diff options
| -rw-r--r-- | ChangeLog | 34 | ||||
| -rw-r--r-- | ext/wavpack/gstwavpackdec.c | 169 | ||||
| -rw-r--r-- | ext/wavpack/gstwavpackparse.c | 972 | ||||
| -rw-r--r-- | ext/wavpack/gstwavpackparse.h | 46 |
4 files changed, 713 insertions, 508 deletions
@@ -1,3 +1,35 @@ +2006-01-29 Tim-Philipp Müller <tim at centricular dot net> + + * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_setcaps), + (gst_wavpack_dec_base_init), (gst_wavpack_dec_dispose), + (gst_wavpack_dec_class_init), (gst_wavpack_dec_sink_event), + (gst_wavpack_dec_init), (gst_wavpack_dec_format_samples), + (gst_wavpack_dec_chain), (gst_wavpack_dec_plugin_init): + Add debug category, use boilerplate macros, fix handling + of widths of 32 bits. + + * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init), + (gst_wavpack_parse_dispose), (gst_wavpack_parse_class_init), + (gst_wavpack_parse_index_get_last_entry), + (gst_wavpack_parse_index_get_entry_from_sample), + (gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset), + (gst_wavpack_parse_src_query), + (gst_wavpack_parse_scan_to_find_sample), + (gst_wavpack_parse_send_newsegment), + (gst_wavpack_parse_handle_seek_event), + (gst_wavpack_parse_src_event), (gst_wavpack_parse_init), + (gst_wavpack_parse_get_upstream_length), + (gst_wavpack_parse_pull_buffer), + (gst_wavpack_parse_create_src_pad), (gst_wavpack_parse_loop), + (gst_wavpack_parse_change_state), + (gst_wavepack_parse_sink_activate), + (gst_wavepack_parse_sink_activate_pull), + (gst_wavpack_parse_plugin_init): + * ext/wavpack/gstwavpackparse.h: + Rewrite a bit, mostly to fix flow logic and to make seeking work. + Fix buffer/event refcounting. Add some debug statements. Add + width of 32 to source pad template caps. Use boilerplate macros. + 2006-01-28 Edward Hervey <edward@fluendo.com> * sys/glsink/Makefile.am: @@ -17,7 +49,7 @@ * ext/faad/gstfaad.c: (gst_faad_setcaps), (gst_faad_chanpos_to_gst), (gst_faad_sync), (gst_faad_chain): Handle 'framed' field in caps; Port syncing for raw streams - from 0.8 branch (for AAC+ radio streams) (#328722). + from 0.8 branch (for AAC+ radio streams) (#328854, #328721). 2006-01-27 Jan Schmidt <thaytan@mad.scientist.com> diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c index d8a4308e..a7df49c6 100644 --- a/ext/wavpack/gstwavpackdec.c +++ b/ext/wavpack/gstwavpackdec.c @@ -28,16 +28,8 @@ #include "gstwavpackdec.h" #include "gstwavpackcommon.h" -/* Filter signals and args */ -enum -{ - LAST_SIGNAL -}; - -enum -{ - ARG_0 -}; +GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug); +#define GST_CAT_DEFAULT gst_wavpack_dec_debug static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, @@ -59,63 +51,66 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " - "width = (int) { 8, 16, 24 }, " - "depth = (int) { 8, 16, 24 }, " + "width = (int) { 8, 16, 24, 32 }, " + "depth = (int) { 8, 16, 24, 32 }, " "channels = (int) { 1, 2 }, " "rate = (int) [ 6000, 192000 ], " - "endianness = (int) LITTLE_ENDIAN, " - "signed = (boolean) true;" + "endianness = (int) LITTLE_ENDIAN, " "signed = (boolean) true") +/* "audio/x-raw-float, " "width = (int) 32, " "channels = (int) { 1, 2 }, " - "rate = (int) [ 6000, 192000 ], " "endianness = (int) LITTLE_ENDIAN") + "rate = (int) [ 6000, 192000 ], " "endianness = (int) LITTLE_ENDIAN" +*/ ); -static void gst_wavpack_dec_class_init (GstWavpackDecClass * klass); -static void gst_wavpack_dec_base_init (GstWavpackDecClass * klass); -static void gst_wavpack_dec_init (GstWavpackDec * wavpackdec); - static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer); +static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event); -static GstElementClass *parent = NULL; +GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT) -static GstPadLinkReturn -gst_wavpack_dec_link (GstPad * pad, GstPad * peer) + static gboolean gst_wavpack_dec_setcaps (GstPad * pad, GstCaps * caps) { GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad)); GstStructure *structure; GstCaps *srccaps; - gint bits; + gint bits, rate, channels; - if (!gst_caps_is_fixed (GST_PAD_CAPS (peer))) - return GST_PAD_LINK_REFUSED; + structure = gst_caps_get_structure (caps, 0); + if (!gst_structure_get_int (structure, "rate", &rate) || + !gst_structure_get_int (structure, "channels", &channels) || + !gst_structure_get_int (structure, "width", &bits)) { + return FALSE; + } - structure = gst_caps_get_structure (GST_PAD_CAPS (peer), 0); - gst_structure_get_int (structure, "rate", - (gint32 *) & wavpackdec->samplerate); - gst_structure_get_int (structure, "channels", - (gint *) & wavpackdec->channels); - gst_structure_get_int (structure, "width", &bits); + wavpackdec->samplerate = rate; + wavpackdec->channels = channels; wavpackdec->width = bits; - if (bits != 32) { - srccaps = gst_caps_new_simple ("audio/x-raw-int", - "rate", G_TYPE_INT, wavpackdec->samplerate, - "channels", G_TYPE_INT, wavpackdec->channels, - "depth", G_TYPE_INT, bits, - "width", G_TYPE_INT, bits, - "endianness", G_TYPE_INT, LITTLE_ENDIAN, - "signed", G_TYPE_BOOLEAN, TRUE, NULL); +/* 32-bit output seems to be in fact 32 bit int (e.g. Prod_Girls.wv) */ +/* if (bits != 32) { */ + srccaps = gst_caps_new_simple ("audio/x-raw-int", + "rate", G_TYPE_INT, wavpackdec->samplerate, + "channels", G_TYPE_INT, wavpackdec->channels, + "depth", G_TYPE_INT, bits, + "width", G_TYPE_INT, bits, + "endianness", G_TYPE_INT, G_LITTLE_ENDIAN, + "signed", G_TYPE_BOOLEAN, TRUE, NULL); +/* } else { srccaps = gst_caps_new_simple ("audio/x-raw-float", "rate", G_TYPE_INT, wavpackdec->samplerate, "channels", G_TYPE_INT, wavpackdec->channels, - "width", G_TYPE_INT, 32, "endianness", G_TYPE_INT, LITTLE_ENDIAN, NULL); + "width", G_TYPE_INT, 32, + "endianness", G_TYPE_INT, G_LITTLE_ENDIAN, NULL); } +*/ +/* gst_pad_set_caps (wavpackdec->sinkpad, caps); */ + gst_pad_set_caps (wavpackdec->srcpad, srccaps); gst_pad_use_fixed_caps (wavpackdec->srcpad); - return GST_PAD_LINK_OK; + return TRUE; } #if 0 @@ -129,31 +124,8 @@ gst_wavpack_dec_wvclink (GstPad * pad, GstPad * peer) } #endif -GType -gst_wavpack_dec_get_type (void) -{ - static GType plugin_type = 0; - - if (!plugin_type) { - static const GTypeInfo plugin_info = { - sizeof (GstWavpackDecClass), - (GBaseInitFunc) gst_wavpack_dec_base_init, - NULL, - (GClassInitFunc) gst_wavpack_dec_class_init, - NULL, - NULL, - sizeof (GstWavpackDec), - 0, - (GInstanceInitFunc) gst_wavpack_dec_init, - }; - plugin_type = g_type_register_static (GST_TYPE_ELEMENT, - "GstWavpackDec", &plugin_info, 0); - } - return plugin_type; -} - static void -gst_wavpack_dec_base_init (GstWavpackDecClass * klass) +gst_wavpack_dec_base_init (gpointer klass) { static GstElementDetails plugin_details = { "WAVPACK decoder", @@ -178,8 +150,11 @@ gst_wavpack_dec_dispose (GObject * object) GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (object); g_free (wavpackdec->decodebuf); + wavpackdec->decodebuf = NULL; - G_OBJECT_CLASS (parent)->dispose (object); + /* FIXME: what about wavpackdec->stream and wavpackdec->context? (tpm) */ + + G_OBJECT_CLASS (parent_class)->dispose (object); } static void @@ -191,8 +166,6 @@ gst_wavpack_dec_class_init (GstWavpackDecClass * klass) gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; - parent = g_type_class_ref (GST_TYPE_ELEMENT); - gobject_class->dispose = gst_wavpack_dec_dispose; } @@ -202,8 +175,31 @@ gst_wavpack_dec_src_query (GstPad * pad, GstQuery * query) return gst_pad_query_default (pad, query); } +static gboolean +gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event) +{ + GstWavpackDec *dec; + + dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad)); + + GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event)); + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_NEWSEGMENT:{ + /* TODO: save current segment so we can do clipping, for now + * we'll just leave the clipping to the audio sink */ + break; + } + default: + break; + } + + gst_object_unref (dec); + + return gst_pad_event_default (pad, event); +} + static void -gst_wavpack_dec_init (GstWavpackDec * wavpackdec) +gst_wavpack_dec_init (GstWavpackDec * wavpackdec, GstWavpackDecClass * gklass) { GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpackdec); @@ -212,8 +208,12 @@ gst_wavpack_dec_init (GstWavpackDec * wavpackdec) "sink"), "sink"); gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->sinkpad); - gst_pad_set_chain_function (wavpackdec->sinkpad, gst_wavpack_dec_chain); - gst_pad_set_link_function (wavpackdec->sinkpad, gst_wavpack_dec_link); + gst_pad_set_chain_function (wavpackdec->sinkpad, + GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain)); + gst_pad_set_setcaps_function (wavpackdec->sinkpad, + GST_DEBUG_FUNCPTR (gst_wavpack_dec_setcaps)); + gst_pad_set_event_function (wavpackdec->sinkpad, + GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event)); #if 0 wavpackdec->wvcsinkpad = @@ -229,7 +229,8 @@ gst_wavpack_dec_init (GstWavpackDec * wavpackdec) "src"), "src"); gst_pad_use_fixed_caps (wavpackdec->srcpad); - gst_pad_set_query_function (wavpackdec->srcpad, gst_wavpack_dec_src_query); + gst_pad_set_query_function (wavpackdec->srcpad, + GST_DEBUG_FUNCPTR (gst_wavpack_dec_src_query)); gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->srcpad); @@ -290,6 +291,7 @@ gst_wavpack_dec_format_samples (GstWavpackDec * wavpackdec, int32_t * samples, buf = gst_buffer_new_and_alloc (num_samples * wavpackdec->width / 8 * wavpackdec->channels); + dst = (guint8 *) GST_BUFFER_DATA (buf); switch (wavpackdec->width) { @@ -353,8 +355,7 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf) gst_wavpack_dec_format_samples (wavpackdec, wavpackdec->decodebuf, wavpackdec->context->streams[0]->wphdr.block_samples); - GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf); - GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf); + gst_buffer_stamp (outbuf, buf); gst_buffer_unref (buf); if (cbuf) { @@ -362,17 +363,27 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf) } gst_buffer_set_caps (outbuf, GST_PAD_CAPS (wavpackdec->srcpad)); - if (GST_FLOW_OK != (ret = gst_pad_push (wavpackdec->srcpad, outbuf))) { - gst_buffer_unref (outbuf); + + GST_LOG_OBJECT (wavpackdec, "pushing buffer with time %" GST_TIME_FORMAT, + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf))); + + ret = gst_pad_push (wavpackdec->srcpad, outbuf); + if (ret != GST_FLOW_OK) { + GST_DEBUG_OBJECT (wavpackdec, "pad_push: %s", gst_flow_get_name (ret)); } return ret; - } gboolean gst_wavpack_dec_plugin_init (GstPlugin * plugin) { - return gst_element_register (plugin, "wavpackdec", - GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC); + if (!gst_element_register (plugin, "wavpackdec", + GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC)) + return FALSE; + + GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0, + "wavpack decoder"); + + return TRUE; } diff --git a/ext/wavpack/gstwavpackparse.c b/ext/wavpack/gstwavpackparse.c index 5621d4e6..f00e337e 100644 --- a/ext/wavpack/gstwavpackparse.c +++ b/ext/wavpack/gstwavpackparse.c @@ -1,5 +1,6 @@ /* GStreamer wavpack plugin * (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net> + * (c) 2006 Tim-Philipp Müller <tim centricular net> * * gstwavpackparse.c: wavpack file parser * @@ -31,17 +32,6 @@ GST_DEBUG_CATEGORY_STATIC (gst_wavpack_parse_debug); #define GST_CAT_DEFAULT gst_wavpack_parse_debug -/* Filter signals and args */ -enum -{ - LAST_SIGNAL -}; - -enum -{ - ARG_0 -}; - static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, @@ -54,7 +44,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("audio/x-wavpack, " - "width = (int) { 8, 16, 24 }, " + "width = (int) { 8, 16, 24, 32 }, " "channels = (int) { 1, 2 }, " "rate = (int) [ 6000, 192000 ], " "framed = (boolean) true") ); @@ -65,47 +55,22 @@ static GstStaticPadTemplate wvc_src_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc", GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) true") ); -static void gst_wavpack_parse_class_init (GstWavpackParseClass * klass); -static void gst_wavpack_parse_base_init (GstWavpackParseClass * klass); -static void gst_wavpack_parse_init (GstWavpackParse * wavpackparse); - static