Merge remote-tracking branch 'qatar/master'

* qatar/master: x86: cabac: replace explicit memory references with "m" operands avplay: don't request a stereo downmix wmapro: use av_float2int() lavc: avoid invalid memcpy() in avcodec_default_release_buffer() lavu: replace int/float punning functions lavfi: install libavfilter/vsrc_buffer.h Remove extraneous semicolons sdp: Restore the original mp4 format h264 extradata if converted rtpenc: Add support for mp4 format h264 rtpenc: Simplify code by introducing a separate end pointer movenc: Use the actual converted sample for RTP hinting Fix a bunch of common typos. Conflicts: doc/developer.texi doc/eval.texi doc/filters.texi doc/protocols.texi ffmpeg.c ffplay.c libavcodec/mpegvideo.h libavcodec/x86/cabac.h libavfilter/Makefile libavformat/avformat.h libavformat/cafdec.c libavformat/flvdec.c libavformat/flvenc.c libavformat/gxfenc.c libavformat/img2.c libavformat/movenc.c libavformat/mpegts.c libavformat/rtpenc_h264.c libavformat/utils.c libavformat/wtv.c Merged-by: Michael Niedermayer <[email protected]>
linjiansheng · Dec 12, 2011 · 7fad19a · 7fad19a
2 parents 16abd68 + 599b4c6
commit 7fad19a
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Showing 78 changed files with 388 additions and 286 deletions.
diff --git a/avconv.c b/avconv.c
@@ -3783,7 +3783,7 @@ static void opt_output_file(void *optctx, const char *filename)
     }
 
     if (!(oc->oformat->flags & AVFMT_NOFILE)) {
-        /* test if it already exists to avoid loosing precious files */
+        /* test if it already exists to avoid losing precious files */
         assert_file_overwrite(filename);
 
         /* open the file */

diff --git a/cmdutils.h b/cmdutils.h
@@ -91,7 +91,7 @@ int opt_timelimit(const char *opt, const char *arg);
  * parsed or the corresponding value is invalid.
  *
  * @param context the context of the value to be set (e.g. the
- * corresponding commandline option name)
+ * corresponding command line option name)
  * @param numstr the string to be parsed
  * @param type the type (OPT_INT64 or OPT_FLOAT) as which the
  * string should be parsed
@@ -106,7 +106,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type, do
  * the string cannot be correctly parsed.
  *
  * @param context the context of the value to be set (e.g. the
- * corresponding commandline option name)
+ * corresponding command line option name)
  * @param timestr the string to be parsed
  * @param is_duration a flag which tells how to interpret timestr, if
  * not zero timestr is interpreted as a duration, otherwise as a
@@ -189,7 +189,7 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
 int parse_option(void *optctx, const char *opt, const char *arg, const OptionDef *options);
 
 /**
- * Find the '-loglevel' option in the commandline args and apply it.
+ * Find the '-loglevel' option in the command line args and apply it.
  */
 void parse_loglevel(int argc, char **argv, const OptionDef *options);
 

diff --git a/doc/avconv.texi b/doc/avconv.texi
@@ -29,7 +29,7 @@ rates and resize video on the fly with a high quality polyphase filter.
 avconv reads from an arbitrary number of input "files" (which can be regular
 files, pipes, network streams, grabbing devices, etc.), specified by the
 @code{-i} option, and writes to an arbitrary number of output "files", which are
-specified by a plain output filename. Anything found on the commandline which
+specified by a plain output filename. Anything found on the command line which
 cannot be interpreted as an option is considered to be an output filename.
 
 Each input or output file can in principle contain any number of streams of

diff --git a/doc/ffserver.texi b/doc/ffserver.texi
@@ -147,7 +147,7 @@ that only captures in stereo and also requires that one channel be flipped.
 If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
 starting ffmpeg.
 
-@subsection The audio and video loose sync after a while.
+@subsection The audio and video lose sync after a while.
 
