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ffmpeg.wasm-core
Commit f9a2d0c3 authored by Michael Niedermayer's avatar Michael Niedermayer
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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: 's avatarMichael Niedermayer <michaelni@gmx.at>
parents a3a5c61c daf98908
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Showing with 203 additions and 27 deletions
avconv.c
View file @ f9a2d0c3
......@@ -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);
}
}
}
......
doc/filters.texi
View file @ f9a2d0c3
......@@ -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.
......
ffmpeg.c
View file @ f9a2d0c3
......@@ -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);
}
}
}
......
libavcodec/arm/dsputil_init_neon.c
View file @ f9a2d0c3
......@@ -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;
......
libavcodec/arm/dsputil_neon.S
View file @ f9a2d0c3
......@@ -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]
......
libavcodec/dca.c
View file @ f9a2d0c3
......@@ -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) {
......
libavcodec/dsputil.c
View file @ f9a2d0c3
......@@ -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;
......
libavcodec/dsputil.h
View file @ f9a2d0c3
......@@ -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
......
libavfilter/avfilter.h
View file @ f9a2d0c3
......@@ -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, \
......
libavfilter/avfiltergraph.h
View file @ f9a2d0c3
......@@ -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 */
libavformat/latmenc.c
View file @ f9a2d0c3
......@@ -28,6 +28,7 @@
#include "rawenc.h"
typedef struct {
AVClass *av_class;
int off;
int channel_conf;
int object_type;
......
libavformat/matroskadec.c
View file @ f9a2d0c3
......@@ -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) {
......
tests/fate-run.sh
View file @ f9a2d0c3
......@@ -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"
......
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