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Merge remote-tracking branch 'qatar/master' 

* qatar/master: (23 commits)
  build: cosmetics: Reorder some lists in a more logical fashion
  x86: pngdsp: Fix assembly for OS/2
  fate: add test for RTjpeg in nuv with frameheader
  rtmp: send check_bw as notification
  g723_1: clip argument for 15-bit version of normalize_bits()
  g723_1: use all LPC vectors in formant postfilter
  id3v2: Support v2.2 PIC
  avplay: fix build with lavfi disabled.
  avconv: split configuring filter configuration to a separate file.
  avconv: split option parsing into a separate file.
  mpc8: do not leave padding after last frame in buffer for the next decode call
  mpegaudioenc: list supported channel layouts.
  mpegaudiodec: don't print an error on > 1 frame in a packet.
  api-example: update to new audio encoding API.
  configure: add --enable/disable-random option
  doc: cygwin: Update list of FATE package requirements
  build: Remove all installed headers and header directories on uninstall
  build: change checkheaders to use regular build rules
  rtmp: Add a new option 'rtmp_subscribe'
  rtmp: Add support for subscribing live streams
  ...

Conflicts:
	Makefile
	common.mak
	configure
	doc/examples/decoding_encoding.c
	ffmpeg.c
	libavcodec/g723_1.c
	libavcodec/mpegaudiodec.c
	libavcodec/x86/pngdsp.asm
	libavformat/version.h
	library.mak
	tests/fate/video.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-08-09 00:26:38 +02:00
parent 4270d8c04d 5864eb427f
commit 11a1033c9f
27 changed files with 3826 additions and 3472 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
.gitignore vendored
View File

@ -4,7 +4,7 @@
*.def
*.dll
*.exe
*.ho
*.h.c
*.lib
*.pc
*.so

6
Makefile
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@ -18,6 +18,7 @@ PROGS-$(CONFIG_FFSERVER) += ffserver
PROGS := $(PROGS-yes:%=%$(EXESUF))
INSTPROGS = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
OBJS = cmdutils.o
OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
TESTTOOLS = audiogen videogen rotozoom tiny_psnr base64
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
TOOLS = qt-faststart trasher
@ -68,8 +69,9 @@ config.h: .config
SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS \
ARCH_HEADERS BUILT_HEADERS SKIPHEADERS \
ALTIVEC-OBJS ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS MMI-OBJS \
MMX-OBJS NEON-OBJS VIS-OBJS YASM-OBJS \
ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS NEON-OBJS \
MMI-OBJS ALTIVEC-OBJS VIS-OBJS \
MMX-OBJS YASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS \
OBJS TESTOBJS

14
common.mak
View File

@ -10,8 +10,9 @@ ifndef SUBDIR
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX AS YASM AR LD HOSTCC STRIP CP
SILENT = DEPCC DEPAS DEPHOSTCC DEPYASM RM RANLIB
BRIEF = CC CXX HOSTCC AS YASM AR LD STRIP CP
SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
MSG = $@
M = @$(call ECHO,$(TAG),$@);
$(foreach VAR,$(BRIEF), \
@ -98,8 +99,9 @@ DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME)
ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
HEADEROBJS := $(filter-out $(SKIPHEADERS:.h=.ho),$(ALLHEADERS:.h=.ho))
checkheaders: $(HEADEROBJS)
HOBJS = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
checkheaders: $(HOBJS)
.SECONDARY: $(HOBJS:.o=.c)
alltools: $(TOOLS)
@ -117,8 +119,8 @@ $(TOOLOBJS): | tools
OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS) $(HEADEROBJS))
CLEANSUFFIXES = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
CLEANSUFFIXES = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d))
-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))

2
configure vendored
View File

@ -1211,6 +1211,7 @@ HAVE_LIST="
alsa_asoundlib_h
altivec_h
arpa_inet_h
asm_mod_q
asm_mod_y
asm_types_h
attribute_may_alias
@ -3119,6 +3120,7 @@ EOF
enabled neon && check_asm neon '"vadd.i16 q0, q0, q0"'
enabled vfpv3 && check_asm vfpv3 '"vmov.f32 s0, #1.0"'
check_asm asm_mod_q '"add r0, %Q0, %R0" :: "r"((long long)0)'
check_asm asm_mod_y '"vmul.i32 d0, d0, %y0" :: "x"(0)'
enabled_all armv6t2 shared !pic && enable_pic

136
doc/examples/decoding_encoding.c
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@ -34,6 +34,7 @@
#include <libavutil/imgutils.h>
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/audioconvert.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
@ -41,6 +42,59 @@
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channells = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channells) {
best_ch_layout = *p;
best_nb_channells = nb_channels;
}
p++;
}
return best_ch_layout;
}
/*
* Audio encoding example
*/
@ -48,11 +102,13 @@ static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame_size, i, j, out_size, outbuf_size;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f;
short *samples;
uint16_t *samples;
float t, tincr;
uint8_t *outbuf;
printf("Encode audio file %s\n", filename);
@ -67,9 +123,19 @@ static void audio_encode_example(const char *filename)
/* put sample parameters */
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "encoder does not support %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
@ -77,35 +143,71 @@ static void audio_encode_example(const char *filename)
exit(1);
}
/* the codec gives us the frame size, in samples */
frame_size = c->frame_size;
samples = malloc(frame_size * 2 * c->channels);
outbuf_size = 10000;
outbuf = malloc(outbuf_size);
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<200;i++) {
for(j=0;j<frame_size;j++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
samples[2*j+1] = samples[2*j];
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
fwrite(outbuf, 1, out_size, f);
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
fclose(f);
free(outbuf);
free(samples);
av_freep(&samples);
av_freep(&frame);
avcodec_close(c);
av_free(c);
}

4
doc/platform.texi
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@ -318,9 +318,9 @@ following "Devel" ones:
binutils, gcc4-core, make, git, mingw-runtime, texi2html
@end example
And the following "Utils" one:
In order to run FATE you will also need the following "Utils" packages:
@example
diffutils
bc, diffutils
@end example
Then run

5
doc/protocols.texi
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@ -267,6 +267,11 @@ value will be sent.
Stream identifier to play or to publish. This option overrides the
parameter specified in the URI.
@item rtmp_subscribe
Name of live stream to subscribe to. By default no value will be sent.
It is only sent if the option is specified or if rtmp_live
is set to live.
@item rtmp_swfurl
URL of the SWF player for the media. By default no value will be sent.

3377
ffmpeg.c
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File diff suppressed because it is too large Load Diff

