gstreamer/gst/avi/gstavimux.c

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/* AVI muxer plugin for GStreamer
* Copyright (C) 2002 Ronald Bultje <rbultje@ronald.bitfreak.net>
* (C) 2006 Mark Nauwelaerts <manauw@skynet.be>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/* based on:
* - the old avimuxer (by Wim Taymans)
* - xawtv's aviwriter (by Gerd Knorr)
* - mjpegtools' avilib (by Rainer Johanni)
* - openDML large-AVI docs
*/
/**
* SECTION:element-avimux
*
* <refsect2>
* <para>
* Muxes raw or compressed audio and/or video streams into an AVI file.
* </para>
* <title>Example launch line</title>
* <para>
* (write everything in one line, without the backslash characters)
* <programlisting>
* gst-launch-0.10 videotestsrc num-buffers=250 \
* ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240,framerate=(fraction)25/1' \
* ! queue ! mux. \
* audiotestsrc num-buffers=440 ! audioconvert \
* ! 'audio/x-raw-int,rate=44100,channels=2' ! queue ! mux. \
* avimux name=mux ! filesink location=test.avi
* </programlisting>
* This will create an .AVI file containing an uncompressed video stream
* with a test picture and an uncompressed audio stream containing a
* test sound.
* </para>
* <title>Another example launch line</title>
* <para>
* (write everything in one line, without the backslash characters)
* <programlisting>
* gst-launch-0.10 videotestsrc num-buffers=250 \
* ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240,framerate=(fraction)25/1' \
* ! xvidenc ! queue ! mux. \
* audiotestsrc num-buffers=440 ! audioconvert ! 'audio/x-raw-int,rate=44100,channels=2' \
* ! lame ! queue ! mux. \
* avimux name=mux ! filesink location=test.avi
* </programlisting>
* This will create an .AVI file containing the same test video and sound
* as above, only that both streams will be compressed this time. This will
* only work if you have the necessary encoder elements installed of course.
* </para>
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gst/gst-i18n-plugin.h"
#include <stdlib.h>
#include <string.h>
#include <gst/video/video.h>
#include "gstavimux.h"
GST_DEBUG_CATEGORY_STATIC (avimux_debug);
#define GST_CAT_DEFAULT avimux_debug
enum
{
ARG_0,
ARG_BIGFILE
};
#define DEFAULT_BIGFILE TRUE
static const GstElementDetails gst_avi_mux_details =
GST_ELEMENT_DETAILS ("Avi muxer",
"Codec/Muxer",
"Muxes audio and video into an avi stream",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-msvideo")
);
static GstStaticPadTemplate video_sink_factory =
GST_STATIC_PAD_TEMPLATE ("video_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("video/x-raw-yuv, "
"format = (fourcc) { YUY2, I420 }, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], "
"framerate = (fraction) [ 0, MAX ]; "
"image/jpeg, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], "
"framerate = (fraction) [ 0, MAX ]; "
"video/x-divx, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], "
"framerate = (fraction) [ 0, MAX ], "
"divxversion = (int) [ 3, 5 ]; "
"video/x-xvid, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], "
"framerate = (fraction) [ 0, MAX ]; "
"video/x-3ivx, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], "
"framerate = (fraction) [ 0, MAX ]; "
"video/x-msmpeg, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], "
"framerate = (fraction) [ 0, MAX ], "
"msmpegversion = (int) [ 41, 43 ]; "
"video/mpeg, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], "
"framerate = (fraction) [ 0, MAX ], "
"mpegversion = (int) 1, "
"systemstream = (boolean) FALSE; "
"video/x-h263, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], "
"framerate = (fraction) [ 0, MAX ]; "
"video/x-h264, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], "
"framerate = (fraction) [ 0, MAX ]; "
"video/x-dv, "
"width = (int) 720, "
"height = (int) { 576, 480 }, "
"framerate = (fraction) [ 0, MAX ], "
"systemstream = (boolean) FALSE; "
"video/x-huffyuv, "
"width = (int) [ 16, 4096 ], "
"height = (int) [ 16, 4096 ], " "framerate = (fraction) [ 0, MAX ]")
);
static GstStaticPadTemplate audio_sink_factory =
GST_STATIC_PAD_TEMPLATE ("audio_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) LITTLE_ENDIAN, "
"signed = (boolean) { TRUE, FALSE }, "
"width = (int) { 8, 16 }, "
"depth = (int) { 8, 16 }, "
"rate = (int) [ 1000, 96000 ], "
"channels = (int) [ 1, 2 ]; "
"audio/mpeg, "
"mpegversion = (int) 1, "
"layer = (int) [ 1, 3 ], "
"rate = (int) [ 1000, 96000 ], "
"channels = (int) [ 1, 2 ]; "
"audio/x-vorbis, "
"rate = (int) [ 1000, 96000 ], "
"channels = (int) [ 1, 2 ]; "
"audio/x-ac3, "
"rate = (int) [ 1000, 96000 ], " "channels = (int) [ 1, 2 ]")
);
static void gst_avi_mux_base_init (gpointer g_class);
static void gst_avi_mux_class_init (GstAviMuxClass * klass);
static void gst_avi_mux_init (GstAviMux * avimux);
static GstFlowReturn gst_avi_mux_collect_pads (GstCollectPads * pads,
GstAviMux * avimux);
static gboolean gst_avi_mux_handle_event (GstPad * pad, GstEvent * event);
static GstPad *gst_avi_mux_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name);
static void gst_avi_mux_release_pad (GstElement * element, GstPad * pad);
static void gst_avi_mux_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_avi_mux_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_avi_mux_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
GType
gst_avi_mux_get_type (void)
{
static GType avimux_type = 0;
if (!avimux_type) {
static const GTypeInfo avimux_info = {
sizeof (GstAviMuxClass),
gst_avi_mux_base_init,
NULL,
(GClassInitFunc) gst_avi_mux_class_init,
NULL,
NULL,
sizeof (GstAviMux),
0,
(GInstanceInitFunc) gst_avi_mux_init,
};
static const GInterfaceInfo tag_setter_info = {
NULL,
NULL,
NULL
};
avimux_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstAviMux", &avimux_info, 0);
g_type_add_interface_static (avimux_type, GST_TYPE_TAG_SETTER,
&tag_setter_info);
}
return avimux_type;
}
static void
gst_avi_mux_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&audio_sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&video_sink_factory));
gst_element_class_set_details (element_class, &gst_avi_mux_details);
GST_DEBUG_CATEGORY_INIT (avimux_debug, "avimux", 0, "Muxer for AVI streams");
}
static void
gst_avi_mux_finalize (GObject * object)
{
GstAviMux *mux = GST_AVI_MUX (object);
g_free (mux->idx);
mux->idx = NULL;
g_free (mux->vids_idx);
mux->vids_idx = NULL;
g_free (mux->auds_idx);
mux->auds_idx = NULL;
gst_object_unref (mux->collect);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_avi_mux_class_init (GstAviMuxClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/aalib/gstaasink.c: (gst_aasink_class_init): * ext/esd/esdsink.c: (gst_esdsink_class_init): * ext/flac/gstflactag.c: (gst_flac_tag_class_init): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init): * ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init): * ext/libcaca/gstcacasink.c: (gst_cacasink_class_init): * ext/libmng/gstmngdec.c: (gst_mngdec_class_init): * ext/libmng/gstmngenc.c: (gst_mngenc_class_init): * ext/libpng/gstpngdec.c: (gst_pngdec_class_init): * ext/libpng/gstpngenc.c: (gst_pngenc_class_init): * ext/mikmod/gstmikmod.c: (gst_mikmod_class_init): * ext/shout2/gstshout2.c: (gst_shout2send_class_init): * ext/speex/gstspeexenc.c: (gst_speexenc_class_init): * gst/alpha/gstalpha.c: (gst_alpha_class_init): * gst/avi/gstavimux.c: (gst_avimux_class_init): * gst/debug/efence.c: (gst_efence_class_init): * gst/debug/negotiation.c: (gst_negotiation_class_init): * gst/flx/gstflxdec.c: (gst_flxdec_class_init): * gst/goom/gstgoom.c: (gst_goom_class_init): * gst/id3demux/gstid3demux.c: (gst_id3demux_class_init): * gst/interleave/deinterleave.c: (deinterleave_class_init): * gst/interleave/interleave.c: (interleave_class_init): * gst/law/alaw-decode.c: (gst_alawdec_class_init): * gst/law/alaw-encode.c: (gst_alawenc_class_init): * gst/law/mulaw-encode.c: (gst_mulawenc_class_init): * gst/median/gstmedian.c: (gst_median_class_init): * gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init): * gst/multipart/multipartmux.c: (gst_multipart_mux_class_init): * gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init): * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init): * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init): * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init): * gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init): * gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init): * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init): * gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init): * gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init): * gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init): * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init): * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init): * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init): * gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init): * gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init): * gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init): * gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init): * gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init): * gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init): * gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init): * gst/smpte/gstsmpte.c: (gst_smpte_class_init): * gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init): * gst/udp/gstudpsink.c: (gst_udpsink_class_init): * gst/videomixer/videomixer.c: (gst_videomixer_class_init): * gst/wavenc/gstwavenc.c: (gst_wavenc_class_init): * sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init): * sys/oss/gstosssink.c: (gst_oss_sink_class_init): * sys/osxaudio/gstosxaudioelement.c: (gst_osxaudioelement_class_init): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init): * sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:21:45 +00:00
parent_class = g_type_class_peek_parent (klass);
gobject_class->get_property = gst_avi_mux_get_property;
gobject_class->set_property = gst_avi_mux_set_property;
gobject_class->finalize = gst_avi_mux_finalize;
g_object_class_install_property (gobject_class, ARG_BIGFILE,
g_param_spec_boolean ("bigfile", "Bigfile Support (>2GB)",
"Support for openDML-2.0 (big) AVI files", DEFAULT_BIGFILE,
G_PARAM_READWRITE));
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_avi_mux_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_avi_mux_release_pad);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_avi_mux_change_state);
}
static void
gst_avi_mux_init (GstAviMux * avimux)
{
avimux->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_pad_use_fixed_caps (avimux->srcpad);
gst_element_add_pad (GST_ELEMENT (avimux), avimux->srcpad);
avimux->audiocollectdata = NULL;
avimux->audio_pad_connected = FALSE;
avimux->videocollectdata = NULL;
avimux->video_pad_connected = FALSE;
avimux->num_frames = 0;
/* audio/video/AVI header initialisation */
memset (&(avimux->avi_hdr), 0, sizeof (gst_riff_avih));
memset (&(avimux->vids_hdr), 0, sizeof (gst_riff_strh));
memset (&(avimux->vids), 0, sizeof (gst_riff_strf_vids));
memset (&(avimux->auds_hdr), 0, sizeof (gst_riff_strh));
memset (&(avimux->auds), 0, sizeof (gst_riff_strf_auds));
avimux->vids_hdr.type = GST_MAKE_FOURCC ('v', 'i', 'd', 's');
avimux->vids_hdr.rate = 1;
avimux->avi_hdr.max_bps = 10000000;
avimux->auds_hdr.type = GST_MAKE_FOURCC ('a', 'u', 'd', 's');
avimux->vids_hdr.quality = 0xFFFFFFFF;
avimux->auds_hdr.quality = 0xFFFFFFFF;
avimux->vids_codec_data = NULL;
avimux->tags = NULL;
avimux->tags_snap = NULL;
avimux->idx = NULL;
avimux->vids_idx = g_new (gst_avi_superindex_entry, GST_AVI_SUPERINDEX_COUNT);
avimux->auds_idx = g_new (gst_avi_superindex_entry, GST_AVI_SUPERINDEX_COUNT);
avimux->write_header = TRUE;
avimux->enable_large_avi = DEFAULT_BIGFILE;
avimux->collect = gst_collect_pads_new ();
gst_collect_pads_set_function (avimux->collect,
(GstCollectPadsFunction) (GST_DEBUG_FUNCPTR (gst_avi_mux_collect_pads)),
avimux);
}
static gboolean
gst_avi_mux_vidsink_set_caps (GstPad * pad, GstCaps * vscaps)
{
GstAviMux *avimux;
GstStructure *structure;
const gchar *mimetype;
const GValue *fps;
const GValue *codec_data;
gint width, height;
avimux = GST_AVI_MUX (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (avimux, "%s:%s, caps=%" GST_PTR_FORMAT,
GST_DEBUG_PAD_NAME (pad), vscaps);
structure = gst_caps_get_structure (vscaps, 0);
mimetype = gst_structure_get_name (structure);
/* global */
avimux->vids.size = sizeof (gst_riff_strf_vids);
avimux->vids.planes = 1;
if (!gst_structure_get_int (structure, "width", &width) ||
!gst_structure_get_int (structure, "height", &height)) {
goto refuse_caps;
close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps): * ext/esd/esdmon.c: (gst_esdmon_get): * ext/flac/gstflactag.c: (gst_flac_tag_chain): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_sink_getcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps), (gst_jpegenc_setcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps): * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink), (gst_mngdec_src_getcaps): * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink), (gst_mngenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink): * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_src_event), (speex_dec_chain): * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect), (gst_avimux_audsinkconnect), (gst_avimux_handle_event): * gst/debug/negotiation.c: (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain): * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/interleave/deinterleave.c: (deinterleave_sink_link), (deinterleave_chain): * gst/law/mulaw-encode.c: (mulawenc_setcaps): * gst/median/gstmedian.c: (gst_median_link): * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect): * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get): close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
2006-03-13 15:49:08 +00:00
}
avimux->vids.width = width;
avimux->vids.height = height;
fps = gst_structure_get_value (structure, "framerate");
if (fps == NULL || !GST_VALUE_HOLDS_FRACTION (fps))
goto refuse_caps;
avimux->vids_hdr.rate = gst_value_get_fraction_numerator (fps);
avimux->vids_hdr.scale = gst_value_get_fraction_denominator (fps);
/* codec initialization data, if any */
codec_data = gst_structure_get_value (structure, "codec_data");
if (codec_data) {
avimux->vids_codec_data = gst_value_get_buffer (codec_data);
gst_buffer_ref (avimux->vids_codec_data);
}
if (!strcmp (mimetype, "video/x-raw-yuv")) {
guint32 format;
gst_structure_get_fourcc (structure, "format", &format);
avimux->vids.compression = format;
switch (format) {
case GST_MAKE_FOURCC ('Y', 'U', 'Y', '2'):
avimux->vids.bit_cnt = 16;
break;
case GST_MAKE_FOURCC ('I', '4', '2', '0'):
avimux->vids.bit_cnt = 12;
break;
}
} else {
avimux->vids.bit_cnt = 24;
avimux->vids.compression = 0;
/* find format */
if (!strcmp (mimetype, "video/x-huffyuv")) {
avimux->vids.compression = GST_MAKE_FOURCC ('H', 'F', 'Y', 'U');
} else if (!strcmp (mimetype, "image/jpeg")) {
avimux->vids.compression = GST_MAKE_FOURCC ('M', 'J', 'P', 'G');
} else if (!strcmp (mimetype, "video/x-divx")) {
gint divxversion;
gst_structure_get_int (structure, "divxversion", &divxversion);
switch (divxversion) {
case 3:
avimux->vids.compression = GST_MAKE_FOURCC ('D', 'I', 'V', '3');
break;
case 4:
avimux->vids.compression = GST_MAKE_FOURCC ('D', 'I', 'V', 'X');
break;
case 5:
