From e1d778e559a402f0b4ebe2b84978f3542807a630 Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Thu, 21 May 2009 00:41:47 +0100 Subject: mpegdemux: Add support for outputting sub-picture streams found in files. Output subpicture streams when they are found on the private stream ID. Don't strip off the first byte of such packets when pushing. --- gst/mpegdemux/gstmpegdemux.c | 39 +++++++++++++++++++++++++++++++-------- 1 file changed, 31 insertions(+), 8 deletions(-) (limited to 'gst/mpegdemux/gstmpegdemux.c') diff --git a/gst/mpegdemux/gstmpegdemux.c b/gst/mpegdemux/gstmpegdemux.c index 439bb563..552bae5a 100644 --- a/gst/mpegdemux/gstmpegdemux.c +++ b/gst/mpegdemux/gstmpegdemux.c @@ -39,6 +39,7 @@ * Fluendo, S.L. All Rights Reserved. * * Contributor(s): Wim Taymans + * Jan Schmidt */ #ifdef HAVE_CONFIG_H @@ -177,6 +178,13 @@ static GstStaticPadTemplate audio_template = "audio/x-private1-ac3;" "audio/x-private1-dts;" "audio/ac3") ); +static GstStaticPadTemplate subpicture_template = +GST_STATIC_PAD_TEMPLATE ("subpicture_%02x", + GST_PAD_SRC, + GST_PAD_SOMETIMES, + GST_STATIC_CAPS ("video/x-dvd-subpicture") + ); + static GstStaticPadTemplate private_template = GST_STATIC_PAD_TEMPLATE ("private_%d", GST_PAD_SRC, @@ -250,10 +258,14 @@ gst_flups_demux_base_init (GstFluPSDemuxClass * klass) klass->sink_template = gst_static_pad_template_get (&sink_template); klass->video_template = gst_static_pad_template_get (&video_template); klass->audio_template = gst_static_pad_template_get (&audio_template); + klass->subpicture_template = + gst_static_pad_template_get (&subpicture_template); klass->private_template = gst_static_pad_template_get (&private_template); gst_element_class_add_pad_template (element_class, klass->video_template); gst_element_class_add_pad_template (element_class, klass->audio_template); + gst_element_class_add_pad_template (element_class, + klass->subpicture_template); gst_element_class_add_pad_template (element_class, klass->private_template); gst_element_class_add_pad_template (element_class, klass->sink_template); @@ -404,6 +416,9 @@ gst_flups_demux_create_stream (GstFluPSDemux * demux, gint id, gint stream_type) caps = gst_caps_new_simple ("audio/x-private1-lpcm", NULL); break; case ST_PS_DVD_SUBPICTURE: + template = klass->subpicture_template; + name = g_strdup_printf ("subpicture_%02x", id); + caps = gst_caps_new_simple ("video/x-dvd-subpicture", NULL); break; case ST_GST_AUDIO_RAWA52: template = klass->audio_template; @@ -1839,18 +1854,26 @@ gst_flups_demux_data_cb (GstPESFilter * filter, gboolean first, } if (G_LIKELY (stream_type == -1)) { - /* new id */ + /* new id is in the first byte */ id = data[offset++]; - /* Number of audio frames in this packet */ - nframes = data[offset++]; - - GST_DEBUG_OBJECT (demux, "private type 0x%02x, %d frames", id, - nframes); - - datalen -= 2; + datalen--; /* and remap */ stream_type = demux->psm[id]; + + /* Now, if it's a subpicture stream - no more, otherwise + * take the first byte too, since it's the frame count in audio + * streams and our backwards compat convention is to strip it off */ + if (stream_type != ST_PS_DVD_SUBPICTURE) { + /* Number of audio frames in this packet */ + nframes = data[offset++]; + datalen--; + GST_DEBUG_OBJECT (demux, "private type 0x%02x, %d frames", id, + nframes); + } else { + GST_DEBUG_OBJECT (demux, "private type 0x%02x, stream type %d", id, + stream_type); + } } } if (stream_type == -1) -- cgit v1.2.1 From