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author | Michael Niedermayer <michaelni@gmx.at> | 2011-05-17 04:47:05 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2011-05-17 04:51:33 +0200 |
commit | f8ae3a2108b612776e886d927b4a7289dde619f1 (patch) | |
tree | c10fc78e3c6340c9513f99faadf077ee6c7d2986 /libavcodec | |
parent | e6e7ba0ce3aadef32f7f16f706c4a0406b5bd70f (diff) | |
parent | 901ff51116f831c9082e14c80c7481dd3999aa30 (diff) | |
download | ffmpeg-f8ae3a2108b612776e886d927b4a7289dde619f1.tar.gz |
Merge remote branch 'qatar/master'
12 files changed, 36 insertions(+), 81 deletions(-)
yes thats 36 new lines in 14 commits
* qatar/master:
ffmpeg: fix -aspect cli option
Restructure video filter implementation in ffmpeg.c.
ffplay: remove audio_write_get_buf_size() forward declaration
lavfi: print key-frame and picture type information in ff_dlog_ref()
mathops: remove ancient confusing comment
cws2fws: Improve error message wording.
tools: Check the return value of write().
mpegaudio: move OUT_FMT macro to mpegaudiodec.c
mpegaudio: remove OUT_MIN/MAX macros
Add missing #includes to mp3_header_(de)compress bsf
dct: fix indentation
dct: bypass table allocation for DCT_II of size 32
h264dsp_mmx: Add #ifdefs around some mmxext functions on x86_64.
Remove unused header mpegaudio3.h.
Conflicts:
ffmpeg.c
libavcodec/mpegaudio.h
libavcodec/mpegaudio3.h
libavfilter/avfilter.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/Makefile | 1 | ||||
-rw-r--r-- | libavcodec/dct.c | 37 | ||||
-rw-r--r-- | libavcodec/mathops.h | 3 | ||||
-rw-r--r-- | libavcodec/mp3_header_compress_bsf.c | 1 | ||||
-rw-r--r-- | libavcodec/mp3_header_decompress_bsf.c | 1 | ||||
-rw-r--r-- | libavcodec/mpegaudio.h | 15 | ||||
-rw-r--r-- | libavcodec/mpegaudio3.h | 53 | ||||
-rw-r--r-- | libavcodec/mpegaudiodec.c | 4 | ||||
-rw-r--r-- | libavcodec/x86/h264dsp_mmx.c | 2 |
9 files changed, 30 insertions, 87 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile index ac16e06df8..2b00575463 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -669,7 +669,6 @@ SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER) += libschroedinger.h SKIPHEADERS-$(CONFIG_VAAPI) += vaapi_internal.h SKIPHEADERS-$(CONFIG_VDPAU) += vdpau.h SKIPHEADERS-$(CONFIG_XVMC) += xvmc.h -SKIPHEADERS += mpegaudio3.h EXAMPLES = api diff --git a/libavcodec/dct.c b/libavcodec/dct.c index 83cf1b4896..ef3cd50a79 100644 --- a/libavcodec/dct.c +++ b/libavcodec/dct.c @@ -180,33 +180,36 @@ av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse) int n = 1 << nbits; int i; + memset(s, 0, sizeof(*s)); + s->nbits = nbits; s->inverse = inverse; - ff_init_ff_cos_tabs(nbits+2); + if (inverse == DCT_II && nbits == 5) { + s->dct_calc = dct32_func; + } else { + ff_init_ff_cos_tabs(nbits+2); - s->costab = ff_cos_tabs[nbits+2]; + s->costab = ff_cos_tabs[nbits+2]; - s->csc2 = av_malloc(n/2 * sizeof(FFTSample)); + s->csc2 = av_malloc(n/2 * sizeof(FFTSample)); - if (ff_rdft_init(&s->rdft, nbits, inverse == DCT_III) < 0) { - av_free(s->csc2); - return -1; - } + if (ff_rdft_init(&s->rdft, nbits, inverse == DCT_III) < 0) { + av_free(s->csc2); + return -1; + } - for (i = 0; i < n/2; i++) - s->csc2[i] = 0.5 / sin((M_PI / (2*n) * (2*i + 1))); + for (i = 0; i < n/2; i++) + s->csc2[i] = 0.5 / sin((M_PI / (2*n) * (2*i + 1))); - switch(inverse) { - case DCT_I : s->dct_calc = ff_dct_calc_I_c; break; - case DCT_II : s->dct_calc = ff_dct_calc_II_c ; break; - case DCT_III: s->dct_calc = ff_dct_calc_III_c; break; - case DST_I : s->dct_calc = ff_dst_calc_I_c; break; + switch(inverse) { + case DCT_I : s->dct_calc = ff_dct_calc_I_c; break; + case DCT_II : s->dct_calc = ff_dct_calc_II_c ; break; + case DCT_III: s->dct_calc = ff_dct_calc_III_c; break; + case DST_I : s->dct_calc = ff_dst_calc_I_c; break; + } } - if (inverse == DCT_II && nbits == 5) - s->dct_calc = dct32_func; - s->dct32 = dct32; if (HAVE_MMX) ff_dct_init_mmx(s); diff --git a/libavcodec/mathops.h b/libavcodec/mathops.h index 4d88ed14c9..29b3cd0acb 100644 --- a/libavcodec/mathops.h +++ b/libavcodec/mathops.h @@ -45,9 +45,6 @@ #endif #ifndef MULH -//gcc 3.4 creates an incredibly bloated mess out of this -//# define MULH(a,b) (((int64_t)(a) * (int64_t)(b))>>32) - static av_always_inline int MULH(int a, int b){ return ((int64_t)(a) * (int64_t)(b))>>32; } diff --git a/libavcodec/mp3_header_compress_bsf.c b/libavcodec/mp3_header_compress_bsf.c index 006a3679e9..b67e78c046 100644 --- a/libavcodec/mp3_header_compress_bsf.c +++ b/libavcodec/mp3_header_compress_bsf.c @@ -18,6 +18,7 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ +#include "libavutil/intreadwrite.h" #include "avcodec.h" #include "mpegaudio.h" diff --git a/libavcodec/mp3_header_decompress_bsf.c b/libavcodec/mp3_header_decompress_bsf.c index f096d5e4fd..7af5a10e12 100644 --- a/libavcodec/mp3_header_decompress_bsf.c +++ b/libavcodec/mp3_header_decompress_bsf.c @@ -18,6 +18,7 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ +#include "libavutil/intreadwrite.h" #include "avcodec.h" #include "mpegaudio.h" #include "mpegaudiodata.h" diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h index 30ef349399..138085366f 100644 --- a/libavcodec/mpegaudio.h +++ b/libavcodec/mpegaudio.h @@ -35,8 +35,6 @@ #include "dsputil.h" #include "dct.h" -#define CONFIG_AUDIO_NONSHORT 0 - /* max frame size, in samples */ #define MPA_FRAME_SIZE 1152 @@ -69,19 +67,9 @@ #if CONFIG_FLOAT typedef float OUT_INT; -#define OUT_FMT AV_SAMPLE_FMT_FLT -#elif CONFIG_MPEGAUDIO_HP && CONFIG_AUDIO_NONSHORT -typedef int32_t OUT_INT; -#define OUT_MAX INT32_MAX -#define OUT_MIN INT32_MIN -#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 31) -#define OUT_FMT AV_SAMPLE_FMT_S32 #else typedef int16_t OUT_INT; -#define OUT_MAX INT16_MAX -#define OUT_MIN INT16_MIN #define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15) -#define OUT_FMT AV_SAMPLE_FMT_S16 #endif #if CONFIG_FLOAT @@ -147,6 +135,9 @@ typedef struct MPADecodeContext { DECLARE_ALIGNED(16, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT]; INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */ GranuleDef granules[2][2]; /* Used in Layer 3 */ +#ifdef DEBUG + int frame_count; +#endif int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3 int dither_state; int error_recognition; diff --git a/libavcodec/mpegaudio3.h b/libavcodec/mpegaudio3.h deleted file mode 100644 index c374a59a76..0000000000 --- a/libavcodec/mpegaudio3.h +++ /dev/null @@ -1,53 +0,0 @@ -/* - * Copyright (c) 2007 Michael Niedermayer - * - * 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 - */ - -/* layer 3 "granule" */ -typedef struct GranuleDef { - uint8_t scfsi; - int part2_3_length; - int big_values; - int global_gain; - int scalefac_compress; - uint8_t block_type; - uint8_t switch_point; - int table_select[3]; - int subblock_gain[3]; - uint8_t scalefac_scale; - uint8_t count1table_select; - int region_size[3]; /* number of huffman codes in each region */ - int preflag; - int short_start, long_end; /* long/short band indexes */ - uint8_t scale_factors[40]; - int32_t sb_hybrid[SBLIMIT * 18]; /* 576 samples */ -} GranuleDef; - -void ff_mp3_init(void); - -/** - * Compute huffman coded region sizes. - */ -void ff_init_short_region(MPADecodeContext *s, GranuleDef *g); - -/** - * Compute huffman coded region sizes. - */ -void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2); - -void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g); diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index b1c9ef93a0..bdff815d4d 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -47,6 +47,7 @@ # define MULH3(x, y, s) ((s)*(y)*(x)) # define MULLx(x, y, s) ((y)*(x)) # define RENAME(a) a ## _float +# define OUT_FMT AV_SAMPLE_FMT_FLT #else # define SHR(a,b) ((a)>>(b)) # define compute_antialias compute_antialias_integer @@ -57,6 +58,7 @@ # define MULH3(x, y, s) MULH((s)*(x), y) # define MULLx(x, y, s) MULL(x,y,s) # define RENAME(a) a +# define OUT_FMT AV_SAMPLE_FMT_S16 #endif /****************/ @@ -490,7 +492,7 @@ static inline int round_sample(int64_t *sum) int sum1; sum1 = (int)((*sum) >> OUT_SHIFT); *sum &= (1<<OUT_SHIFT)-1; - return av_clip(sum1, OUT_MIN, OUT_MAX); + return av_clip_int16(sum1); } # define MULS(ra, rb) MUL64(ra, rb) diff --git a/libavcodec/x86/h264dsp_mmx.c b/libavcodec/x86/h264dsp_mmx.c index b331f94b5e..01791604d6 100644 --- a/libavcodec/x86/h264dsp_mmx.c +++ b/libavcodec/x86/h264dsp_mmx.c @@ -252,6 +252,7 @@ LF_IFUNC(v, chroma_intra, depth, avx) LF_FUNCS( uint8_t, 8) LF_FUNCS(uint16_t, 10) +#if ARCH_X86_32 LF_FUNC (v8, luma, 8, mmxext) static void ff_deblock_v_luma_8_mmxext(uint8_t *pix, int stride, int alpha, int beta, int8_t *tc0) { @@ -266,6 +267,7 @@ static void ff_deblock_v_luma_intra_8_mmxext(uint8_t *pix, int stride, int alpha ff_deblock_v8_luma_intra_8_mmxext(pix+0, stride, alpha, beta); ff_deblock_v8_luma_intra_8_mmxext(pix+8, stride, alpha, beta); } +#endif /* ARCH_X86_32 */ LF_FUNC (v, luma, 10, mmxext) LF_IFUNC(v, luma_intra, 10, mmxext) |