avimux->vids.compression = GST_MAKE_FOURCC ('D', 'X', '5', '0');
break;
}
} else if (!strcmp (mimetype, "video/x-xvid")) {
avimux->vids.compression = GST_MAKE_FOURCC ('X', 'V', 'I', 'D');
} else if (!strcmp (mimetype, "video/x-3ivx")) {
avimux->vids.compression = GST_MAKE_FOURCC ('3', 'I', 'V', '2');
} else if (gst_structure_has_name (structure, "video/x-msmpeg")) {
gint msmpegversion;
gst_structure_get_int (structure, "msmpegversion", &msmpegversion);
switch (msmpegversion) {
case 41:
avimux->vids.compression = GST_MAKE_FOURCC ('M', 'P', 'G', '4');
break;
case 42:
avimux->vids.compression = GST_MAKE_FOURCC ('M', 'P', '4', '2');
break;
case 43:
avimux->vids.compression = GST_MAKE_FOURCC ('M', 'P', '4', '3');
break;
}
} else if (!strcmp (mimetype, "video/x-dv")) {
avimux->vids.compression = GST_MAKE_FOURCC ('D', 'V', 'S', 'D');
} else if (!strcmp (mimetype, "video/x-h263")) {
avimux->vids.compression = GST_MAKE_FOURCC ('H', '2', '6', '3');
} else if (!strcmp (mimetype, "video/mpeg")) {
avimux->vids.compression = GST_MAKE_FOURCC ('M', 'P', 'E', 'G');
}
if (!avimux->vids.compression)
goto refuse_caps;
}
avimux->vids_hdr.fcc_handler = avimux->vids.compression;
avimux->vids.image_size = avimux->vids.height * avimux->vids.width;
avimux->avi_hdr.width = avimux->vids.width;
avimux->avi_hdr.height = avimux->vids.height;
avimux->avi_hdr.us_frame = 1000000. * avimux->vids_hdr.scale /
avimux->vids_hdr.rate;
close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps): * ext/esd/esdmon.c: (gst_esdmon_get): * ext/flac/gstflactag.c: (gst_flac_tag_chain): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_sink_getcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps), (gst_jpegenc_setcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps): * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink), (gst_mngdec_src_getcaps): * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink), (gst_mngenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink): * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_src_event), (speex_dec_chain): * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect), (gst_avimux_audsinkconnect), (gst_avimux_handle_event): * gst/debug/negotiation.c: (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain): * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/interleave/deinterleave.c: (deinterleave_sink_link), (deinterleave_chain): * gst/law/mulaw-encode.c: (mulawenc_setcaps): * gst/median/gstmedian.c: (gst_median_link): * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect): * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get): close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
2006-03-13 15:49:08 +00:00
gst_object_unref (avimux);
return TRUE;
close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps): * ext/esd/esdmon.c: (gst_esdmon_get): * ext/flac/gstflactag.c: (gst_flac_tag_chain): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_sink_getcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps), (gst_jpegenc_setcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps): * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink), (gst_mngdec_src_getcaps): * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink), (gst_mngenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink): * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_src_event), (speex_dec_chain): * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect), (gst_avimux_audsinkconnect), (gst_avimux_handle_event): * gst/debug/negotiation.c: (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain): * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/interleave/deinterleave.c: (deinterleave_sink_link), (deinterleave_chain): * gst/law/mulaw-encode.c: (mulawenc_setcaps): * gst/median/gstmedian.c: (gst_median_link): * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect): * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get): close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
2006-03-13 15:49:08 +00:00
refuse_caps:
{
GST_WARNING_OBJECT (avimux, "refused caps %" GST_PTR_FORMAT, vscaps);
gst_object_unref (avimux);
return FALSE;
}
}
static gboolean
gst_avi_mux_audsink_set_caps (GstPad * pad, GstCaps * vscaps)
{
GstAviMux *avimux;
GstStructure *structure;
const gchar *mimetype;
gint channels, rate;
avimux = GST_AVI_MUX (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (avimux, "%s:%s, caps=%" GST_PTR_FORMAT,
GST_DEBUG_PAD_NAME (pad), vscaps);
structure = gst_caps_get_structure (vscaps, 0);
mimetype = gst_structure_get_name (structure);
/* we want these for all */
if (!gst_structure_get_int (structure, "channels", &channels) ||
!gst_structure_get_int (structure, "rate", &rate)) {
goto refuse_caps;
}
avimux->auds.channels = channels;
avimux->auds.rate = rate;
if (!strcmp (mimetype, "audio/x-raw-int")) {
gint width, depth;
avimux->auds.format = GST_RIFF_WAVE_FORMAT_PCM;
if (!gst_structure_get_int (structure, "width", &width) ||
(width != 8 && !gst_structure_get_int (structure, "depth", &depth))) {
goto refuse_caps;
}
avimux->auds.blockalign = width;
avimux->auds.size = (width == 8) ? 8 : depth;
/* set some more info straight */
avimux->auds.blockalign /= 8;
avimux->auds.blockalign *= avimux->auds.channels;
avimux->auds.av_bps = avimux->auds.blockalign * avimux->auds.rate;
} else if (!strcmp (mimetype, "audio/mpeg") ||
!strcmp (mimetype, "audio/x-vorbis") ||
!strcmp (mimetype, "audio/x-ac3")) {
avimux->auds.format = 0;
if (!strcmp (mimetype, "audio/mpeg")) {
gint layer = 3;
gst_structure_get_int (structure, "layer", &layer);
switch (layer) {
case 3:
avimux->auds.format = GST_RIFF_WAVE_FORMAT_MPEGL3;
break;
case 1:
case 2:
avimux->auds.format = GST_RIFF_WAVE_FORMAT_MPEGL12;
break;
}
} else if (!strcmp (mimetype, "audio/x-vorbis")) {
avimux->auds.format = GST_RIFF_WAVE_FORMAT_VORBIS3;
} else if (!strcmp (mimetype, "audio/x-ac3")) {
avimux->auds.format = GST_RIFF_WAVE_FORMAT_A52;
}
avimux->auds.blockalign = 1;
avimux->auds.av_bps = 0;
avimux->auds.size = 16;
if (!avimux->auds.format)
goto refuse_caps;
}
avimux->auds_hdr.rate = avimux->auds.blockalign * avimux->auds.rate;
avimux->auds_hdr.samplesize = avimux->auds.blockalign;
avimux->auds_hdr.scale = 1;
close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps): * ext/esd/esdmon.c: (gst_esdmon_get): * ext/flac/gstflactag.c: (gst_flac_tag_chain): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_sink_getcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps), (gst_jpegenc_setcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps): * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink), (gst_mngdec_src_getcaps): * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink), (gst_mngenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink): * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_src_event), (speex_dec_chain): * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect), (gst_avimux_audsinkconnect), (gst_avimux_handle_event): * gst/debug/negotiation.c: (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain): * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/interleave/deinterleave.c: (deinterleave_sink_link), (deinterleave_chain): * gst/law/mulaw-encode.c: (mulawenc_setcaps): * gst/median/gstmedian.c: (gst_median_link): * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect): * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get): close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
2006-03-13 15:49:08 +00:00
gst_object_unref (avimux);
return TRUE;
