*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/**
* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink
+ * gst-launch-1.0 filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink
* ]| Read a wav file and output to the soundcard using the ALSA element. The
* wav file is assumed to contain raw uncompressed samples.
* |[
- * gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
+ * gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
* ]| Stream data from a network url.
* </refsect2>
- *
- * Last reviewed on 2007-02-14 (0.10.6)
*/
/*
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
+
#include <string.h>
#include <math.h>
#include "gstwavparse.h"
-#include "gst/riff/riff-ids.h"
#include "gst/riff/riff-media.h"
#include <gst/base/gsttypefindhelper.h>
#include <gst/gst-i18n-plugin.h>
GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
#define GST_CAT_DEFAULT (wavparse_debug)
+#define GST_RIFF_TAG_Fake GST_MAKE_FOURCC ('F','a','k','e')
+
+#define GST_BWF_TAG_iXML GST_MAKE_FOURCC ('i','X','M','L')
+#define GST_BWF_TAG_qlty GST_MAKE_FOURCC ('q','l','t','y')
+#define GST_BWF_TAG_mext GST_MAKE_FOURCC ('m','e','x','t')
+#define GST_BWF_TAG_levl GST_MAKE_FOURCC ('l','e','v','l')
+#define GST_BWF_TAG_link GST_MAKE_FOURCC ('l','i','n','k')
+#define GST_BWF_TAG_axml GST_MAKE_FOURCC ('a','x','m','l')
+
+/* Data size chunk of RF64,
+ * see http://tech.ebu.ch/docs/tech/tech3306-2009.pdf */
+#define GST_RS64_TAG_DS64 GST_MAKE_FOURCC ('d','s','6','4')
+
static void gst_wavparse_dispose (GObject * object);
-static gboolean gst_wavparse_sink_activate (GstPad * sinkpad);
-static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad,
- gboolean active);
+static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
+ GstObject * parent);
+static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
+ GstObject * parent, GstPadMode mode, gboolean active);
static gboolean gst_wavparse_send_event (GstElement * element,
GstEvent * event);
static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
GstStateChange transition);
-static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad);
-static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query);
-static gboolean gst_wavparse_pad_convert (GstPad * pad,
- GstFormat src_format,
+static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
+ GstQuery * query);
+static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
-static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);
-static gboolean gst_wavparse_sink_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
+ GstBuffer * buf);
+static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
+ GstEvent * event);
static void gst_wavparse_loop (GstPad * pad);
-static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
+static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
+ GstEvent * event);
+
+static void gst_wavparse_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_wavparse_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+
+#define DEFAULT_IGNORE_LENGTH FALSE
+
+enum
+{
+ PROP_0,
+ PROP_IGNORE_LENGTH,
+};
static GstStaticPadTemplate sink_template_factory =
-GST_STATIC_PAD_TEMPLATE ("wavparse_sink",
+GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wav")
);
-#define DEBUG_INIT(bla) \
+#define DEBUG_INIT \
GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
-GST_BOILERPLATE_FULL (GstWavParse, gst_wavparse, GstElement,
- GST_TYPE_ELEMENT, DEBUG_INIT);
+#define gst_wavparse_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
+ DEBUG_INIT);
-static void
-gst_wavparse_base_init (gpointer g_class)
+typedef struct
{
- GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
- GstPadTemplate *src_template;
-
- /* register pads */
- gst_element_class_add_static_pad_template (element_class,
- &sink_template_factory);
-
- src_template = gst_pad_template_new ("wavparse_src", GST_PAD_SRC,
- GST_PAD_SOMETIMES, gst_riff_create_audio_template_caps ());
- gst_element_class_add_pad_template (element_class, src_template);
- gst_object_unref (src_template);
-
- gst_element_class_set_details_simple (element_class, "WAV audio demuxer",
- "Codec/Demuxer/Audio",
- "Parse a .wav file into raw audio",
- "Erik Walthinsen <omega@cse.ogi.edu>");
-}
+ /* Offset Size Description Value
+ * 0x00 4 ID unique identification value
+ * 0x04 4 Position play order position
+ * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
+ * 0x0c 4 Chunk Start Byte Offset of Data Chunk *
+ * 0x10 4 Block Start Byte Offset to sample of First Channel
+ * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
+ */
+ guint32 id;
+ guint32 position;
+ guint32 data_chunk_id;
+ guint32 chunk_start;
+ guint32 block_start;
+ guint32 sample_offset;
+} GstWavParseCue;
+
+typedef struct
+{
+ /* Offset Size Description Value
+ * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
+ * 0x0c Text
+ */
+ guint32 cue_point_id;
+ gchar *text;
+} GstWavParseLabl, GstWavParseNote;
static void
gst_wavparse_class_init (GstWavParseClass * klass)
{
GstElementClass *gstelement_class;
GObjectClass *object_class;
+ GstPadTemplate *src_template;
gstelement_class = (GstElementClass *) klass;
object_class = (GObjectClass *) klass;
object_class->dispose = gst_wavparse_dispose;
+ object_class->set_property = gst_wavparse_set_property;
+ object_class->get_property = gst_wavparse_get_property;
+
+ /**
+ * GstWavParse:ignore-length:
+ *
+ * This selects whether the length found in a data chunk
+ * should be ignored. This may be useful for streamed audio
+ * where the length is unknown until the end of streaming,
+ * and various software/hardware just puts some random value
+ * in there and hopes it doesn't break too much.
