diff options
Diffstat (limited to 'multimedia/gstreamer-plugins-ugly/files/patch-gst_mpegaudioparse_gstmpegaudioparse.c')
| -rw-r--r-- | multimedia/gstreamer-plugins-ugly/files/patch-gst_mpegaudioparse_gstmpegaudioparse.c | 733 |
1 files changed, 0 insertions, 733 deletions
diff --git a/multimedia/gstreamer-plugins-ugly/files/patch-gst_mpegaudioparse_gstmpegaudioparse.c b/multimedia/gstreamer-plugins-ugly/files/patch-gst_mpegaudioparse_gstmpegaudioparse.c deleted file mode 100644 index 6c1d5b594cd0..000000000000 --- a/multimedia/gstreamer-plugins-ugly/files/patch-gst_mpegaudioparse_gstmpegaudioparse.c +++ /dev/null @@ -1,733 +0,0 @@ ---- gst/mpegaudioparse/gstmpegaudioparse.c.orig 2007-12-17 00:16:51.000000000 -0500 -+++ gst/mpegaudioparse/gstmpegaudioparse.c 2007-12-17 00:17:13.000000000 -0500 -@@ -29,6 +29,7 @@ - GST_DEBUG_CATEGORY_STATIC (mp3parse_debug); - #define GST_CAT_DEFAULT mp3parse_debug - -+ - /* elementfactory information */ - static GstElementDetails mp3parse_details = { - "MPEG1 Audio Parser", -@@ -71,8 +72,9 @@ enum - - - static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass); --static void gst_mp3parse_base_init (GstMPEGAudioParseClass * klass); --static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse); -+static void gst_mp3parse_base_init (gpointer klass); -+static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse, -+ GstMPEGAudioParseClass * klass); - - static gboolean gst_mp3parse_sink_event (GstPad * pad, GstEvent * event); - static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer); -@@ -95,35 +97,11 @@ static gboolean mp3parse_bytepos_to_time - static gboolean - mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total); - --static GstElementClass *parent_class = NULL; -- - /*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */ - --GType --gst_mp3parse_get_type (void) --{ -- static GType mp3parse_type = 0; -- -- if (!mp3parse_type) { -- static const GTypeInfo mp3parse_info = { -- sizeof (GstMPEGAudioParseClass), -- (GBaseInitFunc) gst_mp3parse_base_init, -- NULL, -- (GClassInitFunc) gst_mp3parse_class_init, -- NULL, -- NULL, -- sizeof (GstMPEGAudioParse), -- 0, -- (GInstanceInitFunc) gst_mp3parse_init, -- }; -+GST_BOILERPLATE (GstMPEGAudioParse, gst_mp3parse, GstElement, GST_TYPE_ELEMENT); - -- mp3parse_type = g_type_register_static (GST_TYPE_ELEMENT, -- "GstMPEGAudioParse", &mp3parse_info, 0); -- } -- return mp3parse_type; --} -- --static guint mp3types_bitrates[2][3][16] = { -+static const guint mp3types_bitrates[2][3][16] = { - { - {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,}, - {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,}, -@@ -136,7 +114,7 @@ static guint mp3types_bitrates[2][3][16] - }, - }; - --static guint mp3types_freqs[3][3] = { {44100, 48000, 32000}, -+static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000}, - {22050, 24000, 16000}, - {11025, 12000, 8000} - }; -@@ -218,13 +196,14 @@ mp3_caps_create (guint layer, guint chan - new = gst_caps_new_simple ("audio/mpeg", - "mpegversion", G_TYPE_INT, 1, - "layer", G_TYPE_INT, layer, -- "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL); -+ "rate", G_TYPE_INT, samplerate, -+ "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL); - - return new; - } - - static void --gst_mp3parse_base_init (GstMPEGAudioParseClass * klass) -+gst_mp3parse_base_init (gpointer klass) - { - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - -@@ -265,8 +244,8 @@ gst_mp3parse_reset (GstMPEGAudioParse * - { - mp3parse->skip = 0; - mp3parse->resyncing = TRUE; -- mp3parse->cur_offset = -1; - mp3parse->next_ts = GST_CLOCK_TIME_NONE; -+ mp3parse->cur_offset = -1; - - mp3parse->tracked_offset = 0; - mp3parse->pending_ts = GST_CLOCK_TIME_NONE; -@@ -276,6 +255,7 @@ gst_mp3parse_reset (GstMPEGAudioParse * - - mp3parse->rate = mp3parse->channels = mp3parse->layer = -1; - mp3parse->version = 1; -+ mp3parse->max_bitreservoir = GST_CLOCK_TIME_NONE; - - mp3parse->avg_bitrate = 0; - mp3parse->bitrate_sum = 0; -@@ -285,21 +265,42 @@ gst_mp3parse_reset (GstMPEGAudioParse * - - mp3parse->xing_flags = 0; - mp3parse->xing_bitrate = 0; -+ -+ if (mp3parse->seek_table) { -+ g_list_foreach (mp3parse->seek_table, (GFunc) g_free, NULL); -+ g_list_free (mp3parse->seek_table); -+ mp3parse->seek_table = NULL; -+ } -+ -+ g_mutex_lock (mp3parse->pending_accurate_seeks_lock); -+ if (mp3parse->pending_accurate_seeks) { -+ g_slist_foreach (mp3parse->pending_accurate_seeks, (GFunc) g_free, NULL); -+ g_slist_free (mp3parse->pending_accurate_seeks); -+ mp3parse->pending_accurate_seeks = NULL; -+ } -+ g_mutex_unlock (mp3parse->pending_accurate_seeks_lock); -+ -+ if (mp3parse->pending_segment) { -+ GstEvent **eventp = &mp3parse->pending_segment; -+ -+ gst_event_replace (eventp, NULL); -+ } -+ -+ mp3parse->exact_position = FALSE; -+ gst_segment_init (&mp3parse->segment, GST_FORMAT_TIME); - } - - static void --gst_mp3parse_init (GstMPEGAudioParse * mp3parse) -+gst_mp3parse_init (GstMPEGAudioParse * mp3parse, GstMPEGAudioParseClass * klass) - { - mp3parse->sinkpad = -- gst_pad_new_from_template (gst_static_pad_template_get -- (&mp3_sink_template), "sink"); -+ gst_pad_new_from_static_template (&mp3_sink_template, "sink"); - gst_pad_set_event_function (mp3parse->sinkpad, gst_mp3parse_sink_event); - gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain); - gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad); - - mp3parse->srcpad = -- gst_pad_new_from_template (gst_static_pad_template_get -- (&mp3_src_template), "src"); -+ gst_pad_new_from_static_template (&mp3_src_template, "src"); - gst_pad_use_fixed_caps (mp3parse->srcpad); - gst_pad_set_event_function (mp3parse->srcpad, mp3parse_src_event); - gst_pad_set_query_function (mp3parse->srcpad, mp3parse_src_query); -@@ -307,6 +308,7 @@ gst_mp3parse_init (GstMPEGAudioParse * m - gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad); - - mp3parse->adapter = gst_adapter_new (); -+ mp3parse->pending_accurate_seeks_lock = g_mutex_new (); - - gst_mp3parse_reset (mp3parse); - } -@@ -316,10 +318,14 @@ gst_mp3parse_dispose (GObject * object) - { - GstMPEGAudioParse *mp3parse = GST_MP3PARSE (object); - -+ gst_mp3parse_reset (mp3parse); -+ - if (mp3parse->adapter) { - g_object_unref (mp3parse->adapter); - mp3parse->adapter = NULL; - } -+ g_mutex_free (mp3parse->pending_accurate_seeks_lock); -+ mp3parse->pending_accurate_seeks_lock = NULL; - - G_OBJECT_CLASS (parent_class)->dispose (object); - } -@@ -329,6 +335,7 @@ gst_mp3parse_sink_event (GstPad * pad, G - { - gboolean res; - GstMPEGAudioParse *mp3parse; -+ GstEvent **eventp; - - mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad)); - -@@ -343,6 +350,59 @@ gst_mp3parse_sink_event (GstPad * pad, G - gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate, - &format, &start, &stop, &pos); - -+ g_mutex_lock (mp3parse->pending_accurate_seeks_lock); -+ if (format == GST_FORMAT_BYTES && mp3parse->pending_accurate_seeks) { -+ MPEGAudioPendingAccurateSeek *seek = NULL; -+ GSList *node; -+ -+ for (node = mp3parse->pending_accurate_seeks; node; node = node->next) { -+ MPEGAudioPendingAccurateSeek *tmp = node->data; -+ -+ if (tmp->upstream_start == pos) { -+ seek = tmp; -+ break; -+ } -+ } -+ if (seek) { -+ GstSegment *s = &seek->segment; -+ -+ event = -+ gst_event_new_new_segment_full (FALSE, s->rate, s->applied_rate, -+ GST_FORMAT_TIME, s->start, s->stop, s->last_stop); -+ -+ mp3parse->segment = seek->segment; -+ -+ mp3parse->resyncing = FALSE; -+ mp3parse->cur_offset = pos; -+ mp3parse->next_ts = seek->timestamp_start; -+ mp3parse->pending_ts = GST_CLOCK_TIME_NONE; -+ mp3parse->tracked_offset = 0; -+ -+ gst_event_parse_new_segment_full (event, &update, &rate, -+ &applied_rate, &format, &start, &stop, &pos); -+ -+ GST_DEBUG_OBJECT (mp3parse, -+ "Pushing accurate newseg rate %g, applied rate %g, " -+ "format %d, start %lld, stop %lld, pos %lld\n", rate, -+ applied_rate, format, start, stop, pos); -+ -+ g_free (seek); -+ mp3parse->pending_accurate_seeks = -+ g_slist_delete_link (mp3parse->pending_accurate_seeks, node); -+ -+ g_mutex_unlock (mp3parse->pending_accurate_seeks_lock); -+ res = gst_pad_push_event (mp3parse->srcpad, event); -+ -+ return res; -+ } else { -+ GST_WARNING_OBJECT (mp3parse, -+ "Accurate seek not possible, didn't get an appropiate upstream segment"); -+ } -+ } -+ g_mutex_unlock (mp3parse->pending_accurate_seeks_lock); -+ -+ mp3parse->exact_position = FALSE; -+ - if (format == GST_FORMAT_BYTES) { - GstClockTime seg_start, seg_stop, seg_pos; - -@@ -356,8 +416,9 @@ gst_mp3parse_sink_event (GstPad * pad, G - GST_FORMAT_TIME, seg_start, seg_stop, seg_pos); - format = GST_FORMAT_TIME; - GST_DEBUG_OBJECT (mp3parse, "Converted incoming segment to TIME. " -- "start = %" G_GINT64_FORMAT ", stop = %" G_GINT64_FORMAT -- "pos = %" G_GINT64_FORMAT, seg_start, seg_stop, seg_pos); -+ "start = %" GST_TIME_FORMAT ", stop = %" GST_TIME_FORMAT -+ ", pos = %" GST_TIME_FORMAT, GST_TIME_ARGS (seg_start), -+ GST_TIME_ARGS (seg_stop), GST_TIME_ARGS (seg_pos)); - } - } - -@@ -368,6 +429,7 @@ gst_mp3parse_sink_event (GstPad * pad, G - event = gst_event_new_new_segment_full (update, rate, applied_rate, - GST_FORMAT_TIME, 0, GST_CLOCK_TIME_NONE, 0); - } -+ - mp3parse->resyncing = TRUE; - mp3parse->cur_offset = -1; - mp3parse->next_ts = GST_CLOCK_TIME_NONE; -@@ -379,12 +441,23 @@ gst_mp3parse_sink_event (GstPad * pad, G - GST_DEBUG_OBJECT (mp3parse, "Pushing newseg rate %g, applied rate %g, " - "format %d, start %lld, stop %lld, pos %lld\n", - rate, applied_rate, format, start, stop, pos); -- res = gst_pad_push_event (mp3parse->srcpad, event); -+ -+ gst_segment_set_newsegment_full (&mp3parse->segment, update, rate, -+ applied_rate, format, start, stop, pos); -+ -+ /* save the segment for later, right before we push a new buffer so that -+ * the caps are fixed and the next linked element can receive the segment. */ -+ eventp = &mp3parse->pending_segment; -+ gst_event_replace (eventp, event); -+ gst_event_unref (event); -+ res = TRUE; - break; - } - case GST_EVENT_FLUSH_STOP: - /* Clear