close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps): * ext/esd/esdmon.c: (gst_esdmon_get): * ext/flac/gstflactag.c: (gst_flac_tag_chain): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_sink_getcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps), (gst_jpegenc_setcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps): * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink), (gst_mngdec_src_getcaps): * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink), (gst_mngenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink): * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_src_event), (speex_dec_chain): * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect), (gst_avimux_audsinkconnect), (gst_avimux_handle_event): * gst/debug/negotiation.c: (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain): * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/interleave/deinterleave.c: (deinterleave_sink_link), (deinterleave_chain): * gst/law/mulaw-encode.c: (mulawenc_setcaps): * gst/median/gstmedian.c: (gst_median_link): * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect): * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get): close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
2006-03-13 15:49:08 +00:00
refuse_caps:
{
GST_WARNING_OBJECT (avimux, "refused caps %" GST_PTR_FORMAT, vscaps);
gst_object_unref (avimux);
return FALSE;
}
}
static void
gst_avi_mux_pad_link (GstPad * pad, GstPad * peer, gpointer data)
{
GstAviMux *avimux = GST_AVI_MUX (data);
if (avimux->audiocollectdata && pad == avimux->audiocollectdata->pad) {
avimux->audio_pad_connected = TRUE;
} else if (avimux->videocollectdata && pad == avimux->videocollectdata->pad) {
avimux->video_pad_connected = TRUE;
} else {
g_assert_not_reached ();
}
GST_DEBUG_OBJECT (avimux, "pad '%s' connected", GST_PAD_NAME (pad));
}
static void
gst_avi_mux_pad_unlink (GstPad * pad, GstPad * peer, gpointer data)
{
GstAviMux *avimux = GST_AVI_MUX (data);
if (avimux->audiocollectdata && pad == avimux->audiocollectdata->pad) {
avimux->audio_pad_connected = FALSE;
avimux->audiocollectdata = NULL;
} else if (avimux->videocollectdata && pad == avimux->videocollectdata->pad) {
avimux->video_pad_connected = FALSE;
avimux->videocollectdata = NULL;
} else {
g_assert_not_reached ();
}
gst_collect_pads_remove_pad (avimux->collect, pad);
GST_DEBUG_OBJECT (avimux, "pad '%s' unlinked and removed from collect",
GST_PAD_NAME (pad));
}
/* TODO GstCollectPads will block if it has to manage a non-linked pad;
* best to upgrade it so it helps all muxers using it */
static GstPad *
gst_avi_mux_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * req_name)
{
GstAviMux *avimux;
GstPad *newpad;
GstElementClass *klass = GST_ELEMENT_GET_CLASS (element);
g_return_val_if_fail (templ != NULL, NULL);
if (templ->direction != GST_PAD_SINK) {
g_warning ("avimux: request pad that is not a SINK pad\n");
return NULL;
}
g_return_val_if_fail (GST_IS_AVI_MUX (element), NULL);
avimux = GST_AVI_MUX (element);
if (templ == gst_element_class_get_pad_template (klass, "audio_%d")) {
if (avimux->audiocollectdata)
return NULL;
newpad = gst_pad_new_from_template (templ, "audio_00");
gst_pad_set_setcaps_function (newpad,
GST_DEBUG_FUNCPTR (gst_avi_mux_audsink_set_caps));
avimux->audiocollectdata = gst_collect_pads_add_pad (avimux->collect,
newpad, sizeof (GstCollectData));
} else if (templ == gst_element_class_get_pad_template (klass, "video_%d")) {
if (avimux->videocollectdata)
return NULL;
newpad = gst_pad_new_from_template (templ, "video_00");
gst_pad_set_setcaps_function (newpad,
GST_DEBUG_FUNCPTR (gst_avi_mux_vidsink_set_caps));
avimux->videocollectdata = gst_collect_pads_add_pad (avimux->collect,
newpad, sizeof (GstCollectData));
} else {
g_warning ("avimux: this is not our template!\n");
return NULL;
}
/* FIXME: hacked way to override/extend the event function of
* GstCollectPads; because it sets its own event function giving the
* element no access to events */
avimux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad);
gst_pad_set_event_function (newpad,
GST_DEBUG_FUNCPTR (gst_avi_mux_handle_event));
g_signal_connect (newpad, "linked",
G_CALLBACK (gst_avi_mux_pad_link), avimux);
g_signal_connect (newpad, "unlinked",
G_CALLBACK (gst_avi_mux_pad_unlink), avimux);
gst_element_add_pad (element, newpad);
return newpad;
}
static void
gst_avi_mux_release_pad (GstElement * element, GstPad * pad)
{
GstAviMux *avimux = GST_AVI_MUX (element);
if (avimux->videocollectdata && pad == avimux->videocollectdata->pad) {
avimux->videocollectdata = NULL;
} else if (avimux->audiocollectdata && pad == avimux->audiocollectdata->pad) {
avimux->audiocollectdata = NULL;
} else {
g_warning ("Unknown pad %s", GST_PAD_NAME (pad));
return;
}
GST_DEBUG_OBJECT (avimux, "removed pad '%s'", GST_PAD_NAME (pad));
gst_collect_pads_remove_pad (avimux->collect, pad);
gst_element_remove_pad (element, pad);
}
/* maybe some of these functions should be moved to riff.h? */
/* DISCLAIMER: this function is ugly. So be it (i.e. it makes the rest easier) */
/* so is this struct */
typedef struct _GstMarkedBuffer
{
guint *highmark;
GstBuffer *buffer;
} GstMarkedBuffer;
static void
gst_avi_mux_write_tag (const GstTagList * list, const gchar * tag,
gpointer data)
{
const struct
{
guint32 fcc;
gchar *tag;
} rifftags[] = {
{
GST_RIFF_INFO_ICMT, GST_TAG_COMMENT}, {
GST_RIFF_INFO_INAM, GST_TAG_TITLE}, {
GST_RIFF_INFO_ISFT, GST_TAG_ENCODER}, {
GST_RIFF_INFO_IGNR, GST_TAG_GENRE}, {
GST_RIFF_INFO_ICOP, GST_TAG_COPYRIGHT}, {
GST_RIFF_INFO_IART, GST_TAG_ARTIST}, {
GST_RIFF_INFO_IARL, GST_TAG_LOCATION}, {
0, NULL}
};
gint n, len, plen;
GstBuffer *buf = ((GstMarkedBuffer *) data)->buffer;
guint *highmark = ((GstMarkedBuffer *) data)->highmark;
guint8 *buffdata = GST_BUFFER_DATA (buf) + *highmark;
gchar *str;
for (n = 0; rifftags[n].fcc != 0; n++) {
if (!strcmp (rifftags[n].tag, tag) &&
gst_tag_list_get_string (list, tag, &str)) {
len = strlen (str);
plen = len + 1;
if (plen & 1)
plen++;
if (GST_BUFFER_SIZE (buf) >= *highmark + 8 + plen) {
GST_WRITE_UINT32_LE (buffdata, rifftags[n].fcc);
GST_WRITE_UINT32_LE (buffdata + 4, len + 1);
memcpy (buffdata + 8, str, len);
buffdata[8 + len] = 0;
*highmark += 8 + plen;
GST_DEBUG ("writing tag in buffer %p, highmark at %d", buf, *highmark);
}
g_free (str);
break;
}
}
}
#define ODML_SUPERINDEX_SIZE \
(32 + GST_AVI_SUPERINDEX_COUNT * sizeof (gst_avi_superindex_entry))
static GstBuffer *
gst_avi_mux_riff_get_avi_header (GstAviMux * avimux)
{
GstTagList *tags;
const GstTagList *iface_tags;
GstBuffer *buffer;
guint8 *buffdata;
guint size = 0;
guint codec_size = 0;
guint highmark = 0;
/* do we have some extra data for the codec */
if (avimux->vids_codec_data)
codec_size = GST_BUFFER_SIZE (avimux->vids_codec_data);
/* first, let's see what actually needs to be in the buffer */
size += 32 + sizeof (gst_riff_avih); /* avi header */
if (avimux->video_pad_connected) { /* we have video */
size += 28 + sizeof (gst_riff_strh) + sizeof (gst_riff_strf_vids); /* vid hdr */
size += codec_size; /* codec data */
size += 24; /* odml header */
size += ODML_SUPERINDEX_SIZE; /* vids superindex */
}
if (avimux->audio_pad_connected) { /* we have audio */
size += 28 + sizeof (gst_riff_strh) + sizeof (gst_riff_strf_auds); /* aud hdr */
size += ODML_SUPERINDEX_SIZE; /* auds superindex */
}
/* this is the "riff size" */
avimux->header_size = size;
size += 12; /* avi data header */
GST_DEBUG ("creating avi header, header_size %u, data_size %u, idx_size %u",
avimux->header_size, avimux->data_size, avimux->idx_size);
/* tags */
iface_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (avimux));
if ((iface_tags || avimux->tags) && !avimux->tags_snap) {
if (iface_tags && avimux->tags) {
tags = gst_tag_list_merge (iface_tags, avimux->tags,
GST_TAG_MERGE_APPEND);
} else if (iface_tags) {
tags = gst_tag_list_copy (iface_tags);
} else {
tags = gst_tag_list_copy (avimux->tags);
}
