[FFmpeg-cvslog] libavcodec: Implementation of AC3 fixedpoint decoder
Nedeljko Babic
git at videolan.org
Tue Apr 1 20:19:56 CEST 2014
ffmpeg | branch: master | Nedeljko Babic <nbabic at mips.com> | Tue Apr 1 16:31:08 2014 +0200| [696e34a6e15d9d9d655191a953779d06dc3b5897] | committer: Michael Niedermayer
libavcodec: Implementation of AC3 fixedpoint decoder
Signed-off-by: Nedeljko Babic <nbabic at mips.com>
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=696e34a6e15d9d9d655191a953779d06dc3b5897
---
Changelog | 1 +
configure | 1 +
doc/general.texi | 2 +-
libavcodec/Makefile | 3 +-
libavcodec/ac3.h | 46 +++++++++++
libavcodec/ac3dec.c | 197 ++++++++++++++++++++++++---------------------
libavcodec/ac3dec.h | 33 ++++----
libavcodec/ac3dec_fixed.c | 176 ++++++++++++++++++++++++++++++++++++++++
libavcodec/ac3dec_float.c | 89 ++++++++++++++++++++
libavcodec/ac3dsp.c | 26 ++++++
libavcodec/ac3dsp.h | 3 +
libavcodec/allcodecs.c | 2 +-
libavcodec/kbdwin.c | 10 +++
libavcodec/kbdwin.h | 1 +
libavcodec/version.h | 2 +-
15 files changed, 483 insertions(+), 109 deletions(-)
diff --git a/Changelog b/Changelog
index 0e38ec8..b8147c9 100644
--- a/Changelog
+++ b/Changelog
@@ -2,6 +2,7 @@ Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version <next>:
+- added AC3 fixed-point decoding
- shuffleplanes filter
- subfile protocol
- Phantom Cine demuxer
diff --git a/configure b/configure
index b82891c..848dc7e 100755
--- a/configure
+++ b/configure
@@ -1951,6 +1951,7 @@ aac_decoder_select="mdct sinewin"
aac_encoder_select="audio_frame_queue mdct sinewin"
aac_latm_decoder_select="aac_decoder aac_latm_parser"
ac3_decoder_select="mdct ac3dsp ac3_parser dsputil"
+ac3_fixed_decoder_select="mdct ac3dsp ac3_parser dsputil"
ac3_encoder_select="mdct ac3dsp dsputil"
ac3_fixed_encoder_select="mdct ac3dsp dsputil"
aic_decoder_select="dsputil golomb"
diff --git a/doc/general.texi b/doc/general.texi
index e9087df..9f17d48 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -820,7 +820,7 @@ following image formats are supported:
@tab encoding supported through external library libaacplus
@item AAC @tab E @tab X
@tab encoding supported through external library libfaac and libvo-aacenc
- at item AC-3 @tab IX @tab X
+ at item AC-3 @tab IX @tab IX
@item ADPCM 4X Movie @tab @tab X
@item ADPCM CDROM XA @tab @tab X
@item ADPCM Creative Technology @tab @tab X
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 70a6e25..f68371a 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -94,7 +94,8 @@ OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
psymodel.o iirfilter.o \
mpeg4audio.o kbdwin.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
-OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
+OBJS-$(CONFIG_AC3_DECODER) += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o
+OBJS-$(CONFIG_AC3_FIXED_DECODER) += ac3dec_fixed.o ac3dec_data.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3enc.o ac3tab.o \
ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_FIXED_ENCODER) += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
diff --git a/libavcodec/ac3.h b/libavcodec/ac3.h
index 0cc9e2c..ac3936d 100644
--- a/libavcodec/ac3.h
+++ b/libavcodec/ac3.h
@@ -51,6 +51,52 @@
#define EXP_D25 2
#define EXP_D45 3
+#ifndef CONFIG_AC3_FIXED
+#define CONFIG_AC3_FIXED 0
+#endif
+
+#if CONFIG_AC3_FIXED
+
+#define FFT_FLOAT 0
+
+#define FIXR(a) ((int)((a) * 0 + 0.5))
+#define FIXR12(a) ((int)((a) * 4096 + 0.5))
+#define FIXR15(a) ((int)((a) * 32768 + 0.5))
+#define ROUND15(x) ((x) + 16384) >> 15
+
+#define AC3_RENAME(x) x ## _fixed
+#define AC3_NORM(norm) (1<<24)/(norm)
+#define AC3_MUL(a,b) ((((int64_t) (a)) * (b))>>12)
+#define AC3_RANGE(x) (x)
+#define AC3_DYNAMIC_RANGE(x) (x)
+#define AC3_SPX_BLEND(x) (x)
+#define AC3_DYNAMIC_RANGE1 0
+
+#define INTFLOAT int
+#define SHORTFLOAT int16_t
+
+#else /* CONFIG_AC3_FIXED */
+