391
ffmpeg.h Normal file
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@ -0,0 +1,391 @@
/*
* 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
*/
#ifndef FFMPEG_H
#define FFMPEG_H
#include "config.h"
#include <stdint.h>
#include <stdio.h>
#include <signal.h>
#if HAVE_PTHREADS
#include <pthread.h>
#endif
#include "cmdutils.h"
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavfilter/avfilter.h"
#include "libavfilter/avfiltergraph.h"
#include "libavutil/avutil.h"
#include "libavutil/dict.h"
#include "libavutil/fifo.h"
#include "libavutil/pixfmt.h"
#include "libavutil/rational.h"
#include "libswresample/swresample.h"
#define VSYNC_AUTO -1
#define VSYNC_PASSTHROUGH 0
#define VSYNC_CFR 1
#define VSYNC_VFR 2
#define VSYNC_DROP 0xff
#define MAX_STREAMS 1024 /* arbitrary sanity check value */
/* select an input stream for an output stream */
typedef struct StreamMap {
int disabled; /** 1 is this mapping is disabled by a negative map */
int file_index;
int stream_index;
int sync_file_index;
int sync_stream_index;
char *linklabel; /** name of an output link, for mapping lavfi outputs */
} StreamMap;
typedef struct {
int file_idx, stream_idx, channel_idx; // input
int ofile_idx, ostream_idx; // output
} AudioChannelMap;
typedef struct OptionsContext {
/* input/output options */
int64_t start_time;
const char *format;
SpecifierOpt *codec_names;
int nb_codec_names;
SpecifierOpt *audio_channels;
int nb_audio_channels;
SpecifierOpt *audio_sample_rate;
int nb_audio_sample_rate;
SpecifierOpt *frame_rates;
int nb_frame_rates;
SpecifierOpt *frame_sizes;
int nb_frame_sizes;
SpecifierOpt *frame_pix_fmts;
int nb_frame_pix_fmts;
/* input options */
int64_t input_ts_offset;
int rate_emu;
SpecifierOpt *ts_scale;
int nb_ts_scale;
SpecifierOpt *dump_attachment;
int nb_dump_attachment;
/* output options */
StreamMap *stream_maps;
int nb_stream_maps;
AudioChannelMap *audio_channel_maps; /* one info entry per -map_channel */
int nb_audio_channel_maps; /* number of (valid) -map_channel settings */
int metadata_global_manual;
int metadata_streams_manual;
int metadata_chapters_manual;
const char **attachments;
int nb_attachments;
int chapters_input_file;
int64_t recording_time;
uint64_t limit_filesize;
float mux_preload;
float mux_max_delay;
int video_disable;
int audio_disable;
int subtitle_disable;
int data_disable;
/* indexed by output file stream index */
int *streamid_map;
int nb_streamid_map;
SpecifierOpt *metadata;
int nb_metadata;
SpecifierOpt *max_frames;
int nb_max_frames;
SpecifierOpt *bitstream_filters;
int nb_bitstream_filters;
SpecifierOpt *codec_tags;
int nb_codec_tags;
SpecifierOpt *sample_fmts;
int nb_sample_fmts;
SpecifierOpt *qscale;
int nb_qscale;
SpecifierOpt *forced_key_frames;
int nb_forced_key_frames;
SpecifierOpt *force_fps;
int nb_force_fps;
SpecifierOpt *frame_aspect_ratios;
int nb_frame_aspect_ratios;
SpecifierOpt *rc_overrides;
int nb_rc_overrides;
SpecifierOpt *intra_matrices;
int nb_intra_matrices;
SpecifierOpt *inter_matrices;
int nb_inter_matrices;
SpecifierOpt *top_field_first;
int nb_top_field_first;
SpecifierOpt *metadata_map;
int nb_metadata_map;
SpecifierOpt *presets;
int nb_presets;
SpecifierOpt *copy_initial_nonkeyframes;
int nb_copy_initial_nonkeyframes;
SpecifierOpt *filters;
int nb_filters;
} OptionsContext;
typedef struct InputFilter {
AVFilterContext *filter;
struct InputStream *ist;
struct FilterGraph *graph;
uint8_t *name;
} InputFilter;
typedef struct OutputFilter {
AVFilterContext *filter;
struct OutputStream *ost;
struct FilterGraph *graph;
uint8_t *name;
/* temporary storage until stream maps are processed */
AVFilterInOut *out_tmp;
} OutputFilter;
typedef struct FilterGraph {
int index;
const char *graph_desc;
AVFilterGraph *graph;
InputFilter **inputs;
int nb_inputs;
OutputFilter **outputs;
int nb_outputs;
} FilterGraph;
typedef struct InputStream {
int file_index;
AVStream *st;
int discard; /* true if stream data should be discarded */
int decoding_needed; /* true if the packets must be decoded in 'raw_fifo' */
AVCodec *dec;
AVFrame *decoded_frame;
int64_t start; /* time when read started */
/* predicted dts of the next packet read for this stream or (when there are
* several frames in a packet) of the next frame in current packet (in AV_TIME_BASE units) */
int64_t next_dts;
int64_t dts; ///< dts of the last packet read for this stream (in AV_TIME_BASE units)
int64_t next_pts; ///< synthetic pts for the next decode frame (in AV_TIME_BASE units)
int64_t pts; ///< current pts of the decoded frame (in AV_TIME_BASE units)
int wrap_correction_done;
double ts_scale;
int is_start; /* is 1 at the start and after a discontinuity */
int saw_first_ts;
int showed_multi_packet_warning;
AVDictionary *opts;
AVRational framerate; /* framerate forced with -r */
int top_field_first;
int resample_height;
int resample_width;
int resample_pix_fmt;
int resample_sample_fmt;
int resample_sample_rate;
int resample_channels;
uint64_t resample_channel_layout;
struct sub2video {
int64_t last_pts;
AVFilterBufferRef *ref;
int w, h;
} sub2video;
/* a pool of free buffers for decoded data */
FrameBuffer *buffer_pool;
int dr1;
/* decoded data from this stream goes into all those filters
* currently video and audio only */
InputFilter **filters;
int nb_filters;
} InputStream;
typedef struct InputFile {
AVFormatContext *ctx;
int eof_reached; /* true if eof reached */
int unavailable; /* true if the file is unavailable (possibly temporarily) */
int ist_index; /* index of first stream in input_streams */
int64_t ts_offset;
int nb_streams; /* number of stream that ffmpeg is aware of; may be different
from ctx.nb_streams if new streams appear during av_read_frame() */
int nb_streams_warn; /* number of streams that the user was warned of */
int rate_emu;
#if HAVE_PTHREADS
pthread_t thread; /* thread reading from this file */
int finished; /* the thread has exited */
int joined; /* the thread has been joined */
pthread_mutex_t fifo_lock; /* lock for access to fifo */
pthread_cond_t fifo_cond; /* the main thread will signal on this cond after reading from fifo */
AVFifoBuffer *fifo; /* demuxed packets are stored here; freed by the main thread */
#endif
} InputFile;
typedef struct OutputStream {
int file_index; /* file index */
int index; /* stream index in the output file */
int source_index; /* InputStream index */
AVStream *st; /* stream in the output file */
int encoding_needed; /* true if encoding needed for this stream */
int frame_number;
/* input pts and corresponding output pts
for A/V sync */
struct InputStream *sync_ist; /* input stream to sync against */
int64_t sync_opts; /* output frame counter, could be changed to some true timestamp */ // FIXME look at frame_number
/* pts of the first frame encoded for this stream, used for limiting
* recording time */
int64_t first_pts;
AVBitStreamFilterContext *bitstream_filters;
AVCodec *enc;
int64_t max_frames;
AVFrame *filtered_frame;
/* video only */
AVRational frame_rate;
int force_fps;
int top_field_first;
float frame_aspect_ratio;
float last_quality;
/* forced key frames */
int64_t *forced_kf_pts;
int forced_kf_count;
int forced_kf_index;
char *forced_keyframes;
/* audio only */
int audio_channels_map[SWR_CH_MAX]; /* list of the channels id to pick from the source stream */
int audio_channels_mapped; /* number of channels in audio_channels_map */
FILE *logfile;
OutputFilter *filter;
char *avfilter;
int64_t sws_flags;
int64_t swr_dither_method;
double swr_dither_scale;
AVDictionary *opts;
int is_past_recording_time;
int unavailable; /* true if the steram is unavailable (possibly temporarily) */
int stream_copy;
const char *attachment_filename;
int copy_initial_nonkeyframes;
int keep_pix_fmt;
} OutputStream;
typedef struct OutputFile {
AVFormatContext *ctx;
AVDictionary *opts;
int ost_index; /* index of the first stream in output_streams */
int64_t recording_time; ///< desired length of the resulting file in microseconds == AV_TIME_BASE units
int64_t start_time; ///< start time in microseconds == AV_TIME_BASE units
uint64_t limit_filesize; /* filesize limit expressed in bytes */
} OutputFile;
extern InputStream **input_streams;
extern int nb_input_streams;
extern InputFile **input_files;
extern int nb_input_files;
extern OutputStream **output_streams;
extern int nb_output_streams;
extern OutputFile **output_files;
extern int nb_output_files;
extern FilterGraph **filtergraphs;
extern int nb_filtergraphs;
extern const char *pass_logfilename_prefix;
extern char *vstats_filename;
extern float audio_drift_threshold;
extern float dts_delta_threshold;
extern float dts_error_threshold;
extern int audio_volume;
extern int audio_sync_method;
extern int video_sync_method;
extern int do_benchmark;
extern int do_benchmark_all;
extern int do_deinterlace;
extern int do_hex_dump;
extern int do_pkt_dump;
extern int copy_ts;
extern int copy_tb;
extern int debug_ts;
extern int opt_shortest;
extern int exit_on_error;
extern int print_stats;
extern int qp_hist;
extern int same_quant;
extern int stdin_interaction;
extern int frame_bits_per_raw_sample;
extern AVIOContext *progress_avio;
extern const AVIOInterruptCB int_cb;
extern const OptionDef options[];
void term_init(void);
void term_exit(void);
void reset_options(OptionsContext *o, int is_input);
void show_usage(void);
void opt_output_file(void *optctx, const char *filename);
void assert_avoptions(AVDictionary *m);
int guess_input_channel_layout(InputStream *ist);
enum PixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum PixelFormat target);
void choose_sample_fmt(AVStream *st, AVCodec *codec);
int configure_filtergraph(FilterGraph *fg);
int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out);
int ist_in_filtergraph(FilterGraph *fg, InputStream *ist);
FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost);
#endif /* FFMPEG_H */

