Commit 4354788a authored by Michael Niedermayer's avatar Michael Niedermayer

Merge remote-tracking branch 'qatar/master'

* qatar/master:
  tls: Use ERR_get_error() in do_tls_poll
  indeo3: Fix a fencepost error.
  mxfdec: Fix comparison of unsigned expression < 0.
  mpegts: set stream id on just created stream, not an unrelated variable
  ra288: return error if input buffer is too small
  ra288: utilize DSPContext.vector_fmul()
  ra288: use memcpy() to copy decoded samples to output
  mace: only calculate output buffer size once
  Remove redundant filename self-references inside files.
  indeo3data: add missing config.h #include for HAVE_BIGENDIAN
  x86: drop pointless ARCH_X86 #ifdef from files in x86 subdirectory
  avplay: reset rdft when closing stream.
  doc/git-howto: expand format-patch and send-email notes.
  lavf: expand doxy for some AVFormatContext fields.
Merged-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parents 0827222b f38f3b88
......@@ -205,8 +205,19 @@ I. BASICS:
git format-patch <commit> [-o directory]
will generate a set of patches out of the current branch starting from
commit. By default the patches are created in the current directory.
will generate a set of patches for each commit between <commit> and
current HEAD. E.g.
git format-patch origin/master
will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
git format-patch -n
which will generate patches from last n commits.
By default the patches are created in the current directory.
11. Sending patches for review
......@@ -215,6 +226,8 @@ I. BASICS:
will send the patches created by git format-patch or directly generates
them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. git-email
package on Debian-based distros).
12. Pushing changes to remote trees
......
......@@ -2348,6 +2348,8 @@ static void stream_component_close(VideoState *is, int stream_index)
if (is->rdft) {
av_rdft_end(is->rdft);
av_freep(&is->rdft_data);
is->rdft = NULL;
is->rdft_bits = 0;
}
break;
case AVMEDIA_TYPE_VIDEO:
......
/*
* simple_idct_arm.S
* Copyright (C) 2002 Frederic 'dilb' Boulay
*
* Author: Frederic Boulay <dilb@handhelds.org>
......
......@@ -444,7 +444,7 @@ static int decode_cell_data(Cell *cell, uint8_t *block, uint8_t *ref_block,
BUFFER_PRECHECK;
dyad1 = bytestream_get_byte(data_ptr);
dyad2 = code;
if (dyad1 > delta_tab->num_dyads || dyad1 >= 248)
if (dyad1 >= delta_tab->num_dyads || dyad1 >= 248)
return IV3_BAD_DATA;
} else {
/* process QUADS */
......
......@@ -24,6 +24,8 @@
#include <stdint.h>
#include "config.h"
/*
* Define compressed VQ tables.
*/
......
/*
* jfdctfst.c
*
* This file is part of the Independent JPEG Group's software.
*
* The authors make NO WARRANTY or representation, either express or implied,
......
/*
* jfdctint.c
*
* This file is part of the Independent JPEG Group's software.
*
* The authors make NO WARRANTY or representation, either express or implied,
......
/*
* jrevdct.c
*
* This file is part of the Independent JPEG Group's software.
*
* The authors make NO WARRANTY or representation, either express or implied,
......
......@@ -20,7 +20,7 @@
/**
* @file
* data structures common to libdiracenc.c and libdiracdec.c
* data structures common to libdirac encoder and decoder
*/
#ifndef AVCODEC_LIBDIRAC_H
......
......@@ -20,7 +20,7 @@
/**
* @file
* function definitions common to libschroedingerdec.c and libschroedingerenc.c
* function definitions common to libschroedinger decoder and encoder
*/
#include "libdirac_libschro.h"
......
......@@ -20,7 +20,7 @@
/**
* @file
* data structures common to libschroedingerdec.c and libschroedingerenc.c
* data structures common to libschroedinger decoder and encoder
*/
#ifndef AVCODEC_LIBSCHROEDINGER_H
......