our adapter and set up for a new position */ - gst_adapter_clear (mp3parse->adapter); -+ eventp = &mp3parse->pending_segment; -+ gst_event_replace (eventp, NULL); - res = gst_pad_push_event (mp3parse->srcpad, event); - break; - default: -@@ -397,14 +470,26 @@ gst_mp3parse_sink_event (GstPad * pad, G - return res; - } - -+static MPEGAudioSeekEntry * -+mp3parse_seek_table_last_entry (GstMPEGAudioParse * mp3parse) -+{ -+ MPEGAudioSeekEntry *ret = NULL; -+ -+ if (mp3parse->seek_table) { -+ ret = mp3parse->seek_table->data; -+ } -+ -+ return ret; -+} -+ - /* Prepare a buffer of the indicated size, timestamp it and output */ - static GstFlowReturn - gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size) - { - GstBuffer *outbuf; - guint bitrate; -- -- GST_DEBUG_OBJECT (mp3parse, "pushing buffer of %d bytes", size); -+ GstFlowReturn ret = GST_FLOW_OK; -+ GstClockTime push_start; - - outbuf = gst_adapter_take_buffer (mp3parse->adapter, size); - -@@ -456,6 +541,22 @@ gst_mp3parse_emit_frame (GstMPEGAudioPar - } - } - -+ if (GST_BUFFER_TIMESTAMP (outbuf) == 0) -+ mp3parse->exact_position = TRUE; -+ -+ if (mp3parse->exact_position && (!mp3parse->seek_table || -+ (mp3parse_seek_table_last_entry (mp3parse))->byte < -+ GST_BUFFER_OFFSET (outbuf))) { -+ MPEGAudioSeekEntry *entry = g_new0 (MPEGAudioSeekEntry, 1); -+ -+ entry->byte = mp3parse->cur_offset; -+ entry->timestamp = GST_BUFFER_TIMESTAMP (outbuf); -+ mp3parse->seek_table = g_list_prepend (mp3parse->seek_table, entry); -+ GST_DEBUG_OBJECT (mp3parse, "Adding index entry %" GST_TIME_FORMAT -+ " @ offset 0x%08" G_GINT64_MODIFIER "x", -+ GST_TIME_ARGS (entry->timestamp), entry->byte); -+ } -+ - /* Update our byte offset tracking */ - if (mp3parse->cur_offset != -1) { - mp3parse->cur_offset += size; -@@ -483,7 +584,53 @@ gst_mp3parse_emit_frame (GstMPEGAudioPar - mp3parse->srcpad, taglist); - } - -- return gst_pad_push (mp3parse->srcpad, outbuf); -+ /* We start pushing 9 frames earlier (29 frames for MPEG2) than -+ * segment start to be able to decode the first frame we want. -+ * 9 (29) frames are the theoretical maximum of frames that contain -+ * data for the current frame (bit reservoir). -+ */ -+ -+ if (mp3parse->segment.start == 0) { -+ push_start = 0; -+ } else if (GST_CLOCK_TIME_IS_VALID (mp3parse->max_bitreservoir)) { -+ if (mp3parse->segment.start > mp3parse->max_bitreservoir) -+ push_start = mp3parse->segment.start - mp3parse->max_bitreservoir; -+ else -+ push_start = 0; -+ } else { -+ push_start = mp3parse->segment.start; -+ } -+ -+ if (G_UNLIKELY ((GST_CLOCK_TIME_IS_VALID (push_start) && -+ GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf) -+ < push_start) -+ || (GST_CLOCK_TIME_IS_VALID (mp3parse->segment.stop) -+ && GST_BUFFER_TIMESTAMP (outbuf) >= mp3parse->segment.stop))) { -+ GST_DEBUG_OBJECT (mp3parse, -+ "Buffer outside of configured segment range %" GST_TIME_FORMAT -+ " to %" GST_TIME_FORMAT ", dropping, timestamp %" -+ GST_TIME_FORMAT ", offset 0x%08" G_GINT64_MODIFIER "x", -+ GST_TIME_ARGS (push_start), GST_TIME_ARGS (mp3parse->segment.stop), -+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), -+ GST_BUFFER_OFFSET (outbuf)); -+ gst_buffer_unref (outbuf); -+ ret = GST_FLOW_OK; -+ } else { -+ GST_DEBUG_OBJECT (mp3parse, -+ "pushing buffer of %d bytes, timestamp %" GST_TIME_FORMAT -+ ", offset 0x%08" G_GINT64_MODIFIER "x", size, -+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), -+ GST_BUFFER_OFFSET (outbuf)); -+ mp3parse->segment.last_stop = GST_BUFFER_TIMESTAMP (outbuf); -+ /* push any pending segment now */ -+ if (mp3parse->pending_segment) { -+ gst_pad_push_event (mp3parse->srcpad, mp3parse->pending_segment); -+ mp3parse->pending_segment = NULL; -+ } -+ ret = gst_pad_push (mp3parse->srcpad, outbuf); -+ } -+ -+ return ret; - } - - #define XING_FRAMES_FLAG 0x0001 -@@ -610,14 +757,41 @@ gst_mp3parse_handle_first_frame (GstMPEG - mp3parse->xing_bytes = 0; - - if (xing_flags & XING_TOC_FLAG) { -- gint i; -+ int i, percent = 0; -+ guchar *table = mp3parse->xing_seek_table; -+ -+ /* xing seek table: percent time -> 1/256 bytepos */ -+ memcpy (mp3parse->xing_seek_table, data, 100); - -- for (i = 0; i < 100; i++) { -- mp3parse->xing_seek_table[i] = data[0]; -- data++; -+ /* build inverse table: 1/256 bytepos -> 1/100 percent time */ -+ for (i = 0; i < 256; i++) { -+ while (percent < 99 && table[percent + 1] <= i) -+ percent++; -+ -+ if (table[percent] == i) { -+ mp3parse->xing_seek_table_inverse[i] = percent * 100; -+ } else if (table[percent] < i && percent < 99) { -+ gdouble fa, fb, fx; -+ gint a = percent, b = percent + 1; -+ -+ fa = table[a]; -+ fb = table[b]; -+ fx = (b - a) / (fb - fa) * (i - fa) + a; -+ mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100); -+ } else if (percent == 98 && table[percent + 1] <= i) { -+ gdouble fa, fb, fx; -+ gint a = percent + 1, b = 100; -+ -+ fa = table[a]; -+ fb = 256.0; -+ fx = (b - a) / (fb - fa) * (i - fa) + a; -+ mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100); -+ } - } -+ data += 100; - } else { - memset (mp3parse->xing_seek_table, 0, 100); -+ memset (mp3parse->xing_seek_table_inverse, 0, 256); - } - - if (xing_flags & XING_VBR_SCALE_FLAG) { -@@ -777,6 +951,9 @@ gst_mp3parse_chain (GstPad * pad, GstBuf - mp3parse->spf = 1152; - } - -+ mp3parse->max_bitreservoir = gst_util_uint64_scale (GST_SECOND, -+ ((version == 1) ? 10 : 30) * mp3parse->spf, mp3parse->rate); -+ - /* Check the first frame for a Xing header to get our total length */ - if (mp3parse->frame_count == 0) { - /* For the first frame in the file, look for a Xing frame after -@@ -930,6 +1107,45 @@ gst_mp3parse_change_state (GstElement * - return result; - } - -+static gboolean -+mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total) -+{ -+ GstFormat fmt = GST_FORMAT_BYTES; -+ -+ if (gst_pad_query_peer_duration (mp3parse->sinkpad, &fmt, total)) -+ return TRUE; -+ -+ if (mp3parse->xing_flags & XING_BYTES_FLAG) { -+ *total = mp3parse->xing_bytes; -+ return TRUE; -+ } -+ -+ return FALSE; -+} -+ -+static gboolean -+mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total) -+{ -+ gint64 total_bytes; -+ -+ *total = GST_CLOCK_TIME_NONE; -+ -+ if (mp3parse->xing_flags & XING_FRAMES_FLAG) { -+ *total = mp3parse->xing_total_time; -+ return TRUE; -+ } -+ -+ /* Calculate time from the measured bitrate */ -+ if (!mp3parse_total_bytes (mp3parse, &total_bytes)) -+ return FALSE; -+ -+ if (total_bytes != -1 -+ && !mp3parse_bytepos_to_time (mp3parse, total_bytes, total)) -+ return FALSE; -+ -+ return TRUE; -+} -+ - /* Convert a timestamp to the file position required to start decoding that - * timestamp. For now, this just uses the avg bitrate. Later, use an - * incrementally