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Code Issues Releases Activity

Merge remote-tracking branch 'qatar/master'

Browse Source 
* qatar/master:
  ppc: fix some pointer to integer casts
  ppc: fix 32-bit PIC build
  vmdaudio: fix decoding of 16-bit audio format.
  lavf: do not set codec_tag for rawvideo
  h264: check for out of bounds reads in ff_h264_decode_extradata().
  flvdec: Check for overflow before allocating arrays
  avconv: use correct output stream index when checking max_frames
  avconv: remove fake coded_frame on streamcopy hack

Conflicts:
	avconv.c
	libavcodec/h264.c
	libavcodec/ppc/asm.S
	libavcodec/vmdav.c
	libavformat/flvdec.c
	libavformat/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2011-09-25 23:30:25 +02:00
commit 537a9e5cc2
6 changed files with 106 additions and 69 deletions
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9
avconv.c
View File

@ -1826,7 +1826,6 @@ static int output_packet(InputStream *ist, int ist_index,
abort();
}
} else {
AVFrame avframe; //FIXME/XXX remove this
AVPicture pict;
AVPacket opkt;
int64_t ost_tb_start_time= av_rescale_q(of->start_time, AV_TIME_BASE_Q, ost->st->time_base);
@ -1842,10 +1841,6 @@ static int output_packet(InputStream *ist, int ist_index,
/* no reencoding needed : output the packet directly */
/* force the input stream PTS */
avcodec_get_frame_defaults(&avframe);
ost->st->codec->coded_frame= &avframe;
avframe.key_frame = pkt->flags & AV_PKT_FLAG_KEY;
if(ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
audio_size += data_size;
else if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
@ -2455,8 +2450,8 @@ static int transcode(OutputFile *output_files,
}
if (ost->frame_number >= ost->max_frames) {
int j;
for (j = of->ost_index; j < of->ctx->nb_streams; j++)
output_streams[j].is_past_recording_time = 1;
for (j = 0; j < of->ctx->nb_streams; j++)
output_streams[of->ost_index + j].is_past_recording_time = 1;
continue;
}
}

9
ffmpeg.c
View File

@ -1845,7 +1845,6 @@ static int output_packet(InputStream *ist, int ist_index,
abort();
}
} else {
AVFrame avframe; //FIXME/XXX remove this
AVPicture pict;
AVPacket opkt;
int64_t ost_tb_start_time= av_rescale_q(of->start_time, AV_TIME_BASE_Q, ost->st->time_base);
@ -1861,10 +1860,6 @@ static int output_packet(InputStream *ist, int ist_index,
/* no reencoding needed : output the packet directly */
/* force the input stream PTS */
avcodec_get_frame_defaults(&avframe);
ost->st->codec->coded_frame= &avframe;
avframe.key_frame = pkt->flags & AV_PKT_FLAG_KEY;
if(ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
audio_size += data_size;
else if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
@ -2503,8 +2498,8 @@ static int transcode(OutputFile *output_files, int nb_output_files,
}
if (ost->frame_number >= ost->max_frames) {
int j;
for (j = of->ost_index; j < of->ctx->nb_streams; j++)
output_streams[j].is_past_recording_time = 1;
for (j = 0; j < of->ctx->nb_streams; j++)
output_streams[of->ost_index + j].is_past_recording_time = 1;
continue;
}
}

23
libavcodec/ppc/asm.S
View File

@ -44,10 +44,13 @@ X(\name):
L(\name):
.endm
.macro movrel rd, sym
.macro movrel rd, sym, gp
ld \rd, \sym@got(r2)
.endm
.macro get_got rd
.endm
#else /* ARCH_PPC64 */
#define PTR .int
@ -65,19 +68,25 @@ X(\name):
\name:
.endm
.macro movrel rd, sym
.macro movrel rd, sym, gp
#if CONFIG_PIC
bcl 20, 31, lab_pic_\@
lab_pic_\@:
mflr \rd
addis \rd, \rd, (\sym - lab_pic_\@)@ha
addi \rd, \rd, (\sym - lab_pic_\@)@l
lwz \rd, \sym@got(\gp)
#else
lis \rd, \sym@ha
la \rd, \sym@l(\rd)
#endif
.endm
.macro get_got rd
#if CONFIG_PIC
bcl 20, 31, .Lgot\@
.Lgot\@:
mflr \rd
addis \rd, \rd, _GLOBAL_OFFSET_TABLE_ - .Lgot\@@ha
addi \rd, \rd, _GLOBAL_OFFSET_TABLE_ - .Lgot\@@l
#endif
.endm
#endif /* ARCH_PPC64 */
#if HAVE_IBM_ASM