789
ffmpeg_filter.c Normal file
View File

@ -0,0 +1,789 @@
/*
* ffmpeg filter configuration
*
* 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 "ffmpeg.h"
#include "libavfilter/avfilter.h"
#include "libavfilter/avfiltergraph.h"
#include "libavfilter/buffersink.h"
#include "libavutil/audioconvert.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/bprint.h"
#include "libavutil/pixdesc.h"
#include "libavutil/pixfmt.h"
#include "libavutil/imgutils.h"
#include "libavutil/samplefmt.h"
enum PixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum PixelFormat target)
{
if (codec && codec->pix_fmts) {
const enum PixelFormat *p = codec->pix_fmts;
int has_alpha= av_pix_fmt_descriptors[target].nb_components % 2 == 0;
enum PixelFormat best= PIX_FMT_NONE;
if (st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
if (st->codec->codec_id == AV_CODEC_ID_MJPEG) {
p = (const enum PixelFormat[]) { PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUV420P, PIX_FMT_YUV422P, PIX_FMT_NONE };
} else if (st->codec->codec_id == AV_CODEC_ID_LJPEG) {
p = (const enum PixelFormat[]) { PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUVJ444P, PIX_FMT_YUV420P,
PIX_FMT_YUV422P, PIX_FMT_YUV444P, PIX_FMT_BGRA, PIX_FMT_NONE };
}
}
for (; *p != PIX_FMT_NONE; p++) {
best= avcodec_find_best_pix_fmt2(best, *p, target, has_alpha, NULL);
if (*p == target)
break;
}
if (*p == PIX_FMT_NONE) {
if (target != PIX_FMT_NONE)
av_log(NULL, AV_LOG_WARNING,
"Incompatible pixel format '%s' for codec '%s', auto-selecting format '%s'\n",
av_pix_fmt_descriptors[target].name,
codec->name,
av_pix_fmt_descriptors[best].name);
return best;
}
}
return target;
}
void choose_sample_fmt(AVStream *st, AVCodec *codec)
{
if (codec && codec->sample_fmts) {
const enum AVSampleFormat *p = codec->sample_fmts;
for (; *p != -1; p++) {
if (*p == st->codec->sample_fmt)
break;
}
if (*p == -1) {
if((codec->capabilities & CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
av_log(NULL, AV_LOG_ERROR, "Conversion will not be lossless.\n");
if(av_get_sample_fmt_name(st->codec->sample_fmt))
av_log(NULL, AV_LOG_WARNING,
"Incompatible sample format '%s' for codec '%s', auto-selecting format '%s'\n",
av_get_sample_fmt_name(st->codec->sample_fmt),
codec->name,
av_get_sample_fmt_name(codec->sample_fmts[0]));
st->codec->sample_fmt = codec->sample_fmts[0];
}
}
}
static char *choose_pix_fmts(OutputStream *ost)
{
if (ost->keep_pix_fmt) {
if (ost->filter)
avfilter_graph_set_auto_convert(ost->filter->graph->graph,
AVFILTER_AUTO_CONVERT_NONE);
if (ost->st->codec->pix_fmt == PIX_FMT_NONE)
return NULL;
return av_strdup(av_get_pix_fmt_name(ost->st->codec->pix_fmt));
}
if (ost->st->codec->pix_fmt != PIX_FMT_NONE) {
return av_strdup(av_get_pix_fmt_name(choose_pixel_fmt(ost->st, ost->enc, ost->st->codec->pix_fmt)));
} else if (ost->enc && ost->enc->pix_fmts) {
const enum PixelFormat *p;
AVIOContext *s = NULL;
uint8_t *ret;
int len;
if (avio_open_dyn_buf(&s) < 0)
exit_program(1);
p = ost->enc->pix_fmts;
if (ost->st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
if (ost->st->codec->codec_id == AV_CODEC_ID_MJPEG) {
p = (const enum PixelFormat[]) { PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUV420P, PIX_FMT_YUV422P, PIX_FMT_NONE };
} else if (ost->st->codec->codec_id == AV_CODEC_ID_LJPEG) {
p = (const enum PixelFormat[]) { PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUVJ444P, PIX_FMT_YUV420P,
PIX_FMT_YUV422P, PIX_FMT_YUV444P, PIX_FMT_BGRA, PIX_FMT_NONE };
}
}
for (; *p != PIX_FMT_NONE; p++) {
const char *name = av_get_pix_fmt_name(*p);
avio_printf(s, "%s:", name);
}
len = avio_close_dyn_buf(s, &ret);
ret[len - 1] = 0;
return ret;
} else
return NULL;
}
/**
* Define a function for building a string containing a list of
* allowed formats,
*/
#define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name, separator)\
static char *choose_ ## var ## s(OutputStream *ost) \
{ \
if (ost->st->codec->var != none) { \
get_name(ost->st->codec->var); \
return av_strdup(name); \
} else if (ost->enc->supported_list) { \
const type *p; \
AVIOContext *s = NULL; \
uint8_t *ret; \
int len; \
\
if (avio_open_dyn_buf(&s) < 0) \
exit_program(1); \
\
for (p = ost->enc->supported_list; *p != none; p++) { \
get_name(*p); \
avio_printf(s, "%s" separator, name); \
} \
len = avio_close_dyn_buf(s, &ret); \
ret[len - 1] = 0; \
return ret; \
} else \
return NULL; \
}
#define GET_PIX_FMT_NAME(pix_fmt)\
const char *name = av_get_pix_fmt_name(pix_fmt);
// DEF_CHOOSE_FORMAT(enum PixelFormat, pix_fmt, pix_fmts, PIX_FMT_NONE,
// GET_PIX_FMT_NAME, ":")
#define GET_SAMPLE_FMT_NAME(sample_fmt)\
const char *name = av_get_sample_fmt_name(sample_fmt)
DEF_CHOOSE_FORMAT(enum AVSampleFormat, sample_fmt, sample_fmts,
AV_SAMPLE_FMT_NONE, GET_SAMPLE_FMT_NAME, ",")
#define GET_SAMPLE_RATE_NAME(rate)\
char name[16];\
snprintf(name, sizeof(name), "%d", rate);
DEF_CHOOSE_FORMAT(int, sample_rate, supported_samplerates, 0,
GET_SAMPLE_RATE_NAME, ",")
#define GET_CH_LAYOUT_NAME(ch_layout)\
char name[16];\
snprintf(name, sizeof(name), "0x%"PRIx64, ch_layout);
DEF_CHOOSE_FORMAT(uint64_t, channel_layout, channel_layouts, 0,
GET_CH_LAYOUT_NAME, ",")
FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost)
{
FilterGraph *fg = av_mallocz(sizeof(*fg));
if (!fg)
exit_program(1);
fg->index = nb_filtergraphs;
fg->outputs = grow_array(fg->outputs, sizeof(*fg->outputs), &fg->nb_outputs,
fg->nb_outputs + 1);
if (!(fg->outputs[0] = av_mallocz(sizeof(*fg->outputs[0]))))
exit_program(1);
fg->outputs[0]->ost = ost;
fg->outputs[0]->graph = fg;
ost->filter = fg->outputs[0];
fg->inputs = grow_array(fg->inputs, sizeof(*fg->inputs), &fg->nb_inputs,
fg->nb_inputs + 1);
if (!(fg->inputs[0] = av_mallocz(sizeof(*fg->inputs[0]))))
exit_program(1);
fg->inputs[0]->ist = ist;
fg->inputs[0]->graph = fg;
ist->filters = grow_array(ist->filters, sizeof(*ist->filters),
&ist->nb_filters, ist->nb_filters + 1);
ist->filters[ist->nb_filters - 1] = fg->inputs[0];
filtergraphs = grow_array(filtergraphs, sizeof(*filtergraphs),
&nb_filtergraphs, nb_filtergraphs + 1);
filtergraphs[nb_filtergraphs - 1] = fg;
return fg;
}
static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
{
InputStream *ist = NULL;
enum AVMediaType type = avfilter_pad_get_type(in->filter_ctx->input_pads, in->pad_idx);
int i;
// TODO: support other filter types
if (type != AVMEDIA_TYPE_VIDEO && type != AVMEDIA_TYPE_AUDIO) {
av_log(NULL, AV_LOG_FATAL, "Only video and audio filters supported "
"currently.\n");
exit_program(1);
}
if (in->name) {
AVFormatContext *s;
AVStream *st = NULL;
char *p;
int file_idx = strtol(in->name, &p, 0);
if (file_idx < 0 || file_idx >= nb_input_files) {
av_log(NULL, AV_LOG_FATAL, "Invalid file index %d in filtergraph description %s.\n",
file_idx, fg->graph_desc);
exit_program(1);
}
s = input_files[file_idx]->ctx;
for (i = 0; i < s->nb_streams; i++) {
enum AVMediaType stream_type = s->streams[i]->codec->codec_type;
if (stream_type != type &&
!(stream_type == AVMEDIA_TYPE_SUBTITLE &&
type == AVMEDIA_TYPE_VIDEO /* sub2video hack */))
continue;
if (check_stream_specifier(s, s->streams[i], *p == ':' ? p + 1 : p) == 1) {
st = s->streams[i];
break;
}
}
if (!st) {
av_log(NULL, AV_LOG_FATAL, "Stream specifier '%s' in filtergraph description %s "
"matches no streams.\n", p, fg->graph_desc);
exit_program(1);
}
