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authorMichael Niedermayer <michaelni@gmx.at>2012-10-15 14:04:35 +0200
committerMichael Niedermayer <michaelni@gmx.at>2012-10-15 14:09:35 +0200
commitae237a117acbe958bea798e32249f4c2baeca5f9 (patch)
treec45fc400e59ab03d5346a82580bfc2054b9efe0f
parent4f5e5a05132be4946a655b991850bf81b7497893 (diff)
parent95cd815c3663603871a1f2da95846e8f72d4ea96 (diff)
downloadffmpeg-ae237a117acbe958bea798e32249f4c2baeca5f9.tar.gz
Merge remote-tracking branch 'qatar/master'
* qatar/master: swscale: try to use mmap only if available configure: check for mprotect wmapro: use planar sample format wmalossless: output in planar sample format wmadec: use float planar sample format output shorten: use planar sample format lavc: update documentation for AVFrame.extended_data Conflicts: libavcodec/shorten.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
-rwxr-xr-xconfigure2
-rw-r--r--libavcodec/avcodec.h2
-rw-r--r--libavcodec/shorten.c15
-rw-r--r--libavcodec/wmadec.c65
-rw-r--r--libavcodec/wmalosslessdec.c17
-rw-r--r--libavcodec/wmaprodec.c17
-rw-r--r--libswscale/utils.c8
7 files changed, 60 insertions, 66 deletions
diff --git a/configure b/configure
index eead2d66be..33dd3b30f1 100755
--- a/configure
+++ b/configure
@@ -1344,6 +1344,7 @@ HAVE_LIST="
mkstemp
mm_empty
mmap
+ mprotect
msvcrt
nanosleep
PeekNamedPipe
@@ -3540,6 +3541,7 @@ check_func localtime_r
check_func ${malloc_prefix}memalign && enable memalign
check_func mkstemp
check_func mmap
+check_func mprotect
check_func ${malloc_prefix}posix_memalign && enable posix_memalign
check_func_headers malloc.h _aligned_malloc && enable aligned_malloc
check_func setrlimit
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 84a1d0ba67..0b3a19af19 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -1066,7 +1066,7 @@ typedef struct AVFrame {
* extended_data must be used by the decoder in order to access all
* channels.
*
- * encoding: unused
+ * encoding: set by user
* decoding: set by AVCodecContext.get_buffer()
*/
uint8_t **extended_data;
diff --git a/libavcodec/shorten.c b/libavcodec/shorten.c
index d04011e468..c36bb9ee2b 100644
--- a/libavcodec/shorten.c
+++ b/libavcodec/shorten.c
@@ -195,7 +195,7 @@ static int init_offset(ShortenContext *s)
break;
case TYPE_S16HL:
case TYPE_S16LH:
- s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
break;
default:
av_log(s->avctx, AV_LOG_ERROR, "unknown audio type\n");
@@ -587,11 +587,11 @@ static int shorten_decode_frame(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- samples_u8 = (uint8_t *)s->frame.data[0];
- samples_s16 = (int16_t *)s->frame.data[0];
- /* interleave output */
- for (i = 0; i < s->blocksize; i++) {
- for (chan = 0; chan < s->channels; chan++) {
+
+ for (chan = 0; chan < s->channels; chan++) {
+ samples_u8 = ((uint8_t **)s->frame.extended_data)[chan];
+ samples_s16 = ((int16_t **)s->frame.extended_data)[chan];
+ for (i = 0; i < s->blocksize; i++) {
switch (s->internal_ftype) {
case TYPE_U8:
*samples_u8++ = av_clip_uint8(s->decoded[chan][i]);
@@ -604,6 +604,7 @@ static int shorten_decode_frame(AVCodecContext *avctx, void *data,
}
}
+
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
}
@@ -655,4 +656,6 @@ AVCodec ff_shorten_decoder = {
.decode = shorten_decode_frame,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Shorten"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