7
libavcodec/ppc/fft_altivec_s.S
View File

@ -353,6 +353,7 @@ extfunc ff_fft_calc\interleave\()_altivec
mflr r0
stp r0, 2*PS(r1)
stpu r1, -(160+16*PS)(r1)
get_got r11
addi r6, r1, 16*PS
stvm r6, v20, v21, v22, v23, v24, v25, v26, v27, v28, v29
mfvrsave r0
@ -360,14 +361,14 @@ extfunc ff_fft_calc\interleave\()_altivec
li r6, 0xfffffffc
mtvrsave r6
movrel r6, fft_data
movrel r6, fft_data, r11
lvm r6, v14, v15, v16, v17, v18, v19, v20, v21
lvm r6, v22, v23, v24, v25, v26, v27, v28, v29
li r9, 16
movrel r12, X(ff_cos_tabs)
movrel r12, X(ff_cos_tabs), r11
movrel r6, fft_dispatch_tab\interleave\()_altivec
movrel r6, fft_dispatch_tab\interleave\()_altivec, r11
lwz r3, 0(r3)
subi r3, r3, 2
slwi r3, r3, 2+ARCH_PPC64

121
libavcodec/vmdav.c
View File

@ -465,9 +465,8 @@ static av_cold int vmdvideo_decode_end(AVCodecContext *avctx)
#define BLOCK_TYPE_SILENCE 3
typedef struct VmdAudioContext {
AVCodecContext *avctx;
int out_bps;
int predictors[2];
int chunk_size;
} VmdAudioContext;
static const uint16_t vmdaudio_table[128] = {
@ -490,13 +489,23 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
{
VmdAudioContext *s = avctx->priv_data;
s->avctx = avctx;
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
return AVERROR(EINVAL);
}
if (avctx->block_align < 1) {
av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
return AVERROR(EINVAL);
}
if (avctx->bits_per_coded_sample == 16)
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
else
avctx->sample_fmt = AV_SAMPLE_FMT_U8;
s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2);
av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
"block align = %d, sample rate = %d\n",
avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
@ -505,52 +514,47 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
return 0;
}
static void vmdaudio_decode_audio(VmdAudioContext *s, unsigned char *data,
const uint8_t *buf, int buf_size, int stereo)
static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
int channels)
{
int i;
int chan = 0;
int16_t *out = (int16_t*)data;
int ch;
const uint8_t *buf_end = buf + buf_size;
int predictor[2];
int st = channels - 1;
for(i = 0; i < buf_size; i++) {
if(buf[i] & 0x80)
s->predictors[chan] -= vmdaudio_table[buf[i] & 0x7F];
/* decode initial raw sample */
for (ch = 0; ch < channels; ch++) {
predictor[ch] = (int16_t)AV_RL16(buf);
buf += 2;
*out++ = predictor[ch];
}
/* decode DPCM samples */
ch = 0;
while (buf < buf_end) {
uint8_t b = *buf++;
if (b & 0x80)
predictor[ch] -= vmdaudio_table[b & 0x7F];
else
s->predictors[chan] += vmdaudio_table[buf[i]];
s->predictors[chan] = av_clip_int16(s->predictors[chan]);
out[i] = s->predictors[chan];
chan ^= stereo;
predictor[ch] += vmdaudio_table[b];
predictor[ch] = av_clip_int16(predictor[ch]);
*out++ = predictor[ch];
ch ^= st;
}
}
static int vmdaudio_loadsound(VmdAudioContext *s, unsigned char *data,
const uint8_t *buf, int silent_chunks, int data_size)
{
int silent_size = s->avctx->block_align * silent_chunks * s->out_bps;
if (silent_chunks) {
memset(data, s->out_bps == 2 ? 0x00 : 0x80, silent_size);
data += silent_size;
}
if (s->avctx->bits_per_coded_sample == 16)
vmdaudio_decode_audio(s, data, buf, data_size, s->avctx->channels == 2);
else {
/* just copy the data */
memcpy(data, buf, data_size);
}
return silent_size + data_size * s->out_bps;
}
static int vmdaudio_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
const uint8_t *buf_end;
int buf_size = avpkt->size;
VmdAudioContext *s = avctx->priv_data;
int block_type, silent_chunks;
unsigned char *output_samples = (unsigned char *)data;
int block_type, silent_chunks, audio_chunks;
int nb_samples, out_size;
uint8_t *output_samples_u8 = data;
int16_t *output_samples_s16 = data;
if (buf_size < 16) {
av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
@ -566,13 +570,16 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
buf += 16;
buf_size -= 16;
/* get number of silent chunks */
silent_chunks = 0;
if (block_type == BLOCK_TYPE_INITIAL) {
uint32_t flags;
if (buf_size < 4)
return -1;
flags = AV_RB32(buf);
silent_chunks = av_popcount(flags);
if (buf_size < 4) {
av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
return AVERROR(EINVAL);
}
flags = AV_RB32(buf);
silent_chunks = av_popcount(flags);
buf += 4;
buf_size -= 4;
} else if (block_type == BLOCK_TYPE_SILENCE) {
@ -581,11 +588,41 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
}
/* ensure output buffer is large enough */
if (*data_size < (avctx->block_align*silent_chunks + buf_size) * s->out_bps)
audio_chunks = buf_size / s->chunk_size;
nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) / avctx->channels;
out_size = nb_samples * avctx->channels * s->out_bps;
if (*data_size < out_size)
return -1;
*data_size = vmdaudio_loadsound(s, output_samples, buf, silent_chunks, buf_size);
/* decode silent chunks */
if (silent_chunks > 0) {
int silent_size = avctx->block_align * silent_chunks;
if (s->out_bps == 2) {
memset(output_samples_s16, 0x00, silent_size * 2);
output_samples_s16 += silent_size;
} else {
memset(output_samples_u8, 0x80, silent_size);
output_samples_u8 += silent_size;
}
}
/* decode audio chunks */
if (audio_chunks > 0) {
buf_end = buf + buf_size;
while (buf < buf_end) {
if (s->out_bps == 2) {
decode_audio_s16(output_samples_s16, buf, s->chunk_size,
avctx->channels);
output_samples_s16 += avctx->block_align;
} else {
memcpy(output_samples_u8, buf, s->chunk_size);
output_samples_u8 += avctx->block_align;
}
buf += s->chunk_size;
}
}
*data_size = out_size;
return avpkt->size;
}

