Merge remote-tracking branch 'qatar/master'

* qatar/master:
  doxygen: misc consistency, spelling and wording fixes
  vcr1: drop unnecessary emms_c() calls without MMX code
  Replace all uses of av_close_input_file() with avformat_close_input().
  lavf: add avformat_close_input().
  lavf: deprecate av_close_input_stream().
  lavf doxy: add some basic demuxing documentation.
  lavf doxy: add some general lavf information.
  lavf doxy: add misc utility functions to a group.
  lavf doxy: add av_guess_codec/format to the encoding group.
  lavf doxy: add core functions to a doxy group.
  Add basic libavdevice documentation.
  lavc: convert error_recognition to err_recognition.
  avconv: update -map option help text
  x86: Require 7 registers for the cabac asm
  x86: bswap: remove test for bswap instruction
  bswap: make generic implementation more compiler-friendly
  h264: remove useless cast
  proresdec: fix decode_slice() prototype

Conflicts:
	configure
	doc/APIchanges
	ffprobe.c
	libavcodec/avcodec.h
	libavcodec/celp_math.h
	libavcodec/h264.c
	libavfilter/src_movie.c
	libavformat/anm.c
	libavformat/avformat.h
	libavformat/version.h
	libavutil/avstring.h
	libavutil/bswap.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>

avconv.c

@@ -610,7 +610,7 @@ void exit_program(int ret)
         av_dict_free(&output_files[i].opts);
     }
     for(i=0;i<nb_input_files;i++) {
-        av_close_input_file(input_files[i].ctx);
+        avformat_close_input(&input_files[i].ctx);
     }
     for (i = 0; i < nb_input_streams; i++) {
         av_freep(&input_streams[i].decoded_frame);
@@ -3140,7 +3140,7 @@ static int opt_input_file(OptionsContext *o, const char *opt, const char *filena
     ret = avformat_find_stream_info(ic, opts);
     if (ret < 0) {
         av_log(NULL, AV_LOG_FATAL, "%s: could not find codec parameters\n", filename);
-        av_close_input_file(ic);
+        avformat_close_input(&ic);
         exit_program(1);
     }
 
@@ -4230,7 +4230,7 @@ static const OptionDef options[] = {
     { "c", HAS_ARG | OPT_STRING | OPT_SPEC, {.off = OFFSET(codec_names)}, "codec name", "codec" },
     { "codec", HAS_ARG | OPT_STRING | OPT_SPEC, {.off = OFFSET(codec_names)}, "codec name", "codec" },
     { "pre", HAS_ARG | OPT_STRING | OPT_SPEC, {.off = OFFSET(presets)}, "preset name", "preset" },
-    { "map", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map}, "set input stream mapping", "file.stream[:syncfile.syncstream]" },
+    { "map", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map}, "set input stream mapping", "[-]input_file_id[:stream_specifier][,sync_file_id[:stream_specifier]]" },
     { "map_metadata", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map_metadata}, "set metadata information of outfile from infile",
       "outfile[,metadata]:infile[,metadata]" },
     { "map_chapters",  OPT_INT | HAS_ARG | OPT_EXPERT | OPT_OFFSET, {.off = OFFSET(chapters_input_file)},  "set chapters mapping", "input_file_index" },

configure

@@ -1139,7 +1139,6 @@ HAVE_LIST="
     asm_mod_y
     attribute_may_alias
     attribute_packed
-    bswap
     cbrtf
     closesocket
     cmov
@@ -2884,8 +2883,6 @@ EOF
     enabled ssse3 && check_asm ssse3 '"pabsw %xmm0, %xmm0"'
     enabled mmx2  && check_asm mmx2  '"pmaxub %mm0, %mm1"'
 
-    check_asm bswap '"bswap %%eax" ::: "%eax"'
-
     if ! disabled_any asm mmx yasm; then
         if check_cmd $yasmexe --version; then
             enabled x86_64 && yasm_extra="-m amd64"

doc/APIchanges

@@ -31,6 +31,10 @@ API changes, most recent first:
 2011-10-20 - b35e9e1 - lavu 51.22.0
   Add av_strtok() to avstring.h.
 