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_ENCODER,
PACKAGE_STRING " AVI muxer", NULL);
} else {
tags = avimux->tags_snap;
}
avimux->tags_snap = tags;
if (avimux->tags_snap)
size += 1024;
/* allocate the buffer */
buffer = gst_buffer_new_and_alloc (size);
buffdata = GST_BUFFER_DATA (buffer);
highmark = 0;
GST_DEBUG ("creating buffer %p, size %d, highmark at 0",
buffer, GST_BUFFER_SIZE (buffer));
/* avi header metadata */
memcpy (buffdata + 0, "RIFF", 4);
GST_WRITE_UINT32_LE (buffdata + 4,
avimux->header_size + avimux->idx_size + avimux->data_size +
avimux->tag_size);
memcpy (buffdata + 8, "AVI ", 4);
memcpy (buffdata + 12, "LIST", 4);
GST_WRITE_UINT32_LE (buffdata + 16, avimux->header_size - 4 * 5);
memcpy (buffdata + 20, "hdrl", 4);
memcpy (buffdata + 24, "avih", 4);
GST_WRITE_UINT32_LE (buffdata + 28, sizeof (gst_riff_avih));
buffdata += 32;
highmark += 32;
/* the AVI header itself */
GST_WRITE_UINT32_LE (buffdata + 0, avimux->avi_hdr.us_frame);
GST_WRITE_UINT32_LE (buffdata + 4, avimux->avi_hdr.max_bps);
GST_WRITE_UINT32_LE (buffdata + 8, avimux->avi_hdr.pad_gran);
GST_WRITE_UINT32_LE (buffdata + 12, avimux->avi_hdr.flags);
GST_WRITE_UINT32_LE (buffdata + 16, avimux->avi_hdr.tot_frames);
GST_WRITE_UINT32_LE (buffdata + 20, avimux->avi_hdr.init_frames);
GST_WRITE_UINT32_LE (buffdata + 24, avimux->avi_hdr.streams);
GST_WRITE_UINT32_LE (buffdata + 28, avimux->avi_hdr.bufsize);
GST_WRITE_UINT32_LE (buffdata + 32, avimux->avi_hdr.width);
GST_WRITE_UINT32_LE (buffdata + 36, avimux->avi_hdr.height);
GST_WRITE_UINT32_LE (buffdata + 40, avimux->avi_hdr.scale);
GST_WRITE_UINT32_LE (buffdata + 44, avimux->avi_hdr.rate);
GST_WRITE_UINT32_LE (buffdata + 48, avimux->avi_hdr.start);
GST_WRITE_UINT32_LE (buffdata + 52, avimux->avi_hdr.length);
buffdata += 56;
highmark += 56;
if (avimux->video_pad_connected) {
/* video header metadata */
memcpy (buffdata + 0, "LIST", 4);
GST_WRITE_UINT32_LE (buffdata + 4,
sizeof (gst_riff_strh) + sizeof (gst_riff_strf_vids)
+ codec_size + 4 * 5 + ODML_SUPERINDEX_SIZE);
memcpy (buffdata + 8, "strl", 4);
/* generic header */
memcpy (buffdata + 12, "strh", 4);
GST_WRITE_UINT32_LE (buffdata + 16, sizeof (gst_riff_strh));
/* the actual header */
GST_WRITE_UINT32_LE (buffdata + 20, avimux->vids_hdr.type);
GST_WRITE_UINT32_LE (buffdata + 24, avimux->vids_hdr.fcc_handler);
GST_WRITE_UINT32_LE (buffdata + 28, avimux->vids_hdr.flags);
GST_WRITE_UINT32_LE (buffdata + 32, avimux->vids_hdr.priority);
GST_WRITE_UINT32_LE (buffdata + 36, avimux->vids_hdr.init_frames);
GST_WRITE_UINT32_LE (buffdata + 40, avimux->vids_hdr.scale);
GST_WRITE_UINT32_LE (buffdata + 44, avimux->vids_hdr.rate);
GST_WRITE_UINT32_LE (buffdata + 48, avimux->vids_hdr.start);
GST_WRITE_UINT32_LE (buffdata + 52, avimux->vids_hdr.length);
GST_WRITE_UINT32_LE (buffdata + 56, avimux->vids_hdr.bufsize);
GST_WRITE_UINT32_LE (buffdata + 60, avimux->vids_hdr.quality);
GST_WRITE_UINT32_LE (buffdata + 64, avimux->vids_hdr.samplesize);
/* the video header */
memcpy (buffdata + 68, "strf", 4);
GST_WRITE_UINT32_LE (buffdata + 72,
sizeof (gst_riff_strf_vids) + codec_size);
/* the actual header */
GST_WRITE_UINT32_LE (buffdata + 76, avimux->vids.size + codec_size);
GST_WRITE_UINT32_LE (buffdata + 80, avimux->vids.width);
GST_WRITE_UINT32_LE (buffdata + 84, avimux->vids.height);
GST_WRITE_UINT16_LE (buffdata + 88, avimux->vids.planes);
GST_WRITE_UINT16_LE (buffdata + 90, avimux->vids.bit_cnt);
GST_WRITE_UINT32_LE (buffdata + 92, avimux->vids.compression);
GST_WRITE_UINT32_LE (buffdata + 96, avimux->vids.image_size);
GST_WRITE_UINT32_LE (buffdata + 100, avimux->vids.xpels_meter);
GST_WRITE_UINT32_LE (buffdata + 104, avimux->vids.ypels_meter);
GST_WRITE_UINT32_LE (buffdata + 108, avimux->vids.num_colors);
GST_WRITE_UINT32_LE (buffdata + 112, avimux->vids.imp_colors);
buffdata += 116;
highmark += 116;
/* include codec data, if any */
if (codec_size) {
memcpy (buffdata, GST_BUFFER_DATA (avimux->vids_codec_data), codec_size);
buffdata += codec_size;
highmark += codec_size;
}
/* odml superindex chunk */
if (avimux->vids_idx_index > 0)
memcpy (buffdata, "indx", 4);
else
memcpy (buffdata, "JUNK", 4);
GST_WRITE_UINT32_LE (buffdata + 4, ODML_SUPERINDEX_SIZE - 8); /* chunk size */
GST_WRITE_UINT16_LE (buffdata + 8, 4); /* bytes per entry */
buffdata[10] = 0; /* index subtype */
buffdata[11] = GST_AVI_INDEX_OF_INDEXES; /* index type */
GST_WRITE_UINT32_LE (buffdata + 12, avimux->vids_idx_index); /* entries in use */
memcpy (buffdata + 16, "00db", 4); /* stream id */
GST_WRITE_UINT32_LE (buffdata + 20, 0); /* reserved */
GST_WRITE_UINT32_LE (buffdata + 24, 0); /* reserved */
GST_WRITE_UINT32_LE (buffdata + 28, 0); /* reserved */
memcpy (buffdata + 32, avimux->vids_idx,
GST_AVI_SUPERINDEX_COUNT * sizeof (gst_avi_superindex_entry));
buffdata += ODML_SUPERINDEX_SIZE;
highmark += ODML_SUPERINDEX_SIZE;
}
if (avimux->audio_pad_connected) {
/* audio header */
memcpy (buffdata + 0, "LIST", 4);
GST_WRITE_UINT32_LE (buffdata + 4,
sizeof (gst_riff_strh) + sizeof (gst_riff_strf_auds) + 4 * 5
+ ODML_SUPERINDEX_SIZE);
memcpy (buffdata + 8, "strl", 4);
/* generic header */
memcpy (buffdata + 12, "strh", 4);
GST_WRITE_UINT32_LE (buffdata + 16, sizeof (gst_riff_strh));
/* the actual header */
GST_WRITE_UINT32_LE (buffdata + 20, avimux->auds_hdr.type);
GST_WRITE_UINT32_LE (buffdata + 24, avimux->auds_hdr.fcc_handler);
GST_WRITE_UINT32_LE (buffdata + 28, avimux->auds_hdr.flags);
GST_WRITE_UINT32_LE (buffdata + 32, avimux->auds_hdr.priority);
GST_WRITE_UINT32_LE (buffdata + 36, avimux->auds_hdr.init_frames);
GST_WRITE_UINT32_LE (buffdata + 40, avimux->auds_hdr.scale);
GST_WRITE_UINT32_LE (buffdata + 44, avimux->auds_hdr.rate);
GST_WRITE_UINT32_LE (buffdata + 48, avimux->auds_hdr.start);
GST_WRITE_UINT32_LE (buffdata + 52, avimux->auds_hdr.length);
GST_WRITE_UINT32_LE (buffdata + 56, avimux->auds_hdr.bufsize);
GST_WRITE_UINT32_LE (buffdata + 60, avimux->auds_hdr.quality);
GST_WRITE_UINT32_LE (buffdata + 64, avimux->auds_hdr.samplesize);
/* the audio header */
memcpy (buffdata + 68, "strf", 4);
GST_WRITE_UINT32_LE (buffdata + 72, sizeof (gst_riff_strf_auds));
/* the actual header */
GST_WRITE_UINT16_LE (buffdata + 76, avimux->auds.format);
GST_WRITE_UINT16_LE (buffdata + 78, avimux->auds.channels);
GST_WRITE_UINT32_LE (buffdata + 80, avimux->auds.rate);
GST_WRITE_UINT32_LE (buffdata + 84, avimux->auds.av_bps);
GST_WRITE_UINT16_LE (buffdata + 88, avimux->auds.blockalign);
GST_WRITE_UINT16_LE (buffdata + 90, avimux->auds.size);
buffdata += 92;
highmark += 92;
/* odml superindex chunk */
if (avimux->auds_idx_index > 0)
memcpy (buffdata, "indx", 4);
else
memcpy (buffdata, "JUNK", 4);
GST_WRITE_UINT32_LE (buffdata + 4, ODML_SUPERINDEX_SIZE - 8); /* chunk size */
GST_WRITE_UINT16_LE (buffdata + 8, 4); /* bytes per entry */
buffdata[10] = 0; /* index subtype */
buffdata[11] = GST_AVI_INDEX_OF_INDEXES; /* index type */
GST_WRITE_UINT32_LE (buffdata + 12, avimux->auds_idx_index); /* entries in use */
memcpy (buffdata + 16, "01wb", 4); /* stream id */
GST_WRITE_UINT32_LE (buffdata + 20, 0); /* reserved */
GST_WRITE_UINT32_LE (buffdata + 24, 0); /* reserved */
GST_WRITE_UINT32_LE (buffdata + 28, 0); /* reserved */
memcpy (buffdata + 32, avimux->auds_idx,
GST_AVI_SUPERINDEX_COUNT * sizeof (gst_avi_superindex_entry));
buffdata += ODML_SUPERINDEX_SIZE;
highmark += ODML_SUPERINDEX_SIZE;
}
if (avimux->video_pad_connected) {
/* odml header */
memcpy (buffdata + 0, "LIST", 4);
GST_WRITE_UINT32_LE (buffdata + 4, sizeof (guint32) + 4 * 3);
memcpy (buffdata + 8, "odml", 4);
memcpy (buffdata + 12, "dmlh", 4);
GST_WRITE_UINT32_LE (buffdata + 16, sizeof (guint32));
GST_WRITE_UINT32_LE (buffdata + 20, avimux->total_frames);
buffdata += 24;
highmark += 24;
}
/* tags */