accumulated seek table */ -@@ -937,12 +1153,38 @@ static gboolean - mp3parse_time_to_bytepos (GstMPEGAudioParse * mp3parse, GstClockTime ts, - gint64 * bytepos) - { -+ gint64 total_bytes; -+ GstClockTime total_time; -+ - /* -1 always maps to -1 */ - if (ts == -1) { - *bytepos = -1; - return TRUE; - } - -+ /* If XING seek table exists use this for time->byte conversion */ -+ if ((mp3parse->xing_flags & XING_TOC_FLAG) && -+ mp3parse_total_bytes (mp3parse, &total_bytes) && -+ mp3parse_total_time (mp3parse, &total_time)) { -+ gdouble fa, fb, fx; -+ gdouble percent = -+ CLAMP ((100.0 * gst_util_guint64_to_gdouble (ts)) / -+ gst_util_guint64_to_gdouble (total_time), 0.0, 100.0); -+ gint index = CLAMP (percent, 0, 99); -+ -+ fa = mp3parse->xing_seek_table[index]; -+ if (index < 99) -+ fb = mp3parse->xing_seek_table[index + 1]; -+ else -+ fb = 256.0; -+ -+ fx = fa + (fb - fa) * (percent - index); -+ -+ *bytepos = (1.0 / 256.0) * fx * total_bytes; -+ -+ return TRUE; -+ } -+ - if (mp3parse->avg_bitrate == 0) - goto no_bitrate; - -@@ -958,6 +1200,9 @@ static gboolean - mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse, - gint64 bytepos, GstClockTime * ts) - { -+ gint64 total_bytes; -+ GstClockTime total_time; -+ - if (bytepos == -1) { - *ts = GST_CLOCK_TIME_NONE; - return TRUE; -@@ -968,61 +1213,33 @@ mp3parse_bytepos_to_time (GstMPEGAudioPa - return TRUE; - } - -- /* Cannot convert anything except 0 if we don't have a bitrate yet */ -- if (mp3parse->avg_bitrate == 0) -- return FALSE; -- -- *ts = (GstClockTime) gst_util_uint64_scale (GST_SECOND, bytepos * 8, -- mp3parse->avg_bitrate); -- return TRUE; --} -- --static gboolean --mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total) --{ -- GstQuery *query; -- GstPad *peer; -- -- if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) { -- query = gst_query_new_duration (GST_FORMAT_BYTES); -- gst_query_set_duration (query, GST_FORMAT_BYTES, -1); -- -- if (gst_pad_query (peer, query)) { -- gst_object_unref (peer); -- gst_query_parse_duration (query, NULL, total); -- return TRUE; -- } -- gst_object_unref (peer); -- } -- -- if (mp3parse->xing_flags & XING_BYTES_FLAG) { -- *total = mp3parse->xing_bytes; -- return TRUE; -- } -- -- return FALSE; --} -+ /* If XING seek table exists use this for byte->time conversion */ -+ if ((mp3parse->xing_flags & XING_TOC_FLAG) && -+ mp3parse_total_bytes (mp3parse, &total_bytes) && -+ mp3parse_total_time (mp3parse, &total_time)) { -+ gdouble fa, fb, fx; -+ gdouble pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0); -+ gint index = CLAMP (pos, 0, 255); -+ -+ fa = mp3parse->xing_seek_table_inverse[index]; -+ if (index < 255) -+ fb = mp3parse->xing_seek_table_inverse[index + 1]; -+ else -+ fb = 10000.0; - --static gboolean --mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total) --{ -- gint64 total_bytes; -+ fx = fa + (fb - fa) * (pos - index); - -- *total = GST_CLOCK_TIME_NONE; -+ *ts = (1.0 / 10000.0) * fx * gst_util_guint64_to_gdouble (total_time); - -- if (mp3parse->xing_flags & XING_FRAMES_FLAG) { -- *total = mp3parse->xing_total_time; - return TRUE; - } - -- /* Calculate time from the measured bitrate */ -- if (!mp3parse_total_bytes (mp3parse, &total_bytes)) -- return FALSE; -- -- if (total_bytes != -1 -- && !mp3parse_bytepos_to_time (mp3parse, total_bytes, total)) -+ /* Cannot convert anything