+2011-xx-xx - xxxxxxx - lavf 53.17.0
+  Add avformat_open_input().
+  Deprecate av_close_input_file() and av_close_input_stream().
+
 2011-xx-xx - xxxxxxx - lavc 53.25.0
   Add nb_samples and extended_data fields to AVFrame.
   Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE.

ffmpeg.c

@@ -657,7 +657,7 @@ void av_noreturn exit_program(int ret)
         av_dict_free(&output_files[i].opts);
     }
     for(i=0;i<nb_input_files;i++) {
-        av_close_input_file(input_files[i].ctx);
+        avformat_close_input(&input_files[i].ctx);
     }
     for (i = 0; i < nb_input_streams; i++) {
         av_freep(&input_streams[i].decoded_frame);
@@ -3423,7 +3423,7 @@ static int opt_input_file(OptionsContext *o, const char *opt, const char *filena
     ret = avformat_find_stream_info(ic, opts);
     if (ret < 0) {
         av_log(NULL, AV_LOG_FATAL, "%s: could not find codec parameters\n", filename);
-        av_close_input_file(ic);
+        avformat_close_input(&ic);
         exit_program(1);
     }
 
@@ -4667,7 +4667,7 @@ static const OptionDef options[] = {
     { "c", HAS_ARG | OPT_STRING | OPT_SPEC, {.off = OFFSET(codec_names)}, "codec name", "codec" },
     { "codec", HAS_ARG | OPT_STRING | OPT_SPEC, {.off = OFFSET(codec_names)}, "codec name", "codec" },
     { "pre", HAS_ARG | OPT_STRING | OPT_SPEC, {.off = OFFSET(presets)}, "preset name", "preset" },
-    { "map", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map}, "set input stream mapping", "file.stream[:syncfile.syncstream]" },
+    { "map", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map}, "set input stream mapping", "[-]input_file_id[:stream_specifier][,sync_file_id[:stream_specifier]]" },
     { "map_channel", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map_channel}, "map an audio channel from one stream to another", "file.stream.channel[:syncfile.syncstream]" },
     { "map_meta_data", HAS_ARG | OPT_EXPERT | OPT_FUNC2, {(void*)opt_map_meta_data}, "DEPRECATED set meta data information of outfile from infile",
       "outfile[,metadata]:infile[,metadata]" },

ffplay.c

@@ -2679,8 +2679,7 @@ static int read_thread(void *arg)
     if (is->subtitle_stream >= 0)
         stream_component_close(is, is->subtitle_stream);
     if (is->ic) {
-        av_close_input_file(is->ic);
-        is->ic = NULL; /* safety */
+        avformat_close_input(&is->ic);
     }
     avio_set_interrupt_cb(NULL);
 

ffprobe.c

@@ -1160,7 +1160,7 @@ static int probe_file(const char *filename)
     PRINT_CHAPTER(format);
     writer_print_footer(wctx);
 
-    av_close_input_file(fmt_ctx);
+    avformat_close_input(&fmt_ctx);
     writer_close(&wctx);
 
 end:

ffserver.c

@@ -849,7 +849,7 @@ static void close_connection(HTTPContext *c)
             if (st->codec->codec)
                 avcodec_close(st->codec);
         }
-        av_close_input_file(c->fmt_in);
+        avformat_close_input(&c->fmt_in);
     }
 
     /* free RTP output streams if any */
@@ -2169,7 +2169,7 @@ static int open_input_stream(HTTPContext *c, const char *info)
     c->fmt_in = s;
     if (strcmp(s->iformat->name, "ffm") && avformat_find_stream_info(c->fmt_in, NULL) < 0) {
         http_log("Could not find stream info '%s'\n", input_filename);
-        av_close_input_file(s);
+        avformat_close_input(&s);
         return -1;
     }
 
@@ -2311,8 +2311,7 @@ static int http_prepare_data(HTTPContext *c)
                     return 0;
                 } else {
                     if (c->stream->loop) {
-                        av_close_input_file(c->fmt_in);
-                        c->fmt_in = NULL;
+                        avformat_close_input(&c->fmt_in);
                         if (open_input_stream(c, "") < 0)
                             goto no_loop;
                         goto redo;
@@ -2736,7 +2735,7 @@ static int http_receive_data(HTTPContext *c)
 
             /* Now we have the actual streams */
             if (s->nb_streams != feed->nb_streams) {
-                av_close_input_stream(s);
+                avformat_close_input(&s);
                 av_free(pb);
                 http_log("Feed '%s' stream number does not match registered feed\n",
                          c->stream->feed_filename);
@@ -2749,7 +2748,7 @@ static int http_receive_data(HTTPContext *c)
                 avcodec_copy_context(fst->codec, st->codec);
             }
 