 Yes, they do.
 

diff --git a/doc/protocols.texi b/doc/protocols.texi
@@ -155,8 +155,8 @@ be seekable, so they will fail with the pipe output protocol.
 
 Real-Time Messaging Protocol.
 
-The Real-Time Messaging Protocol (RTMP) is used for streaming
-multimedia content across a TCP/IP network.
+The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
+content across a TCP/IP network.
 
 The required syntax is:
 @example

diff --git a/doc/soc.txt b/doc/soc.txt
@@ -18,7 +18,7 @@ essential that changes to their codebase are publicly visible, clean and
 easy reviewable that again leads us to:
 * use of a revision control system like git
 * separation of cosmetic from non-cosmetic changes (this is almost entirely
-  ignored by mentors and students in soc 2006 which might lead to a suprise
+  ignored by mentors and students in soc 2006 which might lead to a surprise
   when the code will be reviewed at the end before a possible inclusion in
   FFmpeg, individual changes were generally not reviewable due to cosmetics).
 * frequent commits, so that comments can be provided early
diff --git a/ffmpeg.c b/ffmpeg.c
@@ -4162,7 +4162,7 @@ static void opt_output_file(void *optctx, const char *filename)
     }
 
     if (!(oc->oformat->flags & AVFMT_NOFILE)) {
-        /* test if it already exists to avoid loosing precious files */
+        /* test if it already exists to avoid losing precious files */
         assert_file_overwrite(filename);
 
         /* open the file */

diff --git a/ffplay.c b/ffplay.c
@@ -2468,8 +2468,6 @@ static int read_thread(void *arg)
     if(genpts)
         ic->flags |= AVFMT_FLAG_GENPTS;
 
-    av_dict_set(&codec_opts, "request_channels", "2", 0);
-
     opts = setup_find_stream_info_opts(ic, codec_opts);
     orig_nb_streams = ic->nb_streams;
 

diff --git a/ffserver.c b/ffserver.c
@@ -475,7 +475,7 @@ static void start_children(FFStream *feed)
                     slash++;
                 strcpy(slash, "ffmpeg");
 
-                http_log("Launch commandline: ");
+                http_log("Launch command line: ");
                 http_log("%s ", pathname);
                 for (i = 1; feed->child_argv[i] && feed->child_argv[i][0]; i++)
                     http_log("%s ", feed->child_argv[i]);

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
@@ -1348,7 +1348,7 @@ typedef struct AVCodecContext {
      * Some codecs need additional format info. It is stored here.
      * If any muxer uses this then ALL demuxers/parsers AND encoders for the
      * specific codec MUST set it correctly otherwise stream copy breaks.
-     * In general use of this field by muxers is not recommanded.
+     * In general use of this field by muxers is not recommended.
      * - encoding: Set by libavcodec.
      * - decoding: Set by libavcodec. (FIXME: Is this OK?)
      */
@@ -3541,7 +3541,7 @@ typedef struct ReSampleContext ReSampleContext;
  * @param linear           if 1 then the used FIR filter will be linearly interpolated
                            between the 2 closest, if 0 the closest will be used
  * @param cutoff           cutoff frequency, 1.0 corresponds to half the output sampling rate
- * @return allocated ReSampleContext, NULL if error occured
+ * @return allocated ReSampleContext, NULL if error occurred
  */
 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
                                         int output_rate, int input_rate,

diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c
@@ -35,7 +35,7 @@
 #include "dct.h"
 #include "rdft.h"
 #include "fmtconvert.h"
-#include "libavutil/intfloat_readwrite.h"
+#include "libavutil/intfloat.h"
 
 extern const uint16_t ff_wma_critical_freqs[25];
 
