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Merge remote-tracking branch 'qatar/master' 

* qatar/master: (23 commits)
  avconv: Reformat s16 volume adjustment.
  ARM: NEON optimised vector_fmac_scalar()
  dca: use vector_fmac_scalar from dsputil
  dsputil: add vector_fmac_scalar()
  latmenc: Fix private options
  vf_unsharp: store hsub/vsub in the filter context
  vf_unsharp: adopt a more natural order of params in apply_unsharp()
  vf_unsharp: rename method "unsharpen" to "apply_unsharp"
  vf_scale: apply the same transform to the aspect during init that is applied per frame
  vf_pad: fix "vsub" variable value computation
  vf_scale: add a "sar" variable
  lavfi: fix realloc size computation in avfilter_add_format()
  vsrc_color: use internal timebase
  lavfi: fix signature for avfilter_graph_parse() and avfilter_graph_config()
  graphparser: prefer void * over AVClass * for log contexts
  avfiltergraph: use meaningful error codes
  avconv: Initialize return value for codec copy path.
  fate: use 'run' helper for seek-test
  fate: remove seek-mpeg2reuse test
  Fix memory (re)allocation in matroskadec.c, related to MSVR-11-0080.
  ...

Conflicts:
	doc/filters.texi
	libavfilter/avfilter.h
	libavfilter/avfiltergraph.c
	libavfilter/avfiltergraph.h
	libavfilter/graphparser.c
	libavfilter/vf_scale.c
	libavfilter/vsrc_color.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2011-09-29 01:03:02 +02:00
parent a3a5c61c61 daf9890811
commit f9a2d0c3fe
13 changed files with 203 additions and 27 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

64
avconv.c
View File

@ -153,7 +153,7 @@ static uint8_t *audio_buf;
static uint8_t *audio_out;
static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
static short *samples;
static void *samples;
#define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"
@ -1586,7 +1586,7 @@ static int output_packet(InputStream *ist, int ist_index,
{
AVFormatContext *os;
OutputStream *ost;
int ret, i;
int ret = 0, i;
int got_output;
void *buffer_to_free = NULL;
static unsigned int samples_size= 0;
@ -1641,8 +1641,8 @@ static int output_packet(InputStream *ist, int ist_index,
if (ist->decoding_needed) {
switch(ist->st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:{
if(pkt && samples_size < FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
samples_size = FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE);
if(pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);
av_free(samples);
samples= av_malloc(samples_size);
}
@ -1731,11 +1731,57 @@ static int output_packet(InputStream *ist, int ist_index,
// preprocess audio (volume)
if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
if (audio_volume != 256) {
short *volp;
volp = samples;
for(i=0;i<(decoded_data_size / sizeof(short));i++) {
int v = ((*volp) * audio_volume + 128) >> 8;
*volp++ = av_clip_int16(v);
switch (ist->st->codec->sample_fmt) {
case AV_SAMPLE_FMT_U8:
{
uint8_t *volp = samples;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;
*volp++ = av_clip_uint8(v);
}
break;
}
case AV_SAMPLE_FMT_S16:
{
int16_t *volp = samples;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
int v = ((*volp) * audio_volume + 128) >> 8;
*volp++ = av_clip_int16(v);
}
break;
}
case AV_SAMPLE_FMT_S32:
{
int32_t *volp = samples;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);
*volp++ = av_clipl_int32(v);
}
break;
}
case AV_SAMPLE_FMT_FLT:
{
float *volp = samples;
float scale = audio_volume / 256.f;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
*volp++ *= scale;
}
break;
}
case AV_SAMPLE_FMT_DBL:
{
double *volp = samples;
double scale = audio_volume / 256.;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
*volp++ *= scale;
}
break;
}
default:
av_log(NULL, AV_LOG_FATAL,
"Audio volume adjustment on sample format %s is not supported.\n",
av_get_sample_fmt_name(ist->st->codec->sample_fmt));
exit_program(1);
}
}
}

3
doc/filters.texi
View File

@ -1687,6 +1687,9 @@ input sample aspect ratio
@item dar
input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar}
@item sar
input sample aspect ratio
@item hsub, vsub
horizontal and vertical chroma subsample values. For example for the
pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1.

