1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00

Merge commit 'cbcd497f384f0f8ef3f76f85b29b644b900d6b9f'

* commit 'cbcd497f384f0f8ef3f76f85b29b644b900d6b9f':
  adxdec: use planar sample format
  adpcmdec: use planar sample format for adpcm_thp
  adpcmdec: use planar sample format for adpcm_ea_xas
  adpcmdec: use planar sample format for adpcm_ea_r1/r2/r3
  adpcmdec: use planar sample format for adpcm_xa
  adpcmdec: use planar sample format for adpcm_ima_ws for vqa version 3
  adpcmdec: use planar sample format for adpcm_4xm
  adpcmdec: use planar sample format for adpcm_ima_wav
  adpcmdec: use planar sample format for adpcm_ima_qt
  pcmdec: use planar sample format for pcm_lxf
  mace: use planar sample format
  atrac1: use planar sample format
  build: non-x86: Only compile mpegvideo optimizations when necessary
  rtpdec_mpeg4: au_headers is a single array, simple av_free is enough
  avcodec: free extended_data instead address of it
  fate: Add tests of the ff_make_absolute_url function
  url: Handle relative urls starting with two slashes
  url: Handle relative urls being just a new query string
  url: Don't treat slashes in query parameters as directory separators

Conflicts:
	libavcodec/adxdec.c
	libavcodec/mips/Makefile
	libavcodec/pcm.c
	libavcodec/utils.c
	libavformat/Makefile
	libavformat/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-10-10 13:00:28 +02:00
commit eadba3e94d
18 changed files with 239 additions and 159 deletions