......@@ -243,11 +243,14 @@ static int mace_decode_frame(AVCodecContext *avctx,
int16_t *samples = data;
MACEContext *ctx = avctx->priv_data;
int i, j, k, l;
int out_size;
int is_mace3 = (avctx->codec_id == CODEC_ID_MACE3);
if (*data_size < (3 * buf_size << (2-is_mace3))) {
av_log(avctx, AV_LOG_ERROR, "Output buffer too small!\n");
return -1;
out_size = 3 * (buf_size << (1 - is_mace3)) *
av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < out_size) {
av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
return AVERROR(EINVAL);
}
for(i = 0; i < avctx->channels; i++) {
......@@ -274,7 +277,7 @@ static int mace_decode_frame(AVCodecContext *avctx,
}
}
*data_size = 3 * buf_size << (2-is_mace3);
*data_size = out_size;
return buf_size;
}
......
......@@ -26,6 +26,7 @@
#include "lpc.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "dsputil.h"
#define MAX_BACKWARD_FILTER_ORDER 36
#define MAX_BACKWARD_FILTER_LEN 40
......@@ -35,8 +36,9 @@
#define RA288_BLOCKS_PER_FRAME 32
typedef struct {
float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
DSPContext dsp;
DECLARE_ALIGNED(16, float, sp_lpc)[FFALIGN(36, 8)]; ///< LPC coefficients for speech data (spec: A)
DECLARE_ALIGNED(16, float, gain_lpc)[FFALIGN(10, 8)]; ///< LPC coefficients for gain (spec: GB)
/** speech data history (spec: SB).
* Its first 70 coefficients are updated only at backward filtering.
......@@ -57,16 +59,12 @@ typedef struct {
static av_cold int ra288_decode_init(AVCodecContext *avctx)
{
RA288Context *ractx = avctx->priv_data;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
dsputil_init(&ractx->dsp, avctx);
return 0;
}
static void apply_window(float *tgt, const float *m1, const float *m2, int n)
{
while (n--)
*tgt++ = *m1++ * *m2++;
}
static void convolve(float *tgt, const float *src, int len, int n)
{
for (; n >= 0; n--)
......@@ -123,15 +121,18 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
* @param out2 pointer to the recursive part of the output
* @param window pointer to the windowing function table
*/
static void do_hybrid_window(int order, int n, int non_rec, float *out,
static void do_hybrid_window(RA288Context *ractx,
int order, int n, int non_rec, float *out,
float *hist, float *out2, const float *window)
{
int i;
float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC];
LOCAL_ALIGNED_16(float, work)[FFALIGN(MAX_BACKWARD_FILTER_ORDER +
MAX_BACKWARD_FILTER_LEN +
MAX_BACKWARD_FILTER_NONREC, 8)];
apply_window(work, window, hist, order + n + non_rec);
ractx->dsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 8));
convolve(buffer1, work + order , n , order);
convolve(buffer2, work + order + n, non_rec, order);
......@@ -148,16 +149,17 @@ static void do_hybrid_window(int order, int n, int non_rec, float *out,
/**
* Backward synthesis filter, find the LPC coefficients from past speech data.
*/
static void backward_filter(float *hist, float *rec, const float *window,
static void backward_filter(RA288Context *ractx,
float *hist, float *rec, const float *window,
float *lpc, const float *tab,
int order, int n, int non_rec, int move_size)
{
float temp[MAX_BACKWARD_FILTER_ORDER+1];
do_hybrid_window(order, n, non_rec, temp, hist, rec, window);
do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
apply_window(lpc, lpc, tab, order);
ractx->dsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 8));
memmove(hist, hist + n, move_size*sizeof(*hist));
}
......@@ -168,7 +170,7 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
float *out = data;
int i, j, out_size;
int i, out_size;
RA288Context *ractx = avctx->priv_data;
GetBitContext gb;
......@@ -176,7 +178,7 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data,
av_log(avctx, AV_LOG_ERROR,
"Error! Input buffer is too small [%d<%d]\n",
buf_size, avctx->block_align);
return 0;
return AVERROR_INVALIDDATA;
}
out_size = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME *
......@@ -194,14 +196,14 @@ static int ra288_decode_frame(AVCodecContext * avctx, void *data,
decode(ractx, gain, cb_coef);
for (j=0; j < RA288_BLOCK_SIZE; j++)
*(out++) = ractx->sp_hist[70 + 36 + j];
memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
out += RA288_BLOCK_SIZE;
if ((i & 7) == 3) {
backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
}
}
......