+#define FIXR(x) ((float)(x))
+#define FIXR12(x) ((float)(x))
+#define FIXR15(x) ((float)(x))
+#define ROUND15(x) (x)
+
+#define AC3_RENAME(x) x
+#define AC3_NORM(norm) (1.0f/(norm))
+#define AC3_MUL(a,b) ((a) * (b))
+#define AC3_RANGE(x) (dynamic_range_tab[(x)])
+#define AC3_DYNAMIC_RANGE(x) (powf(x, s->drc_scale))
+#define AC3_SPX_BLEND(x) (x)* (1.0f/32)
+#define AC3_DYNAMIC_RANGE1 1.0f
+
+#define INTFLOAT float
+#define SHORTFLOAT float
+
+#endif /* CONFIG_AC3_FIXED */
+
+#define AC3_LEVEL(x) ROUND15((x) * FIXR15(0.7071067811865476))
+
/* pre-defined gain values */
#define LEVEL_PLUS_3DB 1.4142135623730950
#define LEVEL_PLUS_1POINT5DB 1.1892071150027209
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 3a6a7ad..1d97df9 100644
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -179,14 +179,23 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
ac3_tables_init();
ff_mdct_init(&s->imdct_256, 8, 1, 1.0);
ff_mdct_init(&s->imdct_512, 9, 1, 1.0);
- ff_kbd_window_init(s->window, 5.0, 256);
+ AC3_RENAME(ff_kbd_window_init)(s->window, 5.0, 256);
ff_dsputil_init(&s->dsp, avctx);
+
+#if (CONFIG_AC3_FIXED)
+ s->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & CODEC_FLAG_BITEXACT);
+#else
avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+#endif
+
ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
- avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+ if (CONFIG_AC3_FIXED)
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+ else
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* allow downmixing to stereo or mono */
#if FF_API_REQUEST_CHANNELS
@@ -345,40 +354,45 @@ static void set_downmix_coeffs(AC3DecodeContext *s)
float cmix = gain_levels[s-> center_mix_level];
float smix = gain_levels[s->surround_mix_level];
float norm0, norm1;
+ float downmix_coeffs[AC3_MAX_CHANNELS][2];
for (i = 0; i < s->fbw_channels; i++) {
- s->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
- s->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]];
+ downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
+ downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]];
}
if (s->channel_mode > 1 && s->channel_mode & 1) {
- s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = cmix;
+ downmix_coeffs[1][0] = downmix_coeffs[1][1] = cmix;
}
if (s->channel_mode == AC3_CHMODE_2F1R || s->channel_mode == AC3_CHMODE_3F1R) {
int nf = s->channel_mode - 2;
- s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB;
+ downmix_coeffs[nf][0] = downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB;
}
if (s->channel_mode == AC3_CHMODE_2F2R || s->channel_mode == AC3_CHMODE_3F2R) {
int nf = s->channel_mode - 4;
- s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = smix;
+ downmix_coeffs[nf][0] = downmix_coeffs[nf+1][1] = smix;
}
/* renormalize */
norm0 = norm1 = 0.0;
for (i = 0; i < s->fbw_channels; i++) {
- norm0 += s->downmix_coeffs[i][0];
- norm1 += s->downmix_coeffs[i][1];
+ norm0 += downmix_coeffs[i][0];
+ norm1 += downmix_coeffs[i][1];
}
norm0 = 1.0f / norm0;
norm1 = 1.0f / norm1;
for (i = 0; i < s->fbw_channels; i++) {
- s->downmix_coeffs[i][0] *= norm0;
- s->downmix_coeffs[i][1] *= norm1;
+ downmix_coeffs[i][0] *= norm0;
+ downmix_coeffs[i][1] *= norm1;
}
if (s->output_mode == AC3_CHMODE_MONO) {
for (i = 0; i < s->fbw_channels; i++)
- s->downmix_coeffs[i][0] = (s->downmix_coeffs[i][0] +
- s->downmix_coeffs[i][1]) * LEVEL_MINUS_3DB;
+ downmix_coeffs[i][0] = (downmix_coeffs[i][0] +
+ downmix_coeffs[i][1]) * LEVEL_MINUS_3DB;
+ }
+ for (i = 0; i < s->fbw_channels; i++) {
+ s->downmix_coeffs[i][0] = FIXR12(downmix_coeffs[i][0]);
+ s->downmix_coeffs[i][1] = FIXR12(downmix_coeffs[i][1]);
}
}
@@ -646,20 +660,30 @@ static inline void do_imdct(AC3DecodeContext *s, int channels)
for (ch = 1; ch <= channels; ch++) {
if (s->block_switch[ch]) {
int i;
- float *x = s->tmp_output + 128;