View File

@ -134,7 +134,26 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
default:
break;
}
switch(avctx->codec->id) {
case AV_CODEC_ID_ADPCM_IMA_QT:
case AV_CODEC_ID_ADPCM_IMA_WAV:
case AV_CODEC_ID_ADPCM_4XM:
case AV_CODEC_ID_ADPCM_XA:
case AV_CODEC_ID_ADPCM_EA_R1:
case AV_CODEC_ID_ADPCM_EA_R2:
case AV_CODEC_ID_ADPCM_EA_R3:
case AV_CODEC_ID_ADPCM_EA_XAS:
case AV_CODEC_ID_ADPCM_THP:
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
break;
case AV_CODEC_ID_ADPCM_IMA_WS:
avctx->sample_fmt = c->vqa_version == 3 ? AV_SAMPLE_FMT_S16P :
AV_SAMPLE_FMT_S16;
break;
default:
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
}
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
@ -264,17 +283,22 @@ static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned c
return c->predictor;
}
static int xa_decode(AVCodecContext *avctx,
short *out, const unsigned char *in,
ADPCMChannelStatus *left, ADPCMChannelStatus *right, int inc)
static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
const uint8_t *in, ADPCMChannelStatus *left,
ADPCMChannelStatus *right, int channels, int sample_offset)
{
int i, j;
int shift,filter,f0,f1;
int s_1,s_2;
int d,s,t;
for(i=0;i<4;i++) {
out0 += sample_offset;
if (channels == 1)
out1 = out0 + 28;
else
out1 += sample_offset;
for(i=0;i<4;i++) {
shift = 12 - (in[4+i*2] & 15);
filter = in[4+i*2] >> 4;
if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table)) {
@ -294,16 +318,14 @@ static int xa_decode(AVCodecContext *avctx,
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
*out = s_1;
out += inc;
out0[j] = s_1;
}
if (inc==2) { /* stereo */
if (channels == 2) {
left->sample1 = s_1;
left->sample2 = s_2;
s_1 = right->sample1;
s_2 = right->sample2;
out = out + 1 - 28*2;
}
shift = 12 - (in[5+i*2] & 15);
@ -323,18 +345,19 @@ static int xa_decode(AVCodecContext *avctx,
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
s_2 = s_1;
s_1 = av_clip_int16(s);
*out = s_1;
out += inc;
out1[j] = s_1;
}
if (inc==2) { /* stereo */
if (channels == 2) {
right->sample1 = s_1;
right->sample2 = s_2;
out -= 1;
} else {
left->sample1 = s_1;
left->sample2 = s_2;
}
out0 += 28 * (3 - channels);
out1 += 28 * (3 - channels);
}
return 0;
@ -573,6 +596,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
ADPCMChannelStatus *cs;
int n, m, channel, i;
short *samples;
int16_t **samples_p;
int st; /* stereo */
int count1, count2;
int nb_samples, coded_samples, ret;
@ -592,6 +616,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
return ret;
}
samples = (short *)c->frame.data[0];
samples_p = (int16_t **)c->frame.extended_data;
/* use coded_samples when applicable */
/* it is always <= nb_samples, so the output buffer will be large enough */
@ -636,21 +661,19 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
return AVERROR_INVALIDDATA;
}
samples = (short *)c->frame.data[0] + channel;
samples = samples_p[channel];
for (m = 0; m < 32; m++) {
for (m = 0; m < 64; m += 2) {
int byte = bytestream2_get_byteu(&gb);
*samples = adpcm_ima_qt_expand_nibble(cs, byte & 0x0F, 3);
samples += avctx->channels;
*samples = adpcm_ima_qt_expand_nibble(cs, byte >> 4 , 3);
samples += avctx->channels;
samples[m ] = adpcm_ima_qt_expand_nibble(cs, byte & 0x0F, 3);
samples[m + 1] = adpcm_ima_qt_expand_nibble(cs, byte >> 4 , 3);
}
}
break;
case AV_CODEC_ID_ADPCM_IMA_WAV:
for(i=0; i<avctx->channels; i++){
cs = &(c->status[i]);
cs->predictor = *samples++ = sign_extend(bytestream2_get_le16u(&gb), 16);
cs->predictor = samples_p[i][0] = sign_extend(bytestream2_get_le16u(&gb), 16);
cs->step_index = sign_extend(bytestream2_get_le16u(&gb), 16);
if (cs->step_index > 88u){
@ -660,19 +683,16 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
}
}
for (n = (nb_samples - 1) / 8; n > 0; n--) {
for (n = 0; n < (nb_samples - 1) / 8; n++) {
for (i = 0; i < avctx->channels; i++) {
cs = &c->status[i];
for (m = 0; m < 4; m++) {
samples = &samples_p[i][1 + n * 8];
for (m = 0; m < 8; m += 2) {
int v = bytestream2_get_byteu(&gb);
*samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 3);
samples += avctx->channels;
*samples = adpcm_ima_expand_nibble(cs, v >> 4 , 3);
samples += avctx->channels;
samples[m ] = adpcm_ima_expand_nibble(cs, v & 0x0F, 3);
samples[m + 1] = adpcm_ima_expand_nibble(cs, v >> 4 , 3);
}
samples -= 8 * avctx->channels - 1;
}
samples += 7 * avctx->channels;
}
break;
case AV_CODEC_ID_ADPCM_4XM:
@ -689,14 +709,12 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
}
for (i = 0; i < avctx->channels; i++) {
samples = (short *)c->frame.data[0] + i;
samples = (int16_t *)c->frame.data[i];
cs = &c->status[i];
for (n = nb_samples >> 1; n > 0; n--) {
int v = bytestream2_get_byteu(&gb);
*samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 4);
samples += avctx->channels;