@@ -193,8 +193,8 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
         if (s->version_b) {
             if (get_bits_left(gb) < 64)
                 return AVERROR_INVALIDDATA;
-            coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
-            coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root;
+            coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
+            coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
         } else {
             if (get_bits_left(gb) < 58)
                 return AVERROR_INVALIDDATA;

diff --git a/libavcodec/bitstream.c b/libavcodec/bitstream.c
@@ -103,7 +103,7 @@ static int alloc_table(VLC *vlc, int size, int use_static)
     vlc->table_size += size;
     if (vlc->table_size > vlc->table_allocated) {
         if(use_static)
-            abort(); //cant do anything, init_vlc() is used with too little memory
+            abort(); // cannot do anything, init_vlc() is used with too little memory
         vlc->table_allocated += (1 << vlc->bits);
         vlc->table = av_realloc_f(vlc->table,
                                   vlc->table_allocated, sizeof(VLC_TYPE) * 2);

diff --git a/libavcodec/eatgv.c b/libavcodec/eatgv.c
@@ -157,7 +157,7 @@ static int tgv_decode_inter(TgvContext * s, const uint8_t *buf, const uint8_t *b
     vector_bits       = AV_RL16(&buf[6]);
     buf += 12;
 
-    /* allocate codebook buffers as neccessary */
+    /* allocate codebook buffers as necessary */
     if (num_mvs > s->num_mvs) {
         s->mv_codebook = av_realloc(s->mv_codebook, num_mvs*2*sizeof(int));
         s->num_mvs = num_mvs;
@@ -293,7 +293,7 @@ static int tgv_decode_frame(AVCodecContext *avctx,
         s->frame.buffer_hints = FF_BUFFER_HINTS_VALID;
         s->frame.linesize[0] = s->width;
 
-        /* allocate additional 12 bytes to accomodate av_memcpy_backptr() OUTBUF_PADDED optimisation */
+        /* allocate additional 12 bytes to accommodate av_memcpy_backptr() OUTBUF_PADDED optimisation */
         s->frame.data[0] = av_malloc(s->width*s->height + 12);
         if (!s->frame.data[0])
             return AVERROR(ENOMEM);

diff --git a/libavcodec/error_resilience.c b/libavcodec/error_resilience.c
@@ -801,7 +801,7 @@ void ff_er_frame_end(MpegEncContext *s){
     if(!s->error_recognition || s->error_count==0 || s->avctx->lowres ||
        s->avctx->hwaccel ||
        s->avctx->codec->capabilities&CODEC_CAP_HWACCEL_VDPAU ||
-       s->picture_structure != PICT_FRAME || // we dont support ER of field pictures yet, though it should not crash if enabled
+       s->picture_structure != PICT_FRAME || // we do not support ER of field pictures yet, though it should not crash if enabled
        s->error_count==3*s->mb_width*(s->avctx->skip_top + s->avctx->skip_bottom)) return;
 
     if (s->current_picture.f.motion_val[0] == NULL) {

diff --git a/libavcodec/g722enc.c b/libavcodec/g722enc.c
@@ -153,7 +153,7 @@ static int g722_encode_trellis(AVCodecContext *avctx,
 
         for (j = 0; j < frontier && nodes[0][j]; j++) {
             /* Only k >> 2 affects the future adaptive state, therefore testing
-             * small steps that don't change k >> 2 is useless, the orignal
+             * small steps that don't change k >> 2 is useless, the original
              * value from encode_low is better than them. Since we step k
              * in steps of 4, make sure range is a multiple of 4, so that
              * we don't miss the original value from encode_low. */

diff --git a/libavcodec/h264.c b/libavcodec/h264.c
@@ -2201,8 +2201,8 @@ static av_always_inline void hl_decode_mb_444_internal(H264Context *h, int simpl
 static void hl_decode_mb_simple_ ## bits(H264Context *h){ \
     hl_decode_mb_internal(h, 1, sh); \
 }
-hl_decode_mb_simple(0, 8);
-hl_decode_mb_simple(1, 16);
+hl_decode_mb_simple(0, 8)
+hl_decode_mb_simple(1, 16)
 
