mirror of
https://github.com/xenia-project/FFmpeg.git
synced 2024-11-24 12:09:55 +00:00
Merge remote-tracking branch 'qatar/master'
* qatar/master: configure: add functions for testing code fragments af_amix: avoid spurious EAGAIN. af_amix: return AVERROR(EAGAIN) when request_frame didn't produce output. af_amix: only consider negative return codes as errors. avconv: use only meaningful timestamps in start time check. avconv: fix the check for -ss as an output option. mss3: add forgotten 'static' qualifier for private table lavc: options: add planar names for request_sample_fmt flacdec: add planar output support flvdec: Treat all nellymoser versions as the same codec Conflicts: ffmpeg.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
commit
46a47077ee
39
configure
vendored
39
configure
vendored
@ -716,6 +716,20 @@ check_ld(){
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check_cmd $ld $LDFLAGS $flags -o $TMPE $TMPO $libs $extralibs
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}
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check_code(){
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log check_code "$@"
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check=$1
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headers=$2
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code=$3
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shift 3
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{
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for hdr in $headers; do
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echo "#include <$hdr>"
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done
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echo "int main(void) { $code; return 0; }"
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} | check_$check "$@"
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}
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check_cppflags(){
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log check_cppflags "$@"
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set -- $($filter_cppflags "$@")
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@ -912,15 +926,7 @@ check_type(){
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type=$2
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shift 2
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disable_safe "$type"
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incs=""
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for hdr in $headers; do
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incs="$incs
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#include <$hdr>"
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done
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check_cc "$@" <<EOF && enable_safe "$type"
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$incs
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$type v;
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EOF
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check_code cc "$headers" "$type v" "$@" && enable_safe "$type"
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}
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check_struct(){
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@ -930,15 +936,8 @@ check_struct(){
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member=$3
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shift 3
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disable_safe "${struct}_${member}"
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incs=""
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for hdr in $headers; do
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incs="$incs
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#include <$hdr>"
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done
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check_cc "$@" <<EOF && enable_safe "${struct}_${member}"
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$incs
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const void *p = &(($struct *)0)->$member;
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EOF
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check_code cc "$headers" "const void *p = &(($struct *)0)->$member" "$@" &&
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enable_safe "${struct}_${member}"
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}
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require(){
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@ -2689,9 +2688,7 @@ case "$arch" in
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;;
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x86)
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subarch="x86_32"
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check_cc <<EOF && subarch="x86_64"
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int test[(int)sizeof(char*) - 7];
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EOF
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check_code cc "" "int test[(int)sizeof(char*) - 7]" && subarch="x86_64"
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if test "$subarch" = "x86_64"; then
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spic=$shared
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fi
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@ -104,11 +104,22 @@ int avpriv_flac_is_extradata_valid(AVCodecContext *avctx,
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static void flac_set_bps(FLACContext *s)
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{
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if (s->bps > 16) {
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
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enum AVSampleFormat req = s->avctx->request_sample_fmt;
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int need32 = s->bps > 16;
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int want32 = av_get_bytes_per_sample(req) > 2;
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int planar = av_sample_fmt_is_planar(req);
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if (need32 || want32) {
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if (planar)
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
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else
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
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s->sample_shift = 32 - s->bps;
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} else {
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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if (planar)
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
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else
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s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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s->sample_shift = 16 - s->bps;
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}
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}
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@ -132,7 +143,7 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)
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avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
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allocate_buffers(s);
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flac_set_bps(s);
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ff_flacdsp_init(&s->dsp, avctx->sample_fmt);
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ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps);
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s->got_streaminfo = 1;
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avcodec_get_frame_defaults(&s->frame);
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@ -233,7 +244,7 @@ static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)
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avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]);
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allocate_buffers(s);
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flac_set_bps(s);
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ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt);
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ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
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s->got_streaminfo = 1;
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return 0;
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@ -492,9 +503,11 @@ static int decode_frame(FLACContext *s)
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"supported\n");
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return -1;
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}
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s->bps = s->avctx->bits_per_raw_sample = fi.bps;
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flac_set_bps(s);
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if (!s->bps) {
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s->bps = s->avctx->bits_per_raw_sample = fi.bps;
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flac_set_bps(s);
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}
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if (!s->max_blocksize)
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s->max_blocksize = FLAC_MAX_BLOCKSIZE;