*samples = adpcm_ima_expand_nibble(cs, v >> 4 , 4);
samples += avctx->channels;
*samples++ = adpcm_ima_expand_nibble(cs, v & 0x0F, 4);
*samples++ = adpcm_ima_expand_nibble(cs, v >> 4 , 4);
}
}
break;
@ -858,14 +876,12 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
case AV_CODEC_ID_ADPCM_IMA_WS:
if (c->vqa_version == 3) {
for (channel = 0; channel < avctx->channels; channel++) {
int16_t *smp = samples + channel;
int16_t *smp = samples_p[channel];
for (n = nb_samples / 2; n > 0; n--) {
int v = bytestream2_get_byteu(&gb);
*smp = adpcm_ima_expand_nibble(&c->status[channel], v >> 4 , 3);
smp += avctx->channels;
*smp = adpcm_ima_expand_nibble(&c->status[channel], v & 0x0F, 3);
smp += avctx->channels;
*smp++ = adpcm_ima_expand_nibble(&c->status[channel], v >> 4 , 3);
*smp++ = adpcm_ima_expand_nibble(&c->status[channel], v & 0x0F, 3);
}
}
} else {
@ -881,14 +897,21 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
bytestream2_seek(&gb, 0, SEEK_END);
break;
case AV_CODEC_ID_ADPCM_XA:
{
int16_t *out0 = samples_p[0];
int16_t *out1 = samples_p[1];
int samples_per_block = 28 * (3 - avctx->channels) * 4;
int sample_offset = 0;
while (bytestream2_get_bytes_left(&gb) >= 128) {
if ((ret = xa_decode(avctx, samples, buf + bytestream2_tell(&gb), &c->status[0],
&c->status[1], avctx->channels)) < 0)
if ((ret = xa_decode(avctx, out0, out1, buf + bytestream2_tell(&gb),
&c->status[0], &c->status[1],
avctx->channels, sample_offset)) < 0)
return ret;
bytestream2_skipu(&gb, 128);
samples += 28 * 8;
sample_offset += samples_per_block;
}
break;
}
case AV_CODEC_ID_ADPCM_IMA_EA_EACS:
for (i=0; i<=st; i++) {
c->status[i].step_index = bytestream2_get_le32u(&gb);
@ -1022,7 +1045,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
for (channel=0; channel<avctx->channels; channel++) {
bytestream2_seek(&gb, offsets[channel], SEEK_SET);
samplesC = samples + channel;
samplesC = samples_p[channel];
if (avctx->codec->id == AV_CODEC_ID_ADPCM_EA_R1) {
current_sample = sign_extend(bytestream2_get_le16(&gb), 16);
@ -1038,10 +1061,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
current_sample = sign_extend(bytestream2_get_be16(&gb), 16);
previous_sample = sign_extend(bytestream2_get_be16(&gb), 16);
for (count2=0; count2<28; count2++) {
*samplesC = sign_extend(bytestream2_get_be16(&gb), 16);
samplesC += avctx->channels;
}
for (count2=0; count2<28; count2++)
*samplesC++ = sign_extend(bytestream2_get_be16(&gb), 16);
} else {
coeff1 = ea_adpcm_table[ byte >> 4 ];
coeff2 = ea_adpcm_table[(byte >> 4) + 4];
@ -1061,8 +1082,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
previous_sample = current_sample;
current_sample = next_sample;
*samplesC = current_sample;
samplesC += avctx->channels;
*samplesC++ = current_sample;
}
}
}
@ -1086,8 +1106,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
case AV_CODEC_ID_ADPCM_EA_XAS:
for (channel=0; channel<avctx->channels; channel++) {
int coeff[2][4], shift[4];
short *s2, *s = &samples[channel];
for (n=0; n<4; n++, s+=32*avctx->channels) {
int16_t *s = samples_p[channel];
for (n = 0; n < 4; n++, s += 32) {
int val = sign_extend(bytestream2_get_le16u(&gb), 16);
for (i=0; i<2; i++)
coeff[i][n] = ea_adpcm_table[(val&0x0F)+4*i];
@ -1095,19 +1115,22 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
val = sign_extend(bytestream2_get_le16u(&gb), 16);
shift[n] = 20 - (val & 0x0F);
s[avctx->channels] = val & ~0x0F;
s[1] = val & ~0x0F;
}
for (m=2; m<32; m+=2) {
s = &samples[m*avctx->channels + channel];
for (n=0; n<4; n++, s+=32*avctx->channels) {
s = &samples_p[channel][m];
for (n = 0; n < 4; n++, s += 32) {
int level, pred;
int byte = bytestream2_get_byteu(&gb);
for (s2=s, i=0; i<8; i+=4, s2+=avctx->channels) {
int level = sign_extend(byte >> (4 - i), 4) << shift[n];
int pred = s2[-1*avctx->channels] * coeff[0][n]
+ s2[-2*avctx->channels] * coeff[1][n];
s2[0] = av_clip_int16((level + pred + 0x80) >> 8);
}
level = sign_extend(byte >> 4, 4) << shift[n];
pred = s[-1] * coeff[0][n] + s[-2] * coeff[1][n];
s[0] = av_clip_int16((level + pred + 0x80) >> 8);
level = sign_extend(byte, 4) << shift[n];
pred = s[0] * coeff[0][n] + s[-1] * coeff[1][n];
s[1] = av_clip_int16((level + pred + 0x80) >> 8);
}
}
}
@ -1221,7 +1244,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
prev[i][n] = sign_extend(bytestream2_get_be16u(&gb), 16);
for (ch = 0; ch <= st; ch++) {
samples = (short *)c->frame.data[0] + ch;
samples = samples_p[ch];
/* Read in every sample for this channel. */
for (i = 0; i < nb_samples / 14; i++) {
@ -1247,10 +1270,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
*samples = av_clip_int16(sampledat);
prev[ch][1] = prev[ch][0];
prev[ch][0] = *samples++;
/* In case of stereo, skip one sample, this sample
is for the other channel. */
samples += st;
