mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
Ported to 0.9 (faad, amrnb, mpeg2dec)
Original commit message from CVS: Ported to 0.9 (faad, amrnb, mpeg2dec)
This commit is contained in:
parent
a29a07acfb
commit
a853a4b182
3 changed files with 186 additions and 62 deletions
56
configure.ac
56
configure.ac
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@ -378,6 +378,50 @@ GST_CHECK_FEATURE(AALIB, [aasink plug-in], aasink, [
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AS_SCRUB_INCLUDE(AALIB_CFLAGS)
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])
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dnl *** AMR-NB ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_AMRNB, true)
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GST_CHECK_FEATURE(AMRNB, [AMR-NB], amrnbdec amrnbenc, [
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GST_CHECK_LIBHEADER(AMRNB, amrnb,
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Decoder_Interface_init, -lm,
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amrnb/interf_dec.h,
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AMRNB_LIBS="-lamrnb -lm"
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AC_SUBST(AMRNB_LIBS))
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])
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dnl **** Free AAC Decoder (FAAD) ****
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translit(dnm, m, l) AM_CONDITIONAL(USE_FAAD, true)
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GST_CHECK_FEATURE(FAAD, [AAC decoder plug-in], faad, [
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HAVE_FAAD="yes"
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GST_CHECK_LIBHEADER(FAAD, faad, faacDecOpen, -lm, faad.h, FAAD_LIBS="-lfaad -lm", HAVE_FAAD="no")
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if test $HAVE_FAAD = "yes"; then
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AC_MSG_CHECKING([Checking for FAAD >= 2])
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AC_TRY_RUN([
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#include <faad.h>
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#if !defined(FAAD2_VERSION) || !defined(FAAD_FMT_DOUBLE)
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#error Not faad2
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#else
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#include <string.h>
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int main()
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{
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char version[9] = FAAD2_VERSION;
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// a release candidate of 2.0 is not enought for us
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if ( strstr( version, "2.0 RC" ) ) { return 1; }
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return 0;
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}
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#endif
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],
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[ HAVE_FAAD="yes" && AC_MSG_RESULT(yes)],
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[ HAVE_FAAD="no" && AC_MSG_RESULT(no)])
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fi;
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AS_SCRUB_INCLUDE(FAAD_CFLAGS)
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AC_SUBST(FAAD_LIBS)
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])
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dnl *** libcaca ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_LIBCACA, true)
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GST_CHECK_FEATURE(LIBCACA, [libcaca], libcaca, [
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@ -434,6 +478,15 @@ GST_CHECK_FEATURE(MAD, [mad mp3 decoder], mad, [
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])
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AC_SUBST(MAD_LIBS)
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dnl *** mpeg2dec ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_MPEG2DEC, true)
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GST_CHECK_FEATURE(MPEG2DEC, [mpeg2dec], mpeg2dec, [
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PKG_CHECK_MODULES(MPEG2DEC, libmpeg2 >= 0.4.0,
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HAVE_MPEG2DEC="yes", HAVE_MPEG2DEC="no")
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AC_SUBST(MPEG2DEC_CFLAGS)
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AC_SUBST(MPEG2DEC_LIBS)
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])
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dnl *** dv1394 ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_DV1394, true)
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GST_CHECK_FEATURE(DV1394, [raw1394 and avc1394 library], dv1394src, [
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@ -555,6 +608,9 @@ ext/raw1394/Makefile
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ext/shout2/Makefile
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ext/sidplay/Makefile
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ext/esd/Makefile
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ext/mpeg2dec/Makefile
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ext/faad/Makefile
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ext/amrnb/Makefile
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po/Makefile.in
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common/Makefile
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common/m4/Makefile
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@ -10,11 +10,11 @@ else
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AALIB_DIR=
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endif
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# if USE_AMRNB
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# AMRNB_DIR=amrnb
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# else
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AMRNB_DIR=
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# endif
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if USE_AMRNB
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AMRNB_DIR=amrnb
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else
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AMRNB_DIR=
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endif
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# if USE_ARTS
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# ARTS_DIR=arts
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@ -106,11 +106,11 @@ endif
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FAAC_DIR=
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# endif
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# if USE_FAAD
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# FAAD_DIR=faad
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# else
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FAAD_DIR=
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# endif
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if USE_FAAD
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FAAD_DIR=faad
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else