 /**
  * Process a macroblock; this handles edge cases, such as interlacing.
@@ -2926,7 +2926,7 @@ static int decode_slice_header(H264Context *h, H264Context *h0){
              * FIXME: avoiding a memcpy would be nice, but ref handling makes many assumptions
              * about there being no actual duplicates.
              * FIXME: this doesn't copy padding for out-of-frame motion vectors.  Given we're
-             * concealing a lost frame, this probably isn't noticable by comparison, but it should
+             * concealing a lost frame, this probably isn't noticeable by comparison, but it should
              * be fixed. */
             if (h->short_ref_count) {
                 if (prev) {
@@ -3354,7 +3354,7 @@ static av_always_inline void fill_filter_caches_inter(H264Context *h, MpegEncCon
 
 /**
  *
- * @return non zero if the loop filter can be skiped
+ * @return non zero if the loop filter can be skipped
  */
 static int fill_filter_caches(H264Context *h, int mb_type){
     MpegEncContext * const s = &h->s;
@@ -3879,7 +3879,7 @@ static int decode_nal_units(H264Context *h, const uint8_t *buf, int buf_size){
                 av_log(h->s.avctx, AV_LOG_ERROR, "Invalid mix of idr and non-idr slices");
                 return -1;
             }
-            idr(h); //FIXME ensure we don't loose some frames if there is reordering
+            idr(h); // FIXME ensure we don't lose some frames if there is reordering
         case NAL_SLICE:
             init_get_bits(&hx->s.gb, ptr, bit_length);
             hx->intra_gb_ptr=

diff --git a/libavcodec/h264pred_template.c b/libavcodec/h264pred_template.c
@@ -388,9 +388,9 @@ static void FUNCC(pred16x16_##n##_dc)(uint8_t *_src, int stride){\
     PREDICT_16x16_DC(PIXEL_SPLAT_X4(v));\
 }
 
-PRED16x16_X(127, (1<<(BIT_DEPTH-1))-1);
-PRED16x16_X(128, (1<<(BIT_DEPTH-1))+0);
-PRED16x16_X(129, (1<<(BIT_DEPTH-1))+1);
+PRED16x16_X(127, (1<<(BIT_DEPTH-1))-1)
+PRED16x16_X(128, (1<<(BIT_DEPTH-1))+0)
+PRED16x16_X(129, (1<<(BIT_DEPTH-1))+1)
 
 static inline void FUNCC(pred16x16_plane_compat)(uint8_t *p_src, int p_stride, const int svq3, const int rv40){
   int i, j, k;
@@ -502,9 +502,9 @@ static void FUNCC(pred8x8_##n##_dc)(uint8_t *_src, int stride){\
     }\
 }
 
-PRED8x8_X(127, (1<<(BIT_DEPTH-1))-1);
-PRED8x8_X(128, (1<<(BIT_DEPTH-1))+0);
-PRED8x8_X(129, (1<<(BIT_DEPTH-1))+1);
+PRED8x8_X(127, (1<<(BIT_DEPTH-1))-1)
+PRED8x8_X(128, (1<<(BIT_DEPTH-1))+0)
+PRED8x8_X(129, (1<<(BIT_DEPTH-1))+1)
 
 static void FUNCC(pred8x16_128_dc)(uint8_t *_src, int stride){
     FUNCC(pred8x8_128_dc)(_src, stride);

diff --git a/libavcodec/jfdctint_template.c b/libavcodec/jfdctint_template.c
@@ -340,7 +340,7 @@ FUNC(ff_jpeg_fdct_islow)(DCTELEM *data)
 
 /*
  * The secret of DCT2-4-8 is really simple -- you do the usual 1-DCT
- * on the rows and then, instead of doing even and odd, part on the colums
+ * on the rows and then, instead of doing even and odd, part on the columns
  * you do even part two times.
  */
 GLOBAL(void)

diff --git a/libavcodec/lpc.h b/libavcodec/lpc.h
@@ -67,7 +67,7 @@ typedef struct LPCContext {
      * Perform autocorrelation on input samples with delay of 0 to lag.
      * @param data  input samples.
      *              constraints: no alignment needed, but must have have at
-     *              least lag*sizeof(double) valid bytes preceeding it, and
+     *              least lag*sizeof(double) valid bytes preceding it, and
      *              size must be at least (len+1)*sizeof(double) if data is
      *              16-byte aligned or (len+2)*sizeof(double) if data is
      *              unaligned.