64
ffmpeg.c
View File

@ -160,7 +160,7 @@ static uint8_t *audio_buf;
static uint8_t *audio_out;
static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
static short *samples;
static void *samples;
static uint8_t *input_tmp= NULL;
#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"
@ -1596,7 +1596,7 @@ static int output_packet(InputStream *ist, int ist_index,
{
AVFormatContext *os;
OutputStream *ost;
int ret, i;
int ret = 0, i;
int got_output;
void *buffer_to_free = NULL;
static unsigned int samples_size= 0;
@ -1651,8 +1651,8 @@ static int output_packet(InputStream *ist, int ist_index,
if (ist->decoding_needed) {
switch(ist->st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:{
if(pkt && samples_size < FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
samples_size = FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE);
if(pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) {
samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE);
av_free(samples);
samples= av_malloc(samples_size);
}
@ -1758,11 +1758,57 @@ static int output_packet(InputStream *ist, int ist_index,
// preprocess audio (volume)
if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
if (audio_volume != 256) {
short *volp;
volp = samples;
for(i=0;i<(decoded_data_size / sizeof(short));i++) {
int v = ((*volp) * audio_volume + 128) >> 8;
*volp++ = av_clip_int16(v);
switch (ist->st->codec->sample_fmt) {
case AV_SAMPLE_FMT_U8:
{
uint8_t *volp = samples;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;
*volp++ = av_clip_uint8(v);
}
break;
}
case AV_SAMPLE_FMT_S16:
{
int16_t *volp = samples;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
int v = ((*volp) * audio_volume + 128) >> 8;
*volp++ = av_clip_int16(v);
}
break;
}
case AV_SAMPLE_FMT_S32:
{
int32_t *volp = samples;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);
*volp++ = av_clipl_int32(v);
}
break;
}
case AV_SAMPLE_FMT_FLT:
{
float *volp = samples;
float scale = audio_volume / 256.f;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
*volp++ *= scale;
}
break;
}
case AV_SAMPLE_FMT_DBL:
{
double *volp = samples;
double scale = audio_volume / 256.;
for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {
*volp++ *= scale;
}
break;
}
default:
av_log(NULL, AV_LOG_FATAL,
"Audio volume adjustment on sample format %s is not supported.\n",
av_get_sample_fmt_name(ist->st->codec->sample_fmt));
exit_program(1);
}
}
}

3
libavcodec/arm/dsputil_init_neon.c
View File

@ -143,6 +143,8 @@ void ff_vector_fmul_window_neon(float *dst, const float *src0,
const float *src1, const float *win, int len);
void ff_vector_fmul_scalar_neon(float *dst, const float *src, float mul,
int len);
void ff_vector_fmac_scalar_neon(float *dst, const float *src, float mul,
int len);
void ff_butterflies_float_neon(float *v1, float *v2, int len);
float ff_scalarproduct_float_neon(const float *v1, const float *v2, int len);
void ff_vector_fmul_reverse_neon(float *dst, const float *src0,
@ -305,6 +307,7 @@ void ff_dsputil_init_neon(DSPContext *c, AVCodecContext *avctx)
c->vector_fmul = ff_vector_fmul_neon;
c->vector_fmul_window = ff_vector_fmul_window_neon;
c->vector_fmul_scalar = ff_vector_fmul_scalar_neon;
c->vector_fmac_scalar = ff_vector_fmac_scalar_neon;
c->butterflies_float = ff_butterflies_float_neon;
c->scalarproduct_float = ff_scalarproduct_float_neon;
c->vector_fmul_reverse = ff_vector_fmul_reverse_neon;

48
libavcodec/arm/dsputil_neon.S
View File

@ -587,6 +587,54 @@ NOVFP vdup.32 q8, r2
.unreq len
endfunc
function ff_vector_fmac_scalar_neon, export=1
VFP len .req r2
VFP acc .req r3
NOVFP len .req r3
NOVFP acc .req r2
VFP vdup.32 q15, d0[0]
NOVFP vdup.32 q15, r2
bics r12, len, #15
mov acc, r0
beq 3f
vld1.32 {q0}, [r1,:128]!
vld1.32 {q8}, [acc,:128]!
vld1.32 {q1}, [r1,:128]!
vld1.32 {q9}, [acc,:128]!
1: vmla.f32 q8, q0, q15
vld1.32 {q2}, [r1,:128]!
vld1.32 {q10}, [acc,:128]!
vmla.f32 q9, q1, q15
vld1.32 {q3}, [r1,:128]!
vld1.32 {q11}, [acc,:128]!
vmla.f32 q10, q2, q15
vst1.32 {q8}, [r0,:128]!
vmla.f32 q11, q3, q15
vst1.32 {q9}, [r0,:128]!
subs r12, r12, #16
beq 2f
vld1.32 {q0}, [r1,:128]!
vld1.32 {q8}, [acc,:128]!
vst1.32 {q10}, [r0,:128]!
vld1.32 {q1}, [r1,:128]!
vld1.32 {q9}, [acc,:128]!
vst1.32 {q11}, [r0,:128]!
b 1b
2: vst1.32 {q10}, [r0,:128]!
vst1.32 {q11}, [r0,:128]!
ands len, len, #15
it eq
bxeq lr
3: vld1.32 {q0}, [r1,:128]!
vld1.32 {q8}, [acc,:128]!
vmla.f32 q8, q0, q15
vst1.32 {q8}, [r0,:128]!
subs len, len, #4
bgt 3b
bx lr
.unreq len
endfunc
function ff_butterflies_float_neon, export=1
1: vld1.32 {q0},[r0,:128]
vld1.32 {q1},[r1,:128]