ist = input_streams[input_files[file_idx]->ist_index + st->index];
} else {
/* find the first unused stream of corresponding type */
for (i = 0; i < nb_input_streams; i++) {
ist = input_streams[i];
if (ist->st->codec->codec_type == type && ist->discard)
break;
}
if (i == nb_input_streams) {
av_log(NULL, AV_LOG_FATAL, "Cannot find a matching stream for "
"unlabeled input pad %d on filter %s\n", in->pad_idx,
in->filter_ctx->name);
exit_program(1);
}
}
av_assert0(ist);
ist->discard = 0;
ist->decoding_needed = 1;
ist->st->discard = AVDISCARD_NONE;
fg->inputs = grow_array(fg->inputs, sizeof(*fg->inputs),
&fg->nb_inputs, fg->nb_inputs + 1);
if (!(fg->inputs[fg->nb_inputs - 1] = av_mallocz(sizeof(*fg->inputs[0]))))
exit_program(1);
fg->inputs[fg->nb_inputs - 1]->ist = ist;
fg->inputs[fg->nb_inputs - 1]->graph = fg;
ist->filters = grow_array(ist->filters, sizeof(*ist->filters),
&ist->nb_filters, ist->nb_filters + 1);
ist->filters[ist->nb_filters - 1] = fg->inputs[fg->nb_inputs - 1];
}
static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
{
char *pix_fmts;
OutputStream *ost = ofilter->ost;
AVCodecContext *codec = ost->st->codec;
AVFilterContext *last_filter = out->filter_ctx;
int pad_idx = out->pad_idx;
int ret;
char name[255];
AVBufferSinkParams *buffersink_params = av_buffersink_params_alloc();
snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
ret = avfilter_graph_create_filter(&ofilter->filter,
avfilter_get_by_name("buffersink"),
name, NULL, NULL/*buffersink_params*/, fg->graph);
av_freep(&buffersink_params);
if (ret < 0)
return ret;
if (codec->width || codec->height) {
char args[255];
AVFilterContext *filter;
snprintf(args, sizeof(args), "%d:%d:flags=0x%X",
codec->width,
codec->height,
(unsigned)ost->sws_flags);
snprintf(name, sizeof(name), "scaler for output stream %d:%d",
ost->file_index, ost->index);
if ((ret = avfilter_graph_create_filter(&filter, avfilter_get_by_name("scale"),
name, args, NULL, fg->graph)) < 0)
return ret;
if ((ret = avfilter_link(last_filter, pad_idx, filter, 0)) < 0)
return ret;
last_filter = filter;
pad_idx = 0;
}
if ((pix_fmts = choose_pix_fmts(ost))) {
AVFilterContext *filter;
snprintf(name, sizeof(name), "pixel format for output stream %d:%d",
ost->file_index, ost->index);
if ((ret = avfilter_graph_create_filter(&filter,
avfilter_get_by_name("format"),
"format", pix_fmts, NULL,
fg->graph)) < 0)
return ret;
if ((ret = avfilter_link(last_filter, pad_idx, filter, 0)) < 0)
return ret;
last_filter = filter;
pad_idx = 0;
av_freep(&pix_fmts);
}
if (ost->frame_rate.num && 0) {
AVFilterContext *fps;
char args[255];
snprintf(args, sizeof(args), "fps=%d/%d", ost->frame_rate.num,
ost->frame_rate.den);
snprintf(name, sizeof(name), "fps for output stream %d:%d",
ost->file_index, ost->index);
ret = avfilter_graph_create_filter(&fps, avfilter_get_by_name("fps"),
name, args, NULL, fg->graph);
if (ret < 0)
return ret;
ret = avfilter_link(last_filter, pad_idx, fps, 0);
if (ret < 0)
return ret;
last_filter = fps;
pad_idx = 0;
}
if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
return ret;
return 0;
}
static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
{
OutputStream *ost = ofilter->ost;
AVCodecContext *codec = ost->st->codec;
AVFilterContext *last_filter = out->filter_ctx;
int pad_idx = out->pad_idx;
char *sample_fmts, *sample_rates, *channel_layouts;
char name[255];
int ret;
snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
ret = avfilter_graph_create_filter(&ofilter->filter,
avfilter_get_by_name("abuffersink"),
name, NULL, NULL, fg->graph);
if (ret < 0)
return ret;
#define AUTO_INSERT_FILTER(opt_name, filter_name, arg) do { \
AVFilterContext *filt_ctx; \
\
av_log(NULL, AV_LOG_INFO, opt_name " is forwarded to lavfi " \
"similarly to -af " filter_name "=%s.\n", arg); \
\
ret = avfilter_graph_create_filter(&filt_ctx, \
avfilter_get_by_name(filter_name), \
filter_name, arg, NULL, fg->graph); \
if (ret < 0) \
return ret; \
\
ret = avfilter_link(last_filter, pad_idx, filt_ctx, 0); \
if (ret < 0) \
return ret; \
\
last_filter = filt_ctx; \
pad_idx = 0; \
} while (0)
if (ost->audio_channels_mapped) {
int i;
AVBPrint pan_buf;
av_bprint_init(&pan_buf, 256, 8192);
av_bprintf(&pan_buf, "0x%"PRIx64,
av_get_default_channel_layout(ost->audio_channels_mapped));
for (i = 0; i < ost->audio_channels_mapped; i++)
if (ost->audio_channels_map[i] != -1)
av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]);
AUTO_INSERT_FILTER("-map_channel", "pan", pan_buf.str);
av_bprint_finalize(&pan_buf, NULL);
}
if (codec->channels && !codec->channel_layout)
codec->channel_layout = av_get_default_channel_layout(codec->channels);
sample_fmts = choose_sample_fmts(ost);
sample_rates = choose_sample_rates(ost);
channel_layouts = choose_channel_layouts(ost);
if (sample_fmts || sample_rates || channel_layouts) {
AVFilterContext *format;
char args[256];
int len = 0;
if (sample_fmts)
len += snprintf(args + len, sizeof(args) - len, "sample_fmts=%s:",
sample_fmts);
if (sample_rates)
len += snprintf(args + len, sizeof(args) - len, "sample_rates=%s:",
sample_rates);
if (channel_layouts)
len += snprintf(args + len, sizeof(args) - len, "channel_layouts=%s:",
channel_layouts);
args[len - 1] = 0;
av_freep(&sample_fmts);
av_freep(&sample_rates);
av_freep(&channel_layouts);
snprintf(name, sizeof(name), "audio format for output stream %d:%d",
ost->file_index, ost->index);
ret = avfilter_graph_create_filter(&format,
avfilter_get_by_name("aformat"),
name, args, NULL, fg->graph);
if (ret < 0)
return ret;
ret = avfilter_link(last_filter, pad_idx, format, 0);
if (ret < 0)
return ret;
last_filter = format;
pad_idx = 0;
}
if (audio_volume != 256 && 0) {
char args[256];
snprintf(args, sizeof(args), "%f", audio_volume / 256.);
AUTO_INSERT_FILTER("-vol", "volume", args);
}
if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
return ret;
return 0;
}
#define DESCRIBE_FILTER_LINK(f, inout, in) \
{ \
AVFilterContext *ctx = inout->filter_ctx; \
AVFilterPad *pads = in ? ctx->input_pads : ctx->output_pads; \
int nb_pads = in ? ctx->input_count : ctx->output_count; \
AVIOContext *pb; \
\
if (avio_open_dyn_buf(&pb) < 0) \
exit_program(1); \
\
avio_printf(pb, "%s", ctx->filter->name); \
if (nb_pads > 1) \
avio_printf(pb, ":%s", avfilter_pad_get_name(pads, inout->pad_idx));\
avio_w8(pb, 0); \
avio_close_dyn_buf(pb, &f->name); \
}
int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
{
av_freep(&ofilter->name);
DESCRIBE_FILTER_LINK(ofilter, out, 0);
switch (avfilter_pad_get_type(out->filter_ctx->output_pads, out->pad_idx)) {
case AVMEDIA_TYPE_VIDEO: return configure_output_video_filter(fg, ofilter, out);
case AVMEDIA_TYPE_AUDIO: return configure_output_audio_filter(fg, ofilter, out);
default: av_assert0(0);
}
}
static int sub2video_prepare(InputStream *ist)
{
AVFormatContext *avf = input_files[ist->file_index]->ctx;
int i, ret, w, h;
uint8_t *image[4];
int linesize[4];
/* Compute the size of the canvas for the subtitles stream.
If the subtitles codec has set a size, use it. Otherwise use the
maximum dimensions of the video streams in the same file. */
w = ist->st->codec->width;
h = ist->st->codec->height;
if (!(w && h)) {
for (i = 0; i < avf->nb_streams; i++) {
if (avf->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
w = FFMAX(w, avf->streams[i]->codec->width);
h = FFMAX(h, avf->streams[i]->codec->height);
}
}