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@ -520,7 +533,7 @@ static int decode_frame(FLACContext *s)
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if (!s->got_streaminfo) {
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allocate_buffers(s);
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ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt);
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ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
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s->got_streaminfo = 1;
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dump_headers(s->avctx, (FLACStreaminfo *)s);
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}
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@ -628,4 +641,9 @@ AVCodec ff_flac_decoder = {
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.decode = flac_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S32,
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AV_SAMPLE_FMT_S32P,
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-1 },
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};
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@ -23,10 +23,21 @@
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#include "flacdsp.h"
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#define SAMPLE_SIZE 16
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#define PLANAR 0
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#include "flacdsp_template.c"
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#undef PLANAR
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#define PLANAR 1
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#include "flacdsp_template.c"
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#undef SAMPLE_SIZE
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#undef PLANAR
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#define SAMPLE_SIZE 32
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#define PLANAR 0
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#include "flacdsp_template.c"
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#undef PLANAR
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#define PLANAR 1
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#include "flacdsp_template.c"
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static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
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@ -72,15 +83,27 @@ static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
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}
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av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt)
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av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
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int bps)
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{
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if (bps > 16)
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c->lpc = flac_lpc_32_c;
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else
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c->lpc = flac_lpc_16_c;
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switch (fmt) {
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case AV_SAMPLE_FMT_S32:
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c->decorrelate[0] = flac_decorrelate_indep_c_32;
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c->decorrelate[1] = flac_decorrelate_ls_c_32;
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c->decorrelate[2] = flac_decorrelate_rs_c_32;
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c->decorrelate[3] = flac_decorrelate_ms_c_32;
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c->lpc = flac_lpc_32_c;
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break;
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case AV_SAMPLE_FMT_S32P:
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c->decorrelate[0] = flac_decorrelate_indep_c_32p;
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c->decorrelate[1] = flac_decorrelate_ls_c_32p;
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c->decorrelate[2] = flac_decorrelate_rs_c_32p;
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c->decorrelate[3] = flac_decorrelate_ms_c_32p;
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break;
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case AV_SAMPLE_FMT_S16:
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@ -88,7 +111,13 @@ av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt)
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c->decorrelate[1] = flac_decorrelate_ls_c_16;
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c->decorrelate[2] = flac_decorrelate_rs_c_16;
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c->decorrelate[3] = flac_decorrelate_ms_c_16;
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c->lpc = flac_lpc_16_c;
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break;
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case AV_SAMPLE_FMT_S16P:
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c->decorrelate[0] = flac_decorrelate_indep_c_16p;
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c->decorrelate[1] = flac_decorrelate_ls_c_16p;
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c->decorrelate[2] = flac_decorrelate_rs_c_16p;
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c->decorrelate[3] = flac_decorrelate_ms_c_16p;
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break;
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}
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}
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@ -29,6 +29,6 @@ typedef struct FLACDSPContext {
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int qlevel, int len);
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} FLACDSPContext;
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void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt);
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void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int bps);
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#endif /* AVCODEC_FLACDSP_H */
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@ -19,68 +19,85 @@
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*/
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#include <stdint.h>
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#include "libavutil/avutil.h"
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#undef FUNC
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#undef FSUF
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#undef sample
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#undef sample_type
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#undef OUT
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#undef S
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#if SAMPLE_SIZE == 32
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# define FUNC(n) n ## _32
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# define sample int32_t
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# define sample_type int32_t
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#else
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# define FUNC(n) n ## _16
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# define sample int16_t
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# define sample_type int16_t
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#endif
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#if PLANAR
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# define FSUF AV_JOIN(SAMPLE_SIZE, p)
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# define sample sample_type *
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# define OUT(n) n
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# define S(s, c, i) (s[c][i])
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#else
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# define FSUF SAMPLE_SIZE
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# define sample sample_type
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# define OUT(n) n[0]
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# define S(s, c, i) (*s++)
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#endif
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#define FUNC(n) AV_JOIN(n ## _, FSUF)
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static void FUNC(flac_decorrelate_indep_c)(uint8_t **out, int32_t **in,
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int channels, int len, int shift)
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{
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sample *samples = (sample *) out[0];
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sample *samples = (sample *) OUT(out);
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int i, j;
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for (j = 0; j < len; j++)
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for (i = 0; i < channels; i++)
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*samples++ = in[i][j] << shift;
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S(samples, i, j) = in[i][j] << shift;
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}
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static void FUNC(flac_decorrelate_ls_c)(uint8_t **out, int32_t **in,