+ FFTSample *x = s->tmp_output + 128;
for (i = 0; i < 128; i++)
x[i] = s->transform_coeffs[ch][2 * i];
s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
+#if CONFIG_AC3_FIXED
+ s->fdsp->vector_fmul_window_scaled(s->outptr[ch - 1], s->delay[ch - 1],
+ s->tmp_output, s->window, 128, 8);
+#else
s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
s->tmp_output, s->window, 128);
+#endif
for (i = 0; i < 128; i++)
x[i] = s->transform_coeffs[ch][2 * i + 1];
s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch - 1], x);
} else {
s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
+#if CONFIG_AC3_FIXED
+ s->fdsp->vector_fmul_window_scaled(s->outptr[ch - 1], s->delay[ch - 1],
+ s->tmp_output, s->window, 128, 8);
+#else
s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
s->tmp_output, s->window, 128);
- memcpy(s->delay[ch - 1], s->tmp_output + 128, 128 * sizeof(float));
+#endif
+ memcpy(s->delay[ch - 1], s->tmp_output + 128, 128 * sizeof(FFTSample));
}
}
}
@@ -794,13 +818,13 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
if (get_bits1(gbc)) {
/* Allow asymmetric application of DRC when drc_scale > 1.
Amplification of quiet sounds is enhanced */
- float range = dynamic_range_tab[get_bits(gbc, 8)];
+ INTFLOAT range = AC3_RANGE(get_bits(gbc, 8));
if (range > 1.0 || s->drc_scale <= 1.0)
- s->dynamic_range[i] = powf(range, s->drc_scale);
+ s->dynamic_range[i] = AC3_DYNAMIC_RANGE(range);
else
s->dynamic_range[i] = range;
} else if (blk == 0) {
- s->dynamic_range[i] = 1.0f;
+ s->dynamic_range[i] = AC3_DYNAMIC_RANGE1;
}
} while (i--);
@@ -826,6 +850,9 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
if (start_subband > 7)
start_subband += start_subband - 7;
end_subband = get_bits(gbc, 3) + 5;
+#if CONFIG_AC3_FIXED
+ s->spx_dst_end_freq = end_freq_inv_tab[end_subband];
+#endif
if (end_subband > 7)
end_subband += end_subband - 7;
dst_start_freq = dst_start_freq * 12 + 25;
@@ -846,7 +873,9 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
s->spx_dst_start_freq = dst_start_freq;
s->spx_src_start_freq = src_start_freq;
+#if !CONFIG_AC3_FIXED
s->spx_dst_end_freq = dst_end_freq;
+#endif
decode_band_structure(gbc, blk, s->eac3, 0,
start_subband, end_subband,
@@ -866,18 +895,40 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
for (ch = 1; ch <= fbw_channels; ch++) {
if (s->channel_uses_spx[ch]) {
if (s->first_spx_coords[ch] || get_bits1(gbc)) {
- float spx_blend;
+ INTFLOAT spx_blend;
int bin, master_spx_coord;
s->first_spx_coords[ch] = 0;
- spx_blend = get_bits(gbc, 5) * (1.0f/32);
+ spx_blend = AC3_SPX_BLEND(get_bits(gbc, 5));
master_spx_coord = get_bits(gbc, 2) * 3;
bin = s->spx_src_start_freq;
for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
int bandsize;
int spx_coord_exp, spx_coord_mant;
- float nratio, sblend, nblend, spx_coord;
+ INTFLOAT nratio, sblend, nblend;
+#if CONFIG_AC3_FIXED
+ int64_t accu;
+ /* calculate blending factors */
+ bandsize = s->spx_band_sizes[bnd];
+ accu = (int64_t)((bin << 23) + (bandsize << 22)) * s->spx_dst_end_freq;
+ nratio = (int)(accu >> 32);
+ nratio -= spx_blend << 18;
+
+ if (nratio < 0) {
+ nblend = 0;
+ sblend = 0x800000;
+ } else if (nratio > 0x7fffff) {
+ nblend = 0x800000;
+ sblend = 0;
+ } else {
+ nblend = fixed_sqrt(nratio, 23);
+ accu = (int64_t)nblend * 1859775393;
+ nblend = (int)((accu + (1<<29)) >> 30);
+ sblend = fixed_sqrt(0x800000 - nratio, 23);
+ }
+#else
+ float spx_coord;
/* calculate blending factors */
bandsize = s->spx_band_sizes[bnd];
@@ -886,6 +937,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
nblend = sqrtf(3.0f * nratio); // noise is scaled by sqrt(3)
// to give unity variance
sblend = sqrtf(1.0f - nratio);
+#endif
bin += bandsize;
/* decode spx coordinates */
@@ -894,11 +946,18 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
if (spx_coord_exp == 15) spx_coord_mant <<= 1;
else spx_coord_mant += 4;