diff --git a/libavcodec/wmadec.c b/libavcodec/wmadec.c
index 1bcf0da453..ca12d24031 100644
--- a/libavcodec/wmadec.c
+++ b/libavcodec/wmadec.c
@@ -48,20 +48,6 @@
static void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len);
#ifdef TRACE
-static void dump_shorts(WMACodecContext *s, const char *name, const short *tab, int n)
-{
- int i;
-
- tprintf(s->avctx, "%s[%d]:\n", name, n);
- for(i=0;i<n;i++) {
- if ((i & 7) == 0)
- tprintf(s->avctx, "%4d: ", i);
- tprintf(s->avctx, " %5d.0", tab[i]);
- if ((i & 7) == 7)
- tprintf(s->avctx, "\n");
- }
-}
-
static void dump_floats(WMACodecContext *s, const char *name, int prec, const float *tab, int n)
{
int i;
@@ -112,7 +98,7 @@ static int wma_decode_init(AVCodecContext * avctx)
/* init MDCT */
for(i = 0; i < s->nb_block_sizes; i++)
- ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 1, 1.0);
+ ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 1, 1.0 / 32768.0);
if (s->use_noise_coding) {
init_vlc(&s->hgain_vlc, HGAINVLCBITS, sizeof(ff_wma_hgain_huffbits),
@@ -128,7 +114,7 @@ static int wma_decode_init(AVCodecContext * avctx)
wma_lsp_to_curve_init(s, s->frame_len);
}
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
@@ -774,10 +760,10 @@ next:
}
/* decode a frame of frame_len samples */
-static int wma_decode_frame(WMACodecContext *s, int16_t *samples)
+static int wma_decode_frame(WMACodecContext *s, float **samples,
+ int samples_offset)
{
- int ret, n, ch, incr;
- const float *output[MAX_CHANNELS];
+ int ret, ch;
#ifdef TRACE
tprintf(s->avctx, "***decode_frame: %d size=%d\n", s->frame_count++, s->frame_len);
@@ -794,20 +780,19 @@ static int wma_decode_frame(WMACodecContext *s, int16_t *samples)
break;
}
- /* convert frame to integer */
- n = s->frame_len;
- incr = s->nb_channels;
- for (ch = 0; ch < MAX_CHANNELS; ch++)
- output[ch] = s->frame_out[ch];
- s->fmt_conv.float_to_int16_interleave(samples, output, n, incr);
- for (ch = 0; ch < incr; ch++) {
+ for (ch = 0; ch < s->nb_channels; ch++) {
+ /* copy current block to output */
+ memcpy(samples[ch] + samples_offset, s->frame_out[ch],
+ s->frame_len * sizeof(*s->frame_out[ch]));
/* prepare for next block */
- memmove(&s->frame_out[ch][0], &s->frame_out[ch][n], n * sizeof(float));
- }
+ memmove(&s->frame_out[ch][0], &s->frame_out[ch][s->frame_len],
+ s->frame_len * sizeof(*s->frame_out[ch]));
#ifdef TRACE
- dump_shorts(s, "samples", samples, n * s->nb_channels);
+ dump_floats(s, "samples", 6, samples[ch] + samples_offset, s->frame_len);
#endif
+ }
+
return 0;
}
@@ -819,7 +804,8 @@ static int wma_decode_superframe(AVCodecContext *avctx, void *data,
WMACodecContext *s = avctx->priv_data;
int nb_frames, bit_offset, i, pos, len, ret;
uint8_t *q;
- int16_t *samples;
+ float **samples;
+ int samples_offset;
tprintf(avctx, "***decode_superframe:\n");
@@ -852,7 +838,8 @@ static int wma_decode_superframe(AVCodecContext *avctx, void *data,
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- samples = (int16_t *)s->frame.data[0];
+ samples = (float **)s->frame.extended_data;
+ samples_offset = 0;
if (s->use_bit_reservoir) {
bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3);
@@ -886,9 +873,9 @@ static int wma_decode_superframe(AVCodecContext *avctx, void *data,
skip_bits(&s->gb, s->last_bitoffset);
/* this frame is stored in the last superframe and in the