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FAAD_DIR=
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endif
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## if USE_FESTIVAL
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## FESTIVAL_DIR=festival
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@ -238,11 +238,11 @@ endif
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MIKMOD_DIR=
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# endif
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# if USE_MPEG2DEC
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# MPEG2DEC_DIR=mpeg2dec
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# else
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MPEG2DEC_DIR=
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#endif
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if USE_MPEG2DEC
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MPEG2DEC_DIR=mpeg2dec
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else
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MPEG2DEC_DIR=
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endif
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# if USE_MPEG2ENC
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# MPEG2ENC_DIR=mpeg2enc
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@ -440,4 +440,7 @@ DIST_SUBDIRS=\
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libcaca \
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esd \
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raw1394 \
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dv
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dv \
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amrnb \
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faad \
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mpeg2dec
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@ -79,12 +79,15 @@ static void gst_faad_base_init (GstFaadClass * klass);
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static void gst_faad_class_init (GstFaadClass * klass);
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static void gst_faad_init (GstFaad * faad);
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/*
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static GstPadLinkReturn
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gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps);
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static GstPadLinkReturn
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gst_faad_srcconnect (GstPad * pad, const GstCaps * caps);
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gst_faad_srcconnect (GstPad * pad, const GstCaps * caps);*/
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static gboolean gst_faad_setcaps (GstPad * pad, GstCaps * caps);
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static GstCaps *gst_faad_srcgetcaps (GstPad * pad);
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static void gst_faad_chain (GstPad * pad, GstData * data);
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static gboolean gst_faad_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_faad_chain (GstPad * pad, GstBuffer * buffer);
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static GstElementStateReturn gst_faad_change_state (GstElement * element);
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static GstElementClass *parent_class = NULL;
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@ -155,28 +158,59 @@ gst_faad_init (GstFaad * faad)
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faad->channel_positions = NULL;
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faad->init = FALSE;
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GST_FLAG_SET (faad, GST_ELEMENT_EVENT_AWARE);
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/* GST_FLAG_SET (faad, GST_ELEMENT_EVENT_AWARE); */
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faad->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get (&sink_template),
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"sink");
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gst_element_add_pad (GST_ELEMENT (faad), faad->sinkpad);
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gst_pad_set_event_function (faad->sinkpad, gst_faad_event);
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gst_pad_set_setcaps_function (faad->sinkpad, gst_faad_setcaps);
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gst_pad_set_chain_function (faad->sinkpad, gst_faad_chain);
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gst_pad_set_link_function (faad->sinkpad, gst_faad_sinkconnect);
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/*gst_pad_set_link_function (faad->sinkpad, gst_faad_sinkconnect); */
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faad->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get (&src_template),
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"src");
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gst_element_add_pad (GST_ELEMENT (faad), faad->srcpad);
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gst_pad_set_link_function (faad->srcpad, gst_faad_srcconnect);
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gst_pad_use_fixed_caps (faad->srcpad);
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/*gst_pad_set_link_function (faad->srcpad, gst_faad_srcconnect); */
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gst_pad_set_getcaps_function (faad->srcpad, gst_faad_srcgetcaps);
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}
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static gboolean
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gst_faad_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstStructure *structure;
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GstFaad *faad;
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GstCaps *copy;
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faad = GST_FAAD (GST_PAD_PARENT (pad));
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structure = gst_caps_get_structure (caps, 0);
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/* get channel count */
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gst_structure_get_int (structure, "channels", &faad->channels);
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gst_structure_get_int (structure, "rate", &faad->samplerate);
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/* create reverse caps */
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copy = gst_caps_new_simple ("audio/x-raw-float",
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"channels", G_TYPE_INT, faad->channels,
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"depth", G_TYPE_INT, G_STRINGIFY (bpp),
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"rate", G_TYPE_INT, faad->samplerate);
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gst_pad_set_caps (faad->srcpad, copy);
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gst_caps_unref (copy);
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return TRUE;
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}
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/*
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* Channel identifier conversion - caller g_free()s result!