spx_coord_mant <<= (25 - spx_coord_exp - master_spx_coord);
- spx_coord = spx_coord_mant * (1.0f / (1 << 23));
/* multiply noise and signal blending factors by spx coordinate */
+#if CONFIG_AC3_FIXED
+ accu = (int64_t)nblend * spx_coord_mant;
+ s->spx_noise_blend[ch][bnd] = (int)((accu + (1<<22)) >> 23);
+ accu = (int64_t)sblend * spx_coord_mant;
+ s->spx_signal_blend[ch][bnd] = (int)((accu + (1<<22)) >> 23);
+#else
+ spx_coord = spx_coord_mant * (1.0f / (1 << 23));
s->spx_noise_blend [ch][bnd] = nblend * spx_coord;
s->spx_signal_blend[ch][bnd] = sblend * spx_coord;
+#endif
}
}
} else {
@@ -1255,14 +1314,19 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
/* apply scaling to coefficients (headroom, dynrng) */
for (ch = 1; ch <= s->channels; ch++) {
- float gain = 1.0 / 4194304.0f;
- if (s->channel_mode == AC3_CHMODE_DUALMONO) {
- gain *= s->dynamic_range[2 - ch];
+ INTFLOAT gain;
+ if(s->channel_mode == AC3_CHMODE_DUALMONO) {
+ gain = s->dynamic_range[2-ch];
} else {
- gain *= s->dynamic_range[0];
+ gain = s->dynamic_range[0];
}
+#if CONFIG_AC3_FIXED
+ scale_coefs(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256);
+#else
+ gain *= 1.0 / 4194304.0f;
s->fmt_conv.int32_to_float_fmul_scalar(s->transform_coeffs[ch],
s->fixed_coeffs[ch], gain, 256);
+#endif
}
/* apply spectral extension to high frequency bins */
@@ -1287,19 +1351,24 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
do_imdct(s, s->channels);
if (downmix_output) {
+#if CONFIG_AC3_FIXED
+ ac3_downmix_c_fixed16(s->outptr, s->downmix_coeffs,
+ s->out_channels, s->fbw_channels, 256);
+#else
s->ac3dsp.downmix(s->outptr, s->downmix_coeffs,
s->out_channels, s->fbw_channels, 256);
+#endif
}
} else {
if (downmix_output) {
- s->ac3dsp.downmix(s->xcfptr + 1, s->downmix_coeffs,
- s->out_channels, s->fbw_channels, 256);
+ s->ac3dsp.AC3_RENAME(downmix)(s->xcfptr + 1, s->downmix_coeffs,
+ s->out_channels, s->fbw_channels, 256);
}
if (downmix_output && !s->downmixed) {
s->downmixed = 1;
- s->ac3dsp.downmix(s->dlyptr, s->downmix_coeffs, s->out_channels,
- s->fbw_channels, 128);
+ s->ac3dsp.AC3_RENAME(downmix)(s->dlyptr, s->downmix_coeffs,
+ s->out_channels, s->fbw_channels, 128);
}
do_imdct(s, s->out_channels);
@@ -1320,7 +1389,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
AC3DecodeContext *s = avctx->priv_data;
int blk, ch, err, ret;
const uint8_t *channel_map;
- const float *output[AC3_MAX_CHANNELS];
+ const SHORTFLOAT *output[AC3_MAX_CHANNELS];
enum AVMatrixEncoding matrix_encoding;
AVDownmixInfo *downmix_info;
@@ -1447,7 +1516,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
}
for (ch = 0; ch < s->channels; ch++) {
if (ch < s->out_channels)
- s->outptr[channel_map[ch]] = (float *)frame->data[ch];
+ s->outptr[channel_map[ch]] = (SHORTFLOAT *)frame->data[ch];
}
for (blk = 0; blk < s->num_blocks; blk++) {
if (!err && decode_audio_block(s, blk)) {
@@ -1456,7 +1525,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
}
if (err)
for (ch = 0; ch < s->out_channels; ch++)
- memcpy(((float*)frame->data[ch]) + AC3_BLOCK_SIZE*blk, output[ch], sizeof(**output) * AC3_BLOCK_SIZE);
+ memcpy(((SHORTFLOAT*)frame->data[ch]) + AC3_BLOCK_SIZE*blk, output[ch], AC3_BLOCK_SIZE*sizeof(SHORTFLOAT));
for (ch = 0; ch < s->out_channels; ch++)
output[ch] = s->outptr[channel_map[ch]];
for (ch = 0; ch < s->out_channels; ch++) {
@@ -1469,7 +1538,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
/* keep last block for error concealment in next frame */
for (ch = 0; ch < s->out_channels; ch++)
- memcpy(s->output[ch], output[ch], sizeof(**output) * AC3_BLOCK_SIZE);
+ memcpy(s->output[ch], output[ch], AC3_BLOCK_SIZE*sizeof(SHORTFLOAT));
/*
* AVMatrixEncoding
@@ -1540,66 +1609,12 @@ static av_cold int ac3_decode_end(AVCodecContext *avctx)
AC3DecodeContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct_512);
ff_mdct_end(&s->imdct_256);