-            av_close_input_stream(s);
+            avformat_close_input(&s);
             av_free(pb);
         }
         c->buffer_ptr = c->buffer;
@@ -3629,7 +3628,7 @@ static void build_file_streams(void)
                 if (avformat_find_stream_info(infile, NULL) < 0) {
                     http_log("Could not find codec parameters from '%s'\n",
                              stream->feed_filename);
-                    av_close_input_file(infile);
+                    avformat_close_input(&infile);
                     goto fail;
                 }
                 extract_mpeg4_header(infile);
@@ -3637,7 +3636,7 @@ static void build_file_streams(void)
                 for(i=0;i<infile->nb_streams;i++)
                     add_av_stream1(stream, infile->streams[i]->codec, 1);
 
-                av_close_input_file(infile);
+                avformat_close_input(&infile);
             }
         }
     }
@@ -3727,7 +3726,7 @@ static void build_feed_streams(void)
                     http_log("Deleting feed file '%s' as stream counts differ (%d != %d)\n",
                         feed->feed_filename, s->nb_streams, feed->nb_streams);
 
-                av_close_input_file(s);
+                avformat_close_input(&s);
             } else
                 http_log("Deleting feed file '%s' as it appears to be corrupt\n",
                         feed->feed_filename);

libavcodec/aacpsy.c

@@ -216,7 +216,7 @@ static const float psy_fir_coeffs[] = {
 };
 
 /**
- * calculates the attack threshold for ABR from the above table for the LAME psy model
+ * Calculate the ABR attack threshold from the above LAME psymodel table.
  */
 static float lame_calc_attack_threshold(int bitrate)
 {

libavcodec/amrwbdec.c

@@ -111,7 +111,7 @@ static av_cold int amrwb_decode_init(AVCodecContext *avctx)
 
 /**
  * Decode the frame header in the "MIME/storage" format. This format
- * is simpler and does not carry the auxiliary information of the frame
+ * is simpler and does not carry the auxiliary frame information.
  *
  * @param[in] ctx                  The Context
  * @param[in] buf                  Pointer to the input buffer
@@ -133,7 +133,7 @@ static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
 }
 
 /**
- * Decodes quantized ISF vectors using 36-bit indexes (6K60 mode only)
+ * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
  *
  * @param[in]  ind                 Array of 5 indexes
  * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
@@ -160,7 +160,7 @@ static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
 }
 
 /**
- * Decodes quantized ISF vectors using 46-bit indexes (except 6K60 mode)
+ * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
  *
  * @param[in]  ind                 Array of 7 indexes
  * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
@@ -193,8 +193,8 @@ static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
 }
 
 /**
- * Apply mean and past ISF values using the prediction factor
- * Updates past ISF vector
+ * Apply mean and past ISF values using the prediction factor.
+ * Updates past ISF vector.
  *
  * @param[in,out] isf_q            Current quantized ISF
  * @param[in,out] isf_past         Past quantized ISF
@@ -215,7 +215,7 @@ static void isf_add_mean_and_past(float *isf_q, float *isf_past)
 
 /**
  * Interpolate the fourth ISP vector from current and past frames
- * to obtain a ISP vector for each subframe
+ * to obtain an ISP vector for each subframe.
  *
  * @param[in,out] isp_q            ISPs for each subframe
  * @param[in]     isp4_past        Past ISP for subframe 4
@@ -232,9 +232,9 @@ static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
 }
 
 /**
- * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes)
- * Calculate integer lag and fractional lag always using 1/4 resolution
- * In 1st and 3rd subframes the index is relative to last subframe integer lag
+ * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
+ * Calculate integer lag and fractional lag always using 1/4 resolution.
+ * In 1st and 3rd subframes the index is relative to last subframe integer lag.
  *
  * @param[out]    lag_int          Decoded integer pitch lag
  * @param[out]    lag_frac         Decoded fractional pitch lag
@@ -271,9 +271,9 @@ static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
 }
 
 /**
- * Decode a adaptive codebook index into pitch lag for 8k85 and 6k60 modes
- * Description is analogous to decode_pitch_lag_high, but in 6k60 relative
- * index is used for all subframes except the first
+ * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
+ * The description is analogous to decode_pitch_lag_high, but in 6k60 the
+ * relative index is used for all subframes except the first.
  */
 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
                                  uint8_t *base_lag_int, int subframe, enum Mode mode)
@@ -298,7 +298,7 @@ static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
 