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*/
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/*
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static guchar *
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gst_faad_chanpos_from_gst (GstAudioChannelPosition * pos, guint num)
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{
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@ -223,7 +257,7 @@ gst_faad_chanpos_from_gst (GstAudioChannelPosition * pos, guint num)
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return fpos;
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}
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*/
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static GstAudioChannelPosition *
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gst_faad_chanpos_to_gst (guchar * fpos, guint num)
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{
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@ -274,6 +308,7 @@ gst_faad_chanpos_to_gst (guchar * fpos, guint num)
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return pos;
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}
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/*
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static GstPadLinkReturn
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gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
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{
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@ -282,18 +317,18 @@ gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
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const GValue *value;
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GstBuffer *buf;
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/* Assume raw stream */
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// Assume raw stream
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faad->packetised = FALSE;
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if ((value = gst_structure_get_value (str, "codec_data"))) {
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gulong samplerate;
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guchar channels;
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/* We have codec data, means packetised stream */
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// We have codec data, means packetised stream
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faad->packetised = TRUE;
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buf = g_value_get_boxed (value);
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/* someone forgot that char can be unsigned when writing the API */
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// someone forgot that char can be unsigned when writing the API
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if ((gint8) faacDecInit2 (faad->handle, GST_BUFFER_DATA (buf),
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GST_BUFFER_SIZE (buf), &samplerate, &channels) < 0)
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return GST_PAD_LINK_REFUSED;
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@ -312,11 +347,11 @@ gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
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faad->need_channel_setup = TRUE;
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/* if there's no decoderspecificdata, it's all fine. We cannot know
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* much more at this point... */
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// if there's no decoderspecificdata, it's all fine. We cannot know
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// * much more at this point...
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return GST_PAD_LINK_OK;
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}
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*/
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static GstCaps *
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gst_faad_srcgetcaps (GstPad * pad)
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{
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@ -395,7 +430,7 @@ gst_faad_srcgetcaps (GstPad * pad)
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if (faad->channels != -1) {
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gst_structure_set (str, "channels", G_TYPE_INT, faad->channels, NULL);
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/* put channel information here */
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// put channel information here */
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if (faad->channel_positions) {
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GstAudioChannelPosition *pos;
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@ -437,6 +472,7 @@ gst_faad_srcgetcaps (GstPad * pad)
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return templ;
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}
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/**
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static GstPadLinkReturn
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gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
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{
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@ -455,15 +491,15 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
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mimetype = gst_structure_get_name (structure);
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/* Samplerate and channels are normally provided through
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* the getcaps function */
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// Samplerate and channels are normally provided through
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// * the getcaps function
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if (!gst_structure_get_int (structure, "channels", &channels) ||
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!gst_structure_get_int (structure, "rate", &rate) ||
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rate != faad->samplerate || channels != faad->channels) {
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return GST_PAD_LINK_REFUSED;
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}
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/* Another internal checkup. */
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// Another internal checkup.
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if (faad->need_channel_setup) {
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GstAudioChannelPosition *pos;
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guchar *fpos;
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@ -533,28 +569,55 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
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if (faacDecSetConfiguration (faad->handle, conf) == 0)
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return GST_PAD_LINK_REFUSED;
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/* FIXME: handle return value, how? */
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// FIXME: handle return value, how?
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faad->bps = depth / 8;
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return GST_PAD_LINK_OK;
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}
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return GST_PAD_LINK_REFUSED;
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}*/
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/*
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* Data reading.