+#if (CONFIG_AC3_FIXED)
+ av_free(s->fdsp);
+#endif
return 0;
}
#define OFFSET(x) offsetof(AC3DecodeContext, x)
#define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
-static const AVOption options[] = {
- { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 6.0, PAR },
-
-{"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 2, 0, "dmix_mode"},
-{"ltrt_cmixlev", "Lt/Rt Center Mix Level", OFFSET(ltrt_center_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-{"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-{"loro_cmixlev", "Lo/Ro Center Mix Level", OFFSET(loro_center_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-{"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-
- { NULL},
-};
-
-static const AVClass ac3_decoder_class = {
- .class_name = "AC3 decoder",
- .item_name = av_default_item_name,
- .option = options,
- .version = LIBAVUTIL_VERSION_INT,
-};
-
-AVCodec ff_ac3_decoder = {
- .name = "ac3",
- .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_AC3,
- .priv_data_size = sizeof (AC3DecodeContext),
- .init = ac3_decode_init,
- .close = ac3_decode_end,
- .decode = ac3_decode_frame,
- .capabilities = CODEC_CAP_DR1,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_NONE },
- .priv_class = &ac3_decoder_class,
-};
-
-#if CONFIG_EAC3_DECODER
-static const AVClass eac3_decoder_class = {
- .class_name = "E-AC3 decoder",
- .item_name = av_default_item_name,
- .option = options,
- .version = LIBAVUTIL_VERSION_INT,
-};
-
-AVCodec ff_eac3_decoder = {
- .name = "eac3",
- .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_EAC3,
- .priv_data_size = sizeof (AC3DecodeContext),
- .init = ac3_decode_init,
- .close = ac3_decode_end,
- .decode = ac3_decode_frame,
- .capabilities = CODEC_CAP_DR1,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_NONE },
- .priv_class = &eac3_decoder_class,
-};
-#endif
diff --git a/libavcodec/ac3dec.h b/libavcodec/ac3dec.h
index 58d8ee6..255b9df 100644
--- a/libavcodec/ac3dec.h
+++ b/libavcodec/ac3dec.h
@@ -51,6 +51,7 @@
#define AVCODEC_AC3DEC_H
#include "libavutil/float_dsp.h"
+#include "libavutil/fixed_dsp.h"
#include "libavutil/lfg.h"
#include "ac3.h"
#include "ac3dsp.h"
@@ -138,8 +139,8 @@ typedef struct AC3DecodeContext {
int num_spx_bands; ///< number of spx bands (nspxbnds)
uint8_t spx_band_sizes[SPX_MAX_BANDS]; ///< number of bins in each spx band
uint8_t first_spx_coords[AC3_MAX_CHANNELS]; ///< first spx coordinates states (firstspxcos)
- float spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor (nblendfact)
- float spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact)
+ INTFLOAT spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor (nblendfact)
+ INTFLOAT spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact)
///@}
///@name Adaptive hybrid transform
@@ -151,15 +152,15 @@ typedef struct AC3DecodeContext {
int fbw_channels; ///< number of full-bandwidth channels
int channels; ///< number of total channels
int lfe_ch; ///< index of LFE channel
- float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
+ SHORTFLOAT downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
int downmixed; ///< indicates if coeffs are currently downmixed
int output_mode; ///< output channel configuration
int out_channels; ///< number of output channels
///@}
///@name Dynamic range
- float dynamic_range[2]; ///< dynamic range
- float drc_scale; ///< percentage of dynamic range compression to be applied
+ INTFLOAT dynamic_range[2]; ///< dynamic range
+ INTFLOAT drc_scale; ///< percentage of dynamic range compression to be applied
///@}
///@name Bandwidth
@@ -207,22 +208,26 @@ typedef struct AC3DecodeContext {
///@name Optimization
DSPContext dsp; ///< for optimization
+#if CONFIG_AC3_FIXED
+ AVFixedDSPContext *fdsp;
+#else
AVFloatDSPContext fdsp;
+#endif
AC3DSPContext ac3dsp;
FmtConvertContext fmt_conv; ///< optimized conversion functions