+ */
+ g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
+ g_param_spec_boolean ("ignore-length",
+ "Ignore length",
+ "Ignore length from the Wave header",
+ DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
+ );
+
gstelement_class->change_state = gst_wavparse_change_state;
gstelement_class->send_event = gst_wavparse_send_event;
+
+ /* register pads */
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&sink_template_factory));
+
+ src_template = gst_pad_template_new ("src", GST_PAD_SRC,
+ GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
+ gst_element_class_add_pad_template (gstelement_class, src_template);
+
+ gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
+ "Codec/Demuxer/Audio",
+ "Parse a .wav file into raw audio",
+ "Erik Walthinsen <omega@cse.ogi.edu>");
}
static void
wav->adapter = NULL;
}
if (wav->tags)
- gst_tag_list_free (wav->tags);
+ gst_tag_list_unref (wav->tags);
wav->tags = NULL;
+ if (wav->toc)
+ gst_toc_unref (wav->toc);
+ wav->toc = NULL;
+ if (wav->cues)
+ g_list_free_full (wav->cues, g_free);
+ wav->cues = NULL;
+ if (wav->labls)
+ g_list_free_full (wav->labls, g_free);
+ wav->labls = NULL;
if (wav->caps)
gst_caps_unref (wav->caps);
wav->caps = NULL;
if (wav->start_segment)
gst_event_unref (wav->start_segment);
wav->start_segment = NULL;
- if (wav->close_segment)
- gst_event_unref (wav->close_segment);
- wav->close_segment = NULL;
}
static void
}
static void
-gst_wavparse_init (GstWavParse * wavparse, GstWavParseClass * g_class)
+gst_wavparse_init (GstWavParse * wavparse)
{
gst_wavparse_reset (wavparse);
gst_pad_new_from_static_template (&sink_template_factory, "sink");
gst_pad_set_activate_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
- gst_pad_set_activatepull_function (wavparse->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull));
+ gst_pad_set_activatemode_function (wavparse->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
gst_pad_set_chain_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_chain));
gst_pad_set_event_function (wavparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
- /* src, will be created later */
- wavparse->srcpad = NULL;
-}
-
-static void
-gst_wavparse_destroy_sourcepad (GstWavParse * wavparse)
-{
- if (wavparse->srcpad) {
- gst_element_remove_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
- wavparse->srcpad = NULL;
- }
-}
-
-static void
-gst_wavparse_create_sourcepad (GstWavParse * wavparse)
-{
- GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavparse);
- GstPadTemplate *src_template;
-
- /* destroy previous one */
- gst_wavparse_destroy_sourcepad (wavparse);
-
- /* source */
- src_template = gst_element_class_get_pad_template (klass, "wavparse_src");
- wavparse->srcpad = gst_pad_new_from_template (src_template, "src");
+ /* src */
+ wavparse->srcpad =
+ gst_pad_new_from_template (gst_element_class_get_pad_template
+ (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
gst_pad_use_fixed_caps (wavparse->srcpad);
- gst_pad_set_query_type_function (wavparse->srcpad,
- GST_DEBUG_FUNCPTR (gst_wavparse_get_query_types));
gst_pad_set_query_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
gst_pad_set_event_function (wavparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
-
- GST_DEBUG_OBJECT (wavparse, "srcpad created");
-}
-
-/* Compute (value * nom) % denom, avoiding overflow. This can be used
- * to perform ceiling or rounding division together with
- * gst_util_uint64_scale[_int]. */
-#define uint64_scale_modulo(val, nom, denom) \
- ((val % denom) * (nom % denom) % denom)
-
-/* Like gst_util_uint64_scale, but performs ceiling division. */
-static guint64
-uint64_ceiling_scale_int (guint64 val, gint num, gint denom)
-{
- guint64 result = gst_util_uint64_scale_int (val, num, denom);
-
- if (uint64_scale_modulo (val, num, denom) == 0)
- return result;
- else
- return result + 1;
-}
-
-/* Like gst_util_uint64_scale, but performs ceiling division. */
-static guint64
-uint64_ceiling_scale (guint64 val, guint64 num, guint64 denom)
-{
- guint64 result = gst_util_uint64_scale (val, num, denom);