 /**
  * Find the pitch vector by interpolating the past excitation at the
- * pitch delay, which is obtained in this function
+ * pitch delay, which is obtained in this function.
  *
  * @param[in,out] ctx              The context
  * @param[in]     amr_subframe     Current subframe data
@@ -351,10 +351,10 @@ static void decode_pitch_vector(AMRWBContext *ctx,
 /**
  * The next six functions decode_[i]p_track decode exactly i pulses
  * positions and amplitudes (-1 or 1) in a subframe track using
- * an encoded pulse indexing (TS 26.190 section 5.8.2)
+ * an encoded pulse indexing (TS 26.190 section 5.8.2).
  *
  * The results are given in out[], in which a negative number means
- * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) )
+ * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
  *
  * @param[out] out                 Output buffer (writes i elements)
  * @param[in]  code                Pulse index (no. of bits varies, see below)
@@ -470,7 +470,7 @@ static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bi
 
 /**
  * Decode the algebraic codebook index to pulse positions and signs,
- * then construct the algebraic codebook vector
+ * then construct the algebraic codebook vector.
  *
  * @param[out] fixed_vector        Buffer for the fixed codebook excitation
  * @param[in]  pulse_hi            MSBs part of the pulse index array (higher modes only)
@@ -541,7 +541,7 @@ static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
 }
 
 /**
- * Decode pitch gain and fixed gain correction factor
+ * Decode pitch gain and fixed gain correction factor.
  *
  * @param[in]  vq_gain             Vector-quantized index for gains
  * @param[in]  mode                Mode of the current frame
@@ -559,7 +559,7 @@ static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
 }
 
 /**
- * Apply pitch sharpening filters to the fixed codebook vector
+ * Apply pitch sharpening filters to the fixed codebook vector.
  *
  * @param[in]     ctx              The context
  * @param[in,out] fixed_vector     Fixed codebook excitation
@@ -580,7 +580,7 @@ static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
 }
 
 /**
- * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced)
+ * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
  *
  * @param[in] p_vector, f_vector   Pitch and fixed excitation vectors
  * @param[in] p_gain, f_gain       Pitch and fixed gains
@@ -599,8 +599,8 @@ static float voice_factor(float *p_vector, float p_gain,
 }
 
 /**
- * Reduce fixed vector sparseness by smoothing with one of three IR filters
- * Also known as "adaptive phase dispersion"
+ * Reduce fixed vector sparseness by smoothing with one of three IR filters,
+ * also known as "adaptive phase dispersion".
  *
  * @param[in]     ctx              The context
  * @param[in,out] fixed_vector     Unfiltered fixed vector
@@ -670,7 +670,7 @@ static float *anti_sparseness(AMRWBContext *ctx,
 
 /**
  * Calculate a stability factor {teta} based on distance between
- * current and past isf. A value of 1 shows maximum signal stability
+ * current and past isf. A value of 1 shows maximum signal stability.
  */
 static float stability_factor(const float *isf, const float *isf_past)
 {
@@ -687,7 +687,7 @@ static float stability_factor(const float *isf, const float *isf_past)
 
 /**
  * Apply a non-linear fixed gain smoothing in order to reduce
- * fluctuation in the energy of excitation
+ * fluctuation in the energy of excitation.
  *
  * @param[in]     fixed_gain       Unsmoothed fixed gain
  * @param[in,out] prev_tr_gain     Previous threshold gain (updated)
@@ -718,7 +718,7 @@ static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
 }
 
 /**
- * Filter the fixed_vector to emphasize the higher frequencies
+ * Filter the fixed_vector to emphasize the higher frequencies.
  *
  * @param[in,out] fixed_vector     Fixed codebook vector
  * @param[in]     voice_fac        Frame voicing factor
@@ -742,7 +742,7 @@ static void pitch_enhancer(float *fixed_vector, float voice_fac)
 }
 
 /**
- * Conduct 16th order linear predictive coding synthesis from excitation
+ * Conduct 16th order linear predictive coding synthesis from excitation.
  *
  * @param[in]     ctx              Pointer to the AMRWBContext
  * @param[in]     lpc              Pointer to the LPC coefficients
@@ -802,7 +802,7 @@ static void de_emphasis(float *out, float *in, float m, float mem[1])
 
 /**
  * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
- * a FIR interpolation filter. Uses past data from before *in address
+ * a FIR interpolation filter. Uses past data from before *in address.
  *
  * @param[out] out                 Buffer for interpolated signal
  * @param[in]  in                  Current signal data (length 0.8*o_size)
@@ -832,7 +832,7 @@ static void upsample_5_4(float *out, const float *in, int o_size)
 