///@}
- float *outptr[AC3_MAX_CHANNELS];
- float *xcfptr[AC3_MAX_CHANNELS];
- float *dlyptr[AC3_MAX_CHANNELS];
+ SHORTFLOAT *outptr[AC3_MAX_CHANNELS];
+ INTFLOAT *xcfptr[AC3_MAX_CHANNELS];
+ INTFLOAT *dlyptr[AC3_MAX_CHANNELS];
///@name Aligned arrays
- DECLARE_ALIGNED(16, int32_t, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< fixed-point transform coefficients
- DECLARE_ALIGNED(32, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< transform coefficients
- DECLARE_ALIGNED(32, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< delay - added to the next block
- DECLARE_ALIGNED(32, float, window)[AC3_BLOCK_SIZE]; ///< window coefficients
- DECLARE_ALIGNED(32, float, tmp_output)[AC3_BLOCK_SIZE]; ///< temporary storage for output before windowing
- DECLARE_ALIGNED(32, float, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< output after imdct transform and windowing
+ DECLARE_ALIGNED(16, int, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< fixed-point transform coefficients
+ DECLARE_ALIGNED(32, INTFLOAT, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< transform coefficients
+ DECLARE_ALIGNED(32, INTFLOAT, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< delay - added to the next block
+ DECLARE_ALIGNED(32, INTFLOAT, window)[AC3_BLOCK_SIZE]; ///< window coefficients
+ DECLARE_ALIGNED(32, INTFLOAT, tmp_output)[AC3_BLOCK_SIZE]; ///< temporary storage for output before windowing
+ DECLARE_ALIGNED(32, SHORTFLOAT, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< output after imdct transform and windowing
DECLARE_ALIGNED(32, uint8_t, input_buffer)[AC3_FRAME_BUFFER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; ///< temp buffer to prevent overread
///@}
} AC3DecodeContext;
diff --git a/libavcodec/ac3dec_fixed.c b/libavcodec/ac3dec_fixed.c
new file mode 100644
index 0000000..1eb9a0b
--- /dev/null
+++ b/libavcodec/ac3dec_fixed.c
@@ -0,0 +1,176 @@
+/*
+ * Copyright (c) 2012
+ * MIPS Technologies, Inc., California.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ * Author: Stanislav Ocovaj (socovaj at mips.com)
+ *
+ * AC3 fixed-point decoder for MIPS platforms
+ *
+ * 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
+ */
+
+#define FFT_FLOAT 0
+#define CONFIG_AC3_FIXED 1
+#define FFT_FIXED_32 1
+#include "ac3dec.h"
+
+
+/**
+ * Table for center mix levels
+ * reference: Section 5.4.2.4 cmixlev
+ */
+static const uint8_t center_levels[4] = { 4, 5, 6, 5 };
+
+/**
+ * Table for surround mix levels
+ * reference: Section 5.4.2.5 surmixlev
+ */
+static const uint8_t surround_levels[4] = { 4, 6, 7, 6 };
+
+int end_freq_inv_tab[8] =
+{
+ 50529027, 44278013, 39403370, 32292987, 27356480, 23729101, 20951060, 18755316
+};
+
+static void scale_coefs (
+ int32_t *dst,
+ const int32_t *src,
+ int dynrng,
+ int len)
+{
+ int i, shift, round;
+ int16_t mul;
+ int temp, temp1, temp2, temp3, temp4, temp5, temp6, temp7;
+
+ mul = (dynrng & 0x1f) + 0x20;
+ shift = 4 - ((dynrng << 24) >> 29);
+ round = 1 << (shift-1);
+ for (i=0; i<len; i+=8) {
+
+ temp = src[i] * mul;
+ temp1 = src[i+1] * mul;
+ temp = temp + round;
+ temp2 = src[i+2] * mul;
+
+ temp1 = temp1 + round;
+ dst[i] = temp >> shift;
+ temp3 = src[i+3] * mul;
+ temp2 = temp2 + round;
+
+ dst[i+1] = temp1 >> shift;
+ temp4 = src[i + 4] * mul;
+ temp3 = temp3 + round;
+ dst[i+2] = temp2 >> shift;
+
+ temp5 = src[i+5] * mul;
+ temp4 = temp4 + round;
+ dst[i+3] = temp3 >> shift;
+ temp6 = src[i+6] * mul;
+
+ dst[i+4] = temp4 >> shift;
+ temp5 = temp5 + round;
+ temp7 = src[i+7] * mul;
+ temp6 = temp6 + round;
+
+ dst[i+5] = temp5 >> shift;
+ temp7 = temp7 + round;
+ dst[i+6] = temp6 >> shift;
+ dst[i+7] = temp7 >> shift;
+
+ }
+}
+
+/**
+ * Downmix samples from original signal to stereo or mono (this is for 16-bit samples
+ * and fixed point decoder - original (for 32-bit samples) is in ac3dsp.c).