}
}
}
@ -1270,6 +1289,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
static const enum AVSampleFormat sample_fmts_s16[] = { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE };
static const enum AVSampleFormat sample_fmts_s16p[] = { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE };
static const enum AVSampleFormat sample_fmts_both[] = { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE };
#define ADPCM_DECODER(id_, sample_fmts_, name_, long_name_) \
AVCodec ff_ ## name_ ## _decoder = { \
@ -1285,14 +1309,14 @@ AVCodec ff_ ## name_ ## _decoder = { \
}
/* Note: Do not forget to add new entries to the Makefile as well. */
ADPCM_DECODER(AV_CODEC_ID_ADPCM_4XM, sample_fmts_s16, adpcm_4xm, "ADPCM 4X Movie");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_4XM, sample_fmts_s16p, adpcm_4xm, "ADPCM 4X Movie");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_CT, sample_fmts_s16, adpcm_ct, "ADPCM Creative Technology");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA, sample_fmts_s16, adpcm_ea, "ADPCM Electronic Arts");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_MAXIS_XA, sample_fmts_s16, adpcm_ea_maxis_xa, "ADPCM Electronic Arts Maxis CDROM XA");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R1, sample_fmts_s16, adpcm_ea_r1, "ADPCM Electronic Arts R1");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R2, sample_fmts_s16, adpcm_ea_r2, "ADPCM Electronic Arts R2");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R3, sample_fmts_s16, adpcm_ea_r3, "ADPCM Electronic Arts R3");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_XAS, sample_fmts_s16, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R1, sample_fmts_s16p, adpcm_ea_r1, "ADPCM Electronic Arts R1");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R2, sample_fmts_s16p, adpcm_ea_r2, "ADPCM Electronic Arts R2");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R3, sample_fmts_s16p, adpcm_ea_r3, "ADPCM Electronic Arts R3");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_XAS, sample_fmts_s16p, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_AMV, sample_fmts_s16, adpcm_ima_amv, "ADPCM IMA AMV");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_APC, sample_fmts_s16, adpcm_ima_apc, "ADPCM IMA CRYO APC");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_DK3, sample_fmts_s16, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
@ -1300,15 +1324,15 @@ ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_DK4, sample_fmts_s16, adpcm_ima_dk4,
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_EA_EACS, sample_fmts_s16, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_EA_SEAD, sample_fmts_s16, adpcm_ima_ea_sead, "ADPCM IMA Electronic Arts SEAD");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_ISS, sample_fmts_s16, adpcm_ima_iss, "ADPCM IMA Funcom ISS");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_QT, sample_fmts_s16, adpcm_ima_qt, "ADPCM IMA QuickTime");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_QT, sample_fmts_s16p, adpcm_ima_qt, "ADPCM IMA QuickTime");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_SMJPEG, sample_fmts_s16, adpcm_ima_smjpeg, "ADPCM IMA Loki SDL MJPEG");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WAV, sample_fmts_s16, adpcm_ima_wav, "ADPCM IMA WAV");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WS, sample_fmts_s16, adpcm_ima_ws, "ADPCM IMA Westwood");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WAV, sample_fmts_s16p, adpcm_ima_wav, "ADPCM IMA WAV");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WS, sample_fmts_both, adpcm_ima_ws, "ADPCM IMA Westwood");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_MS, sample_fmts_s16, adpcm_ms, "ADPCM Microsoft");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_2, sample_fmts_s16, adpcm_sbpro_2, "ADPCM Sound Blaster Pro 2-bit");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_3, sample_fmts_s16, adpcm_sbpro_3, "ADPCM Sound Blaster Pro 2.6-bit");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SBPRO_4, sample_fmts_s16, adpcm_sbpro_4, "ADPCM Sound Blaster Pro 4-bit");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_SWF, sample_fmts_s16, adpcm_swf, "ADPCM Shockwave Flash");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_THP, sample_fmts_s16, adpcm_thp, "ADPCM Nintendo Gamecube THP");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_XA, sample_fmts_s16, adpcm_xa, "ADPCM CDROM XA");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_THP, sample_fmts_s16p, adpcm_thp, "ADPCM Nintendo Gamecube THP");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_XA, sample_fmts_s16p, adpcm_xa, "ADPCM CDROM XA");
ADPCM_DECODER(AV_CODEC_ID_ADPCM_YAMAHA, sample_fmts_s16, adpcm_yamaha, "ADPCM Yamaha");

