}"
}

require(){
@@ -2689,9 +2688,7 @@ case "$arch" in
;;
x86)
subarch="x86_32"
check_cc <<EOF && subarch="x86_64"
int test[(int)sizeof(char*) - 7];
EOF
check_code cc "" "int test[(int)sizeof(char*) - 7]" && subarch="x86_64"
if test "$subarch" = "x86_64"; then
spic=$shared
fi


+ 26
- 8
libavcodec/flacdec.c View File

@@ -104,11 +104,22 @@ int avpriv_flac_is_extradata_valid(AVCodecContext *avctx,

static void flac_set_bps(FLACContext *s)
{
if (s->bps > 16) {
s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
enum AVSampleFormat req = s->avctx->request_sample_fmt;
int need32 = s->bps > 16;
int want32 = av_get_bytes_per_sample(req) > 2;
int planar = av_sample_fmt_is_planar(req);

if (need32 || want32) {
if (planar)
s->avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
else
s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
s->sample_shift = 32 - s->bps;
} else {
s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
if (planar)
s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
else
s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
s->sample_shift = 16 - s->bps;
}
}
@@ -132,7 +143,7 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)
avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
allocate_buffers(s);
flac_set_bps(s);
ff_flacdsp_init(&s->dsp, avctx->sample_fmt);
ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps);
s->got_streaminfo = 1;

avcodec_get_frame_defaults(&s->frame);
@@ -233,7 +244,7 @@ static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)
avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]);
allocate_buffers(s);
flac_set_bps(s);
ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt);
ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
s->got_streaminfo = 1;

return 0;
@@ -492,9 +503,11 @@ static int decode_frame(FLACContext *s)
"supported\n");
return -1;
}
s->bps = s->avctx->bits_per_raw_sample = fi.bps;

flac_set_bps(s);
if (!s->bps) {
s->bps = s->avctx->bits_per_raw_sample = fi.bps;
flac_set_bps(s);
}

if (!s->max_blocksize)
s->max_blocksize = FLAC_MAX_BLOCKSIZE;
@@ -520,7 +533,7 @@ static int decode_frame(FLACContext *s)

if (!s->got_streaminfo) {
allocate_buffers(s);
ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt);
ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
s->got_streaminfo = 1;
dump_headers(s->avctx, (FLACStreaminfo *)s);
}
@@ -628,4 +641,9 @@ AVCodec ff_flac_decoder = {
.decode = flac_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_S32P,
-1 },
};

+ 32
- 3
libavcodec/flacdsp.c View File

@@ -23,10 +23,21 @@
#include "flacdsp.h"

#define SAMPLE_SIZE 16
#define PLANAR 0
#include "flacdsp_template.c"

#undef PLANAR
#define PLANAR 1
#include "flacdsp_template.c"

#undef SAMPLE_SIZE
#undef PLANAR
#define SAMPLE_SIZE 32
#define PLANAR 0
#include "flacdsp_template.c"

#undef PLANAR
#define PLANAR 1
#include "flacdsp_template.c"

static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
@@ -72,15 +83,27 @@ static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],

}

av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt)
av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
int bps)
{
if (bps > 16)
c->lpc = flac_lpc_32_c;
else
c->lpc = flac_lpc_16_c;

switch (fmt) {
case AV_SAMPLE_FMT_S32:
c->decorrelate[0] = flac_decorrelate_indep_c_32;
c->decorrelate[1] = flac_decorrelate_ls_c_32;
c->decorrelate[2] = flac_decorrelate_rs_c_32;
c->decorrelate[3] = flac_decorrelate_ms_c_32;
c->lpc = flac_lpc_32_c;
break;

case AV_SAMPLE_FMT_S32P:
c->decorrelate[0] = flac_decorrelate_indep_c_32p;
c->decorrelate[1] = flac_decorrelate_ls_c_32p;
c->decorrelate[2] = flac_decorrelate_rs_c_32p;
c->decorrelate[3] = flac_decorrelate_ms_c_32p;
break;

case AV_SAMPLE_FMT_S16:
@@ -88,7 +111,13 @@ av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt)
c->decorrelate[1] = flac_decorrelate_ls_c_16;
c->decorrelate[2] = flac_decorrelate_rs_c_16;
c->decorrelate[3] = flac_decorrelate_ms_c_16;
c->lpc = flac_lpc_16_c;
break;

case AV_SAMPLE_FMT_S16P:
c->decorrelate[0] = flac_decorrelate_indep_c_16p;
c->decorrelate[1] = flac_decorrelate_ls_c_16p;
c->decorrelate[2] = flac_decorrelate_rs_c_16p;
c->decorrelate[3] = flac_decorrelate_ms_c_16p;
break;
}
}

+ 1
- 1
libavcodec/flacdsp.h View File

@@ -29,6 +29,6 @@ typedef struct FLACDSPContext {
int qlevel, int len);
} FLACDSPContext;

void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt);
void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int bps);

#endif /* AVCODEC_FLACDSP_H */

+ 32
- 15
libavcodec/flacdsp_template.c View File

@@ -19,68 +19,85 @@
*/

#include <stdint.h>
#include "libavutil/avutil.h"

#undef FUNC
#undef FSUF
#undef sample
#undef sample_type
#undef OUT
#undef S

#if SAMPLE_SIZE == 32
# define FUNC(n) n ## _32
# define sample int32_t
# define sample_type int32_t
#else
# define FUNC(n) n ## _16
# define sample int16_t
# define sample_type int16_t
#endif