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int channels, int len, int shift)
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{
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sample *samples = (sample *) out[0];
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sample *samples = (sample *) OUT(out);
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int i;
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for (i = 0; i < len; i++) {
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int a = in[0][i];
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int b = in[1][i];
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*samples++ = a << shift;
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*samples++ = (a - b) << shift;
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S(samples, 0, i) = a << shift;
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S(samples, 1, i) = (a - b) << shift;
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}
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}
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static void FUNC(flac_decorrelate_rs_c)(uint8_t **out, int32_t **in,
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int channels, int len, int shift)
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{
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sample *samples = (sample *) out[0];
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sample *samples = (sample *) OUT(out);
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int i;
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for (i = 0; i < len; i++) {
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int a = in[0][i];
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int b = in[1][i];
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*samples++ = (a + b) << shift;
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*samples++ = b << shift;
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S(samples, 0, i) = (a + b) << shift;
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S(samples, 1, i) = b << shift;
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}
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}
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static void FUNC(flac_decorrelate_ms_c)(uint8_t **out, int32_t **in,
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int channels, int len, int shift)
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{
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sample *samples = (sample *) out[0];
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sample *samples = (sample *) OUT(out);
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int i;
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for (i = 0; i < len; i++) {
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int a = in[0][i];
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int b = in[1][i];
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a -= b >> 1;
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*samples++ = (a + b) << shift;
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*samples++ = a << shift;
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S(samples, 0, i) = (a + b) << shift;
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S(samples, 1, i) = a << shift;
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}
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}
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|
@ -141,7 +141,7 @@ static const uint8_t mss3_chroma_quant[64] = {
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99, 99, 99, 99, 99, 99, 99, 99
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};
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const uint8_t zigzag_scan[64] = {
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static const uint8_t zigzag_scan[64] = {
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0, 1, 8, 16, 9, 2, 3, 10,
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17, 24, 32, 25, 18, 11, 4, 5,
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12, 19, 26, 33, 40, 48, 41, 34,
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|
@ -400,6 +400,11 @@ static const AVOption options[]={
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{"s32", "32-bit signed integer", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S32 }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
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{"flt", "32-bit float", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_FLT }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
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{"dbl", "64-bit double", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_DBL }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
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{"u8p" , "8-bit unsigned integer planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_U8P }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
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{"s16p", "16-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S16P }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
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{"s32p", "32-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_S32P }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
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{"fltp", "32-bit float planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_FLTP }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
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{"dblp", "64-bit double planar", 0, AV_OPT_TYPE_CONST, {.dbl = AV_SAMPLE_FMT_DBLP }, INT_MIN, INT_MAX, A|D, "request_sample_fmt"},
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{NULL},
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};
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|
@ -354,7 +354,7 @@ static int request_samples(AVFilterContext *ctx, int min_samples)
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s->input_state[i] = INPUT_OFF;
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continue;
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}
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} else if (ret)
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} else if (ret < 0)
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return ret;
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}
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return 0;
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@ -403,7 +403,7 @@ static int request_frame(AVFilterLink *outlink)
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available_samples = get_available_samples(s);
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if (!available_samples)
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return 0;
|
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return AVERROR(EAGAIN);
|
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return output_frame(outlink, available_samples);
|
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}
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@ -416,7 +416,7 @@ static int request_frame(AVFilterLink *outlink)
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return AVERROR_EOF;
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else
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return AVERROR(EAGAIN);
|
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} else if (ret)
|
||||
} else if (ret < 0)
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||||
return ret;
|
||||
}
|
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av_assert0(s->frame_list->nb_frames > 0);
|
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@ -431,10 +431,12 @@ static int request_frame(AVFilterLink *outlink)
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ret = calc_active_inputs(s);
|
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if (ret < 0)
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return ret;
|
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}
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|
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if (s->active_inputs > 1) {
|
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available_samples = get_available_samples(s);
|
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if (!available_samples)
|
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return 0;
|
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return AVERROR(EAGAIN);
|
||||
available_samples = FFMIN(available_samples, wanted_samples);
|
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} else {
|
||||
available_samples = wanted_samples;
|
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|
@ -113,11 +113,7 @@ static int flv_same_audio_codec(AVCodecContext *acodec, int flags)
|
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case FLV_CODECID_MP3:
|
||||
return acodec->codec_id == CODEC_ID_MP3;
|
||||
case FLV_CODECID_NELLYMOSER_8KHZ_MONO:
|
||||
return acodec->sample_rate == 8000 &&
|
||||
acodec->codec_id == CODEC_ID_NELLYMOSER;
|
||||
case FLV_CODECID_NELLYMOSER_16KHZ_MONO:
|
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return acodec->sample_rate == 16000 &&
|
||||
acodec->codec_id == CODEC_ID_NELLYMOSER;
|
||||
case FLV_CODECID_NELLYMOSER:
|
||||
return acodec->codec_id == CODEC_ID_NELLYMOSER;
|
||||
case FLV_CODECID_PCM_MULAW:
|
||||
|
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