View File

@ -49,7 +49,7 @@ static av_cold int adx_decode_init(AVCodecContext *avctx)
c->header_parsed = 1;
}
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
@ -64,7 +64,8 @@ static av_cold int adx_decode_init(AVCodecContext *avctx)
* 2nd-order LPC filter applied to it to form the output signal for a single
* channel.
*/
static int adx_decode(ADXContext *c, int16_t *out, const uint8_t *in, int ch)
static int adx_decode(ADXContext *c, int16_t *out, int offset,
const uint8_t *in, int ch)
{
ADXChannelState *prev = &c->prev[ch];
GetBitContext gb;
@ -77,6 +78,7 @@ static int adx_decode(ADXContext *c, int16_t *out, const uint8_t *in, int ch)
return -1;
init_get_bits(&gb, in + 2, (BLOCK_SIZE - 2) * 8);
out += offset;
s1 = prev->s1;
s2 = prev->s2;
for (i = 0; i < BLOCK_SAMPLES; i++) {
@ -84,8 +86,7 @@ static int adx_decode(ADXContext *c, int16_t *out, const uint8_t *in, int ch)
s0 = ((d << COEFF_BITS) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
s2 = s1;
s1 = av_clip_int16(s0);
*out = s1;
out += c->channels;
*out++ = s1;
}
prev->s1 = s1;
prev->s2 = s2;
@ -98,7 +99,8 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
{
int buf_size = avpkt->size;
ADXContext *c = avctx->priv_data;
int16_t *samples;
int16_t **samples;
int samples_offset;
const uint8_t *buf = avpkt->data;
const uint8_t *buf_end = buf + avpkt->size;
int num_blocks, ch, ret;
@ -145,11 +147,12 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)c->frame.data[0];
samples = (int16_t **)c->frame.extended_data;
samples_offset = 0;
while (num_blocks--) {
for (ch = 0; ch < c->channels; ch++) {
if (buf_end - buf < BLOCK_SIZE || adx_decode(c, samples + ch, buf, ch)) {
if (buf_end - buf < BLOCK_SIZE || adx_decode(c, samples[ch], samples_offset, buf, ch)) {
c->eof = 1;
buf = avpkt->data + avpkt->size;
break;
@ -157,7 +160,7 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
buf_size -= BLOCK_SIZE;
buf += BLOCK_SIZE;
}
samples += BLOCK_SAMPLES * c->channels;
samples_offset += BLOCK_SAMPLES;
}
*got_frame_ptr = 1;
@ -183,4 +186,6 @@ AVCodec ff_adpcm_adx_decoder = {
.flush = adx_decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};

View File

@ -2,5 +2,6 @@ OBJS += alpha/dsputil_alpha.o \
alpha/dsputil_alpha_asm.o \
alpha/motion_est_alpha.o \
alpha/motion_est_mvi_asm.o \
alpha/mpegvideo_alpha.o \
alpha/simple_idct_alpha.o \
OBJS-$(CONFIG_MPEGVIDEO) += alpha/mpegvideo_alpha.o

View File

@ -14,6 +14,7 @@ OBJS-$(CONFIG_FLAC_DECODER) += arm/flacdsp_init_arm.o \
OBJS-$(CONFIG_MPEGAUDIODSP) += arm/mpegaudiodsp_init_arm.o
ARMV6-OBJS-$(CONFIG_MPEGAUDIODSP) += arm/mpegaudiodsp_fixed_armv6.o
OBJS-$(CONFIG_MPEGVIDEO) += arm/mpegvideo_arm.o
OBJS-$(CONFIG_VP3DSP) += arm/vp3dsp_init_arm.o
OBJS-$(CONFIG_VP5_DECODER) += arm/vp56dsp_init_arm.o
OBJS-$(CONFIG_VP6_DECODER) += arm/vp56dsp_init_arm.o
@ -31,12 +32,12 @@ OBJS += arm/dsputil_init_arm.o \
arm/fft_fixed_init_arm.o \
arm/fmtconvert_init_arm.o \
arm/jrevdct_arm.o \
arm/mpegvideo_arm.o \
arm/simple_idct_arm.o \
ARMV5TE-OBJS += arm/dsputil_init_armv5te.o \
arm/mpegvideo_armv5te.o \
ARMV5TE-OBJS-$(CONFIG_MPEGVIDEO) += arm/mpegvideo_armv5te.o \
arm/mpegvideo_armv5te_s.o \
ARMV5TE-OBJS += arm/dsputil_init_armv5te.o \
arm/simple_idct_armv5te.o \
ARMV6-OBJS += arm/dsputil_init_armv6.o \
@ -70,6 +71,7 @@ NEON-OBJS-$(CONFIG_AAC_DECODER) += arm/sbrdsp_neon.o \
NEON-OBJS-$(CONFIG_DCA_DECODER) += arm/dcadsp_neon.o \
arm/synth_filter_neon.o \
NEON-OBJS-$(CONFIG_MPEGVIDEO) += arm/mpegvideo_neon.o
NEON-OBJS-$(CONFIG_RV30_DECODER) += arm/rv34dsp_init_neon.o \
arm/rv34dsp_neon.o \
@ -92,5 +94,4 @@ NEON-OBJS += arm/dsputil_init_neon.o \
arm/dsputil_neon.o \
arm/fmtconvert_neon.o \
arm/int_neon.o \
arm/mpegvideo_neon.o \
arm/simple_idct_neon.o \