6
libswscale/ppc/swscale_altivec.c
View File

@ -242,7 +242,7 @@ static void hScale_altivec_real(SwsContext *c, int16_t *dst, int dstW,
vector unsigned char src_v1, src_vF;
vector signed short src_v, filter_v;
vector signed int val_vEven, val_s;
if ((((int)src + srcPos)% 16) > 12) {
if ((((uintptr_t)src + srcPos) % 16) > 12) {
src_v1 = vec_ld(srcPos + 16, src);
}
src_vF = vec_perm(src_v0, src_v1, vec_lvsl(srcPos, src));
@ -281,7 +281,7 @@ static void hScale_altivec_real(SwsContext *c, int16_t *dst, int dstW,
vector unsigned char src_v1, src_vF;
vector signed short src_v, filter_v;
vector signed int val_v, val_s;
if ((((int)src + srcPos)% 16) > 8) {
if ((((uintptr_t)src + srcPos) % 16) > 8) {
src_v1 = vec_ld(srcPos + 16, src);
}
src_vF = vec_perm(src_v0, src_v1, vec_lvsl(srcPos, src));
@ -367,7 +367,7 @@ static void hScale_altivec_real(SwsContext *c, int16_t *dst, int dstW,
//vector unsigned char src_v0 = vec_ld(srcPos + j, src);
vector unsigned char src_v1, src_vF;
vector signed short src_v, filter_v1R, filter_v;
if ((((int)src + srcPos)% 16) > 8) {
if ((((uintptr_t)src + srcPos) % 16) > 8) {
src_v1 = vec_ld(srcPos + j + 16, src);
}
src_vF = vec_perm(src_v0, src_v1, permS);