if (tags) {
guint8 *ptr;
guint startsize;
GstMarkedBuffer data = { &highmark, buffer };
memcpy (buffdata + 0, "LIST", 4);
ptr = buffdata + 4; /* fill in later */
startsize = highmark + 4;
memcpy (buffdata + 8, "INFO", 4);
buffdata += 12;
highmark += 12;
/* 12 bytes is needed for data header */
GST_BUFFER_SIZE (buffer) -= 12;
gst_tag_list_foreach (tags, gst_avi_mux_write_tag, &data);
/* do not free tags here, as it refers to the tag snapshot */
GST_BUFFER_SIZE (buffer) += 12;
buffdata = GST_BUFFER_DATA (buffer) + highmark;
/* update list size */
GST_WRITE_UINT32_LE (ptr, highmark - startsize - 4);
avimux->tag_size = highmark - startsize + 4;
}
/* avi data header */
memcpy (buffdata + 0, "LIST", 4);
GST_WRITE_UINT32_LE (buffdata + 4, avimux->data_size);
memcpy (buffdata + 8, "movi", 4);
buffdata += 12;
highmark += 12;
{ /* only the part that is filled in actually makes up the header
* unref the parent as we only need this part from now on */
GstBuffer *subbuffer = gst_buffer_create_sub (buffer, 0, highmark);
gst_buffer_unref (buffer);
return subbuffer;
}
}
static GstBuffer *
gst_avi_mux_riff_get_avix_header (guint32 datax_size)
{
GstBuffer *buffer;
guint8 *buffdata;
buffer = gst_buffer_new_and_alloc (24);
buffdata = GST_BUFFER_DATA (buffer);
memcpy (buffdata + 0, "RIFF", 4);
GST_WRITE_UINT32_LE (buffdata + 4, datax_size + 4 * 4);
memcpy (buffdata + 8, "AVIX", 4);
memcpy (buffdata + 12, "LIST", 4);
GST_WRITE_UINT32_LE (buffdata + 16, datax_size);
memcpy (buffdata + 20, "movi", 4);
return buffer;
}
static GstBuffer *
gst_avi_mux_riff_get_video_header (guint32 video_frame_size)
{
GstBuffer *buffer;
guint8 *buffdata;
buffer = gst_buffer_new_and_alloc (8);
buffdata = GST_BUFFER_DATA (buffer);
memcpy (buffdata + 0, "00db", 4);
GST_WRITE_UINT32_LE (buffdata + 4, video_frame_size);
return buffer;
}
static GstBuffer *
gst_avi_mux_riff_get_audio_header (guint32 audio_sample_size)
{
GstBuffer *buffer;
guint8 *buffdata;
buffer = gst_buffer_new_and_alloc (8);
buffdata = GST_BUFFER_DATA (buffer);
memcpy (buffdata + 0, "01wb", 4);
GST_WRITE_UINT32_LE (buffdata + 4, audio_sample_size);
return buffer;
}
/* write an odml index chunk in the movi list */
static GstFlowReturn
gst_avi_mux_write_avix_index (GstAviMux * avimux, guchar * code,
guchar * chunk, gst_avi_superindex_entry * super_index,
gint * super_index_count)
{
GstFlowReturn res;
GstBuffer *buffer;
guint8 *buffdata, *data;
gst_riff_index_entry *entry;
gint i;
guint32 size, entry_count;
/* allocate the maximum possible */
buffer = gst_buffer_new_and_alloc (32 + 8 * avimux->idx_index);
buffdata = GST_BUFFER_DATA (buffer);
/* general index chunk info */
memcpy (buffdata + 0, chunk, 4); /* chunk id */
GST_WRITE_UINT32_LE (buffdata + 4, 0); /* chunk size; fill later */
GST_WRITE_UINT16_LE (buffdata + 8, 2); /* index entry is 2 words */
buffdata[10] = 0; /* index subtype */
buffdata[11] = GST_AVI_INDEX_OF_CHUNKS; /* index type: AVI_INDEX_OF_CHUNKS */
GST_WRITE_UINT32_LE (buffdata + 12, 0); /* entries in use; fill later */
memcpy (buffdata + 16, code, 4); /* stream to which index refers */
GST_WRITE_UINT64_LE (buffdata + 20, avimux->avix_start); /* base offset */
GST_WRITE_UINT32_LE (buffdata + 28, 0); /* reserved */
buffdata += 32;
/* now the actual index entries */
i = avimux->idx_index;
entry = avimux->idx;
while (i > 0) {
if (memcmp ((guchar *) & entry->id, code, 4) == 0) {
/* enter relative offset to the data (!) */
GST_WRITE_UINT32_LE (buffdata, GUINT32_FROM_LE (entry->offset) + 8);
/* msb is set if not (!) keyframe */
GST_WRITE_UINT32_LE (buffdata + 4, GUINT32_FROM_LE (entry->size)
| (GUINT32_FROM_LE (entry->flags)
& GST_RIFF_IF_KEYFRAME ? 0 : 1 << 31));
buffdata += 8;
}
i--;
entry++;
}
/* ok, now we know the size and no of entries, fill in where needed */
data = GST_BUFFER_DATA (buffer);
GST_BUFFER_SIZE (buffer) = size = buffdata - data;
GST_WRITE_UINT32_LE (data + 4, size - 8);
entry_count = (size - 32) / 8;
GST_WRITE_UINT32_LE (data + 12, entry_count);
/* decorate and send */
gst_buffer_set_caps (buffer, GST_PAD_CAPS (avimux->srcpad));
if ((res = gst_pad_push (avimux->srcpad, buffer)) != GST_FLOW_OK)
return res;
/* keep track of this in superindex (if room) ... */
if (*super_index_count < GST_AVI_SUPERINDEX_COUNT) {
i = *super_index_count;
super_index[i].offset = GUINT64_TO_LE (avimux->total_data);
super_index[i].size = GUINT32_TO_LE (size);
super_index[i].duration = GUINT32_TO_LE (entry_count);
(*super_index_count)++;
} else
GST_WARNING_OBJECT (avimux, "No more room in superindex of stream %s",
code);
/* ... and in size */
avimux->total_data += size;
if (avimux->is_bigfile)
avimux->datax_size += size;
else
avimux->data_size += size;
return GST_FLOW_OK;
}
/* some other usable functions (thankyou xawtv ;-) ) */
static void
gst_avi_mux_add_index (GstAviMux * avimux, guchar * code, guint32 flags,
guint32 size)
{
if (avimux->idx_index == avimux->idx_count) {
avimux->idx_count += 256;
avimux->idx =
g_realloc (avimux->idx,
avimux->idx_count * sizeof (gst_riff_index_entry));
}
memcpy (&(avimux->idx[avimux->idx_index].id), code, 4);
avimux->idx[avimux->idx_index].flags = GUINT32_TO_LE (flags);
avimux->idx[avimux->idx_index].offset = GUINT32_TO_LE (avimux->idx_offset);
avimux->idx[avimux->idx_index].size = GUINT32_TO_LE (size);
avimux->idx_index++;
}
static GstFlowReturn
gst_avi_mux_write_index (GstAviMux * avimux)
{
GstFlowReturn res;
GstBuffer *buffer;
guint8 *buffdata;
buffer = gst_buffer_new_and_alloc (8);
buffdata = GST_BUFFER_DATA (buffer);
memcpy (buffdata + 0, "idx1", 4);
GST_WRITE_UINT32_LE (buffdata + 4,
avimux->idx_index * sizeof (gst_riff_index_entry));
gst_buffer_set_caps (buffer, GST_PAD_CAPS (avimux->srcpad));
res = gst_pad_push (avimux->srcpad, buffer);
if (res != GST_FLOW_OK)
return res;
buffer = gst_buffer_new ();
GST_BUFFER_SIZE (buffer) = avimux->idx_index * sizeof (gst_riff_index_entry);
GST_BUFFER_DATA (buffer) = (guint8 *) avimux->idx;
GST_BUFFER_MALLOCDATA (buffer) = GST_BUFFER_DATA (buffer);
avimux->idx = NULL; /* will be free()'ed by gst_buffer_unref() */
avimux->total_data += GST_BUFFER_SIZE (buffer) + 8;
gst_buffer_set_caps (buffer, GST_PAD_CAPS (avimux->srcpad));
res = gst_pad_push (avimux->srcpad, buffer);
if (res != GST_FLOW_OK)
return res;
avimux->idx_size += avimux->idx_index * sizeof (gst_riff_index_entry) + 8;
/* update header */
avimux->avi_hdr.flags |= GST_RIFF_AVIH_HASINDEX;
return GST_FLOW_OK;
}
static GstFlowReturn
gst_avi_mux_bigfile (GstAviMux * avimux, gboolean last)
{
GstFlowReturn res = GST_FLOW_OK;
GstBuffer *header;
GstEvent *event;
/* first some odml standard index chunk in the movi list */
if (avimux->video_pad_connected) {
res = gst_avi_mux_write_avix_index (avimux, (guchar *) "00db",
(guchar *) "ix00", avimux->vids_idx, &avimux->vids_idx_index);
if (res != GST_FLOW_OK)
return res;
}
if (avimux->audio_pad_connected) {
res = gst_avi_mux_write_avix_index (avimux, (guchar *) "01wb",
(guchar *) "ix01", avimux->auds_idx, &avimux->auds_idx_index);
if (res != GST_FLOW_OK)
return res;
}
if (avimux->is_bigfile) {
/* search back */
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES,
avimux->avix_start, GST_CLOCK_TIME_NONE, avimux->avix_start);
/* if the event succeeds */
gst_pad_push_event (avimux->srcpad, event);
/* rewrite AVIX header */
header = gst_avi_mux_riff_get_avix_header (avimux->datax_size);
gst_buffer_set_caps (header, GST_PAD_CAPS (avimux->srcpad));
res = gst_pad_push (avimux->srcpad, header);
/* go back to current location, at least try */
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES,
avimux->total_data, GST_CLOCK_TIME_NONE, avimux->total_data);
gst_pad_push_event (avimux->srcpad, event);
if (res != GST_FLOW_OK)
return res;
} else { /* write a standard index in the first riff chunk */