gboolean gst_wavepack_parse_sink_activate (GstPad * sinkpad); static gboolean gst_wavepack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active); -static gboolean gst_wavpack_parse_sink_event (GstPad * pad, GstEvent * event); - static void gst_wavpack_parse_loop (GstElement * element); static GstStateChangeReturn gst_wavpack_parse_change_state (GstElement * element, GstStateChange transition); +static void gst_wavpack_parse_reset (GstWavpackParse * wavpackparse); +static gint64 gst_wavpack_parse_get_upstream_length (GstWavpackParse * wvparse); +static GstBuffer *gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse, + gint64 offset, guint size, GstFlowReturn * flow); -static GstElementClass *parent = NULL; - -GType -gst_wavpack_parse_get_type (void) -{ - static GType plugin_type = 0; - - if (!plugin_type) { - static const GTypeInfo plugin_info = { - sizeof (GstWavpackParseClass), - (GBaseInitFunc) gst_wavpack_parse_base_init, - NULL, - (GClassInitFunc) gst_wavpack_parse_class_init, - NULL, - NULL, - sizeof (GstWavpackParse), - 0, - (GInstanceInitFunc) gst_wavpack_parse_init, - }; - plugin_type = g_type_register_static (GST_TYPE_ELEMENT, - "GstWavpackParse", &plugin_info, 0); - } - return plugin_type; -} +GST_BOILERPLATE (GstWavpackParse, gst_wavpack_parse, GstElement, + GST_TYPE_ELEMENT) -static void -gst_wavpack_parse_base_init (GstWavpackParseClass * klass) + static void gst_wavpack_parse_base_init (gpointer klass) { static GstElementDetails plugin_details = { "Wavpack file parser", @@ -127,7 +92,8 @@ gst_wavpack_parse_base_init (GstWavpackParseClass * klass) static void gst_wavpack_parse_dispose (GObject * object) { - G_OBJECT_CLASS (parent)->dispose (object); + gst_wavpack_parse_reset (GST_WAVPACK_PARSE (object)); + G_OBJECT_CLASS (parent_class)->dispose (object); } static void @@ -139,513 +105,703 @@ gst_wavpack_parse_class_init (GstWavpackParseClass * klass) gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; - parent = g_type_class_ref (GST_TYPE_ELEMENT); - gobject_class->dispose = gst_wavpack_parse_dispose; - gstelement_class->change_state = gst_wavpack_parse_change_state; + gstelement_class->change_state = + GST_DEBUG_FUNCPTR (gst_wavpack_parse_change_state); +} + +static GstWavpackParseIndexEntry * +gst_wavpack_parse_index_get_last_entry (GstWavpackParse * wvparse) +{ + gint last; + + g_assert (wvparse->entries != NULL); + g_assert (wvparse->entries->len > 0); + + last = wvparse->entries->len - 1; + return &g_array_index (wvparse->entries, GstWavpackParseIndexEntry, last); +} + +static GstWavpackParseIndexEntry * +gst_wavpack_parse_index_get_entry_from_sample (GstWavpackParse * wvparse, + gint64 sample_offset) +{ + gint i; + + if (wvparse->entries == NULL || wvparse->entries->len == 0) + return NULL; + + for (i = wvparse->entries->len - 1; i >= 0; --i) { + GstWavpackParseIndexEntry *entry; + + entry = &g_array_index (wvparse->entries, GstWavpackParseIndexEntry, i); + + GST_LOG_OBJECT (wvparse, "Index entry %03u: sample %" G_GINT64_FORMAT " @" + " byte %" G_GINT64_FORMAT, entry->sample_offset, entry->byte_offset); + + if (entry->sample_offset <= sample_offset && + sample_offset < entry->sample_offset_end) { + GST_LOG_OBJECT (wvparse, "found match"); + return entry; + } + } + GST_LOG_OBJECT (wvparse, "no match in index"); + return NULL; +} + +static void +gst_wavpack_parse_index_append_entry (GstWavpackParse * wvparse, + gint64 byte_offset, gint64 sample_offset, gint64 num_samples) +{ + GstWavpackParseIndexEntry entry; + + if (wvparse->entries == NULL) { + wvparse->entries = g_array_new (FALSE, TRUE, + sizeof (GstWavpackParseIndexEntry)); + } else { + /* do we have this one already? */ + entry = *gst_wavpack_parse_index_get_last_entry (wvparse); + if (entry.byte_offset >= byte_offset) + return; + } + + GST_LOG_OBJECT (wvparse, "Adding index entry %8" G_GINT64_FORMAT " - %" + GST_TIME_FORMAT " @ offset 0x%08" G_GINT64_MODIFIER "x", sample_offset, + GST_TIME_ARGS (gst_util_uint64_scale_int (sample_offset, + GST_SECOND, wvparse->samplerate)), byte_offset); + + entry.byte_offset = byte_offset; + entry.sample_offset = sample_offset; + entry.sample_offset_end = sample_offset + num_samples; + g_array_append_val (wvparse->entries, entry); +} + +static void +gst_wavpack_parse_reset (GstWavpackParse * wavpackparse) +{ + wavpackparse->total_samples = 0; + wavpackparse->samplerate = 0; + wavpackparse->channels = 0; + + gst_segment_init (&wavpackparse->segment, GST_FORMAT_UNDEFINED); + + wavpackparse->current_offset = 0; + wavpackparse->need_newsegment = TRUE; + wavpackparse->upstream_length = -1; + + if (wavpackparse->entries) { + g_array_free (wavpackparse->entries, TRUE); + wavpackparse->entries = NULL; + } + + if (wavpackparse->srcpad != NULL) { + gboolean res; + + GST_DEBUG_OBJECT (wavpackparse, "Removing src pad"); + res = gst_element_remove_pad (GST_ELEMENT (wavpackparse), + wavpackparse->srcpad); + g_return_if_fail (res != FALSE); + gst_object_unref (wavpackparse->srcpad); + wavpackparse->srcpad = NULL; + } } static gboolean gst_wavpack_parse_src_query (GstPad * pad, GstQuery * query) { GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); - GstFormat format = GST_FORMAT_DEFAULT; - gint64 value; + GstFormat format; gboolean ret = FALSE; switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_POSITION: - gst_query_parse_position (query, &format, &value); - if (format == GST_FORMAT_TIME) { - value = wavpackparse->timestamp; - gst_query_set_duration (query, format, value); - gst_object_unref (wavpackparse); - ret = TRUE; + case GST_QUERY_POSITION:{ + gint64 cur, len; + guint rate; + + GST_OBJECT_LOCK (wavpackparse); + cur = wavpackparse->segment.last_stop; + len = wavpackparse->total_samples; + rate = wavpackparse->samplerate; + GST_OBJECT_UNLOCK (wavpackparse); + + if (len <= 0 || rate == 0) { + GST_DEBUG_OBJECT (wavpackparse, "haven't read header yet"); break; } - break; - case GST_QUERY_DURATION: - gst_query_parse_duration (query, &format, &value); - - if (format == GST_FORMAT_TIME) { - if (wavpackparse->total_samples == 0) { - value = 