......@@ -23,6 +23,7 @@
#define AVCODEC_RA288_H
#include <stdint.h>
#include "dsputil.h"
static const float amptable[8]={
0.515625, 0.90234375, 1.57910156, 2.76342773,
......@@ -96,7 +97,7 @@ static const int16_t codetable[128][5]={
{ 3746, -606, 53, -269, -3301}, { 606, 2018, -1316, 4064, 398}
};
static const float syn_window[111]={
DECLARE_ALIGNED(16, static const float, syn_window)[FFALIGN(111, 8)]={
0.576690972, 0.580838025, 0.585013986, 0.589219987, 0.59345597, 0.597723007,
0.602020264, 0.606384277, 0.610748291, 0.615142822, 0.619598389, 0.624084473,
0.628570557, 0.633117676, 0.637695313, 0.642272949, 0.646911621, 0.651580811,
......@@ -118,7 +119,7 @@ static const float syn_window[111]={
0.142852783, 0.0954284668,0.0477600098
};
static const float gain_window[38]={
DECLARE_ALIGNED(16, static const float, gain_window)[FFALIGN(38, 8)]={
0.505699992, 0.524200022, 0.54339999, 0.563300014, 0.583953857, 0.60534668,
0.627502441, 0.650482178, 0.674316406, 0.699005127, 0.724578857, 0.75112915,
0.778625488, 0.807128906, 0.836669922, 0.86730957, 0.899078369, 0.932006836,
......@@ -129,7 +130,7 @@ static const float gain_window[38]={
};
/** synthesis bandwidth broadening table */
static const float syn_bw_tab[36]={
DECLARE_ALIGNED(16, static const float, syn_bw_tab)[FFALIGN(36, 8)] = {
0.98828125, 0.976699829, 0.965254128, 0.953942537, 0.942763507, 0.931715488,
0.920796931, 0.910006344, 0.899342179, 0.888803005, 0.878387332, 0.868093729,
0.857920766, 0.847867012, 0.837931097, 0.828111589, 0.818407178, 0.808816493,
......@@ -139,7 +140,7 @@ static const float syn_bw_tab[36]={
};
/** gain bandwidth broadening table */
static const float gain_bw_tab[10]={
DECLARE_ALIGNED(16, static const float, gain_bw_tab)[FFALIGN(10, 8)] = {
0.90625, 0.821289063, 0.74432373, 0.674499512, 0.61126709,
0.553955078, 0.50201416, 0.454956055, 0.41229248, 0.373657227
};
......
......@@ -82,7 +82,7 @@
"add "tmp" , "low" \n\t"\
"1: \n\t"
#if ARCH_X86 && HAVE_7REGS && !defined(BROKEN_RELOCATIONS)
#if HAVE_7REGS && !defined(BROKEN_RELOCATIONS)
#define get_cabac_inline get_cabac_inline_x86
static av_always_inline int get_cabac_inline_x86(CABACContext *c,
uint8_t *const state)
......@@ -105,7 +105,7 @@ static av_always_inline int get_cabac_inline_x86(CABACContext *c,
);
return bit & 1;
}
#endif /* ARCH_X86 && HAVE_7REGS && !defined(BROKEN_RELOCATIONS) */
#endif /* HAVE_7REGS && !defined(BROKEN_RELOCATIONS) */
#define get_cabac_bypass_sign get_cabac_bypass_sign_x86
static av_always_inline int get_cabac_bypass_sign_x86(CABACContext *c, int val)
......