diff --git a/libavcodec/mjpegdec.c b/libavcodec/mjpegdec.c
@@ -1114,9 +1114,9 @@ static int mjpeg_decode_app(MJpegDecodeContext *s)
         av_log(s->avctx, AV_LOG_DEBUG, "APPx %8X\n", id);
     }
 
-    /* buggy AVID, it puts EOI only at every 10th frame */
-    /* also this fourcc is used by non-avid files too, it holds some
-       informations, but it's always present in AVID creates files */
+    /* Buggy AVID, it puts EOI only at every 10th frame. */
+    /* Also, this fourcc is used by non-avid files too, it holds some
+       information, but it's always present in AVID-created files. */
     if (id == AV_RL32("AVI1"))
     {
         /* structure:

diff --git a/libavcodec/proresdec_lgpl.c b/libavcodec/proresdec_lgpl.c
@@ -559,7 +559,7 @@ static int decode_slice(AVCodecContext *avctx, ProresThreadData *td)
     sf = sf > 128 ? (sf - 96) << 2 : sf;
 
     /* scale quantization matrixes according with slice's scale factor */
-    /* TODO: this can be SIMD-optimized alot */
+    /* TODO: this can be SIMD-optimized a lot */
     if (ctx->qmat_changed || sf != ctx->prev_slice_sf) {
         ctx->prev_slice_sf = sf;
         for (i = 0; i < 64; i++) {

diff --git a/libavcodec/tscc.c b/libavcodec/tscc.c
@@ -156,7 +156,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
              return -1;
     }
     c->bpp = avctx->bits_per_coded_sample;
-    // buffer size for RLE 'best' case when 2-byte code preceeds each pixel and there may be padding after it too
+    // buffer size for RLE 'best' case when 2-byte code precedes each pixel and there may be padding after it too
     c->decomp_size = (((avctx->width * c->bpp + 7) >> 3) + 3 * avctx->width + 2) * avctx->height + 2;
 
     /* Allocate decompression buffer */

diff --git a/libavcodec/utils.c b/libavcodec/utils.c
@@ -544,7 +544,8 @@ void avcodec_default_release_buffer(AVCodecContext *s, AVFrame *pic){
         avci->buffer_count--;
         last = &avci->buffer[avci->buffer_count];
 
-        FFSWAP(InternalBuffer, *buf, *last);
+        if (buf != last)
+            FFSWAP(InternalBuffer, *buf, *last);
     }
 
     for (i = 0; i < AV_NUM_DATA_POINTERS; i++) {

diff --git a/libavcodec/vaapi_mpeg4.c b/libavcodec/vaapi_mpeg4.c
@@ -98,7 +98,7 @@ static int vaapi_mpeg4_start_frame(AVCodecContext *avctx, av_unused const uint8_
         pic_param->forward_reference_picture  = ff_vaapi_get_surface_id(&s->last_picture);
 
     /* Fill in VAIQMatrixBufferMPEG4 */
-    /* Only the first inverse quantisation method uses the weighthing matrices */
+    /* Only the first inverse quantisation method uses the weighting matrices */
     if (pic_param->vol_fields.bits.quant_type) {
         iq_matrix = ff_vaapi_alloc_iq_matrix(vactx, sizeof(VAIQMatrixBufferMPEG4));
         if (!iq_matrix)

diff --git a/libavcodec/vc1dec.c b/libavcodec/vc1dec.c
@@ -3536,7 +3536,7 @@ static void vc1_apply_p_loop_filter(VC1Context *v)
         vc1_apply_p_v_loop_filter(v, i);
     }
 