View File

@ -36,7 +36,6 @@
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#include "fmtconvert.h"
#include "sinewin.h"
#include "atrac.h"
@ -80,11 +79,9 @@ typedef struct {
DECLARE_ALIGNED(32, float, mid)[256];
DECLARE_ALIGNED(32, float, high)[512];
float* bands[3];
float *out_samples[AT1_MAX_CHANNELS];
FFTContext mdct_ctx[3];
int channels;
DSPContext dsp;
FmtConvertContext fmt_conv;
} AT1Ctx;
/** size of the transform in samples in the long mode for each QMF band */
@ -281,7 +278,6 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
AT1Ctx *q = avctx->priv_data;
int ch, ret;
GetBitContext gb;
float *samples;
if (buf_size < 212 * q->channels) {
@ -295,7 +291,6 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (float *)q->frame.data[0];
for (ch = 0; ch < q->channels; ch++) {
AT1SUCtx* su = &q->SUs[ch];
@ -314,13 +309,7 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
ret = at1_imdct_block(su, q);
if (ret < 0)
return ret;
at1_subband_synthesis(q, su, q->channels == 1 ? samples : q->out_samples[ch]);
}
/* interleave */
if (q->channels == 2) {
q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
AT1_SU_SAMPLES, 2);
at1_subband_synthesis(q, su, (float *)q->frame.extended_data[ch]);
}
*got_frame_ptr = 1;
@ -334,8 +323,6 @@ static av_cold int atrac1_decode_end(AVCodecContext * avctx)
{
AT1Ctx *q = avctx->priv_data;
av_freep(&q->out_samples[0]);
ff_mdct_end(&q->mdct_ctx[0]);
ff_mdct_end(&q->mdct_ctx[1]);
ff_mdct_end(&q->mdct_ctx[2]);
@ -349,7 +336,7 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
AT1Ctx *q = avctx->priv_data;
int ret;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
@ -358,15 +345,6 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
}
q->channels = avctx->channels;
if (avctx->channels == 2) {
q->out_samples[0] = av_malloc(2 * AT1_SU_SAMPLES * sizeof(*q->out_samples[0]));
q->out_samples[1] = q->out_samples[0] + AT1_SU_SAMPLES;
if (!q->out_samples[0]) {
av_freep(&q->out_samples[0]);
return AVERROR(ENOMEM);
}
}
/* Init the mdct transforms */
if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
(ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
@ -381,7 +359,6 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx)
ff_atrac_generate_tables();
ff_dsputil_init(&q->dsp, avctx);
ff_fmt_convert_init(&q->fmt_conv, avctx);
q->bands[0] = q->low;
q->bands[1] = q->mid;
@ -410,4 +387,6 @@ AVCodec ff_atrac1_decoder = {
.decode = atrac1_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};

View File

@ -1,7 +1,8 @@
OBJS += bfin/dsputil_bfin.o \
bfin/fdct_bfin.o \
bfin/idct_bfin.o \
bfin/mpegvideo_bfin.o \
bfin/pixels_bfin.o \
bfin/vp3_bfin.o \
bfin/vp3_idct_bfin.o \
OBJS-$(CONFIG_MPEGVIDEOENC) += bfin/mpegvideo_bfin.o