+ */
+static void ac3_downmix_c_fixed16(int16_t **samples, int16_t (*matrix)[2],
+ int out_ch, int in_ch, int len)
+{
+ int i, j;
+ int v0, v1;
+ if (out_ch == 2) {
+ for (i = 0; i < len; i++) {
+ v0 = v1 = 0;
+ for (j = 0; j < in_ch; j++) {
+ v0 += samples[j][i] * matrix[j][0];
+ v1 += samples[j][i] * matrix[j][1];
+ }
+ samples[0][i] = (v0+2048)>>12;
+ samples[1][i] = (v1+2048)>>12;
+ }
+ } else if (out_ch == 1) {
+ for (i = 0; i < len; i++) {
+ v0 = 0;
+ for (j = 0; j < in_ch; j++)
+ v0 += samples[j][i] * matrix[j][0];
+ samples[0][i] = (v0+2048)>>12;
+ }
+ }
+}
+
+#include "ac3dec.c"
+
+static const AVOption options[] = {
+ { NULL},
+};
+
+static const AVClass ac3_decoder_class = {
+ .class_name = "Fixed-Point AC-3 Decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_ac3_fixed_decoder = {
+ .name = "ac3_fixed",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_AC3,
+ .priv_data_size = sizeof (AC3DecodeContext),
+ .init = ac3_decode_init,
+ .close = ac3_decode_end,
+ .decode = ac3_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
+ .priv_class = &ac3_decoder_class,
+};
diff --git a/libavcodec/ac3dec_float.c b/libavcodec/ac3dec_float.c
new file mode 100644
index 0000000..7108921
--- /dev/null
+++ b/libavcodec/ac3dec_float.c
@@ -0,0 +1,89 @@
+/*
+ * AC-3 Audio Decoder
+ * This code was developed as part of Google Summer of Code 2006.
+ * E-AC-3 support was added as part of Google Summer of Code 2007.
+ *
+ * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com)
+ * Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec at gmail.com>
+ * Copyright (c) 2007 Justin Ruggles <justin.ruggles at gmail.com>
+ *
+ * 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
+ */
+
+/**
+ * Upmix delay samples from stereo to original channel layout.
+ */
+#include "ac3dec.h"
+#include "ac3dec.c"
+
+static const AVOption options[] = {
+ { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 6.0, PAR },
+
+{"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 2, 0, "dmix_mode"},
+{"ltrt_cmixlev", "Lt/Rt Center Mix Level", OFFSET(ltrt_center_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+{"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+{"loro_cmixlev", "Lo/Ro Center Mix Level", OFFSET(loro_center_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+{"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level), AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+
+ { NULL},
+};
+
+static const AVClass ac3_decoder_class = {
+ .class_name = "AC3 decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_ac3_decoder = {
+ .name = "ac3",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_AC3,
+ .priv_data_size = sizeof (AC3DecodeContext),
+ .init = ac3_decode_init,
+ .close = ac3_decode_end,
+ .decode = ac3_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
+ .priv_class = &ac3_decoder_class,
+};
+
+#if CONFIG_EAC3_DECODER
+static const AVClass eac3_decoder_class = {
+ .class_name = "E-AC3 decoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_eac3_decoder = {
+ .name = "eac3",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_EAC3,
+ .priv_data_size = sizeof (AC3DecodeContext),
+ .init = ac3_decode_init,
+ .close = ac3_decode_end,
+ .decode = ac3_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
+ .priv_class = &eac3_decoder_class,
+};
+#endif
diff --git a/libavcodec/ac3dsp.c b/libavcodec/ac3dsp.c
index 0d5ea39..b746817 100644
--- a/libavcodec/ac3dsp.c