#if PLANAR
# define FSUF AV_JOIN(SAMPLE_SIZE, p)
# define sample sample_type *
# define OUT(n) n
# define S(s, c, i) (s[c][i])
#else
# define FSUF SAMPLE_SIZE
# define sample sample_type
# define OUT(n) n[0]
# define S(s, c, i) (*s++)
#endif

#define FUNC(n) AV_JOIN(n ## _, FSUF)

static void FUNC(flac_decorrelate_indep_c)(uint8_t **out, int32_t **in,
int channels, int len, int shift)
{
sample *samples = (sample *) out[0];
sample *samples = (sample *) OUT(out);
int i, j;

for (j = 0; j < len; j++)
for (i = 0; i < channels; i++)
*samples++ = in[i][j] << shift;
S(samples, i, j) = in[i][j] << shift;
}

static void FUNC(flac_decorrelate_ls_c)(uint8_t **out, int32_t **in,
int channels, int len, int shift)
{
sample *samples = (sample *) out[0];
sample *samples = (sample *) OUT(out);
int i;

for (i = 0; i < len; i++) {
int a = in[0][i];
int b = in[1][i];
*samples++ = a << shift;
*samples++ = (a - b) << shift;
S(samples, 0, i) = a << shift;
S(samples, 1, i) = (a - b) << shift;
}
}

static void FUNC(flac_decorrelate_rs_c)(uint8_t **out, int32_t **in,
int channels, int len, int shift)
{
sample *samples = (sample *) out[0];
sample *samples = (sample *) OUT(out);
int i;

for (i = 0; i < len; i++) {
int a = in[0][i];
int b = in[1][i];
*samples++ = (a + b) << shift;
*samples++ = b << shift;
S(samples, 0, i) = (a + b) << shift;
S(samples, 1, i) = b << shift;
}
}

static void FUNC(flac_decorrelate_ms_c)(uint8_t **out, int32_t **in,
int channels, int len, int shift)
{
sample *samples = (sample *) out[0];
sample *samples = (sample *) OUT(out);
int i;

for (i = 0; i < len; i++) {
int a = in[0][i];
int b = in[1][i];
a -= b >> 1;
*samples++ = (a + b) << shift;
*samples++ = a << shift;
S(samples, 0, i) = (a + b) << shift;
S(samples, 1, i) = a << shift;
}
}

+ 1
- 1
libavcodec/mss3.c View File

@@ -141,7 +141,7 @@ static const uint8_t mss3_chroma_quant[64] = {
99, 99, 99, 99, 99, 99, 99, 99
};

const uint8_t zigzag_scan[64] = {
static const uint8_t zigzag_scan[64] = {
0, 1, 8, 16, 9, 2, 3, 10,
17, 24, 32, 25, 18, 11, 4, 5,
12, 19, 26, 33, 40, 48, 41, 34,


+ 5
- 0
libavcodec/options_table.h View File

@@ -400,6 +400,11 @@ static const AVOption options[]={
{"s32", "32-bit signed integer", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S32 }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"flt", "32-bit float", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_FLT }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"dbl", "64-bit double", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_DBL }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"u8p" , "8-bit unsigned integer planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_U8P }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"s16p", "16-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S16P }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"s32p", "32-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S32P }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"fltp", "32-bit float planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_FLTP }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{"dblp", "64-bit double planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_DBLP }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
{NULL},
};



+ 6
- 4
libavfilter/af_amix.c View File

@@ -354,7 +354,7 @@ static int request_samples(AVFilterContext *ctx, int min_samples)
s->input_state[i] = INPUT_OFF;
continue;
}
} else if (ret)
} else if (ret < 0)
return ret;
}
return 0;
@@ -403,7 +403,7 @@ static int request_frame(AVFilterLink *outlink)

available_samples = get_available_samples(s);
if (!available_samples)
return 0;
return AVERROR(EAGAIN);

return output_frame(outlink, available_samples);
}
@@ -416,7 +416,7 @@ static int request_frame(AVFilterLink *outlink)
return AVERROR_EOF;
else
return AVERROR(EAGAIN);
} else if (ret)
} else if (ret < 0)
return ret;
}
av_assert0(s->frame_list->nb_frames > 0);
@@ -431,10 +431,12 @@ static int request_frame(AVFilterLink *outlink)
ret = calc_active_inputs(s);
if (ret < 0)
return ret;
}

if (s->active_inputs > 1) {
available_samples = get_available_samples(s);
if (!available_samples)
return 0;
return AVERROR(EAGAIN);
available_samples = FFMIN(available_samples, wanted_samples);
} else {
available_samples = wanted_samples;


+ 0
- 4
libavformat/flvdec.c View File

@@ -113,11 +113,7 @@ static int flv_same_audio_codec(AVCodecContext *acodec, int flags)
case FLV_CODECID_MP3:
return acodec->codec_id == CODEC_ID_MP3;
case FLV_CODECID_NELLYMOSER_8KHZ_MONO:
return acodec->sample_rate == 8000 &&
acodec->codec_id == CODEC_ID_NELLYMOSER;
case FLV_CODECID_NELLYMOSER_16KHZ_MONO:
return acodec->sample_rate == 16000 &&
acodec->codec_id == CODEC_ID_NELLYMOSER;
case FLV_CODECID_NELLYMOSER:
return acodec->codec_id == CODEC_ID_NELLYMOSER;
case FLV_CODECID_PCM_MULAW:


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