if (!(w && h)) {
w = FFMAX(w, 720);
h = FFMAX(h, 576);
}
av_log(avf, AV_LOG_INFO, "sub2video: using %dx%d canvas\n", w, h);
}
ist->sub2video.w = ist->st->codec->width = w;
ist->sub2video.h = ist->st->codec->height = h;
/* rectangles are PIX_FMT_PAL8, but we have no guarantee that the
palettes for all rectangles are identical or compatible */
ist->st->codec->pix_fmt = PIX_FMT_RGB32;
ret = av_image_alloc(image, linesize, w, h, PIX_FMT_RGB32, 32);
if (ret < 0)
return ret;
memset(image[0], 0, h * linesize[0]);
ist->sub2video.ref = avfilter_get_video_buffer_ref_from_arrays(
image, linesize, AV_PERM_READ | AV_PERM_PRESERVE,
w, h, PIX_FMT_RGB32);
if (!ist->sub2video.ref) {
av_free(image[0]);
return AVERROR(ENOMEM);
}
return 0;
}
static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
AVFilterInOut *in)
{
AVFilterContext *first_filter = in->filter_ctx;
AVFilter *filter = avfilter_get_by_name("buffer");
InputStream *ist = ifilter->ist;
AVRational tb = ist->framerate.num ? av_inv_q(ist->framerate) :
ist->st->time_base;
AVRational fr = ist->framerate.num ? ist->framerate :
ist->st->r_frame_rate;
AVRational sar;
AVBPrint args;
char name[255];
int pad_idx = in->pad_idx;
int ret;
if (ist->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
ret = sub2video_prepare(ist);
if (ret < 0)
return ret;
}
sar = ist->st->sample_aspect_ratio.num ?
ist->st->sample_aspect_ratio :
ist->st->codec->sample_aspect_ratio;
if(!sar.den)
sar = (AVRational){0,1};
av_bprint_init(&args, 0, 1);
av_bprintf(&args,
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:"
"pixel_aspect=%d/%d:sws_param=flags=%d", ist->st->codec->width,
ist->st->codec->height, ist->st->codec->pix_fmt,
tb.num, tb.den, sar.num, sar.den,
SWS_BILINEAR + ((ist->st->codec->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
if (fr.num && fr.den)
av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den);
snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
ist->file_index, ist->st->index);
if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter, name,
args.str, NULL, fg->graph)) < 0)
return ret;
if (ist->framerate.num) {
AVFilterContext *setpts;
snprintf(name, sizeof(name), "force CFR for input from stream %d:%d",
ist->file_index, ist->st->index);
if ((ret = avfilter_graph_create_filter(&setpts,
avfilter_get_by_name("setpts"),
name, "N", NULL,
fg->graph)) < 0)
return ret;
if ((ret = avfilter_link(setpts, 0, first_filter, pad_idx)) < 0)
return ret;
first_filter = setpts;
pad_idx = 0;
}
if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
return ret;
return 0;
}
static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
AVFilterInOut *in)
{
AVFilterContext *first_filter = in->filter_ctx;
AVFilter *filter = avfilter_get_by_name("abuffer");
InputStream *ist = ifilter->ist;
int pad_idx = in->pad_idx;
char args[255], name[255];
int ret;
snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s"
":channel_layout=0x%"PRIx64,
1, ist->st->codec->sample_rate,
ist->st->codec->sample_rate,
av_get_sample_fmt_name(ist->st->codec->sample_fmt),
ist->st->codec->channel_layout);
snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
ist->file_index, ist->st->index);
if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter,
name, args, NULL,
fg->graph)) < 0)
return ret;
#define AUTO_INSERT_FILTER_INPUT(opt_name, filter_name, arg) do { \
AVFilterContext *filt_ctx; \
\
av_log(NULL, AV_LOG_INFO, opt_name " is forwarded to lavfi " \
"similarly to -af " filter_name "=%s.\n", arg); \
\
snprintf(name, sizeof(name), "graph %d %s for input stream %d:%d", \
fg->index, filter_name, ist->file_index, ist->st->index); \
ret = avfilter_graph_create_filter(&filt_ctx, \
avfilter_get_by_name(filter_name), \
name, arg, NULL, fg->graph); \
if (ret < 0) \
return ret; \
\
ret = avfilter_link(filt_ctx, 0, first_filter, pad_idx); \
if (ret < 0) \
return ret; \
\
first_filter = filt_ctx; \
} while (0)
if (audio_sync_method > 0) {
char args[256] = {0};
av_strlcatf(args, sizeof(args), "min_comp=0.001:min_hard_comp=%f", audio_drift_threshold);
if (audio_sync_method > 1)
av_strlcatf(args, sizeof(args), ":max_soft_comp=%f", audio_sync_method/(double)ist->st->codec->sample_rate);
AUTO_INSERT_FILTER_INPUT("-async", "aresample", args);
}
// if (ost->audio_channels_mapped) {
// int i;
// AVBPrint pan_buf;
// av_bprint_init(&pan_buf, 256, 8192);
// av_bprintf(&pan_buf, "0x%"PRIx64,
// av_get_default_channel_layout(ost->audio_channels_mapped));
// for (i = 0; i < ost->audio_channels_mapped; i++)
// if (ost->audio_channels_map[i] != -1)
// av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]);
// AUTO_INSERT_FILTER_INPUT("-map_channel", "pan", pan_buf.str);
// av_bprint_finalize(&pan_buf, NULL);
// }
if (audio_volume != 256) {
char args[256];
snprintf(args, sizeof(args), "%f", audio_volume / 256.);
AUTO_INSERT_FILTER_INPUT("-vol", "volume", args);
}
if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
return ret;
return 0;
}
static int configure_input_filter(FilterGraph *fg, InputFilter *ifilter,
AVFilterInOut *in)
{
av_freep(&ifilter->name);
DESCRIBE_FILTER_LINK(ifilter, in, 1);
switch (avfilter_pad_get_type(in->filter_ctx->input_pads, in->pad_idx)) {
case AVMEDIA_TYPE_VIDEO: return configure_input_video_filter(fg, ifilter, in);
case AVMEDIA_TYPE_AUDIO: return configure_input_audio_filter(fg, ifilter, in);
default: av_assert0(0);
}
}
int configure_filtergraph(FilterGraph *fg)
{
AVFilterInOut *inputs, *outputs, *cur;
int ret, i, init = !fg->graph, simple = !fg->graph_desc;
const char *graph_desc = simple ? fg->outputs[0]->ost->avfilter :
fg->graph_desc;
avfilter_graph_free(&fg->graph);
if (!(fg->graph = avfilter_graph_alloc()))
return AVERROR(ENOMEM);
if (simple) {
OutputStream *ost = fg->outputs[0]->ost;
char args[255];
snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
fg->graph->scale_sws_opts = av_strdup(args);
}
if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0)
return ret;
if (simple && (!inputs || inputs->next || !outputs || outputs->next)) {
av_log(NULL, AV_LOG_ERROR, "Simple filtergraph '%s' does not have "
"exactly one input and output.\n", graph_desc);
return AVERROR(EINVAL);
}
for (cur = inputs; !simple && init && cur; cur = cur->next)
init_input_filter(fg, cur);
for (cur = inputs, i = 0; cur; cur = cur->next, i++)
if ((ret = configure_input_filter(fg, fg->inputs[i], cur)) < 0)
return ret;
avfilter_inout_free(&inputs);
if (!init || simple) {
/* we already know the mappings between lavfi outputs and output streams,
* so we can finish the setup */
for (cur = outputs, i = 0; cur; cur = cur->next, i++)
configure_output_filter(fg, fg->outputs[i], cur);
avfilter_inout_free(&outputs);
if ((ret = avfilter_graph_config(fg->graph, NULL)) < 0)
return ret;
} else {
/* wait until output mappings are processed */
for (cur = outputs; cur;) {
fg->outputs = grow_array(fg->outputs, sizeof(*fg->outputs),
&fg->nb_outputs, fg->nb_outputs + 1);
if (!(fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]))))
exit_program(1);
fg->outputs[fg->nb_outputs - 1]->graph = fg;
fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
cur = cur->next;
fg->outputs[fg->nb_outputs - 1]->out_tmp->next = NULL;
}
}
return 0;
}
int ist_in_filtergraph(FilterGraph *fg, InputStream *ist)
{
int i;
for (i = 0; i < fg->nb_inputs; i++)
if (fg->inputs[i]->ist == ist)
return 1;
return 0;
}