View File

@ -187,9 +187,7 @@ static int16_t read_table(ChannelData *chd, uint8_t val, int tab_idx)
return current;
}
static void chomp3(ChannelData *chd, int16_t *output, uint8_t val,
int tab_idx,
uint32_t numChannels)
static void chomp3(ChannelData *chd, int16_t *output, uint8_t val, int tab_idx)
{
int16_t current = read_table(chd, val, tab_idx);
@ -200,9 +198,7 @@ static void chomp3(ChannelData *chd, int16_t *output, uint8_t val,
*output = QT_8S_2_16S(current);
}
static void chomp6(ChannelData *chd, int16_t *output, uint8_t val,
int tab_idx,
uint32_t numChannels)
static void chomp6(ChannelData *chd, int16_t *output, uint8_t val, int tab_idx)
{
int16_t current = read_table(chd, val, tab_idx);
@ -222,7 +218,7 @@ static void chomp6(ChannelData *chd, int16_t *output, uint8_t val,
output[0] = QT_8S_2_16S(chd->previous + chd->prev2 -
((chd->prev2-current) >> 2));
output[numChannels] = QT_8S_2_16S(chd->previous + current +
output[1] = QT_8S_2_16S(chd->previous + current +
((chd->prev2-current) >> 2));
chd->prev2 = chd->previous;
chd->previous = current;
@ -234,7 +230,7 @@ static av_cold int mace_decode_init(AVCodecContext * avctx)
if (avctx->channels > 2 || avctx->channels <= 0)
return -1;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
avcodec_get_frame_defaults(&ctx->frame);
avctx->coded_frame = &ctx->frame;
@ -247,7 +243,7 @@ static int mace_decode_frame(AVCodecContext *avctx, void *data,
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *samples;
int16_t **samples;
MACEContext *ctx = avctx->priv_data;
int i, j, k, l, ret;
int is_mace3 = (avctx->codec_id == AV_CODEC_ID_MACE3);
@ -258,10 +254,10 @@ static int mace_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)ctx->frame.data[0];
samples = (int16_t **)ctx->frame.extended_data;
for(i = 0; i < avctx->channels; i++) {
int16_t *output = samples + i;
int16_t *output = samples[i];
for (j=0; j < buf_size / (avctx->channels << is_mace3); j++)
for (k=0; k < (1 << is_mace3); k++) {
@ -273,13 +269,11 @@ static int mace_decode_frame(AVCodecContext *avctx, void *data,
for (l=0; l < 3; l++) {
if (is_mace3)
chomp3(&ctx->chd[i], output, val[1][l], l,
avctx->channels);
chomp3(&ctx->chd[i], output, val[1][l], l);
else
chomp6(&ctx->chd[i], output, val[0][l], l,
avctx->channels);
chomp6(&ctx->chd[i], output, val[0][l], l);
output += avctx->channels << (1-is_mace3);
output += 1 << (1-is_mace3);
}
}
}
@ -299,6 +293,8 @@ AVCodec ff_mace3_decoder = {
.decode = mace_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MACE (Macintosh Audio Compression/Expansion) 3:1"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};
AVCodec ff_mace6_decoder = {
@ -310,4 +306,6 @@ AVCodec ff_mace6_decoder = {
.decode = mace_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MACE (Macintosh Audio Compression/Expansion) 6:1"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};

View File

@ -1,6 +1,7 @@
MMI-OBJS += mips/dsputil_mmi.o \
mips/idct_mmi.o \
mips/mpegvideo_mmi.o
MMI-OBJS-$(CONFIG_MPEGVIDEO) += mips/mpegvideo_mmi.o
MIPSFPU-OBJS-$(CONFIG_AMRNB_DECODER) += mips/acelp_filters_mips.o \
mips/celp_filters_mips.o \

View File

@ -446,15 +446,14 @@ static int pcm_decode_frame(AVCodecContext *avctx, void *data,
{
int i;
n /= avctx->channels;
//unpack
for (c = 0; c < avctx->channels; c++) {
dst_int32_t = (int32_t *)s->frame.data[c];
dst_int32_t = (int32_t *)s->frame.extended_data[c];
for (i = 0; i < n; i++) {
// extract low 20 bits and expand to 32 bits
*dst_int32_t++ = (src[2] << 28) |
(src[1] << 20) |
(src[0] << 12) |
((src[2] & 0xF) << 8) |
((src[2] & 0x0F) << 8) |
src[1];
// extract high 20 bits and expand to 32 bits
*dst_int32_t++ = (src[4] << 24) |

View File

@ -7,6 +7,7 @@ ALTIVEC-OBJS-$(CONFIG_FFT) += ppc/fft_altivec.o \
$(FFT-OBJS-yes)
ALTIVEC-OBJS-$(CONFIG_H264DSP) += ppc/h264_altivec.o
ALTIVEC-OBJS-$(CONFIG_MPEGAUDIODSP) += ppc/mpegaudiodec_altivec.o
ALTIVEC-OBJS-$(CONFIG_MPEGVIDEO) += ppc/mpegvideo_altivec.o
ALTIVEC-OBJS-$(CONFIG_VC1_DECODER) += ppc/vc1dsp_altivec.o
ALTIVEC-OBJS-$(CONFIG_VP8_DECODER) += ppc/vp8dsp_altivec.o
@ -17,4 +18,3 @@ ALTIVEC-OBJS += ppc/dsputil_altivec.o \
ppc/gmc_altivec.o \
ppc/idct_altivec.o \
ppc/int_altivec.o \
ppc/mpegvideo_altivec.o \

View File

@ -393,7 +393,7 @@ static int audio_get_buffer(AVCodecContext *avctx, AVFrame *frame)
if (buf->extended_data[0] && buf_size > buf->audio_data_size) {
av_free(buf->extended_data[0]);
if (buf->extended_data != buf->data)
av_freep(&buf->extended_data);
av_free(buf->extended_data);
buf->extended_data = NULL;
buf->data[0] = NULL;
}