......@@ -36,7 +36,7 @@
//FIXME use some macros to avoid duplicating get_cabac (cannot be done yet
//as that would make optimization work hard)
#if ARCH_X86 && HAVE_7REGS && !defined(BROKEN_RELOCATIONS)
#if HAVE_7REGS && !defined(BROKEN_RELOCATIONS)
static int decode_significance_x86(CABACContext *c, int max_coeff,
uint8_t *significant_coeff_ctx_base,
int *index, x86_reg last_off){
......@@ -145,6 +145,6 @@ static int decode_significance_8x8_x86(CABACContext *c,
);
return coeff_count;
}
#endif /* ARCH_X86 && HAVE_7REGS && !defined(BROKEN_RELOCATIONS) */
#endif /* HAVE_7REGS && !defined(BROKEN_RELOCATIONS) */
#endif /* AVCODEC_X86_H264_I386_H */
/*
* idct_mmx.c
* Copyright (C) 1999-2001 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
*
* This file is part of mpeg2dec, a free MPEG-2 video stream decoder.
......
......@@ -765,17 +765,56 @@ typedef struct AVChapter {
* New fields can be added to the end with minor version bumps.
* Removal, reordering and changes to existing fields require a major
* version bump.
* sizeof(AVFormatContext) must not be used outside libav*.
* sizeof(AVFormatContext) must not be used outside libav*, use
* avformat_alloc_context() to create an AVFormatContext.
*/
typedef struct AVFormatContext {
const AVClass *av_class; /**< Set by avformat_alloc_context. */
/* Can only be iformat or oformat, not both at the same time. */
/**
* A class for logging and AVOptions. Set by avformat_alloc_context().
* Exports (de)muxer private options if they exist.
*/
const AVClass *av_class;
/**
* Can only be iformat or oformat, not both at the same time.
*
* decoding: set by avformat_open_input().
* encoding: set by the user.
*/
struct AVInputFormat *iformat;
struct AVOutputFormat *oformat;
/**
* Format private data. This is an AVOptions-enabled struct
* if and only if iformat/oformat.priv_class is not NULL.
*/
void *priv_data;
/*
* I/O context.
*
* decoding: either set by the user before avformat_open_input() (then
* the user must close it manually) or set by avformat_open_input().
* encoding: set by the user.
*
* Do NOT set this field if AVFMT_NOFILE flag is set in
* iformat/oformat.flags. In such a case, the (de)muxer will handle
* I/O in some other way and this field will be NULL.
*/
AVIOContext *pb;
/**
* A list of all streams in the file. New streams are created with
* avformat_new_stream().
*
* decoding: streams are created by libavformat in avformat_open_input().
* If AVFMTCTX_NOHEADER is set in ctx_flags, then new streams may also
* appear in av_read_frame().
* encoding: streams are created by the user before avformat_write_header().
*/
unsigned int nb_streams;
AVStream **streams;
char filename[1024]; /**< input or output filename */
/* stream info */
#if FF_API_TIMESTAMP
......@@ -886,8 +925,8 @@ typedef struct AVFormatContext {
unsigned int probesize;
/**
* Maximum time (in AV_TIME_BASE units) during which the input should
* be analyzed in av_find_stream_info().
* decoding: maximum time (in AV_TIME_BASE units) during which the input should
* be analyzed in avformat_find_stream_info().
*/
int max_analyze_duration;
......
......@@ -283,7 +283,7 @@ static int mxf_decrypt_triplet(AVFormatContext *s, AVPacket *pkt, KLVPacket *klv
MXFContext *mxf = s->priv_data;
AVIOContext *pb = s->pb;
int64_t end = avio_tell(pb) + klv->length;
uint64_t size;
int64_t size;
uint64_t orig_size;
uint64_t plaintext_size;
uint8_t ivec[16];
......
......@@ -87,7 +87,7 @@ static int do_tls_poll(URLContext *h, int ret)
} else if (ret == SSL_ERROR_WANT_WRITE) {
p.events = POLLOUT;
} else {
av_log(NULL, AV_LOG_ERROR, "%s\n", ERR_error_string(ret, NULL));
av_log(NULL, AV_LOG_ERROR, "%s\n", ERR_error_string(ERR_get_error(), NULL));
return AVERROR(EIO);
}
#endif
......
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