2300
ffmpeg_opt.c Normal file
View File

File diff suppressed because it is too large Load Diff

2
libavcodec/g723_1.c
View File

@ -264,6 +264,7 @@ static int scale_vector(int16_t *vector, int length)
for (i = 0; i < length; i++)
max = FFMAX(max, FFABS(vector[i]));
max = FFMIN(max, 0x7FFF);
bits = normalize_bits(max, 15);
scale = shift_table[bits];
@ -913,6 +914,7 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
}
iir_filter(filter_coef[0], filter_coef[1], buf + i,
filter_signal + i, 1);
lpc += LPC_ORDER;
}
memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));

2
libavcodec/mpc8.c
View File

@ -419,6 +419,8 @@ static int mpc8_decode_frame(AVCodecContext * avctx, void *data,
c->cur_frame++;
c->last_bits_used = get_bits_count(gb);
if(get_bits_left(gb) < 8) // we have only padding left
c->last_bits_used = buf_size << 3;
if(c->cur_frame >= c->frames)
c->cur_frame = 0;

2
libavcodec/mpegaudiodec.c
View File

@ -1686,7 +1686,7 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
if (s->frame_size <= 0 || s->frame_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
return AVERROR_INVALIDDATA;
}else if(s->frame_size < buf_size){
} else if (s->frame_size < buf_size) {
av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
buf_size= s->frame_size;
}

5
libavcodec/mpegaudioenc.c
View File

@ -24,6 +24,8 @@
* The simplest mpeg audio layer 2 encoder.
*/
#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "internal.h"
#include "put_bits.h"
@ -792,6 +794,9 @@ AVCodec ff_mp2_encoder = {
.supported_samplerates = (const int[]){
44100, 48000, 32000, 22050, 24000, 16000, 0
},
.channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
0 },
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
.defaults = mp2_defaults,
};

2
libavcodec/x86/pngdsp.asm
View File

@ -28,7 +28,7 @@ SECTION_RODATA
cextern pw_255
SECTION_TEXT 16
SECTION_TEXT
; %1 = nr. of xmm registers used
%macro ADD_BYTES_FN 1