current one */
- if (wma_decode_frame(s, samples) < 0)
+ if (wma_decode_frame(s, samples, samples_offset) < 0)
goto fail;
- samples += s->nb_channels * s->frame_len;
+ samples_offset += s->frame_len;
nb_frames--;
}
@@ -903,9 +890,9 @@ static int wma_decode_superframe(AVCodecContext *avctx, void *data,
s->reset_block_lengths = 1;
for(i=0;i<nb_frames;i++) {
- if (wma_decode_frame(s, samples) < 0)
+ if (wma_decode_frame(s, samples, samples_offset) < 0)
goto fail;
- samples += s->nb_channels * s->frame_len;
+ samples_offset += s->frame_len;
}
/* we copy the end of the frame in the last frame buffer */
@@ -921,9 +908,9 @@ static int wma_decode_superframe(AVCodecContext *avctx, void *data,
memcpy(s->last_superframe, buf + pos, len);
} else {
/* single frame decode */
- if (wma_decode_frame(s, samples) < 0)
+ if (wma_decode_frame(s, samples, samples_offset) < 0)
goto fail;
- samples += s->nb_channels * s->frame_len;
+ samples_offset += s->frame_len;
}
av_dlog(s->avctx, "%d %d %d %d outbytes:%td eaten:%d\n",
@@ -960,6 +947,8 @@ AVCodec ff_wmav1_decoder = {
.flush = flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_WMAV2_DECODER
@@ -974,5 +963,7 @@ AVCodec ff_wmav2_decoder = {
.flush = flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
#endif
diff --git a/libavcodec/wmalosslessdec.c b/libavcodec/wmalosslessdec.c
index 7e09fd0689..6ec1fb4380 100644
--- a/libavcodec/wmalosslessdec.c
+++ b/libavcodec/wmalosslessdec.c
@@ -186,9 +186,9 @@ static av_cold int decode_init(AVCodecContext *avctx)
channel_mask = AV_RL32(edata_ptr + 2);
s->bits_per_sample = AV_RL16(edata_ptr);
if (s->bits_per_sample == 16)
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
else if (s->bits_per_sample == 24) {
- avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
av_log_missing_feature(avctx, "bit-depth higher than 16", 0);
return AVERROR_PATCHWELCOME;
} else {
@@ -984,11 +984,9 @@ static int decode_subframe(WmallDecodeCtx *s)
for (j = 0; j < subframe_len; j++) {
if (s->bits_per_sample == 16) {
- *s->samples_16[c] = (int16_t) s->channel_residues[c][j] << padding_zeroes;
- s->samples_16[c] += s->num_channels;
+ *s->samples_16[c]++ = (int16_t) s->channel_residues[c][j] << padding_zeroes;
} else {
- *s->samples_32[c] = s->channel_residues[c][j] << padding_zeroes;
- s->samples_32[c] += s->num_channels;
+ *s->samples_32[c]++ = s->channel_residues[c][j] << padding_zeroes;
}
}
}
@@ -1025,8 +1023,8 @@ static int decode_frame(WmallDecodeCtx *s)
return ret;
}
for (i = 0; i < s->num_channels; i++) {
- s->samples_16[i] = (int16_t *)s->frame.data[0] + i;
- s->samples_32[i] = (int32_t *)s->frame.data[0] + i;
+ s->samples_16[i] = (int16_t *)s->frame.extended_data[i];
+ s->samples_32[i] = (int32_t *)s->frame.extended_data[i];
}
/* get frame length */
@@ -1296,4 +1294,7 @@ AVCodec ff_wmalossless_decoder = {
.flush = flush,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1 | CODEC_CAP_DELAY,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Lossless"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_NONE },
};
diff --git a/libavcodec/wmaprodec.c b/libavcodec/wmaprodec.c
index 9de8c3861b..4d15e45875 100644
--- a/libavcodec/wmaprodec.c
+++ b/libavcodec/wmaprodec.c
@@ -94,7 +94,6 @@
#include "put_bits.h"
#include "wmaprodata.h"
#include "dsputil.h"
-#include "fmtconvert.h"
#include "sinewin.h"