except 0 if we don't have a bitrate yet */ -+ if (mp3parse->avg_bitrate == 0) - return FALSE; - -+ *ts = (GstClockTime) gst_util_uint64_scale (GST_SECOND, bytepos * 8, -+ mp3parse->avg_bitrate); - return TRUE; - } - -@@ -1036,11 +1253,12 @@ mp3parse_handle_seek (GstMPEGAudioParse - gint64 cur, stop; - gint64 byte_cur, byte_stop; - -- /* FIXME: Use GstSegment for tracking our position */ -- - gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, - &stop_type, &stop); - -+ GST_DEBUG_OBJECT (mp3parse, "Performing seek to %" GST_TIME_FORMAT, -+ GST_TIME_ARGS (cur)); -+ - /* For any format other than TIME, see if upstream handles - * it directly or fail. For TIME, try upstream, but do it ourselves if - * it fails upstream */ -@@ -1055,6 +1273,90 @@ mp3parse_handle_seek (GstMPEGAudioParse - - /* Handle TIME based seeks by converting to a BYTE position */ - -+ /* For accurate seeking get the frame 9 (MPEG1) or 29 (MPEG2) frames -+ * before the one we want to seek to and push them all to the decoder. -+ * -+ * This is necessary because of the bit reservoir. See -+ * http://www.mars.org/mailman/public/mad-dev/2002-May/000634.html -+ * -+ */ -+ -+ if (flags & GST_SEEK_FLAG_ACCURATE) { -+ MPEGAudioPendingAccurateSeek *seek = -+ g_new0 (MPEGAudioPendingAccurateSeek, 1); -+ GstClockTime start; -+ -+ seek->segment = mp3parse->segment; -+ -+ gst_segment_set_seek (&seek->segment, rate, GST_FORMAT_TIME, -+ flags, cur_type, cur, stop_type, stop, NULL); -+ -+ if (!mp3parse->seek_table) { -+ byte_cur = 0; -+ byte_stop = -1; -+ start = 0; -+ } else { -+ MPEGAudioSeekEntry *entry = NULL, *start_entry = NULL, *stop_entry = NULL; -+ GList *start_node, *stop_node; -+ -+ for (start_node = mp3parse->seek_table; start_node; -+ start_node = start_node->next) { -+ entry = start_node->data; -+ -+ if (cur - mp3parse->max_bitreservoir >= entry->timestamp) { -+ start_entry = entry; -+ break; -+ } -+ } -+ -+ if (!start_entry) { -+ start_entry = mp3parse->seek_table->data; -+ start = start_entry->timestamp; -+ byte_cur = start_entry->byte; -+ } else { -+ start = start_entry->timestamp; -+ byte_cur = start_entry->byte; -+ } -+ -+ for (stop_node = mp3parse->seek_table; stop_node; -+ stop_node = stop_node->next) { -+ entry = stop_node->data; -+ -+ if (stop >= entry->timestamp) { -+ stop_node = stop_node->prev; -+ stop_entry = (stop_node) ? stop_node->data : NULL; -+ break; -+ } -+ } -+ -+ if (!stop_entry) { -+ byte_stop = -1; -+ } else { -+ byte_stop = stop_entry->byte; -+ } -+ -+ } -+ g_mutex_lock (mp3parse->pending_accurate_seeks_lock); -+ event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, -+ byte_cur, stop_type, byte_stop); -+ if (gst_pad_push_event (mp3parse->sinkpad, event)) { -+ mp3parse->exact_position = TRUE; -+ seek->upstream_start = byte_cur; -+ seek->timestamp_start = start; -+ mp3parse->pending_accurate_seeks = -+ g_slist_prepend (mp3parse->pending_accurate_seeks, seek); -+ g_mutex_unlock (mp3parse->pending_accurate_seeks_lock); -+ return TRUE; -+ } else { -+ g_mutex_unlock (mp3parse->pending_accurate_seeks_lock); -+ mp3parse->exact_position = TRUE; -+ g_free (seek); -+ return TRUE; -+ } -+ } -+ -+ mp3parse->exact_position = FALSE; -+ - /* Convert the TIME to the appropriate BYTE position at which to resume - * decoding. */ - if (!mp3parse_time_to_bytepos (mp3parse, (GstClockTime) cur, &byte_cur)) |