-    /* V always preceedes H, therefore we run H one MB before V;
+    /* V always precedes H, therefore we run H one MB before V;
      * at the end of a row, we catch up to complete the row */
     if (s->mb_x) {
         for (i = 0; i < 6; i++) {
@@ -3573,7 +3573,7 @@ static int vc1_decode_p_mb(VC1Context *v)
     int skipped, fourmv;
     int block_cbp = 0, pat, block_tt = 0, block_intra = 0;
 
-    mquant = v->pq; /* Loosy initialization */
+    mquant = v->pq; /* lossy initialization */
 
     if (v->mv_type_is_raw)
         fourmv = get_bits1(gb);
@@ -4141,7 +4141,7 @@ static void vc1_decode_b_mb(VC1Context *v)
     int dmv_x[2], dmv_y[2];
     int bmvtype = BMV_TYPE_BACKWARD;
 
-    mquant      = v->pq; /* Loosy initialization */
+    mquant      = v->pq; /* lossy initialization */
     s->mb_intra = 0;
 
     if (v->dmb_is_raw)

diff --git a/libavcodec/vc1dsp.c b/libavcodec/vc1dsp.c
@@ -550,8 +550,8 @@ static av_always_inline int vc1_mspel_ ## DIR ## _filter_16bits(const TYPE *src,
     return 0; /* should not occur */                                    \
 }
 
-VC1_MSPEL_FILTER_16B(ver, uint8_t);
-VC1_MSPEL_FILTER_16B(hor, int16_t);
+VC1_MSPEL_FILTER_16B(ver, uint8_t)
+VC1_MSPEL_FILTER_16B(hor, int16_t)
 
 
 /** Filter used to interpolate fractional pel values

diff --git a/libavcodec/wmaprodec.c b/libavcodec/wmaprodec.c
@@ -86,6 +86,7 @@
  * subframe in order to reconstruct the output samples.
  */
 
+#include "libavutil/intfloat.h"
 #include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 #include "internal.h"
@@ -823,8 +824,8 @@ static int decode_coeffs(WMAProDecodeCtx *s, int c)
                     v1 = get_vlc2(&s->gb, vec1_vlc.table, VLCBITS, VEC1MAXDEPTH);
                     if (v1 == HUFF_VEC1_SIZE - 1)
                         v1 += ff_wma_get_large_val(&s->gb);
-                    vals[i  ] = ((av_alias32){ .f32 = v0 }).u32;
-                    vals[i+1] = ((av_alias32){ .f32 = v1 }).u32;
+                    vals[i  ] = av_float2int(v0);
+                    vals[i+1] = av_float2int(v1);
                 } else {
                     vals[i]   = fval_tab[symbol_to_vec2[idx] >> 4 ];
                     vals[i+1] = fval_tab[symbol_to_vec2[idx] & 0xF];
@@ -1425,7 +1426,7 @@ static int remaining_bits(WMAProDecodeCtx *s, GetBitContext *gb)
  *@param s codec context
  *@param gb bitstream reader context
  *@param len length of the partial frame
- *@param append decides wether to reset the buffer or not
+ *@param append decides whether to reset the buffer or not
  */
 static void save_bits(WMAProDecodeCtx *s, GetBitContext* gb, int len,
                       int append)

diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c
@@ -189,7 +189,7 @@ typedef struct {
      * @{
      */
     int spillover_nbits;          ///< number of bits of the previous packet's
-                                  ///< last superframe preceeding this
+                                  ///< last superframe preceding this
                                   ///< packet's first full superframe (useful
                                   ///< for re-synchronization also)
     int has_residual_lsps;        ///< if set, superframes contain one set of
@@ -1805,7 +1805,7 @@ static int synth_superframe(AVCodecContext *ctx, int *got_frame_ptr)
     s->frame.nb_samples = n_samples;
     samples = (float *)s->frame.data[0];
 
-    /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */
+    /* Parse frames, optionally preceded by per-frame (independent) LSPs. */
     for (n = 0; n < 3; n++) {
         if (!s->has_residual_lsps) {
             int m;