+++ b/libavcodec/ac3dsp.c
@@ -239,6 +239,31 @@ static void ac3_downmix_c(float **samples, float (*matrix)[2],
}
}
+static void ac3_downmix_c_fixed(int32_t **samples, int16_t (*matrix)[2],
+ int out_ch, int in_ch, int len)
+{
+ int i, j;
+ int64_t v0, v1;
+ if (out_ch == 2) {
+ for (i = 0; i < len; i++) {
+ v0 = v1 = 0;
+ for (j = 0; j < in_ch; j++) {
+ v0 += (int64_t)samples[j][i] * matrix[j][0];
+ v1 += (int64_t)samples[j][i] * matrix[j][1];
+ }
+ samples[0][i] = (v0+2048)>>12;
+ samples[1][i] = (v1+2048)>>12;
+ }
+ } else if (out_ch == 1) {
+ for (i = 0; i < len; i++) {
+ v0 = 0;
+ for (j = 0; j < in_ch; j++)
+ v0 += (int64_t)samples[j][i] * matrix[j][0];
+ samples[0][i] = (v0+2048)>>12;
+ }
+ }
+}
+
static void apply_window_int16_c(int16_t *output, const int16_t *input,
const int16_t *window, unsigned int len)
{
@@ -266,6 +291,7 @@ av_cold void ff_ac3dsp_init(AC3DSPContext *c, int bit_exact)
c->sum_square_butterfly_int32 = ac3_sum_square_butterfly_int32_c;
c->sum_square_butterfly_float = ac3_sum_square_butterfly_float_c;
c->downmix = ac3_downmix_c;
+ c->downmix_fixed = ac3_downmix_c_fixed;
c->apply_window_int16 = apply_window_int16_c;
if (ARCH_ARM)
diff --git a/libavcodec/ac3dsp.h b/libavcodec/ac3dsp.h
index f18583c..ed98c8c 100644
--- a/libavcodec/ac3dsp.h
+++ b/libavcodec/ac3dsp.h
@@ -135,6 +135,9 @@ typedef struct AC3DSPContext {
void (*downmix)(float **samples, float (*matrix)[2], int out_ch,
int in_ch, int len);
+ void (*downmix_fixed)(int32_t **samples, int16_t (*matrix)[2], int out_ch,
+ int in_ch, int len);
+
/**
* Apply symmetric window in 16-bit fixed-point.
* @param output destination array
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 10edd39..c55a809 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -323,7 +323,7 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (AAC, aac);
REGISTER_DECODER(AAC_LATM, aac_latm);
REGISTER_ENCDEC (AC3, ac3);
- REGISTER_ENCODER(AC3_FIXED, ac3_fixed);
+ REGISTER_ENCDEC (AC3_FIXED, ac3_fixed);
REGISTER_ENCDEC (ALAC, alac);
REGISTER_DECODER(ALS, als);
REGISTER_DECODER(AMRNB, amrnb);
diff --git a/libavcodec/kbdwin.c b/libavcodec/kbdwin.c
index 5a62e9d..bf32aeb 100644
--- a/libavcodec/kbdwin.c
+++ b/libavcodec/kbdwin.c
@@ -45,3 +45,13 @@ av_cold void ff_kbd_window_init(float *window, float alpha, int n)
for (i = 0; i < n; i++)
window[i] = sqrt(local_window[i] / sum);
}
+
+av_cold void ff_kbd_window_init_fixed(int32_t *window, float alpha, int n)
+{
+ int i;
+ float local_window[FF_KBD_WINDOW_MAX];
+
+ ff_kbd_window_init(local_window, alpha, n);
+ for (i = 0; i < n; i++)
+ window[i] = (int)floor(2147483647.0 * local_window[i] + 0.5);
+}
diff --git a/libavcodec/kbdwin.h b/libavcodec/kbdwin.h
index 4b93975..2e02e10 100644
--- a/libavcodec/kbdwin.h
+++ b/libavcodec/kbdwin.h
@@ -31,5 +31,6 @@
* @param n size of half window, max FF_KBD_WINDOW_MAX
*/
void ff_kbd_window_init(float *window, float alpha, int n);
+void ff_kbd_window_init_fixed(int32_t *window, float alpha, int n);
#endif /* AVCODEC_KBDWIN_H */
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 21f5dea..19b4934 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -29,7 +29,7 @@
#include "libavutil/version.h"
#define LIBAVCODEC_VERSION_MAJOR 55
-#define LIBAVCODEC_VERSION_MINOR 55
+#define LIBAVCODEC_VERSION_MINOR 56
#define LIBAVCODEC_VERSION_MICRO 107
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
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