0; - gst_query_set_duration (query, format, value); - gst_object_unref (wavpackparse); - ret = FALSE; + + gst_query_parse_position (query, &format, NULL); + + switch (format) { + case GST_FORMAT_TIME: + cur = gst_util_uint64_scale_int (cur, GST_SECOND, rate); + gst_query_set_position (query, GST_FORMAT_TIME, cur); + ret = TRUE; + break; + case GST_FORMAT_DEFAULT: + gst_query_set_position (query, GST_FORMAT_DEFAULT, cur); + ret = TRUE; break; - } - value = ((gdouble) wavpackparse->total_samples / - (gdouble) wavpackparse->samplerate) * GST_SECOND; - gst_query_set_duration (query, format, value); - gst_object_unref (wavpackparse); - ret = TRUE; + default: + GST_DEBUG_OBJECT (wavpackparse, "cannot handle position query in " + "%s format", gst_format_get_name (format)); + break; + } + break; + } + case GST_QUERY_DURATION:{ + gint64 len; + guint rate; + + GST_OBJECT_LOCK (wavpackparse); + rate = wavpackparse->samplerate; + len = wavpackparse->total_samples; + GST_OBJECT_UNLOCK (wavpackparse); + + if (len <= 0 || rate == 0) { + GST_DEBUG_OBJECT (wavpackparse, "haven't read header yet"); break; } + + gst_query_parse_duration (query, &format, NULL); + + switch (format) { + case GST_FORMAT_TIME: + len = gst_util_uint64_scale_int (len, GST_SECOND, rate); + gst_query_set_duration (query, GST_FORMAT_TIME, len); + ret = TRUE; + break; + case GST_FORMAT_DEFAULT: + gst_query_set_duration (query, GST_FORMAT_DEFAULT, len); + ret = TRUE; + break; + default: + GST_DEBUG_OBJECT (wavpackparse, "cannot handle duration query in " + "%s format", gst_format_get_name (format)); + break; + } break; - default: - gst_object_unref (wavpackparse); + } + default:{ ret = gst_pad_query_default (pad, query); break; + } } + gst_object_unref (wavpackparse); return ret; } +/* returns TRUE on success, with byte_offset set to the offset of the + * wavpack chunk containing the sample requested. start_sample will be + * set to the first sample in the chunk starting at byte_offset. + * Scanning from the last known header offset to the wanted position + * when seeking forward isn't very clever, but seems fast enough in + * practice and has the nice side effect of populating our index + * table */ static gboolean -gst_wavpack_parse_src_event (GstPad * pad, GstEvent * event) +gst_wavpack_parse_scan_to_find_sample (GstWavpackParse * parse, + gint64 sample, gint64 * byte_offset, gint64 * start_sample) { - GstWavpackParse *wavpackparse; - GstSeekType type; - GstFormat format; - gboolean need_flush; - gint64 offset, dest; - GstSeekFlags flags; - gboolean ret = TRUE; + GstWavpackParseIndexEntry *entry; + GstFlowReturn ret; + gint64 off = 0; + + /* first, check if we have to scan at all */ + entry = gst_wavpack_parse_index_get_entry_from_sample (parse, sample); + if (entry) { + *byte_offset = entry->byte_offset; + *start_sample = entry->sample_offset; + GST_LOG_OBJECT (parse, "Found index entry: sample %" G_GINT64_FORMAT + " @ offset %" G_GINT64_FORMAT, entry->sample_offset, + entry->byte_offset); + return TRUE; + } - wavpackparse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); + GST_LOG_OBJECT (parse, "No matching entry in index, scanning file ..."); - if (GST_EVENT_TYPE (event) != GST_EVENT_SEEK) { - GstPad *peer; + /* if we have an index, we can start scanning from the last known offset + * in there, after all we know our wanted sample is not in the index */ + if (parse->entries && parse->entries->len > 0) { + GstWavpackParseIndexEntry *entry; - if (!(peer = gst_pad_get_peer (wavpackparse->sinkpad))) { - ret = FALSE; - goto done; - } - ret = gst_pad_send_event (peer, event); - gst_object_unref (peer); - goto done; + entry = gst_wavpack_parse_index_get_last_entry (parse); + off = entry->byte_offset; } - gst_event_parse_seek (event, NULL, &format, &flags, &type, &offset, NULL, - NULL); + /* now scan forward until we find the chunk we're looking for or hit EOS */ + do { + WavpackHeader header = { {0,} + , 0, }; + GstBuffer *buf; - need_flush = flags & GST_SEEK_FLAG_FLUSH; + buf = gst_wavpack_parse_pull_buffer (parse, off, sizeof (WavpackHeader), + &ret); + if (buf == NULL) + break; + gst_wavpack_read_header (&header, GST_BUFFER_DATA (buf)); + gst_buffer_unref (buf); - if (offset < 0 || type != GST_SEEK_TYPE_SET) { - ret = FALSE; - goto done; - } + gst_wavpack_parse_index_append_entry (parse, off, header.block_index, + header.block_samples); - if (format == GST_FORMAT_TIME) { - dest = offset * wavpackparse->samplerate / GST_SECOND; - } else if (format == GST_FORMAT_DEFAULT) { - dest = offset; - } else { - ret = FALSE; - goto done; - } + if (header.block_index <= sample && + sample < (header.block_index + header.block_samples)) { + *byte_offset = off; + *start_sample = header.block_index; + return TRUE; + } - wavpackparse->need_discont = TRUE; - wavpackparse->need_flush = need_flush; + off += header.ckSize + 8; + } while (1); - wavpackparse->seek_pending = TRUE; - wavpackparse->seek_offset = dest; + GST_DEBUG_OBJECT (parse, "scan failed: %s (off=0x%08" G_GINT64_MODIFIER "x)", + gst_flow_get_name (ret), off); -done: - gst_event_unref (event); - gst_object_unref (wavpackparse); - return ret; + return FALSE; } -#define BUFSIZE 4096 - -static guint64 -find_header (GstWavpackParse * wavpackparse, guint64 filepos, - WavpackHeader * wphdr) +static gboolean +gst_wavpack_parse_send_newsegment (GstWavpackParse * wvparse, gboolean update) { - guint64 pos = filepos; - gint read = 0; - GstBuffer *buf = NULL; - - while (TRUE) { - guint8 *cur; - - if (GST_FLOW_OK != gst_pad_pull_range (wavpackparse->sinkpad, - wavpackparse->flushed_bytes + filepos, BUFSIZE, &buf)) { - wavpackparse->eos = TRUE; - gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ()); - return 0; - } - read = GST_BUFFER_SIZE (buf); - - if (read == 0) { - wavpackparse->eos = TRUE; - gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ()); - gst_buffer_unref (buf); - return 0; - } + GstSegment *s = &wvparse->segment; + gboolean ret; + gint64 stop_time = -1; + gint64 start_time = 0; + gint64 cur_pos_time; + gint64 diff; + + /* segment is in DEFAULT