2
libavformat/id3v2.c
View File

@ -131,6 +131,8 @@ const CodecMime ff_id3v2_mime_tags[] = {
{"image/png" , AV_CODEC_ID_PNG},
{"image/tiff", AV_CODEC_ID_TIFF},
{"image/bmp", AV_CODEC_ID_BMP},
{"JPG", AV_CODEC_ID_MJPEG}, /* ID3v2.2 */
{"PNG" , AV_CODEC_ID_PNG}, /* ID3v2.2 */
{"", AV_CODEC_ID_NONE},
};

9
libavformat/rtmp.h
View File

@ -32,6 +32,15 @@
#define HMAC_IPAD_VAL 0x36
#define HMAC_OPAD_VAL 0x5C
/**
* A non-zero transaction id requires the server to send back
* a _result or _error response.
* Setting it to 0 marks the message as a notification not
* requiring feedback.
*/
#define RTMP_NOTIFICATION 0
/**
* emulated Flash client version - 9.0.124.2 on Linux
* @{

169
libavformat/rtmpproto.c
View File

@ -91,6 +91,7 @@ typedef struct RTMPContext {
char* flashver; ///< version of the flash plugin
char* swfurl; ///< url of the swf player
char* pageurl; ///< url of the web page
char* subscribe; ///< name of live stream to subscribe
int server_bw; ///< server bandwidth
int client_buffer_time; ///< client buffer time in ms
int flush_interval; ///< number of packets flushed in the same request (RTMPT only)
@ -572,7 +573,7 @@ static int gen_check_bw(URLContext *s, RTMPContext *rt)
p = pkt.data;
ff_amf_write_string(&p, "_checkbw");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_number(&p, RTMP_NOTIFICATION);
ff_amf_write_null(&p);
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
@ -604,6 +605,30 @@ static int gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
return ret;
}
static int gen_fcsubscribe_stream(URLContext *s, RTMPContext *rt,
const char *subscribe)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 27 + strlen(subscribe))) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "FCSubscribe");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, subscribe);
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
return ret;
}
int ff_rtmp_calc_digest(const uint8_t *src, int len, int gap,
const uint8_t *key, int keylen, uint8_t *dst)
{
@ -1011,6 +1036,20 @@ static int handle_invoke(URLContext *s, RTMPPacket *pkt)
}
if ((ret = gen_create_stream(s, rt)) < 0)
return ret;
if (rt->is_input) {
/* Send the FCSubscribe command when the name of live
* stream is defined by the user or if it's a live stream. */
if (rt->subscribe) {
if ((ret = gen_fcsubscribe_stream(s, rt,
rt->subscribe)) < 0)
return ret;
} else if (rt->live == -1) {
if ((ret = gen_fcsubscribe_stream(s, rt,
rt->playpath)) < 0)
return ret;
}
}
break;
case STATE_FCPUBLISH:
rt->state = STATE_CONNECTING;
@ -1593,115 +1632,35 @@ static const AVOption rtmp_options[] = {
{"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
{"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
{"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_subscribe", "Name of live stream to subscribe to. Defaults to rtmp_playpath.", OFFSET(subscribe), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
{"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{ NULL },
};
static const AVClass rtmp_class = {
.class_name = "rtmp",
.item_name = av_default_item_name,
.option = rtmp_options,
.version = LIBAVUTIL_VERSION_INT,
#define RTMP_PROTOCOL(flavor) \
static const AVClass flavor##_class = { \
.class_name = #flavor, \
.item_name = av_default_item_name, \
.option = rtmp_options, \
.version = LIBAVUTIL_VERSION_INT, \
}; \
\
URLProtocol ff_##flavor##_protocol = { \
.name = #flavor, \
.url_open = rtmp_open, \
.url_read = rtmp_read, \
.url_write = rtmp_write, \
.url_close = rtmp_close, \
.priv_data_size = sizeof(RTMPContext), \
.flags = URL_PROTOCOL_FLAG_NETWORK, \
.priv_data_class= &flavor##_class, \
};
URLProtocol ff_rtmp_protocol = {
.name = "rtmp",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.priv_data_size = sizeof(RTMPContext),
.flags = URL_PROTOCOL_FLAG_NETWORK,
.priv_data_class= &rtmp_class,
};
static const AVClass rtmpe_class = {
.class_name = "rtmpe",
.item_name = av_default_item_name,
.option = rtmp_options,
.version = LIBAVUTIL_VERSION_INT,
};
URLProtocol ff_rtmpe_protocol = {
.name = "rtmpe",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.priv_data_size = sizeof(RTMPContext),
.flags = URL_PROTOCOL_FLAG_NETWORK,
.priv_data_class = &rtmpe_class,
};
static const AVClass rtmps_class = {
.class_name = "rtmps",
.item_name = av_default_item_name,
.option = rtmp_options,
.version = LIBAVUTIL_VERSION_INT,
};
URLProtocol ff_rtmps_protocol = {
.name = "rtmps",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.priv_data_size = sizeof(RTMPContext),
.flags = URL_PROTOCOL_FLAG_NETWORK,
.priv_data_class = &rtmps_class,
};
static const AVClass rtmpt_class = {
.class_name = "rtmpt",
.item_name = av_default_item_name,
.option = rtmp_options,
.version = LIBAVUTIL_VERSION_INT,
};
URLProtocol ff_rtmpt_protocol = {
.name = "rtmpt",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.priv_data_size = sizeof(RTMPContext),
.flags = URL_PROTOCOL_FLAG_NETWORK,
.priv_data_class = &rtmpt_class,
};
static const AVClass rtmpte_class = {
.class_name = "rtmpte",
.item_name = av_default_item_name,
.option = rtmp_options,
.version = LIBAVUTIL_VERSION_INT,
};
URLProtocol ff_rtmpte_protocol = {
.name = "rtmpte",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.priv_data_size = sizeof(RTMPContext),
.flags = URL_PROTOCOL_FLAG_NETWORK,
.priv_data_class = &rtmpte_class,
};
static const AVClass rtmpts_class = {
.class_name = "rtmpts",
.item_name = av_default_item_name,
.option = rtmp_options,
.version = LIBAVUTIL_VERSION_INT,
};
URLProtocol ff_rtmpts_protocol = {
.name = "rtmpts",
.url_open = rtmp_open,
.url_read = rtmp_read,
.url_write = rtmp_write,
.url_close = rtmp_close,
.priv_data_size = sizeof(RTMPContext),
.flags = URL_PROTOCOL_FLAG_NETWORK,
.priv_data_class = &rtmpts_class,
};
RTMP_PROTOCOL(rtmp)
RTMP_PROTOCOL(rtmpe)
RTMP_PROTOCOL(rtmps)
RTMP_PROTOCOL(rtmpt)
RTMP_PROTOCOL(rtmpte)
RTMP_PROTOCOL(rtmpts)

2
libavformat/version.h
View File

@ -31,7 +31,7 @@
#define LIBAVFORMAT_VERSION_MAJOR 54
#define LIBAVFORMAT_VERSION_MINOR 22
#define LIBAVFORMAT_VERSION_MICRO 102
#define LIBAVFORMAT_VERSION_MICRO 103
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, \

4
libavutil/arm/intmath.h
View File

@ -97,6 +97,8 @@ static av_always_inline av_const int FASTDIV(int a, int b)
#endif /* HAVE_ARMV6 */
#if HAVE_ASM_MOD_Q
#define av_clipl_int32 av_clipl_int32_arm
static av_always_inline av_const int32_t av_clipl_int32_arm(int64_t a)
{
@ -110,6 +112,8 @@ static av_always_inline av_const int32_t av_clipl_int32_arm(int64_t a)
return x;
}
#endif /* HAVE_ASM_MOD_Q */
#endif /* HAVE_INLINE_ASM */
#endif /* AVUTIL_ARM_INTMATH_H */