View File

@ -412,7 +412,8 @@ OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o
SKIPHEADERS-$(CONFIG_FFRTMPCRYPT_PROTOCOL) += rtmpdh.h
SKIPHEADERS-$(CONFIG_NETWORK) += network.h rtsp.h
TESTPROGS = seek
TESTPROGS = seek \
url \
TOOLS = aviocat \
ismindex \

View File

@ -93,15 +93,7 @@ static PayloadContext *new_context(void)
static void free_context(PayloadContext * data)
{
int i;
for (i = 0; i < data->nb_au_headers; i++) {
/* according to rtp_parse_mp4_au, we treat multiple
* au headers as one, so nb_au_headers is always 1.
* loop anyway in case this changes.
* (note: changes done carelessly might lead to a double free)
*/
av_free(&data->au_headers[i]);
}
av_free(data->au_headers);
av_free(data->mode);
av_free(data);
}

58
libavformat/url-test.c Normal file
View File

@ -0,0 +1,58 @@
/*
* Copyright (c) 2012 Martin Storsjo
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "internal.h"
#undef printf
#undef exit
static void test(const char *base, const char *rel)
{
char buf[200], buf2[200];
ff_make_absolute_url(buf, sizeof(buf), base, rel);
printf("%s\n", buf);
if (base) {
/* Test in-buffer replacement */
snprintf(buf2, sizeof(buf2), "%s", base);
ff_make_absolute_url(buf2, sizeof(buf2), buf2, rel);
if (strcmp(buf, buf2)) {
printf("In-place handling of %s + %s failed\n", base, rel);
exit(1);
}
}
}
int main(void)
{
test(NULL, "baz");
test("/foo/bar", "baz");
test("/foo/bar", "../baz");
test("/foo/bar", "/baz");
test("http://server/foo/", "baz");
test("http://server/foo/bar", "baz");
test("http://server/foo/", "../baz");
test("http://server/foo/bar/123", "../../baz");
test("http://server/foo/bar/123", "/baz");
test("http://server/foo/bar/123", "https://other/url");
test("http://server/foo/bar?param=value/with/slashes", "/baz");
test("http://server/foo/bar?param&otherparam", "?someparam");
test("http://server/foo/bar", "//other/url");
return 0;
}

View File

@ -3824,9 +3824,9 @@ void ff_make_absolute_url(char *buf, int size, const char *base,
sep = strstr(buf, "://");
if (sep) {
/* Take scheme from base url */
if (rel[1] == '/')
if (rel[1] == '/') {
sep[1] = '\0';
else {
} else {
/* Take scheme and host from base url */
sep += 3;
sep = strchr(sep, '/');

View File

@ -69,6 +69,7 @@ include $(SRC_PATH)/tests/fate/h264.mak
include $(SRC_PATH)/tests/fate/image.mak
include $(SRC_PATH)/tests/fate/indeo.mak
include $(SRC_PATH)/tests/fate/libavcodec.mak
include $(SRC_PATH)/tests/fate/libavformat.mak
include $(SRC_PATH)/tests/fate/libavutil.mak
include $(SRC_PATH)/tests/fate/mapchan.mak
include $(SRC_PATH)/tests/fate/lossless-audio.mak
@ -108,6 +109,7 @@ FATE-$(CONFIG_FFMPEG) += $(FATE_FFMPEG)
FATE-$(CONFIG_FFPROBE) += $(FATE_FFPROBE)
FATE-$(CONFIG_AVCODEC) += $(FATE_LIBAVCODEC)
FATE-$(CONFIG_AVFORMAT) += $(FATE_LIBAVFORMAT)
FATE_EXTERN-$(CONFIG_FFMPEG) += $(FATE_SAMPLES_AVCONV) $(FATE_SAMPLES_FFMPEG)
FATE_EXTERN += $(FATE_EXTERN-yes)

View File

@ -0,0 +1,5 @@
FATE_LIBAVFORMAT += fate-url
fate-url: libavformat/url-test$(EXESUF)
fate-url: CMD = run libavformat/url-test
fate-libavformat: $(FATE_LIBAVFORMAT)

13
tests/ref/fate/url Normal file
View File

@ -0,0 +1,13 @@
baz
/foo/baz
/baz
/baz
http://server/foo/baz
http://server/foo/baz
http://server/baz
http://server/baz
http://server/baz
https://other/url
http://server/baz
http://server/foo/bar?someparam
http://other/url