format, but we want to send a TIME newsegment */ + start_time = gst_util_uint64_scale_int (s->start, GST_SECOND, + wvparse->samplerate); + + if (s->stop != -1) { + stop_time = gst_util_uint64_scale_int (s->stop, GST_SECOND, + wvparse->samplerate); + } - cur = GST_BUFFER_DATA (buf); - do { - if (cur[0] == 'w' && cur[1] == 'v' && cur[2] == 'p' && cur[3] == 'k') { - gst_wavpack_read_header (wphdr, cur); - pos += (cur - GST_BUFFER_DATA (buf)); - gst_buffer_unref (buf); - buf = NULL; - return pos; - } - cur++; - } while ((cur - GST_BUFFER_DATA (buf)) < - (BUFSIZE - sizeof (WavpackHeader))); + GST_DEBUG_OBJECT (wvparse, "sending newsegment from %" GST_TIME_FORMAT + " to %" GST_TIME_FORMAT, GST_TIME_ARGS (start_time), + GST_TIME_ARGS (stop_time)); - wavpackparse->flushed_bytes += BUFSIZE - sizeof (WavpackHeader); + /* after a seek, s->last_stop will point to a chunk boundary, ie. from + * which sample we will start sending data again, while s->start will + * point to the sample we actually want to seek to and want to start + * playing right after the seek. Adjust clock-time for the difference + * so we start playing from start_time */ + cur_pos_time = gst_util_uint64_scale_int (s->last_stop, GST_SECOND, + wvparse->samplerate); + diff = start_time - cur_pos_time; - pos += BUFSIZE - sizeof (WavpackHeader); - } + ret = gst_pad_push_event (wvparse->srcpad, + gst_event_new_new_segment (update, s->rate, GST_FORMAT_TIME, + start_time, stop_time, start_time - diff)); - /* never reached */ - - if (buf) { - gst_buffer_unref (buf); - buf = NULL; - } - return wavpackparse->duration - wavpackparse->flushed_bytes; + return ret; } -/* find the position of sample in the input bytestream, adapted from the - * original wavpack find_sample function */ -static guint64 -find_sample (GstWavpackParse * wavpackparse, guint32 sample) +static gboolean +gst_wavpack_parse_handle_seek_event (GstWavpackParse * wvparse, + GstEvent * event) { - WavpackHeader wphdr; - guint64 file_pos1 = 0; - guint64 file_pos2 = wavpackparse->duration - wavpackparse->flushed_bytes; - guint64 sample_pos1 = 0, sample_pos2 = wavpackparse->total_samples; - double ratio = 0.96; - int file_skip = 0; - - if (sample >= wavpackparse->total_samples) { - return wavpackparse->duration - wavpackparse->flushed_bytes; - } - - while (1) { - double bytes_per_sample; - guint64 seek_pos; - - bytes_per_sample = file_pos2 - file_pos1; - bytes_per_sample /= sample_pos2 - sample_pos1; - seek_pos = file_pos1 + (file_skip ? 32 : 0); - seek_pos += (guint64) (bytes_per_sample * (sample - sample_pos1) * ratio); - seek_pos = find_header (wavpackparse, seek_pos, &wphdr); - - if (seek_pos == wavpackparse->duration - wavpackparse->flushed_bytes - || seek_pos >= file_pos2) { - if (ratio > 0.0) { - if ((ratio -= 0.24) < 0.0) - ratio = 0.0; - } else { - return wavpackparse->duration - wavpackparse->flushed_bytes; - } - } else if (wphdr.block_index > sample) { - sample_pos2 = wphdr.block_index; - file_pos2 = seek_pos; - } else if (wphdr.block_index + wphdr.block_samples <= sample) { - if (seek_pos == file_pos1) - file_skip = 1; - else { - sample_pos1 = wphdr.block_index; - file_pos1 = seek_pos; - } - } else { - return seek_pos; - } + GstSeekFlags seek_flags; + GstSeekType start_type; + GstSeekType stop_type; + GstSegment segment; + GstFormat format; + gboolean only_update; + gboolean flush, ret; + gdouble speed; + gint64 stop; + gint64 start; /* sample we want to seek to */ + gint64 byte_offset; /* byte offset the chunk we seek to starts at */ + gint64 chunk_start; /* first sample in chunk we seek to */ + guint rate; + + gst_event_parse_seek (event, &speed, &format, &seek_flags, &start_type, + &start, &stop_type, &stop); + + if (format != GST_FORMAT_DEFAULT && format != GST_FORMAT_TIME) { + GST_DEBUG ("seeking is only supported in TIME or DEFAULT format"); + return FALSE; } -} + if (speed < 0.0) { + GST_DEBUG ("only forward playback supported, rate %f not allowed", speed); + return FALSE; + } -static void -gst_wavpack_parse_seek (GstWavpackParse * wavpackparse) -{ - GstBuffer *buf; - gint num; - WavpackHeader *header = g_malloc (sizeof (WavpackHeader)); - - guint64 offset = find_sample (wavpackparse, wavpackparse->seek_offset); + GST_OBJECT_LOCK (wvparse); - if (offset >= wavpackparse->duration - wavpackparse->flushed_bytes) { - /* seek failed or went beyond the end, go EOS */ - wavpackparse->timestamp = - ((gdouble) wavpackparse->total_samples / - (gdouble) wavpackparse->samplerate) * GST_SECOND; - return; + rate = wvparse->samplerate; + if (rate == 0) { + GST_OBJECT_UNLOCK (wvparse); + GST_DEBUG ("haven't read header yet"); + return FALSE; } - if (GST_FLOW_OK != gst_pad_pull_range (wavpackparse->sinkpad, - wavpackparse->flushed_bytes + offset, sizeof (WavpackHeader), &buf)) { - wavpackparse->eos = TRUE; - gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ()); - return; + /* convert from time to samples if necessary */ + if (format == GST_FORMAT_TIME) { + if (start_type != GST_SEEK_TYPE_NONE) + start = gst_util_uint64_scale_int (start, rate, GST_SECOND); + if (stop_type != GST_SEEK_TYPE_NONE) + stop = gst_util_uint64_scale_int (stop, rate, GST_SECOND); } - num = GST_BUFFER_SIZE (buf); - if (num != sizeof (WavpackHeader)) { - wavpackparse->eos = TRUE; - gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ()); - return; + flush = ((seek_flags & GST_SEEK_FLAG_FLUSH) != 0); + + if (start < 0) { + GST_OBJECT_UNLOCK (wvparse); + GST_DEBUG_OBJECT (wvparse, "Invalid start sample %" G_GINT64_FORMAT, start); + return FALSE; } - gst_wavpack_read_header (header, GST_BUFFER_DATA (buf)); - gst_buffer_unref (buf); + /* operate on segment copy until we know the seek worked */ + segment = wvparse->segment; - if (wavpackparse->need_flush) { - wavpackparse->need_flush = FALSE; - gst_pad_push_event (wavpackparse->srcpad, gst_event_new_flush_start ()); - } + gst_segment_set_seek (&segment, speed, GST_FORMAT_DEFAULT, + seek_flags, start_type, start, stop_type, stop, &only_update); - wavpackparse->need_discont = TRUE; - wavpackparse->timestamp = - ((gdouble) header->block_index / (gdouble) wavpackparse->samplerate) * - GST_SECOND; -} +#if 0 + if (only_update) { + wvparse->segment = segment; + gst_wavpack_parse_send_newsegment (wvparse, TRUE); + goto done; + } +#endif -static void -gst_wavpack_parse_init (GstWavpackParse * wavpackparse) -{ - GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpackparse); + gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_start ()); - wavpackparse->duration = -1; - wavpackparse->flushed_bytes = -1; - wavpackparse->eos = FALSE; + if (flush) { + gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_start ()); + } else { + gst_pad_stop_task (wvparse->sinkpad); + } - wavpackparse->sinkpad = - gst_pad_new_from_template (gst_element_class_get_pad_template (klass, - "sink"), "sink"); + GST_PAD_STREAM_LOCK (wvparse->sinkpad); - gst_pad_set_event_function (wavpackparse->sinkpad, - gst_wavpack_parse_sink_event); + gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_stop ()); - gst_element_add_pad (GST_ELEMENT (wavpackparse), wavpackparse->sinkpad); + if (flush) { + gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_stop ()); + } - gst_pad_set_activate_function (wavpackparse->sinkpad, - gst_wavepack_parse_sink_activate); + GST_DEBUG_OBJECT (wvparse, "Performing seek to %" GST_TIME_FORMAT " sample %" + G_GINT64_FORMAT, GST_TIME_ARGS (segment.start * GST_SECOND / rate), + start); + + ret = gst_wavpack_parse_scan_to_find_sample (wvparse, segment.start, + &byte_offset, &chunk_start); + + if (ret) { + GST_DEBUG_OBJECT (wvparse, "new offset: %" G_GINT64_FORMAT, byte_offset); + wvparse->current_offset = byte_offset; + /* we want to send a newsegment event with the actual seek position + * as start, even though our first buffer might start before the + * configured segment. We leave it up to the decoder or sink to crop + * the output buffers accordingly */ + wvparse->segment = segment; + wvparse->segment.last_stop = chunk_start; + gst_wavpack_parse_send_newsegment (wvparse, FALSE); + } else { + GST_DEBUG_OBJECT (wvparse, "seek failed: don't know where to seek to"); + } - gst_pad_set_activatepull_function (wavpackparse->sinkpad, - gst_wavepack_parse_sink_activate_pull); + GST_PAD_STREAM_UNLOCK (wvparse->sinkpad); + GST_OBJECT_UNLOCK (wvparse); - wavpackparse->srcpad = NULL; + gst_pad_start_task (wvparse->sinkpad, + (GstTaskFunction) gst_wavpack_parse_loop, wvparse); + return ret; } static gboolean -gst_wavpack_parse_sink_event (GstPad * pad, GstEvent * event) +gst_wavpack_parse_src_event (GstPad * pad, GstEvent * event) { - GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); - gboolean res = TRUE; + GstWavpackParse *wavpackparse; + gboolean ret; + + wavpackparse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - wavpackparse->eos = TRUE; - /* fall through */ + case GST_EVENT_SEEK: + ret = gst_wavpack_parse_handle_seek_event (wavpackparse, event); + break; default: - res = gst_pad_event_default (pad, event); - gst_object_unref (wavpackparse); - return res; + ret = gst_pad_event_default (pad, event); break; } + gst_object_unref (wavpackparse); + return ret; } static void -gst_wavpack_parse_loop (GstElement * element) +gst_wavpack_parse_init (GstWavpackParse * wavpackparse, + GstWavpackParseClass * gclass) { - GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (element); - gint num; - GstBuffer *buf; - WavpackHeader *header = g_malloc (sizeof (WavpackHeader)); + GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpackparse); + GstPadTemplate *tmpl; - GST_PAD_STREAM_LOCK (wavpackparse->sinkpad); + tmpl = gst_element_class_get_pad_template (klass, "sink"); + wavpackparse->sinkpad = gst_pad_new_from_template (tmpl, "sink"); - if (wavpackparse->eos) { - goto done; - } + gst_pad_set_activate_function (wavpackparse->sinkpad, + GST_DEBUG_FUNCPTR (gst_wavepack_parse_sink_activate)); - if (wavpackparse->seek_pending) { - gst_wavpack_parse_seek (wavpackparse); - wavpackparse->need_discont = TRUE; - wavpackparse->seek_pending = FALSE; - } + gst_pad_set_activatepull_function (wavpackparse->sinkpad, + GST_DEBUG_FUNCPTR (gst_wavepack_parse_sink_activate_pull)); - if (wavpackparse->need_discont) { - if (GST_IS_PAD (wavpackparse->srcpad)) { - gst_pad_push_event (wavpackparse->srcpad, - gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, - wavpackparse->timestamp, GST_CLOCK_TIME_NONE, 0)); + gst_element_add_pad (GST_ELEMENT (wavpackparse), wavpackparse->sinkpad); - wavpackparse->need_discont = FALSE; + wavpackparse->srcpad = NULL; + gst_wavpack_parse_reset (wavpackparse); +} + +static gint64 +gst_wavpack_parse_get_upstream_length (GstWavpackParse * wavpackparse) +{ + GstPad *peer; + gint64 length = -1; + + peer = gst_pad_get_peer (wavpackparse->sinkpad); + if (peer) { + GstFormat format = GST_FORMAT_BYTES; + + if (!gst_pad_query_duration (peer, &format, &length)) { + length = -1; + } else { + GST_DEBUG ("upstream length: %" G_GINT64_FORMAT, length); } + gst_object_unref (peer); + } else { + GST_DEBUG ("no peer!"); } - if (GST_FLOW_OK != gst_pad_pull_range (wavpackparse->sinkpad, - wavpackparse->flushed_bytes, sizeof (WavpackHeader), &buf)) { - wavpackparse->eos = TRUE; - gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ()); - goto done; - } - num = GST_BUFFER_SIZE (buf); + return length; +} - if (num != sizeof (WavpackHeader)) { - gst_buffer_unref (buf); - wavpackparse->eos = TRUE; - gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ()); - goto done; +static GstBuffer * +gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse, gint64 offset, + guint size, GstFlowReturn * flow) +{ + GstFlowReturn flow_ret; + GstBuffer *buf = NULL; + + if (offset + size >= wvparse->upstream_length) { + wvparse->upstream_length = gst_wavpack_parse_get_upstream_length (wvparse); + if (offset + size >= wvparse->upstream_length) { + GST_DEBUG_OBJECT (wvparse, "EOS: %" G_GINT64_FORMAT " + %u > %" + G_GINT64_FORMAT, offset, size, wvparse->upstream_length); + flow_ret = GST_FLOW_UNEXPECTED; + goto done; + } } - gst_wavpack_read_header (header, GST_BUFFER_DATA (buf)); - gst_buffer_unref (buf); + flow_ret = gst_pad_pull_range (wvparse->sinkpad, offset, size, &buf); - if (GST_FLOW_OK != gst_pad_pull_range (wavpackparse->sinkpad, - wavpackparse->flushed_bytes, header->ckSize + 8, &buf)) { - wavpackparse->eos = TRUE; - gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ()); - goto done; + if (flow_ret != GST_FLOW_OK) { + GST_DEBUG_OBJECT (wvparse, "pull_range (%" G_GINT64_FORMAT ", %u) " + "failed, flow: %s", offset, size, gst_flow_get_name (flow_ret)); + return NULL; } - num = GST_BUFFER_SIZE (buf); - if (num != header->ckSize + 8) { + if (GST_BUFFER_SIZE (buf) < size) { + GST_DEBUG_OBJECT (wvparse, "Short read at offset %" G_GINT64_FORMAT + ", got only %u of %u bytes", offset, GST_BUFFER_SIZE (buf), size); gst_buffer_unref (buf); - wavpackparse->eos = TRUE; - gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ()); - goto done; + buf = NULL; + flow_ret = GST_FLOW_UNEXPECTED; } - if (!GST_IS_PAD (wavpackparse->srcpad)) { +done: + if (flow) + *flow = flow_ret; + return buf; +} + +static gboolean +gst_wavpack_parse_create_src_pad (GstWavpackParse * wvparse, GstBuffer * buf, + WavpackHeader * header) +{ + WavpackMetadata meta; + GstCaps *caps = NULL; + guchar *bufptr; + + g_assert (wvparse->srcpad == NULL); - guchar *bufptr = GST_BUFFER_DATA (buf) + sizeof (WavpackHeader); - GstCaps *caps = NULL; - WavpackMetadata meta; + bufptr = GST_BUFFER_DATA (buf) + sizeof (WavpackHeader); - while (read_metadata_buff (&meta, GST_BUFFER_DATA (buf), &bufptr)) { - if (meta.id == ID_WVC_BITSTREAM) { + while (read_metadata_buff (&meta, GST_BUFFER_DATA (buf), &bufptr)) { + switch (meta.id) { + case ID_WVC_BITSTREAM:{ caps = gst_caps_new_simple ("audio/x-wavpack-correction", "framed", G_TYPE_BOOLEAN, TRUE, NULL); - if (GST_IS_PAD (wavpackparse->srcpad)) { - gst_object_unref (wavpackparse->srcpad); - } - wavpackparse->srcpad = + wvparse->srcpad = gst_pad_new_from_template (gst_element_class_get_pad_template - (GST_ELEMENT_GET_CLASS (wavpackparse), "wvcsrc"), "wvcsrc"); - } else if (meta.id == ID_RIFF_HEADER) { - WaveHeader *wheader = g_malloc (sizeof (WaveHeader)); + (GST_ELEMENT_GET_CLASS (wvparse), "wvcsrc"), "wvcsrc"); + break; + } + case ID_RIFF_HEADER:{ + WaveHeader wheader; /* skip RiffChunkHeader and ChunkHeader */ - g_memmove (wheader, meta.data + 20, sizeof (WaveHeader)); - little_endian_to_native (wheader, WaveHeaderFormat); - wavpackparse->samplerate = wheader->SampleRate; - wavpackparse->channels = wheader->NumChannels; - wavpackparse->total_samples = header->total_samples; + g_memmove (&wheader, meta.data + 20, sizeof (WaveHeader)); + little_endian_to_native (&wheader, WaveHeaderFormat); + wvparse->samplerate = wheader.SampleRate; + wvparse->channels = wheader.NumChannels; + wvparse->total_samples = header->total_samples; caps = gst_caps_new_simple ("audio/x-wavpack", - "width", G_TYPE_INT, wheader->BitsPerSample, - "channels", G_TYPE_INT, wavpackparse->channels, - "rate", G_TYPE_INT, wavpackparse->samplerate, + "width", G_TYPE_INT, wheader.BitsPerSample, + "channels", G_TYPE_INT, wvparse->channels, + "rate", G_TYPE_INT, wvparse->samplerate, "framed", G_TYPE_BOOLEAN, TRUE, NULL); - if (GST_IS_PAD (wavpackparse->srcpad)) { - gst_object_unref (wavpackparse->srcpad); - } - wavpackparse->srcpad = + wvparse->srcpad = gst_pad_new_from_template (gst_element_class_get_pad_template - (GST_ELEMENT_GET_CLASS (wavpackparse), "src"), "src"); + (GST_ELEMENT_GET_CLASS (wvparse), "src"), "src"); + break; + } + default:{ + GST_WARNING_OBJECT (wvparse, "unhandled ID: 0x%02x", meta.id); + break; } } + if (caps != NULL) + break; + } - if (!(caps && GST_IS_PAD (wavpackparse->srcpad))) { - gst_buffer_unref (buf); - goto done; - } + if (caps == NULL || wvparse->srcpad == NULL) + return FALSE; - gst_pad_set_query_function (wavpackparse->srcpad, - gst_wavpack_parse_src_query); - gst_pad_set_event_function (wavpackparse->srcpad, - gst_wavpack_parse_src_event); + GST_DEBUG_OBJECT (wvparse, "Added src pad with caps %" GST_PTR_FORMAT, caps); - gst_pad_set_caps (wavpackparse->srcpad, caps); - gst_pad_use_fixed_caps (wavpackparse->srcpad); + gst_pad_set_query_function (wvparse->srcpad, + GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_query)); + gst_pad_set_event_function (wvparse->srcpad, + GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_event)); - gst_element_add_pad (GST_ELEMENT (wavpackparse), wavpackparse->srcpad); + gst_pad_set_caps (wvparse->srcpad, caps); + gst_pad_use_fixed_caps (wvparse->srcpad); + gst_object_ref (wvparse->srcpad); + gst_element_add_pad (GST_ELEMENT (wvparse), wvparse->srcpad); + gst_element_no_more_pads (GST_ELEMENT (wvparse)); + + return TRUE; +} + +static void +gst_wavpack_parse_loop (GstElement * element) +{ + GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (element); + GstFlowReturn flow_ret; + WavpackHeader header = { {0,}, 0, }; + GstBuffer *buf = NULL; + + GST_LOG_OBJECT (wavpackparse, "Current offset: %" G_GINT64_FORMAT, + wavpackparse->current_offset); + + buf = gst_wavpack_parse_pull_buffer (wavpackparse, + wavpackparse->current_offset, sizeof (WavpackHeader), &flow_ret); + + if (buf == NULL && flow_ret == GST_FLOW_UNEXPECTED) { + goto eos; + } else if (buf == NULL) { + goto pause; } - if (wavpackparse->need_discont) { - if (GST_IS_PAD (wavpackparse->srcpad)) { - gst_pad_push_event (wavpackparse->srcpad, - gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, - wavpackparse->timestamp, GST_CLOCK_TIME_NONE, 0)); + gst_wavpack_read_header (&header, GST_BUFFER_DATA (buf)); + gst_buffer_unref (buf); + + GST_LOG_OBJECT (wavpackparse, "Read header at offset %" G_GINT64_FORMAT + ": chunk size = %u+8", wavpackparse->current_offset, header.ckSize); + + buf = gst_wavpack_parse_pull_buffer (wavpackparse, + wavpackparse->current_offset, header.ckSize + 8, &flow_ret); - wavpackparse->need_discont = FALSE; + if (buf == NULL && flow_ret == GST_FLOW_UNEXPECTED) { + goto eos; + } else if (buf == NULL) { + goto pause; + } + + if (wavpackparse->srcpad == NULL) { + if (!gst_wavpack_parse_create_src_pad (wavpackparse, buf, &header)) { + GST_ELEMENT_ERROR (wavpackparse, STREAM, DECODE, (NULL), (NULL)); + goto pause; } } - wavpackparse->flushed_bytes += header->ckSize + 8; + gst_wavpack_parse_index_append_entry (wavpackparse, + wavpackparse->current_offset, header.block_index, header.block_samples); + + wavpackparse->current_offset += header.ckSize + 8; + + wavpackparse->segment.last_stop = header.block_index; - wavpackparse->timestamp = - ((gdouble) header->block_index / (gdouble) wavpackparse->samplerate) * - GST_SECOND; - GST_BUFFER_TIMESTAMP (buf) = wavpackparse->timestamp; - GST_BUFFER_DURATION (buf) = - ((gdouble) header->block_samples / (gdouble) wavpackparse->samplerate) * - GST_SECOND; + if (wavpackparse->need_newsegment) { + if (gst_wavpack_parse_send_newsegment (wavpackparse, FALSE)) + wavpackparse->need_newsegment = FALSE; + } + + GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (header.block_index, + GST_SECOND, wavpackparse->samplerate); + GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (header.block_samples, + GST_SECOND, wavpackparse->samplerate); + GST_BUFFER_OFFSET (buf) = header.block_index; gst_buffer_set_caps (buf, GST_PAD_CAPS (wavpackparse->srcpad)); - if (GST_FLOW_OK != gst_pad_push (wavpackparse->srcpad, buf)) { - gst_buffer_unref (buf); + GST_LOG_OBJECT (wavpackparse, "Pushing buffer with time %" GST_TIME_FORMAT, + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); + + flow_ret = gst_pad_push (wavpackparse->srcpad, buf); + if (flow_ret != GST_FLOW_OK) { + GST_DEBUG_OBJECT (wavpackparse, "Push failed, flow: %s", + gst_flow_get_name (flow_ret)); + goto pause; } -done: - GST_PAD_STREAM_UNLOCK (wavpackparse->sinkpad); return; +eos: + { + GST_DEBUG_OBJECT (wavpackparse, "sending EOS"); + if (wavpackparse->srcpad) { + gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ()); + } + /* fall through and pause task */ + } +pause: + { + GST_DEBUG_OBJECT (wavpackparse, "Pausing task"); + gst_pad_pause_task (wavpackparse->sinkpad); + return; + } } static GstStateChangeReturn gst_wavpack_parse_change_state (GstElement * element, GstStateChange transition) { - GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (element); + GstWavpackParse *wvparse = GST_WAVPACK_PARSE (element); GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: - wavpackparse->flushed_bytes = 0; - wavpackparse->need_discont = TRUE; - wavpackparse->eos = FALSE; - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - { - GstQuery *query; - GstFormat format = GST_FORMAT_BYTES; - - query = gst_query_new_duration (GST_FORMAT_BYTES); - if (gst_pad_query (GST_PAD_PEER (wavpackparse->sinkpad), query)) { - - gst_query_parse_duration (query, &format, - (gint64 *) & wavpackparse->duration); - - if (format != GST_FORMAT_BYTES) { - wavpackparse->duration = -1; - ret = GST_STATE_CHANGE_FAILURE; - } - - } else { - wavpackparse->duration = -1; - ret = GST_STATE_CHANGE_FAILURE; - } - gst_query_unref (query); - } - break; + gst_segment_init (&wvparse->segment, GST_FORMAT_DEFAULT); + wvparse->segment.last_stop = 0; default: break; } - if (GST_ELEMENT_CLASS (parent)->change_state) - ret = GST_ELEMENT_CLASS (parent)->change_state (element, transition); + if (GST_ELEMENT_CLASS (parent_class)->change_state) + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; case GST_STATE_CHANGE_PAUSED_TO_READY: - wavpackparse->seek_pending = FALSE; + gst_wavpack_parse_reset (wvparse); break; default: break; @@ -658,7 +814,6 @@ gst_wavpack_parse_change_state (GstElement * element, GstStateChange transition) static gboolean gst_wavepack_parse_sink_activate (GstPad * sinkpad) { - if (gst_pad_check_pull_range (sinkpad)) { return gst_pad_activate_pull (sinkpad, TRUE); } else { @@ -669,11 +824,9 @@ gst_wavepack_parse_sink_activate (GstPad * sinkpad) static gboolean gst_wavepack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active) { - gboolean result; if (active) { - result = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavpack_parse_loop, GST_PAD_PARENT (sinkpad)); } else { @@ -681,14 +834,11 @@ gst_wavepack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active) } return result; - - return TRUE; } gboolean gst_wavpack_parse_plugin_init (GstPlugin * plugin) { - if (!gst_element_register (plugin, "wavpackparse", GST_RANK_PRIMARY, GST_TYPE_WAVPACK_PARSE)) { return FALSE; diff --git a/ext/wavpack/gstwavpackparse.h b/ext/wavpack/gstwavpackparse.h index 41bfe494..99f3a89a 100644 --- a/ext/wavpack/gstwavpackparse.h +++ b/ext/wavpack/gstwavpackparse.h @@ -38,28 +38,40 @@ G_BEGIN_DECLS #define GST_IS_WAVPACK_PARSE_CLASS(obj) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_PARSE)) -typedef struct _GstWavpackParse GstWavpackParse; -typedef struct _GstWavpackParseClass GstWavpackParseClass; +typedef struct _GstWavpackParse GstWavpackParse; +typedef struct _GstWavpackParseClass GstWavpackParseClass; +typedef struct _GstWavpackParseIndexEntry GstWavpackParseIndexEntry; + +struct _GstWavpackParseIndexEntry { + gint64 byte_offset; /* byte offset of this chunk */ + gint64 sample_offset; /* first sample in this chunk */ + gint64 sample_offset_end; /* first sample in next chunk */ +}; struct _GstWavpackParse { - GstElement element; + GstElement element; - GstPad *sinkpad, *srcpad; + GstPad *sinkpad; + GstPad *srcpad; - guint32 samplerate; - guint32 channels; - guint32 total_samples; - guint64 timestamp; - - guint64 flushed_bytes; - guint64 duration; - - guint64 seek_offset; - gboolean seek_pending; - gboolean need_discont; - gboolean need_flush; - gboolean eos; + guint samplerate; + guint channels; + guint total_samples; + + gboolean need_newsegment; + + gint64 current_offset; /* byte offset on sink pad */ + gint64 upstream_length; /* length of file in bytes */ + + GstSegment segment; /* the currently configured segment, in + * samples/audio frames (DEFAULT format) */ + + /* Array of GstWavpackParseIndexEntry structs, mapping known + * sample offsets to byte offsets. Is kept increasing without + * gaps (ie. append only and consecutive entries must always + * map to consecutive chunks in the file). */ + GArray *entries; }; struct _GstWavpackParseClass |