-
- if (uint64_scale_modulo (val, num, denom) == 0)
- return result;
- else
- return result + 1;
-}
-
-
-/* FIXME: why is that not in use? */
-#if 0
-static void
-gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
-{
- guint32 got_bytes;
- GstByteStream *bs = wavparse->bs;
- gst_riff_chunk *temp_chunk, chunk;
- guint8 *tempdata;
- struct _gst_riff_labl labl, *temp_labl;
- struct _gst_riff_ltxt ltxt, *temp_ltxt;
- struct _gst_riff_note note, *temp_note;
- char *label_name;
- GstProps *props;
- GstPropsEntry *entry;
- GstCaps *new_caps;
- GList *caps = NULL;
-
- props = wavparse->metadata->properties;
-
- while (len > 0) {
- got_bytes =
- gst_bytestream_peek_bytes (bs, &tempdata, sizeof (gst_riff_chunk));
- if (got_bytes != sizeof (gst_riff_chunk)) {
- return;
- }
- temp_chunk = (gst_riff_chunk *) tempdata;
-
- chunk.id = GUINT32_FROM_LE (temp_chunk->id);
- chunk.size = GUINT32_FROM_LE (temp_chunk->size);
-
- if (chunk.size == 0) {
- gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
- len -= sizeof (gst_riff_chunk);
- continue;
- }
-
- switch (chunk.id) {
- case GST_RIFF_adtl_labl:
- got_bytes =
- gst_bytestream_peek_bytes (bs, &tempdata,
- sizeof (struct _gst_riff_labl));
- if (got_bytes != sizeof (struct _gst_riff_labl)) {
- return;
- }
-
- temp_labl = (struct _gst_riff_labl *) tempdata;
- labl.id = GUINT32_FROM_LE (temp_labl->id);
- labl.size = GUINT32_FROM_LE (temp_labl->size);
- labl.identifier = GUINT32_FROM_LE (temp_labl->identifier);
-
- gst_bytestream_flush (bs, sizeof (struct _gst_riff_labl));
- len -= sizeof (struct _gst_riff_labl);
-
- got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, labl.size - 4);
- if (got_bytes != labl.size - 4) {
- return;
- }
-
- label_name = (char *) tempdata;
-
- gst_bytestream_flush (bs, ((labl.size - 4) + 1) & ~1);
- len -= (((labl.size - 4) + 1) & ~1);
-
- new_caps = gst_caps_new ("label",
- "application/x-gst-metadata",
- gst_props_new ("identifier", G_TYPE_INT (labl.identifier),
- "name", G_TYPE_STRING (label_name), NULL));
-
- if (gst_props_get (props, "labels", &caps, NULL)) {
- caps = g_list_append (caps, new_caps);
- } else {
- caps = g_list_append (NULL, new_caps);
-
- entry = gst_props_entry_new ("labels", GST_PROPS_GLIST (caps));
- gst_props_add_entry (props, entry);
- }
-
- break;
-
- case GST_RIFF_adtl_ltxt:
- got_bytes =
- gst_bytestream_peek_bytes (bs, &tempdata,
- sizeof (struct _gst_riff_ltxt));
- if (got_bytes != sizeof (struct _gst_riff_ltxt)) {
- return;
- }
-
- temp_ltxt = (struct _gst_riff_ltxt *) tempdata;
- ltxt.id = GUINT32_FROM_LE (temp_ltxt->id);
- ltxt.size = GUINT32_FROM_LE (temp_ltxt->size);
- ltxt.identifier = GUINT32_FROM_LE (temp_ltxt->identifier);
- ltxt.length = GUINT32_FROM_LE (temp_ltxt->length);
- ltxt.purpose = GUINT32_FROM_LE (temp_ltxt->purpose);
- ltxt.country = GUINT16_FROM_LE (temp_ltxt->country);
- ltxt.language = GUINT16_FROM_LE (temp_ltxt->language);
- ltxt.dialect = GUINT16_FROM_LE (temp_ltxt->dialect);
- ltxt.codepage = GUINT16_FROM_LE (temp_ltxt->codepage);
-
- gst_bytestream_flush (bs, sizeof (struct _gst_riff_ltxt));
- len -= sizeof (struct _gst_riff_ltxt);
-
- if (ltxt.size - 20 > 0) {
- got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, ltxt.size - 20);
- if (got_bytes != ltxt.size - 20) {
- return;
- }
-
- gst_bytestream_flush (bs, ((ltxt.size - 20) + 1) & ~1);
- len -= (((ltxt.size - 20) + 1) & ~1);
-
- label_name = (char *) tempdata;
- } else {
- label_name = "";
- }
-
- new_caps = gst_caps_new ("ltxt",
- "application/x-gst-metadata",
- gst_props_new ("identifier", G_TYPE_INT (ltxt.identifier),
- "name", G_TYPE_STRING (label_name),
- "length", G_TYPE_INT (ltxt.length), NULL));
-
- if (gst_props_get (props, "ltxts", &caps, NULL)) {
- caps = g_list_append (caps, new_caps);
- } else {
- caps = g_list_append (NULL, new_caps);
-
- entry = gst_props_entry_new ("ltxts", GST_PROPS_GLIST (caps));
- gst_props_add_entry (props, entry);
- }
-
- break;
-
- case GST_RIFF_adtl_note:
- got_bytes =
- gst_bytestream_peek_bytes (bs, &tempdata,
- sizeof (struct _gst_riff_note));