#include "wma.h"
#include "wma_common.h"
@@ -171,7 +170,6 @@ typedef struct WMAProDecodeCtx {
AVCodecContext* avctx; ///< codec context for av_log
AVFrame frame; ///< AVFrame for decoded output
DSPContext dsp; ///< accelerated DSP functions
- FmtConvertContext fmt_conv;
uint8_t frame_data[MAX_FRAMESIZE +
FF_INPUT_BUFFER_PADDING_SIZE];///< compressed frame data
PutBitContext pb; ///< context for filling the frame_data buffer
@@ -283,10 +281,9 @@ static av_cold int decode_init(AVCodecContext *avctx)
s->avctx = avctx;
ff_dsputil_init(&s->dsp, avctx);
- ff_fmt_convert_init(&s->fmt_conv, avctx);
init_put_bits(&s->pb, s->frame_data, MAX_FRAMESIZE);
- avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (avctx->extradata_size >= 18) {
s->decode_flags = AV_RL16(edata_ptr+14);
@@ -1310,8 +1307,6 @@ static int decode_frame(WMAProDecodeCtx *s, int *got_frame_ptr)
int more_frames = 0;
int len = 0;
int i, ret;
- const float *out_ptr[WMAPRO_MAX_CHANNELS];
- float *samples;
/** get frame length */
if (s->len_prefix)
@@ -1384,13 +1379,11 @@ static int decode_frame(WMAProDecodeCtx *s, int *got_frame_ptr)
s->packet_loss = 1;
return 0;
}
- samples = (float *)s->frame.data[0];
- /** interleave samples and write them to the output buffer */
+ /** copy samples to the output buffer */
for (i = 0; i < s->num_channels; i++)
- out_ptr[i] = s->channel[i].out;
- s->fmt_conv.float_interleave(samples, out_ptr, s->samples_per_frame,
- s->num_channels);
+ memcpy(s->frame.extended_data[i], s->channel[i].out,
+ s->samples_per_frame * sizeof(*s->channel[i].out));
for (i = 0; i < s->num_channels; i++) {
/** reuse second half of the IMDCT output for the next frame */
@@ -1643,4 +1636,6 @@ AVCodec ff_wmapro_decoder = {
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Professional"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
diff --git a/libswscale/utils.c b/libswscale/utils.c
index 3310d78dc7..24058c3b0c 100644
--- a/libswscale/utils.c
+++ b/libswscale/utils.c
@@ -1070,6 +1070,8 @@ av_cold int sws_init_context(SwsContext *c, SwsFilter *srcFilter,
}
}
+#define USE_MMAP (HAVE_MMAP && HAVE_MPROTECT && defined MAP_ANONYMOUS)
+
/* precalculate horizontal scaler filter coefficients */
{
#if HAVE_MMXEXT_INLINE
@@ -1080,7 +1082,7 @@ av_cold int sws_init_context(SwsContext *c, SwsFilter *srcFilter,
c->chrMmx2FilterCodeSize = initMMX2HScaler(c->chrDstW, c->chrXInc,
NULL, NULL, NULL, 4);
-#ifdef MAP_ANONYMOUS
+#if USE_MMAP
c->lumMmx2FilterCode = mmap(NULL, c->lumMmx2FilterCodeSize, PROT_READ | PROT_WRITE, MAP_PRIVATE | MAP_ANONYMOUS, -1, 0);
c->chrMmx2FilterCode = mmap(NULL, c->chrMmx2FilterCodeSize, PROT_READ | PROT_WRITE, MAP_PRIVATE | MAP_ANONYMOUS, -1, 0);
#elif HAVE_VIRTUALALLOC
@@ -1111,7 +1113,7 @@ av_cold int sws_init_context(SwsContext *c, SwsFilter *srcFilter,
initMMX2HScaler(c->chrDstW, c->chrXInc, c->chrMmx2FilterCode,
c->hChrFilter, (uint32_t*)c->hChrFilterPos, 4);
-#ifdef MAP_ANONYMOUS
+#if USE_MMAP
mprotect(c->lumMmx2FilterCode, c->lumMmx2FilterCodeSize, PROT_EXEC | PROT_READ);
mprotect(c->chrMmx2FilterCode, c->chrMmx2FilterCodeSize, PROT_EXEC | PROT_READ);
#endif
@@ -1698,7 +1700,7 @@ void sws_freeContext(SwsContext *c)
av_freep(&c->hChrFilterPos);
#if HAVE_MMX_INLINE
-#ifdef MAP_ANONYMOUS
+#if USE_MMAP
if (c->lumMmx2FilterCode)
munmap(c->lumMmx2FilterCode, c->lumMmx2FilterCodeSize);
if (c->chrMmx2FilterCode)