f7eefea47cf770107672d07189843554b10b9c6f Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Thu, 21 May 2009 11:13:54 +0100 Subject: mpegdemux: Add sparse stream filling. First stab at sending new-segment events to effect sparse stream updates. --- gst/mpegdemux/gstmpegdemux.c | 93 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 93 insertions(+) (limited to 'gst/mpegdemux/gstmpegdemux.c') diff --git a/gst/mpegdemux/gstmpegdemux.c b/gst/mpegdemux/gstmpegdemux.c index 552bae5a..3f298fb6 100644 --- a/gst/mpegdemux/gstmpegdemux.c +++ b/gst/mpegdemux/gstmpegdemux.c @@ -57,6 +57,9 @@ #define SCAN_SCR_SZ 12 #define SCAN_PTS_SZ 80 +#define SEGMENT_THRESHOLD (300*GST_MSECOND) +#define VIDEO_SEGMENT_THRESHOLD (500*GST_MSECOND) + typedef enum { SCAN_SCR, @@ -217,6 +220,10 @@ static inline gboolean gst_flups_demux_scan_forward_ts (GstFluPSDemux * demux, static inline gboolean gst_flups_demux_scan_backward_ts (GstFluPSDemux * demux, guint64 * pos, SCAN_MODE mode, guint64 * rts); +static void gst_flups_demux_send_segment_updates (GstFluPSDemux * demux, + GstClockTime new_time); +static void gst_flups_demux_clear_times (GstFluPSDemux * demux); + static GstElementClass *parent_class = NULL; /*static guint gst_flups_demux_signals[LAST_SIGNAL] = { 0 };*/ @@ -352,6 +359,7 @@ gst_flups_demux_create_stream (GstFluPSDemux * demux, gint id, gint stream_type) gchar *name; GstFluPSDemuxClass *klass = GST_FLUPS_DEMUX_GET_CLASS (demux); GstCaps *caps; + GstClockTime threshold = SEGMENT_THRESHOLD; name = NULL; template = NULL; @@ -380,6 +388,7 @@ gst_flups_demux_create_stream (GstFluPSDemux * demux, gint id, gint stream_type) caps = gst_caps_new_simple ("video/mpeg", "mpegversion", G_TYPE_INT, mpeg_version, "systemstream", G_TYPE_BOOLEAN, FALSE, NULL); + threshold = VIDEO_SEGMENT_THRESHOLD; break; } case ST_AUDIO_MPEG1: @@ -399,6 +408,7 @@ gst_flups_demux_create_stream (GstFluPSDemux * demux, gint id, gint stream_type) template = klass->video_template; name = g_strdup_printf ("video_%02x", id); caps = gst_caps_new_simple ("video/x-h264", NULL); + threshold = VIDEO_SEGMENT_THRESHOLD; break; case ST_PS_AUDIO_AC3: template = klass->audio_template; @@ -439,6 +449,7 @@ gst_flups_demux_create_stream (GstFluPSDemux * demux, gint id, gint stream_type) stream->notlinked = FALSE; stream->type = stream_type; stream->pad = gst_pad_new_from_template (template, name); + stream->segment_thresh = threshold; gst_pad_set_event_function (stream->pad, GST_DEBUG_FUNCPTR (gst_flups_demux_src_event)); gst_pad_set_query_function (stream->pad, @@ -574,6 +585,20 @@ gst_flups_demux_send_data (GstFluPSDemux * demux, GstFluPSStream * stream, GST_TIME_ARGS (demux->src_segment.last_stop), GST_TIME_ARGS (MPEGTIME_TO_GSTTIME (demux->current_scr))); + if (demux->src_segment.last_stop != GST_CLOCK_TIME_NONE) { + GstClockTime new_time = demux->base_time + demux->src_segment.last_stop; + + if (stream->last_ts == GST_CLOCK_TIME_NONE || stream->last_ts < new_time) { +#if 0 + g_print ("last_ts update on pad %s to time %" GST_TIME_FORMAT "\n", + GST_PAD_NAME (stream->pad), GST_TIME_ARGS (cur_scr_time)); +#endif + stream->last_ts = new_time; + } + + gst_flups_demux_send_segment_updates (demux, new_time); + } + /* Set the buffer discont flag, and clear discont state on the stream */ if (stream->discont) { GST_DEBUG_OBJECT (demux, "marking discont buffer"); @@ -746,11 +771,75 @@ gst_flups_demux_flush (GstFluPSDemux * demux) gst_adapter_clear (demux->adapter); gst_adapter_clear (demux->rev_adapter); gst_pes_filter_drain (&demux->filter); + gst_flups_demux_clear_times (demux); demux->adapter_offset = G_MAXUINT64; demux->current_scr = G_MAXUINT64; demux->bytes_since_scr = 0; } +static void +gst_flups_demux_clear_times (GstFluPSDemux * demux) +{ + gint id; + + /* Clear the last ts for all streams */ + for (id = 0; id < GST_FLUPS_DEMUX_MAX_STREAMS; id++) { + GstFluPSStream *stream = demux->streams[id]; + + if (stream) { + stream->last_seg_start = stream->last_ts = GST_CLOCK_TIME_NONE; + } + } +} + +static void +gst_flups_demux_send_segment_updates (GstFluPSDemux * demux, + GstClockTime new_time) +{ + /* Advance all lagging streams by sending a segment update */ + gint id; + GstEvent *event = NULL; + + /* FIXME: Handle reverse playback */ + + if (new_time > demux->src_segment.stop) + return; + + for (id = 0; id < GST_FLUPS_DEMUX_MAX_STREAMS; id++) { + GstFluPSStream *stream = demux->streams[id]; + + if (stream) { + if (stream->last_ts == GST_CLOCK_TIME_NONE || + stream->last_ts < demux->src_segment.start) + stream->last_ts = demux->src_segment.start; + if (stream->last_ts + stream->segment_thresh < new_time) { +#if 0 + g_print ("Segment update to pad %s time %" GST_TIME_FORMAT " stop now %" + GST_TIME_FORMAT "\n", GST_PAD_NAME (stream->pad), + GST_TIME_ARGS (new_time), GST_TIME_ARGS (demux->src_segment.stop)); +#endif + GST_DEBUG_OBJECT (demux, + "Segment update to pad %s time %" GST_TIME_FORMAT, + GST_PAD_NAME (stream->pad), GST_TIME_ARGS (new_time)); + if (event == NULL) { + event = gst_event_new_new_segment_full (TRUE, + demux->src_segment.rate, demux->src_segment.applied_rate, + GST_FORMAT_TIME, new_time, + demux->src_segment.stop, + demux->src_segment.time + (new_time - demux->src_segment.start)); + } + gst_event_ref (event); + gst_pad_push_event (stream->pad, event); + stream->last_seg_start = stream->last_ts = new_time; + stream->need_segment = FALSE; + } + } + } + + if (event) + gst_event_unref (event); +} + static void gst_flups_demux_close_segment (GstFluPSDemux * demux) { @@ -762,6 +851,10 @@ gst_flups_demux_close_segment (GstFluPSDemux * demux) &demux->src_segment); #endif + /* FIXME: Need to send a different segment-close to each pad where the + * last_seg_start != clock_time_none, as that indicates a sparse-stream + * event was sent there */ + /* Close the current segment for a linear playback */ if (demux->src_segment.rate >= 0) { /* for forward playback, we played from start to last_stop */ -- cgit v1.2.1 From b46059291736f78e7a86bb5bd1e91c9a6025113f Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Tue, 26 May 2009 18:47:32 +0100 Subject: mpegdemux: Only treat streams from 0xa0 to 0xaf as LPCM, not 0xa0..0xbf Don't treat some streams (Private Stream 2) as LPCM when they're not. Fixes playback of files that have private streams in them now that the PES filter emits such packets. --- gst/mpegdemux/gstmpegdemux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'gst/mpegdemux/gstmpegdemux.c') diff --git a/gst/mpegdemux/gstmpegdemux.c b/gst/mpegdemux/gstmpegdemux.c index 3f298fb6..ce1d0978 100644 --- a/gst/mpegdemux/gstmpegdemux.c +++ b/gst/mpegdemux/gstmpegdemux.c @@ -1414,7 +1414,7 @@ gst_flups_demux_reset_psm (GstFluPSDemux * demux) FILL_TYPE (0x40, 0x7f, -1); FILL_TYPE (0x80, 0x87, ST_PS_AUDIO_AC3); FILL_TYPE (0x88, 0x9f, ST_PS_AUDIO_DTS); - FILL_TYPE (0xa0, 0xbf, ST_PS_AUDIO_LPCM); + FILL_TYPE (0xa0, 0xaf, ST_PS_AUDIO_LPCM); FILL_TYPE (0xbd, 0xbd, -1); FILL_TYPE (0xc0, 0xdf, ST_AUDIO_MPEG1); FILL_TYPE (0xe0, 0xef, ST_GST_VIDEO_MPEG1_OR_2); -- cgit v1.2.1