- if (got_bytes != sizeof (struct _gst_riff_note)) {
- return;
- }
-
- temp_note = (struct _gst_riff_note *) tempdata;
- note.id = GUINT32_FROM_LE (temp_note->id);
- note.size = GUINT32_FROM_LE (temp_note->size);
- note.identifier = GUINT32_FROM_LE (temp_note->identifier);
-
- gst_bytestream_flush (bs, sizeof (struct _gst_riff_note));
- len -= sizeof (struct _gst_riff_note);
-
- got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, note.size - 4);
- if (got_bytes != note.size - 4) {
- return;
- }
-
- gst_bytestream_flush (bs, ((note.size - 4) + 1) & ~1);
- len -= (((note.size - 4) + 1) & ~1);
-
- label_name = (char *) tempdata;
-
- new_caps = gst_caps_new ("note",
- "application/x-gst-metadata",
- gst_props_new ("identifier", G_TYPE_INT (note.identifier),
- "name", G_TYPE_STRING (label_name), NULL));
-
- if (gst_props_get (props, "notes", &caps, NULL)) {
- caps = g_list_append (caps, new_caps);
- } else {
- caps = g_list_append (NULL, new_caps);
-
- entry = gst_props_entry_new ("notes", GST_PROPS_GLIST (caps));
- gst_props_add_entry (props, entry);
- }
-
- break;
-
- default:
- g_print ("Unknown chunk: %" GST_FOURCC_FORMAT "\n",
- GST_FOURCC_ARGS (chunk.id));
- return;
- }
- }
-
- g_object_notify (G_OBJECT (wavparse), "metadata");
-}
-
-static void
-gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
-{
- guint32 got_bytes;
- GstByteStream *bs = wavparse->bs;
- struct _gst_riff_cue *temp_cue, cue;
- struct _gst_riff_cuepoints *points;
- guint8 *tempdata;
- int i;
- GList *cues = NULL;
- GstPropsEntry *entry;
-
- while (len > 0) {
- int required;
-
- got_bytes =
- gst_bytestream_peek_bytes (bs, &tempdata,
- sizeof (struct _gst_riff_cue));
- temp_cue = (struct _gst_riff_cue *) tempdata;
-
- /* fixup for our big endian friends */
- cue.id = GUINT32_FROM_LE (temp_cue->id);
- cue.size = GUINT32_FROM_LE (temp_cue->size);
- cue.cuepoints = GUINT32_FROM_LE (temp_cue->cuepoints);
-
- gst_bytestream_flush (bs, sizeof (struct _gst_riff_cue));
- if (got_bytes != sizeof (struct _gst_riff_cue)) {
- return;
- }
-
- len -= sizeof (struct _gst_riff_cue);
-
- /* -4 because cue.size contains the cuepoints size
- and we've already flushed that out of the system */
- required = cue.size - 4;
- got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, required);
- gst_bytestream_flush (bs, ((required) + 1) & ~1);
- if (got_bytes != required) {
- return;
- }
-
- len -= (((cue.size - 4) + 1) & ~1);
-
- /* now we have an array of struct _gst_riff_cuepoints in tempdata */
- points = (struct _gst_riff_cuepoints *) tempdata;
-
- for (i = 0; i < cue.cuepoints; i++) {
- GstCaps *caps;
-
- caps = gst_caps_new ("cues",
- "application/x-gst-metadata",
- gst_props_new ("identifier", G_TYPE_INT (points[i].identifier),
- "position", G_TYPE_INT (points[i].offset), NULL));
- cues = g_list_append (cues, caps);
- }
-
- entry = gst_props_entry_new ("cues", GST_PROPS_GLIST (cues));
- gst_props_add_entry (wavparse->metadata->properties, entry);
- }
-
- g_object_notify (G_OBJECT (wavparse), "metadata");
-}
-
-/* Read 'fmt ' header */
-static gboolean
-gst_wavparse_fmt (GstWavParse * wav)
-{
- gst_riff_strf_auds *header = NULL;
- GstCaps *caps;
-
- if (!gst_riff_read_strf_auds (wav, &header))
- goto no_fmt;
-
- wav->format = header->format;
- wav->rate = header->rate;
- wav->channels = header->channels;
- if (wav->channels == 0)
- goto no_channels;
-
- wav->blockalign = header->blockalign;
- wav->width = (header->blockalign * 8) / header->channels;
- wav->depth = header->size;
- wav->bps = header->av_bps;
- if (wav->bps <= 0)
- goto no_bps;
-
- /* Note: gst_riff_create_audio_caps might need to fix values in
- * the header header depending on the format, so call it first */
- caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL);
- g_free (header);
-
- if (caps == NULL)
- goto no_caps;
-
- gst_wavparse_create_sourcepad (wav);
- gst_pad_use_fixed_caps (wav->srcpad);
- gst_pad_set_active (wav->srcpad, TRUE);
- gst_pad_set_caps (wav->srcpad, caps);
- gst_caps_free (caps);
- gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad);
- gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