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*/
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static gboolean
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gst_faad_event (GstPad * pad, GstEvent * event)
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{
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GstFaad *faad;
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gboolean res;
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faad = GST_FAAD (gst_pad_get_parent (pad));
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GST_LOG ("handling event %d", GST_EVENT_TYPE (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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case GST_EVENT_DISCONTINUOUS:
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default:
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break;
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}
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res = gst_pad_event_default (faad->sinkpad, event);
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return res;
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}
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static void
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gst_faad_chain (GstPad * pad, GstData * data)
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static GstFlowReturn
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gst_faad_chain (GstPad * pad, GstBuffer * buffer)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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guint input_size;
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guint skip_bytes = 0;
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guchar *input_data;
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GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
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GstBuffer *buf, *outbuf;
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GstBuffer *outbuf;
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faacDecFrameInfo *info;
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guint64 next_ts;
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void *out;
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gboolean run_loop = TRUE;
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/*
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if (GST_IS_EVENT (data)) {
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GstEvent *event = GST_EVENT (data);
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@ -572,14 +635,14 @@ gst_faad_chain (GstPad * pad, GstData * data)
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return;
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}
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}
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*/
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info = g_new0 (faacDecFrameInfo, 1);
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/* buffer + remaining data */
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buf = GST_BUFFER (data);
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next_ts = GST_BUFFER_TIMESTAMP (buf);
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/* buf = GST_BUFFER (data); */
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next_ts = GST_BUFFER_TIMESTAMP (buffer);
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if (faad->tempbuf) {
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buf = gst_buffer_join (faad->tempbuf, buf);
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buffer = gst_buffer_join (faad->tempbuf, buffer);
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faad->tempbuf = NULL;
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}
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@ -590,11 +653,12 @@ gst_faad_chain (GstPad * pad, GstData * data)
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glong init_res;
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init_res = faacDecInit (faad->handle,
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GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), &samplerate, &channels);
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GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), &samplerate,
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&channels);
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if (init_res < 0) {
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GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
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("Failed to init decoder from stream"));
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return;
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return GST_FLOW_UNEXPECTED;
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}
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skip_bytes = init_res;
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faad->init = TRUE;
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@ -609,8 +673,8 @@ gst_faad_chain (GstPad * pad, GstData * data)
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}
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/* decode cycle */
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input_data = GST_BUFFER_DATA (buf);
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input_size = GST_BUFFER_SIZE (buf);
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input_data = GST_BUFFER_DATA (buffer);
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input_size = GST_BUFFER_SIZE (buffer);
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info->bytesconsumed = input_size - skip_bytes;
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if (!faad->packetised) {
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@ -660,7 +724,7 @@ gst_faad_chain (GstPad * pad, GstData * data)
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}
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if (fmt_change) {
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GstPadLinkReturn ret;
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/*GstPadLinkReturn ret; */
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/* store new negotiation information */
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faad->samplerate = info->samplerate;
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@ -671,12 +735,12 @@ gst_faad_chain (GstPad * pad, GstData * data)
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memcpy (faad->channel_positions, info->channel_position,
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faad->channels);
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/* and negotiate */
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/* and negotiate
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ret = gst_pad_renegotiate (faad->srcpad);
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if (GST_PAD_LINK_FAILED (ret)) {
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GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL));
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break;
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}
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} */
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}
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/* play decoded data */
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@ -690,25 +754,27 @@ gst_faad_chain (GstPad * pad, GstData * data)
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|||
if (GST_CLOCK_TIME_IS_VALID (next_ts)) {
|
||||
next_ts += GST_BUFFER_DURATION (outbuf);
|
||||
}
|
||||
gst_pad_push (faad->srcpad, GST_DATA (outbuf));
|
||||
gst_pad_push (faad->srcpad, outbuf);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/* Keep the leftovers in raw stream */
|
||||
if (input_size > 0 && !faad->packetised) {
|
||||
if (input_size < GST_BUFFER_SIZE (buf)) {
|
||||
faad->tempbuf = gst_buffer_create_sub (buf,
|
||||
GST_BUFFER_SIZE (buf) - input_size, input_size);
|
||||
if (input_size < GST_BUFFER_SIZE (buffer)) {
|
||||
faad->tempbuf = gst_buffer_create_sub (buffer,
|
||||
GST_BUFFER_SIZE (buffer) - input_size, input_size);
|
||||
} else {
|
||||
faad->tempbuf = buf;
|
||||
gst_buffer_ref (buf);
|
||||
faad->tempbuf = buffer;
|
||||
gst_buffer_ref (buffer);
|
||||
}
|
||||
}
|
||||
|
||||
gst_buffer_unref (buf);
|
||||
gst_buffer_unref (buffer);
|
||||
|
||||
g_free (info);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static GstElementStateReturn
|
||||
|
@ -758,8 +824,7 @@ gst_faad_change_state (GstElement * element)
|
|||
static gboolean
|
||||
plugin_init (GstPlugin * plugin)
|
||||
{
|
||||
return gst_library_load ("gstaudio") &&
|
||||
gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
|
||||
return gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
|
||||
}
|
||||
|
||||
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
||||
|
|
Loading…
Reference in a new issue