7
libavcodec/dca.c
View File

@ -1833,11 +1833,8 @@ static int dca_decode_frame(AVCodecContext * avctx,
float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;
float* lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
float* rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
int j;
for(j = 0; j < 256; ++j) {
lt_chan[j] -= back_chan[j] * M_SQRT1_2;
rt_chan[j] -= back_chan[j] * M_SQRT1_2;
}
s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {

9
libavcodec/dsputil.c
View File

@ -2443,6 +2443,14 @@ static void vector_fmul_scalar_c(float *dst, const float *src, float mul,
dst[i] = src[i] * mul;
}
static void vector_fmac_scalar_c(float *dst, const float *src, float mul,
int len)
{
int i;
for (i = 0; i < len; i++)
dst[i] += src[i] * mul;
}
static void butterflies_float_c(float *restrict v1, float *restrict v2,
int len)
{
@ -2978,6 +2986,7 @@ av_cold void dsputil_init(DSPContext* c, AVCodecContext *avctx)
c->scalarproduct_float = scalarproduct_float_c;
c->butterflies_float = butterflies_float_c;
c->vector_fmul_scalar = vector_fmul_scalar_c;
c->vector_fmac_scalar = vector_fmac_scalar_c;
c->shrink[0]= av_image_copy_plane;
c->shrink[1]= ff_shrink22;

11
libavcodec/dsputil.h
View File

@ -423,6 +423,17 @@ typedef struct DSPContext {
*/
void (*vector_fmul_scalar)(float *dst, const float *src, float mul,
int len);
/**
* Multiply a vector of floats by a scalar float and add to
* destination vector. Source and destination vectors must
* overlap exactly or not at all.
* @param dst result vector, 16-byte aligned
* @param src input vector, 16-byte aligned
* @param mul scalar value
* @param len length of vector, multiple of 4
*/
void (*vector_fmac_scalar)(float *dst, const float *src, float mul,
int len);
/**
* Calculate the scalar product of two vectors of floats.
* @param v1 first vector, 16-byte aligned

2
libavfilter/avfilter.h
View File

@ -30,7 +30,7 @@
#define LIBAVFILTER_VERSION_MAJOR 2
#define LIBAVFILTER_VERSION_MINOR 43
#define LIBAVFILTER_VERSION_MICRO 5
#define LIBAVFILTER_VERSION_MICRO 6
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \

1
libavfilter/avfiltergraph.h
View File

@ -171,4 +171,5 @@ int avfilter_graph_send_command(AVFilterGraph *graph, const char *target, const
int avfilter_graph_queue_command(AVFilterGraph *graph, const char *target, const char *cmd, const char *arg, int flags, double ts);
#endif /* AVFILTER_AVFILTERGRAPH_H */

1
libavformat/latmenc.c
View File

@ -28,6 +28,7 @@
#include "rawenc.h"
typedef struct {
AVClass *av_class;
int off;
int channel_conf;
int object_type;

15
libavformat/matroskadec.c
View File

@ -967,6 +967,7 @@ static int matroska_decode_buffer(uint8_t** buf, int* buf_size,
uint8_t* data = *buf;
int isize = *buf_size;
uint8_t* pkt_data = NULL;
uint8_t* newpktdata;
int pkt_size = isize;
int result = 0;
int olen;
@ -996,7 +997,12 @@ static int matroska_decode_buffer(uint8_t** buf, int* buf_size,
zstream.avail_in = isize;
do {
pkt_size *= 3;
pkt_data = av_realloc(pkt_data, pkt_size);
newpktdata = av_realloc(pkt_data, pkt_size);
if (!newpktdata) {
inflateEnd(&zstream);
goto failed;
}
pkt_data = newpktdata;
zstream.avail_out = pkt_size - zstream.total_out;
zstream.next_out = pkt_data + zstream.total_out;
if (pkt_data) {
@ -1020,7 +1026,12 @@ static int matroska_decode_buffer(uint8_t** buf, int* buf_size,
bzstream.avail_in = isize;
do {
pkt_size *= 3;
pkt_data = av_realloc(pkt_data, pkt_size);
newpktdata = av_realloc(pkt_data, pkt_size);
if (!newpktdata) {
BZ2_bzDecompressEnd(&bzstream);
goto failed;
}
pkt_data = newpktdata;
bzstream.avail_out = pkt_size - bzstream.total_out_lo32;
bzstream.next_out = pkt_data + bzstream.total_out_lo32;
if (pkt_data) {

2
tests/fate-run.sh
View File

@ -105,7 +105,7 @@ seektest(){
file=$(echo tests/data/$d/$file)
;;
esac
$target_exec $target_path/libavformat/seek-test $target_path/$file
run libavformat/seek-test $target_path/$file
}
mkdir -p "$outdir"