-
- GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
-
- return TRUE;
-
- /* ERRORS */
-no_fmt:
- {
- GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
- ("No FMT tag found"));
- return FALSE;
- }
-no_channels:
- {
- GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
- ("Stream claims to contain zero channels - invalid data"));
- g_free (header);
- return FALSE;
- }
-no_bps:
- {
- GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
- ("Stream claims to bitrate of <= zero - invalid data"));
- g_free (header);
- return FALSE;
- }
-no_caps:
- {
- GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
- return FALSE;
- }
+ gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
}
static gboolean
-gst_wavparse_other (GstWavParse * wav)
-{
- guint32 tag, length;
-
- if (!gst_riff_peek_head (wav, &tag, &length, NULL)) {
- GST_WARNING_OBJECT (wav, "could not peek head");
- return FALSE;
- }
- GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag,
- (gchar *) & tag, length);
-
- switch (tag) {
- case GST_RIFF_TAG_LIST:
- if (!(tag = gst_riff_peek_list (wav))) {
- GST_WARNING_OBJECT (wav, "could not peek list");
- return FALSE;
- }
-
- switch (tag) {
- case GST_RIFF_LIST_INFO:
- if (!gst_riff_read_list (wav, &tag) || !gst_riff_read_info (wav)) {
- GST_WARNING_OBJECT (wav, "could not read list");
- return FALSE;
- }
- break;
-
- case GST_RIFF_LIST_adtl:
- if (!gst_riff_read_skip (wav)) {
- GST_WARNING_OBJECT (wav, "could not read skip");
- return FALSE;
- }
- break;
-
- default:
- GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag,
- (gchar *) & tag);
- if (!gst_riff_read_skip (wav)) {
- GST_WARNING_OBJECT (wav, "could not read skip");
- return FALSE;
- }
- break;
- }
-
- break;
-
- case GST_RIFF_TAG_data:
- if (!gst_bytestream_flush (wav->bs, 8)) {
- GST_WARNING_OBJECT (wav, "could not flush 8 bytes");
- return FALSE;
- }
-
- GST_DEBUG_OBJECT (wav, "switching to data mode");
- wav->state = GST_WAVPARSE_DATA;
- wav->datastart = gst_bytestream_tell (wav->bs);
- if (length == 0) {
- guint64 file_length;
-
- /* length is 0, data probably stretches to the end
- * of file */
- GST_DEBUG_OBJECT (wav, "length is 0 trying to find length");
- /* get length of file */
- file_length = gst_bytestream_length (wav->bs);
- if (file_length == -1) {
- GST_DEBUG_OBJECT (wav,
- "could not get file length, assuming data to eof");
- /* could not get length, assuming till eof */
- length = G_MAXUINT32;
- }
- if (file_length > G_MAXUINT32) {
- GST_DEBUG_OBJECT (wav, "file length %" G_GUINT64_FORMAT
- ", clipping to 32 bits", file_length);
- /* could not get length, assuming till eof */
- length = G_MAXUINT32;
- } else {
- GST_DEBUG_OBJECT (wav, "file length %" G_GUINT64_FORMAT
- ", datalength %u", file_length, length);
- /* substract offset of datastart from length */
- length = file_length - wav->datastart;
- GST_DEBUG_OBJECT (wav, "datalength %u", length);
- }
- }
- wav->datasize = (guint64) length;
- GST_DEBUG_OBJECT (wav, "datasize = %ld", length)
- break;
-
- case GST_RIFF_TAG_cue:
- if (!gst_riff_read_skip (wav)) {
- GST_WARNING_OBJECT (wav, "could not read skip");
- return FALSE;
- }
- break;
-
- default:
- GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, (gchar *) & tag);
- if (!gst_riff_read_skip (wav))
- return FALSE;
- break;
- }
-
- return TRUE;
-}
-#endif
-
-
-static gboolean
gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
{
guint32 doctype;
not_wav:
{
GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
- ("File is not a WAVE file: %" GST_FOURCC_FORMAT,
- GST_FOURCC_ARGS (doctype)));
+ ("File is not a WAVE file: 0x%" G_GINT32_MODIFIER "x", doctype));
return FALSE;
}
}
}
if (wav->bps > 0) {
- *bytepos = uint64_ceiling_scale (ts, (guint64) wav->bps, GST_SECOND);
+ *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
return TRUE;
} else if (wav->fact) {
- guint64 bps =
- gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
- *bytepos = uint64_ceiling_scale (ts, bps, GST_SECOND);
+ guint64 bps = gst_util_uint64_scale (wav->datasize, wav->rate, wav->fact);