4
libavutil/arm/intreadwrite.h
View File

@ -61,6 +61,8 @@ static av_always_inline void AV_WN32(void *p, uint32_t v)
__asm__ ("str %1, %0" : "=m"(*(uint32_t *)p) : "r"(v));
}
#if HAVE_ASM_MOD_Q
#define AV_RN64 AV_RN64
static av_always_inline uint64_t AV_RN64(const void *p)
{
@ -82,6 +84,8 @@ static av_always_inline void AV_WN64(void *p, uint64_t v)
: "r"(v));
}
#endif /* HAVE_ASM_MOD_Q */
#endif /* HAVE_INLINE_ASM */
#endif /* AVUTIL_ARM_INTREADWRITE_H */

4
library.mak
View File

@ -20,7 +20,7 @@ $(SUBDIR)x86/%.o: $(SUBDIR)x86/%.asm
$(DEPYASM) $(YASMFLAGS) -I $(<D)/ -M -o $@ $< > $(@:.o=.d)
$(YASM) $(YASMFLAGS) -I $(<D)/ -o $@ $<
$(OBJS) $(OBJS:.o=.s) $(SUBDIR)%.ho $(TESTOBJS): CPPFLAGS += -DHAVE_AV_CONFIG_H
$(OBJS) $(OBJS:.o=.s) $(SUBDIR)%.h.o $(TESTOBJS): CPPFLAGS += -DHAVE_AV_CONFIG_H
$(TESTOBJS): CPPFLAGS += -DTEST
$(SUBDIR)$(LIBNAME): $(OBJS)
@ -87,7 +87,7 @@ uninstall-libs::
-$(RM) "$(LIBDIR)/$(LIBNAME)"
uninstall-headers::
$(RM) $(addprefix "$(INCINSTDIR)/",$(HEADERS)) $(addprefix "$(INCINSTDIR)/",$(BUILT_HEADERS))
$(RM) $(addprefix "$(INCINSTDIR)/",$(HEADERS) $(BUILT_HEADERS))
$(RM) "$(LIBDIR)/pkgconfig/lib$(NAME).pc"
-rmdir "$(INCINSTDIR)"
endef

10
tests/fate/video.mak
View File

@ -167,8 +167,14 @@ FATE_VIDEO += fate-mpeg2-field-enc
fate-mpeg2-field-enc: CMD = framecrc -flags +bitexact -dct fastint -idct simple -i $(SAMPLES)/mpeg2/mpeg2_field_encoding.ts -an
# FIXME dropped frames in this test because of coarse timebase
FATE_VIDEO += fate-nuv
fate-nuv: CMD = framecrc -idct simple -i $(SAMPLES)/nuv/Today.nuv -an
FATE_NUV += fate-nuv-rtjpeg
fate-nuv-rtjpeg: CMD = framecrc -idct simple -i $(SAMPLES)/nuv/Today.nuv -an
FATE_NUV += fate-nuv-rtjpeg-fh
fate-nuv-rtjpeg-fh: CMD = framecrc -idct simple -i $(SAMPLES)/nuv/rtjpeg_frameheader.nuv -an
FATE_VIDEO += $(FATE_NUV)
fate-nuv: $(FATE_NUV)
FATE_VIDEO += fate-paf-video
fate-paf-video: CMD = framecrc -i $(SAMPLES)/paf/hod1-partial.paf -pix_fmt rgb24 -an

4
tests/ref/fate/acodec-g723_1
View File

@ -1,4 +1,4 @@
dec0deb2425e908d232d2471acff04a3 *tests/data/fate/acodec-g723_1.g723_1
4800 tests/data/fate/acodec-g723_1.g723_1
87fd529c9e41914f73a865d147cc9516 *tests/data/fate/acodec-g723_1.out.wav
stddev: 8425.98 PSNR: 17.82 MAXDIFF:53268 bytes: 95992/ 96000
d70776846d77c652bceed281fcca9cc8 *tests/data/fate/acodec-g723_1.out.wav
stddev: 8423.47 PSNR: 17.82 MAXDIFF:53292 bytes: 95992/ 96000

0
tests/ref/fate/nuv → tests/ref/fate/nuv-rtjpeg
View File

51
tests/ref/fate/nuv-rtjpeg-fh Normal file
View File

@ -0,0 +1,51 @@
#tb 0: 1/50
0, 0, 0, 1, 221184, 0xf48c94f6
0, 2, 2, 1, 221184, 0x89b625b2
0, 3, 3, 1, 221184, 0x37e04714
0, 4, 4, 1, 221184, 0x4f4c5224
0, 5, 5, 1, 221184, 0x9193c9f1
0, 6, 6, 1, 221184, 0x5d1a6197
0, 7, 7, 1, 221184, 0x40cd51e7
0, 8, 8, 1, 221184, 0xb2c1a729
0, 10, 10, 1, 221184, 0x998d6144
0, 11, 11, 1, 221184, 0xf5d52311
0, 12, 12, 1, 221184, 0xea9dd6bf
0, 13, 13, 1, 221184, 0x0e2ed854
0, 14, 14, 1, 221184, 0xe295ba58
0, 15, 15, 1, 221184, 0x8aedbb69
0, 16, 16, 1, 221184, 0x253c9aaa
0, 17, 17, 1, 221184, 0x5eaf9fb1
0, 18, 18, 1, 221184, 0xcdb5a0cb
0, 19, 19, 1, 221184, 0xcdb5a0cb
0, 20, 20, 1, 221184, 0x23f89994
0, 21, 21, 1, 221184, 0x23f89994
0, 22, 22, 1, 221184, 0x10dc98d6
0, 23, 23, 1, 221184, 0x799b9d98
0, 24, 24, 1, 221184, 0xb226996c
0, 25, 25, 1, 221184, 0x0ac59a42
0, 26, 26, 1, 221184, 0x87c2a654
0, 27, 27, 1, 221184, 0xf4c1a711
0, 28, 28, 1, 221184, 0xf60fa72e
0, 29, 29, 1, 221184, 0xc8f8b6fc
0, 30, 30, 1, 221184, 0xd709b813
0, 31, 31, 1, 221184, 0x5fdfb76b
0, 32, 32, 1, 221184, 0x5798b0aa
0, 33, 33, 1, 221184, 0xf572b1c3
0, 34, 34, 1, 221184, 0x14b0afdf
0, 35, 35, 1, 221184, 0x0a66b5b8
0, 36, 36, 1, 221184, 0xe316c620
0, 37, 37, 1, 221184, 0xbc76c5c2
0, 38, 38, 1, 221184, 0x77c7c5e5
0, 39, 39, 1, 221184, 0xfc7ac63e
0, 40, 40, 1, 221184, 0x05a29ffe
0, 41, 41, 1, 221184, 0x9bffbf6c
0, 42, 42, 1, 221184, 0x3c55be40
0, 43, 43, 1, 221184, 0x6f46c14e
0, 44, 44, 1, 221184, 0x9cf4ae70
0, 45, 45, 1, 221184, 0xf205b2f8
0, 46, 46, 1, 221184, 0x7180aff8
0, 47, 47, 1, 221184, 0x125eaffe
0, 48, 48, 1, 221184, 0x6970a32d
0, 49, 49, 1, 221184, 0xaea79f62
0, 50, 50, 1, 221184, 0x48d2a093
0, 51, 51, 1, 221184, 0x10a59eb5