res = gst_avi_mux_write_index (avimux);
/* the index data/buffer is freed by pushing it */
avimux->idx_count = 0;
if (res != GST_FLOW_OK)
return res;
}
avimux->avix_start = avimux->total_data;
if (last)
return res;
avimux->is_bigfile = TRUE;
avimux->numx_frames = 0;
avimux->datax_size = 4; /* movi tag */
avimux->idx_index = 0;
header = gst_avi_mux_riff_get_avix_header (0);
avimux->total_data += GST_BUFFER_SIZE (header);
/* avix_start is used as base offset for the odml index chunk */
avimux->idx_offset = avimux->total_data - avimux->avix_start;
gst_buffer_set_caps (header, GST_PAD_CAPS (avimux->srcpad));
return gst_pad_push (avimux->srcpad, header);
}
/* enough header blabla now, let's go on to actually writing the headers */
static GstFlowReturn
gst_avi_mux_start_file (GstAviMux * avimux)
{
GstFlowReturn res;
GstBuffer *header;
avimux->total_data = 0;
avimux->total_frames = 0;
avimux->data_size = 4; /* movi tag */
avimux->datax_size = 0;
avimux->num_frames = 0;
avimux->numx_frames = 0;
avimux->audio_size = 0;
avimux->audio_time = 0;
avimux->avix_start = 0;
avimux->idx_index = 0;
avimux->idx_offset = 0; /* see 10 lines below */
avimux->idx_size = 0;
avimux->idx_count = 0;
avimux->idx = NULL;
avimux->vids_idx_index = 0;
avimux->auds_idx_index = 0;
avimux->tag_size = 0;
/* header */
avimux->avi_hdr.streams =
(avimux->video_pad_connected ? 1 : 0) +
(avimux->audio_pad_connected ? 1 : 0);
avimux->is_bigfile = FALSE;
header = gst_avi_mux_riff_get_avi_header (avimux);
avimux->total_data += GST_BUFFER_SIZE (header);
gst_buffer_set_caps (header, GST_PAD_CAPS (avimux->srcpad));
res = gst_pad_push (avimux->srcpad, header);
avimux->idx_offset = avimux->total_data;
avimux->write_header = FALSE;
avimux->restart = FALSE;
return res;
}
static GstFlowReturn
gst_avi_mux_stop_file (GstAviMux * avimux)
{
GstFlowReturn res = GST_FLOW_OK;
GstEvent *event;
GstBuffer *header;
/* if bigfile, rewrite header, else write indexes */
/* don't bail out at once if error, still try to re-write header */
if (avimux->video_pad_connected) {
if (avimux->is_bigfile) {
res = gst_avi_mux_bigfile (avimux, TRUE);
} else {
res = gst_avi_mux_write_index (avimux);
}
}
/* we do our best to make it interleaved at least ... */
if (avimux->audio_pad_connected && avimux->video_pad_connected)
avimux->avi_hdr.flags |= GST_RIFF_AVIH_ISINTERLEAVED;
/* set rate and everything having to do with that */
avimux->avi_hdr.max_bps = 0;
if (avimux->audio_pad_connected) {
/* calculate bps if needed */
if (!avimux->auds.av_bps) {
if (avimux->audio_time) {
avimux->auds.av_bps =
(GST_SECOND * avimux->audio_size) / avimux->audio_time;
/* round bps to nearest multiple of 8;
* which is much more likely to be the (cbr) bitrate in use;
* which in turn results in better timestamp calculation on playback */
avimux->auds.av_bps = GST_ROUND_UP_8 (avimux->auds.av_bps - 4);
} else {
GST_ELEMENT_WARNING (avimux, STREAM, MUX,
(_("No or invalid input audio, AVI stream will be corrupt.")),
(NULL));
avimux->auds.av_bps = 0;
}
avimux->auds_hdr.rate = avimux->auds.av_bps * avimux->auds_hdr.scale;
}
avimux->avi_hdr.max_bps += avimux->auds.av_bps;
}
if (avimux->video_pad_connected) {
avimux->avi_hdr.max_bps += ((avimux->vids.bit_cnt + 7) / 8) *
(1000000. / avimux->avi_hdr.us_frame) * avimux->vids.image_size;
}
/* statistics/total_frames/... */
avimux->avi_hdr.tot_frames = avimux->num_frames;
if (avimux->video_pad_connected) {
avimux->vids_hdr.length = avimux->num_frames;
}
if (avimux->audio_pad_connected) {
avimux->auds_hdr.length =
(avimux->audio_time * avimux->auds_hdr.rate) / GST_SECOND;
}
/* seek and rewrite the header */
header = gst_avi_mux_riff_get_avi_header (avimux);
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES,
0, GST_CLOCK_TIME_NONE, 0);
gst_pad_push_event (avimux->srcpad, event);
gst_buffer_set_caps (header, GST_PAD_CAPS (avimux->srcpad));
/* the first error survives */
if (res == GST_FLOW_OK)
res = gst_pad_push (avimux->srcpad, header);
else
gst_pad_push (avimux->srcpad, header);
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES,
avimux->total_data, GST_CLOCK_TIME_NONE, avimux->total_data);
gst_pad_push_event (avimux->srcpad, event);
avimux->write_header = TRUE;
return res;
}
static GstFlowReturn
gst_avi_mux_restart_file (GstAviMux * avimux)
{
GstFlowReturn res;
if ((res = gst_avi_mux_stop_file (avimux)) != GST_FLOW_OK)
return res;
gst_pad_push_event (avimux->srcpad, gst_event_new_eos ());
return gst_avi_mux_start_file (avimux);
}
/* handle events (search) */
static gboolean
gst_avi_mux_handle_event (GstPad * pad, GstEvent * event)
{
GstAviMux *avimux;
GstTagList *list;
gboolean ret;
avimux = GST_AVI_MUX (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_TAG:
gst_event_parse_tag (event, &list);
if (avimux->tags) {
gst_tag_list_insert (avimux->tags, list, GST_TAG_MERGE_PREPEND);
} else {
avimux->tags = gst_tag_list_copy (list);
}
break;
default:
break;
}
/* now GstCollectPads can take care of the rest, e.g. EOS */
ret = avimux->collect_event (pad, event);
close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau. Original commit message from CVS: * ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps): * ext/esd/esdmon.c: (gst_esdmon_get): * ext/flac/gstflactag.c: (gst_flac_tag_chain): * ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps), (gst_gdk_pixbuf_sink_getcaps): * ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps), (gst_jpegenc_setcaps): * ext/jpeg/gstsmokedec.c: (gst_smokedec_chain): * ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps), (gst_smokeenc_setcaps): * ext/libmng/gstmngdec.c: (gst_mngdec_sinklink), (gst_mngdec_src_getcaps): * ext/libmng/gstmngenc.c: (gst_mngenc_sinklink), (gst_mngenc_chain): * ext/libpng/gstpngenc.c: (gst_pngenc_setcaps): * ext/mikmod/gstmikmod.c: (gst_mikmod_srclink): * ext/speex/gstspeexdec.c: (speex_dec_convert), (speex_dec_src_event), (speex_dec_chain): * gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect), (gst_avimux_audsinkconnect), (gst_avimux_handle_event): * gst/debug/negotiation.c: (gst_negotiation_getcaps), (gst_negotiation_pad_link), (gst_negotiation_chain): * gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler), (gst_flxdec_chain): * gst/interleave/deinterleave.c: (deinterleave_sink_link), (deinterleave_chain): * gst/law/mulaw-encode.c: (mulawenc_setcaps): * gst/median/gstmedian.c: (gst_median_link): * gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect), (gst_monoscope_chain): * gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect): * gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps): * sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain): * sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get): close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
2006-03-13 15:49:08 +00:00
gst_object_unref (avimux);
return ret;
}
/* send extra 'padding' data */
static GstFlowReturn
gst_avi_mux_send_pad_data (GstAviMux * avimux, gulong num_bytes)
{
GstBuffer *buffer;
buffer = gst_buffer_new_and_alloc (num_bytes);
memset (GST_BUFFER_DATA (buffer), 0, num_bytes);
gst_buffer_set_caps (buffer, GST_PAD_CAPS (avimux->srcpad));
return gst_pad_push (avimux->srcpad, buffer);
}
/* strip buffer of time/caps meaning, is now only raw data;
* bit of a work-around for the following ...*/
/* TODO on basesink:
* - perhaps use a default format like basesrc (to be chosen by derived element)
* and only act on segment, etc that are of such type
* - in any case, basesink could be more careful in deciding when
* to drop buffers, currently it decides on GST_FORMAT_TIME base
* even when its clip_segment is GST_FORMAT_BYTE based,
* and gst_segment_clip feels this reason enough to drop it
* (reasonable doubt is not enough to let pass :-) )