+ *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
return TRUE;
}
gboolean update;
GstSegment seeksegment = { 0, };
gint64 last_stop;
+ guint32 seqnum = 0;
if (event) {
GST_DEBUG_OBJECT (wav, "doing seek with event");
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
+ seqnum = gst_event_get_seqnum (event);
/* no negative rates yet */
if (rate < 0.0)
if (cur_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (wav->srcpad, format, cur,
- &wav->segment.format, &cur);
+ wav->segment.format, &cur);
if (res && stop_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (wav->srcpad, format, stop,
- &wav->segment.format, &stop);
+ wav->segment.format, &stop);
if (!res)
goto no_format;
if (!event || wav->state != GST_WAVPARSE_DATA) {
if (wav->start_segment)
gst_event_unref (wav->start_segment);
- wav->start_segment =
- gst_event_new_new_segment (FALSE, wav->segment.rate,
- wav->segment.format, wav->segment.last_stop, wav->segment.duration,
- wav->segment.last_stop);
+ wav->start_segment = gst_event_new_segment (&wav->segment);
res = TRUE;
} else {
/* convert seek positions to byte positions in data sections */
/* BYTE seek event */
event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
stop_type, stop);
+ gst_event_set_seqnum (event, seqnum);
res = gst_pad_push_event (wav->sinkpad, event);
}
return res;
* as it completes one iteration (and thus might block when the sink is
* blocking in preroll). */
if (flush) {
- if (wav->srcpad) {
- GST_DEBUG_OBJECT (wav, "sending flush start");
- gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
- }
+ GstEvent *fevent;
+ GST_DEBUG_OBJECT (wav, "sending flush start");
+
+ fevent = gst_event_new_flush_start ();
+ gst_event_set_seqnum (fevent, seqnum);
+ gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
+ gst_pad_push_event (wav->srcpad, fevent);
} else {
gst_pad_pause_task (wav->sinkpad);
}
GST_PAD_STREAM_LOCK (wav->sinkpad);
/* save current position */
- last_stop = wav->segment.last_stop;
+ last_stop = wav->segment.position;
GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
* right values in the segment to perform the seek */
if (event) {
GST_DEBUG_OBJECT (wav, "configuring seek");
- gst_segment_set_seek (&seeksegment, rate, format, flags,
+ gst_segment_do_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
}
/* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
* we can just copy the last_stop. If not, we use the bps to convert TIME to
* bytes. */
- if (!gst_wavparse_time_to_bytepos (wav, seeksegment.last_stop,
+ if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
(gint64 *) & wav->offset))
- wav->offset = seeksegment.last_stop;
+ wav->offset = seeksegment.position;
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
wav->offset -= (wav->offset % wav->bytes_per_sample);
GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
/* make sure filesize is not exceeded due to rounding errors or so,
* same precaution as in _stream_headers */
bformat = GST_FORMAT_BYTES;
- if (gst_pad_query_peer_duration (wav->sinkpad, &bformat, &upstream_size))
+ if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
wav->end_offset = MIN (wav->end_offset, upstream_size);
+ if (wav->datasize > 0 && wav->end_offset > wav->datastart + wav->datasize)
+ wav->end_offset = wav->datastart + wav->datasize;
+
/* this is the range of bytes we will use for playback */
wav->offset = MIN (wav->offset, wav->end_offset);
wav->dataleft = wav->end_offset - wav->offset;
wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
/* prepare for streaming again */
- if (wav->srcpad) {
- if (flush) {
- /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
- GST_DEBUG_OBJECT (wav, "sending flush stop");
- gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop ());
- } else if (wav->segment_running) {
- /* we are running the current segment and doing a non-flushing seek,
- * close the segment first based on the previous last_stop. */
- GST_DEBUG_OBJECT (wav, "closing running segment %" G_GINT64_FORMAT
- " to %" G_GINT64_FORMAT, wav->segment.start, wav->segment.last_stop);
-
