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-rw-r--r--ChangeLog34
-rw-r--r--ext/wavpack/gstwavpackdec.c169
-rw-r--r--ext/wavpack/gstwavpackparse.c972
-rw-r--r--ext/wavpack/gstwavpackparse.h46
4 files changed, 713 insertions, 508 deletions
diff --git a/ChangeLog b/ChangeLog
index 09ebbf7c..04132911 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -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