 /**
  * Calculate the high-band gain based on encoded index (23k85 mode) or
- * on the low-band speech signal and the Voice Activity Detection flag
+ * on the low-band speech signal and the Voice Activity Detection flag.
  *
  * @param[in] ctx                  The context
  * @param[in] synth                LB speech synthesis at 12.8k
@@ -857,7 +857,7 @@ static float find_hb_gain(AMRWBContext *ctx, const float *synth,
 
 /**
  * Generate the high-band excitation with the same energy from the lower
- * one and scaled by the given gain
+ * one and scaled by the given gain.
  *
  * @param[in]  ctx                 The context
  * @param[out] hb_exc              Buffer for the excitation
@@ -880,7 +880,7 @@ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
 }
 
 /**
- * Calculate the auto-correlation for the ISF difference vector
+ * Calculate the auto-correlation for the ISF difference vector.
  */
 static float auto_correlation(float *diff_isf, float mean, int lag)
 {
@@ -896,7 +896,7 @@ static float auto_correlation(float *diff_isf, float mean, int lag)
 
 /**
  * Extrapolate a ISF vector to the 16kHz range (20th order LP)
- * used at mode 6k60 LP filter for the high frequency band
+ * used at mode 6k60 LP filter for the high frequency band.
  *
  * @param[out] out                 Buffer for extrapolated isf
  * @param[in]  isf                 Input isf vector
@@ -981,7 +981,7 @@ static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
 
 /**
  * Conduct 20th order linear predictive coding synthesis for the high
- * frequency band excitation at 16kHz
+ * frequency band excitation at 16kHz.
  *
  * @param[in]     ctx              The context
  * @param[in]     subframe         Current subframe index (0 to 3)
@@ -1019,8 +1019,8 @@ static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
 }
 
 /**
- * Apply to high-band samples a 15th order filter
- * The filter characteristic depends on the given coefficients
+ * Apply a 15th order filter to high-band samples.
+ * The filter characteristic depends on the given coefficients.
  *
  * @param[out]    out              Buffer for filtered output
  * @param[in]     fir_coef         Filter coefficients
@@ -1048,7 +1048,7 @@ static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
 }
 
 /**
- * Update context state before the next subframe
+ * Update context state before the next subframe.
  */
 static void update_sub_state(AMRWBContext *ctx)
 {

libavcodec/avcodec.h

@@ -2667,7 +2667,7 @@ typedef struct AVCodecContext {
 
 #if FF_API_X264_GLOBAL_OPTS
     /**
-     * Influences how often B-frames are used.
+     * Influence how often B-frames are used.
      * - encoding: Set by user.
      * - decoding: unused
      */
@@ -2748,7 +2748,7 @@ typedef struct AVCodecContext {
     int mv0_threshold;
 
     /**
-     * Adjusts sensitivity of b_frame_strategy 1.
+     * Adjust sensitivity of b_frame_strategy 1.
      * - encoding: Set by user.
      * - decoding: unused
      */
@@ -3032,7 +3032,7 @@ typedef struct AVCodecContext {
 
 #if FF_API_FLAC_GLOBAL_OPTS
     /**
-     * Determines which LPC analysis algorithm to use.
+     * Determine which LPC analysis algorithm to use.
      * - encoding: Set by user
      * - decoding: unused
      */
@@ -4263,7 +4263,7 @@ int avcodec_decode_subtitle2(AVCodecContext *avctx, AVSubtitle *sub,
                             AVPacket *avpkt);
 
 /**
- * Frees all allocated data in the given subtitle struct.
+ * Free all allocated data in the given subtitle struct.
  *
  * @param sub AVSubtitle to free.
  */
@@ -4628,8 +4628,16 @@ int av_picture_pad(AVPicture *dst, const AVPicture *src, int height, int width,
 unsigned int av_xiphlacing(unsigned char *s, unsigned int v);
 
 /**
+<<<<<<< HEAD
  * Logs a generic warning message about a missing feature. This function is
  * intended to be used internally by FFmpeg (libavcodec, libavformat, etc.)
+||||||| merged common ancestors
+ * Logs a generic warning message about a missing feature. This function is
+ * intended to be used internally by Libav (libavcodec, libavformat, etc.)
+=======
+ * Log a generic warning message about a missing feature. This function is
+ * intended to be used internally by Libav (libavcodec, libavformat, etc.)
+>>>>>>> qatar/master
  * only, and would normally not be used by applications.
  * @param[in] avc a pointer to an arbitrary struct of which the first field is
  * a pointer to an AVClass struct