- /* queue the segment for sending in the stream thread */
- if (wav->close_segment)
- gst_event_unref (wav->close_segment);
- wav->close_segment = gst_event_new_new_segment (TRUE,
- wav->segment.rate, wav->segment.format,
- wav->segment.start, wav->segment.last_stop, wav->segment.start);
- }
+ if (flush) {
+ GstEvent *fevent;
+
+ /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
+ GST_DEBUG_OBJECT (wav, "sending flush stop");
+
+ fevent = gst_event_new_flush_stop (TRUE);
+ gst_event_set_seqnum (fevent, seqnum);
+ gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
+ gst_pad_push_event (wav->srcpad, fevent);
}
/* now we did the seek and can activate the new segment values */
if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (wav),
gst_message_new_segment_start (GST_OBJECT_CAST (wav),
- wav->segment.format, wav->segment.last_stop));
+ wav->segment.format, wav->segment.position));
}
/* now create the newsegment */
GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
- " to %" G_GINT64_FORMAT, wav->segment.last_stop, stop);
+ " to %" G_GINT64_FORMAT, wav->segment.position, stop);
/* store the newsegment event so it can be sent from the streaming thread. */
if (wav->start_segment)
gst_event_unref (wav->start_segment);
- wav->start_segment =
- gst_event_new_new_segment (FALSE, wav->segment.rate,
- wav->segment.format, wav->segment.last_stop, stop,
- wav->segment.last_stop);
+ wav->start_segment = gst_event_new_segment (&wav->segment);
+ gst_event_set_seqnum (wav->start_segment, seqnum);
/* mark discont if we are going to stream from another position. */
- if (last_stop != wav->segment.last_stop) {
+ if (last_stop != wav->segment.position) {
GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
wav->discont = TRUE;
}
/* and start the streaming task again */
- wav->segment_running = TRUE;
if (!wav->streaming) {
gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
- wav->sinkpad);
+ wav->sinkpad, NULL);
}
GST_PAD_STREAM_UNLOCK (wav->sinkpad);
if (gst_adapter_available (wav->adapter) < 8)
return FALSE;
- data = gst_adapter_peek (wav->adapter, 8);
+ data = gst_adapter_map (wav->adapter, 8);
*tag = GST_READ_UINT32_LE (data);
*size = GST_READ_UINT32_LE (data + 4);
+ gst_adapter_unmap (wav->adapter);
- GST_DEBUG ("Next chunk size is %d bytes, type %" GST_FOURCC_FORMAT, *size,
+ GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
GST_FOURCC_ARGS (*tag));
return TRUE;
* so we throw poor man's exception, which can be caught if caller really
* wants to handle 0 size chunk */
if (!(*size) || (*size) >= (1 << 30)) {
- GST_INFO ("Invalid/unexpected chunk size %d for tag %" GST_FOURCC_FORMAT,
+ GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
*size, GST_FOURCC_ARGS (*tag));
/* chain should give up */
wav->abort_buffering = TRUE;
if (wav->bps > 0) {
GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
wav->duration =
- uint64_ceiling_scale (wav->datasize, GST_SECOND, (guint64) wav->bps);
+ gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
+ (guint64) wav->bps);
GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
GST_TIME_ARGS (wav->duration));
return TRUE;
} else if (wav->fact) {
- wav->duration = uint64_ceiling_scale_int (GST_SECOND, wav->fact, wav->rate);
+ wav->duration =
+ gst_util_uint64_scale_ceil (GST_SECOND, wav->fact, wav->rate);
GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
GST_TIME_ARGS (wav->duration));
return TRUE;
}
- return FALSE;
+ return FALSE;
+}
+
+static gboolean
+gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
+ guint32 size)
+{
+ guint flush;
+
+ if (wav->streaming) {
+ if (!gst_wavparse_peek_chunk (wav, &tag, &size))
+ return FALSE;
+ }
+ GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
+ GST_FOURCC_ARGS (tag));
+ flush = 8 + ((size + 1) & ~1);
+ wav->offset += flush;
+ if (wav->streaming) {
+ gst_adapter_flush (wav->adapter, flush);
+ } else {
+ gst_buffer_unref (buf);
+ }
+
+ return TRUE;
+}
+
+/*
+ * gst_wavparse_cue_chunk:
+ * @wav GstWavParse object
+ * @data holder for data
+ * @size holder for data size
+ *
+ * Parse cue chunk from @data to wav->cues.