*/
static GstBuffer *
gst_avi_mux_strip_buffer (GstAviMux * avimux, GstBuffer * buffer)
{
buffer = gst_buffer_make_metadata_writable (buffer);
GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
gst_buffer_set_caps (buffer, GST_PAD_CAPS (avimux->srcpad));
return buffer;
}
/* do audio buffer */
static GstFlowReturn
gst_avi_mux_do_audio_buffer (GstAviMux * avimux)
{
GstFlowReturn res;
GstBuffer *data, *header;
gulong total_size, pad_bytes = 0;
data = gst_collect_pads_pop (avimux->collect, avimux->audiocollectdata);
/* write a audio header + index entry */
if (GST_BUFFER_SIZE (data) & 1) {
pad_bytes = 2 - (GST_BUFFER_SIZE (data) & 1);
}
header = gst_avi_mux_riff_get_audio_header (GST_BUFFER_SIZE (data));
total_size = GST_BUFFER_SIZE (header) + GST_BUFFER_SIZE (data) + pad_bytes;
if (avimux->is_bigfile) {
avimux->datax_size += total_size;
} else {
avimux->data_size += total_size;
avimux->audio_size += GST_BUFFER_SIZE (data);
avimux->audio_time += GST_BUFFER_DURATION (data);
}
gst_avi_mux_add_index (avimux, (guchar *) "01wb", 0x0,
GST_BUFFER_SIZE (data));
/* prepare buffers for sending */
gst_buffer_set_caps (header, GST_PAD_CAPS (avimux->srcpad));
data = gst_avi_mux_strip_buffer (avimux, data);
if ((res = gst_pad_push (avimux->srcpad, header)) != GST_FLOW_OK)
return res;
if ((res = gst_pad_push (avimux->srcpad, data)) != GST_FLOW_OK)
return res;
if (pad_bytes) {
if ((res = gst_avi_mux_send_pad_data (avimux, pad_bytes)) != GST_FLOW_OK)
return res;
}
/* if any push above fails, we're in trouble with file consistency anyway */
avimux->total_data += total_size;
avimux->idx_offset += total_size;
return res;
}
/* do video buffer */
static GstFlowReturn
gst_avi_mux_do_video_buffer (GstAviMux * avimux)
{
GstFlowReturn res;
GstBuffer *data, *header;
gulong total_size, pad_bytes = 0;
guint flags;
data = gst_collect_pads_pop (avimux->collect, avimux->videocollectdata);
if (avimux->restart) {
if ((res = gst_avi_mux_restart_file (avimux)) != GST_FLOW_OK)
return res;
}
/* write a video header + index entry */
if ((avimux->is_bigfile ? avimux->datax_size : avimux->data_size) +
GST_BUFFER_SIZE (data) > 1024 * 1024 * 2000) {
if (avimux->enable_large_avi) {
if ((res = gst_avi_mux_bigfile (avimux, FALSE)) != GST_FLOW_OK)
return res;
} else {
if ((res = gst_avi_mux_restart_file (avimux)) != GST_FLOW_OK)
return res;
}
}
if (GST_BUFFER_SIZE (data) & 1) {
pad_bytes = 2 - (GST_BUFFER_SIZE (data) & 1);
}
header = gst_avi_mux_riff_get_video_header (GST_BUFFER_SIZE (data));
total_size = GST_BUFFER_SIZE (header) + GST_BUFFER_SIZE (data) + pad_bytes;
avimux->total_frames++;
if (avimux->is_bigfile) {
avimux->datax_size += total_size;
avimux->numx_frames++;
} else {
avimux->data_size += total_size;
avimux->num_frames++;
}
flags = 0x02;
if (!GST_BUFFER_FLAG_IS_SET (data, GST_BUFFER_FLAG_DELTA_UNIT))
flags |= 0x10;
gst_avi_mux_add_index (avimux, (guchar *) "00db", flags,
GST_BUFFER_SIZE (data));
/* prepare buffers for sending */
gst_buffer_set_caps (header, GST_PAD_CAPS (avimux->srcpad));
data = gst_avi_mux_strip_buffer (avimux, data);
if ((res = gst_pad_push (avimux->srcpad, header)) != GST_FLOW_OK)
return res;
if ((res = gst_pad_push (avimux->srcpad, data)) != GST_FLOW_OK)
return res;
if (pad_bytes) {
if ((res = gst_avi_mux_send_pad_data (avimux, pad_bytes)) != GST_FLOW_OK)
return res;
}
/* if any push above fails, we're in trouble with file consistency anyway */
avimux->total_data += total_size;
avimux->idx_offset += total_size;
return res;
}
/* pick the oldest buffer from the pads and push it */
static GstFlowReturn
gst_avi_mux_do_one_buffer (GstAviMux * avimux)
{
GstBuffer *video_buf = NULL;
GstBuffer *audio_buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
GstClockTime video_time = GST_CLOCK_TIME_NONE;
GstClockTime audio_time = GST_CLOCK_TIME_NONE;
if (avimux->videocollectdata && avimux->video_pad_connected) {
video_buf =
gst_collect_pads_peek (avimux->collect, avimux->videocollectdata);
}
if (avimux->audiocollectdata && avimux->audio_pad_connected) {
audio_buf =
gst_collect_pads_peek (avimux->collect, avimux->audiocollectdata);
}
/* segment info is used to translate the incoming timestamps
* to outgoing muxed (running) timeline */
if (video_buf) {
video_time =
gst_segment_to_running_time (&avimux->videocollectdata->segment,
GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (video_buf));
GST_DEBUG ("peeked video buffer %p (time %" GST_TIME_FORMAT ")"
", running %" GST_TIME_FORMAT, video_buf,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (video_buf)),
GST_TIME_ARGS (video_time));
}
if (audio_buf) {
audio_time =
gst_segment_to_running_time (&avimux->audiocollectdata->segment,
GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (audio_buf));
GST_DEBUG ("peeked audio buffer %p (time %" GST_TIME_FORMAT ")"
", running %" GST_TIME_FORMAT, audio_buf,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (audio_buf)),
GST_TIME_ARGS (audio_time));
}
/* now use re-calculated time to choose */
if (video_buf && audio_buf) {
/* either video and audio can be translated, or translate neither */
if (!GST_CLOCK_TIME_IS_VALID (video_time)
|| !GST_CLOCK_TIME_IS_VALID (audio_time)) {
video_time = GST_BUFFER_TIMESTAMP (video_buf);
audio_time = GST_BUFFER_TIMESTAMP (audio_buf);
}
/* invalid time buffers should be rare, pass these first;
* typically contain init data */
if ((video_time <= audio_time && GST_CLOCK_TIME_IS_VALID (audio_time))
|| !GST_CLOCK_TIME_IS_VALID (video_time))
res = gst_avi_mux_do_video_buffer (avimux);
else
res = gst_avi_mux_do_audio_buffer (avimux);
} else if (video_buf) {
res = gst_avi_mux_do_video_buffer (avimux);
} else if (audio_buf) {
res = gst_avi_mux_do_audio_buffer (avimux);
} else {
/* simply finish off the file and send EOS */
gst_avi_mux_stop_file (avimux);
gst_pad_push_event (avimux->srcpad, gst_event_new_eos ());
return GST_FLOW_UNEXPECTED;
}
/* unref the peek obtained above */
if (video_buf)
gst_buffer_unref (video_buf);
if (audio_buf)
gst_buffer_unref (audio_buf);
return res;
}
static GstFlowReturn
gst_avi_mux_collect_pads (GstCollectPads * pads, GstAviMux * avimux)
{
GstFlowReturn res;
if (G_UNLIKELY (avimux->write_header)) {
if ((res = gst_avi_mux_start_file (avimux)) != GST_FLOW_OK)
return res;
}
return gst_avi_mux_do_one_buffer (avimux);
}
static void
gst_avi_mux_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstAviMux *avimux;
avimux = GST_AVI_MUX (object);
switch (prop_id) {
case ARG_BIGFILE:
g_value_set_boolean (value, avimux->enable_large_avi);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_avi_mux_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstAviMux *avimux;
avimux = GST_AVI_MUX (object);
switch (prop_id) {
case ARG_BIGFILE:
avimux->enable_large_avi = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_avi_mux_change_state (GstElement * element, GstStateChange transition)
{
GstAviMux *avimux;
GstStateChangeReturn ret;
avimux = GST_AVI_MUX (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_collect_pads_start (avimux->collect);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* avimux->video_pad_eos = FALSE; */
/* avimux->audio_pad_eos = FALSE; */
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_collect_pads_stop (avimux->collect);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
goto done;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (avimux->tags) {
gst_tag_list_free (avimux->tags);
avimux->tags = NULL;
}
if (avimux->tags_snap) {
gst_tag_list_free (avimux->tags_snap);
avimux->tags_snap = NULL;
}
g_free (avimux->idx);
avimux->idx = NULL;
if (avimux->vids_codec_data)
gst_buffer_unref (avimux->vids_codec_data);
avimux->vids_codec_data = NULL;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
done:
return ret;
}