+ *
+ * Returns: %TRUE when cue chunk is available
+ */
+static gboolean
+gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
+{
+ guint32 i, ncues;
+ GList *cues = NULL;
+ GstWavParseCue *cue;
+
+ if (wav->cues) {
+ GST_WARNING_OBJECT (wav, "found another cue's");
+ return TRUE;
+ }
+
+ ncues = GST_READ_UINT32_LE (data);
+
+ if (size < 4 + ncues * 24) {
+ GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
+ return FALSE;
+ }
+
+ /* parse data */
+ data += 4;
+ for (i = 0; i < ncues; i++) {
+ cue = g_new0 (GstWavParseCue, 1);
+ cue->id = GST_READ_UINT32_LE (data);
+ cue->position = GST_READ_UINT32_LE (data + 4);
+ cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
+ cue->chunk_start = GST_READ_UINT32_LE (data + 12);
+ cue->block_start = GST_READ_UINT32_LE (data + 16);
+ cue->sample_offset = GST_READ_UINT32_LE (data + 20);
+ cues = g_list_append (cues, cue);
+ data += 24;
+ }
+
+ wav->cues = cues;
+
+ return TRUE;
+}
+
+/*
+ * gst_wavparse_labl_chunk:
+ * @wav GstWavParse object
+ * @data holder for data
+ * @size holder for data size
+ *
+ * Parse labl from @data to wav->labls.
+ *
+ * Returns: %TRUE when labl chunk is available
+ */
+static gboolean
+gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
+{
+ GstWavParseLabl *labl;
+
+ if (size < 5)
+ return FALSE;
+
+ labl = g_new0 (GstWavParseLabl, 1);
+
+ /* parse data */
+ data += 8;
+ labl->cue_point_id = GST_READ_UINT32_LE (data);
+ labl->text = g_memdup (data + 4, size - 4);
+
+ wav->labls = g_list_append (wav->labls, labl);
+
+ return TRUE;
+}
+
+/*
+ * gst_wavparse_note_chunk:
+ * @wav GstWavParse object
+ * @data holder for data
+ * @size holder for data size
+ *
+ * Parse note from @data to wav->notes.
+ *
+ * Returns: %TRUE when note chunk is available
+ */
+static gboolean
+gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
+{
+ GstWavParseNote *note;
+
+ if (size < 5)
+ return FALSE;
+
+ note = g_new0 (GstWavParseNote, 1);
+
+ /* parse data */
+ data += 8;
+ note->cue_point_id = GST_READ_UINT32_LE (data);
+ note->text = g_memdup (data + 4, size - 4);
+
+ wav->notes = g_list_append (wav->notes, note);
+
+ return TRUE;
+}
+
+/*
+ * gst_wavparse_smpl_chunk:
+ * @wav GstWavParse object
+ * @data holder for data
+ * @size holder for data size
+ *
+ * Parse smpl chunk from @data.
+ *
+ * Returns: %TRUE when cue chunk is available
+ */
+static gboolean
+gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
+{
+ guint32 note_number;
+
+ /*
+ manufacturer_id = GST_READ_UINT32_LE (data);
+ product_id = GST_READ_UINT32_LE (data + 4);
+ sample_period = GST_READ_UINT32_LE (data + 8);
+ */
+ note_number = GST_READ_UINT32_LE (data + 12);
+ /*
+ pitch_fraction = GST_READ_UINT32_LE (data + 16);
+ SMPTE_format = GST_READ_UINT32_LE (data + 20);
+ SMPTE_offset = GST_READ_UINT32_LE (data + 24);
+ num_sample_loops = GST_READ_UINT32_LE (data + 28);
+ List of Sample Loops, 24 bytes each
+ */
+
+ if (!wav->tags)
+ wav->tags = gst_tag_list_new_empty ();
+ gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
+ GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL);
+ return TRUE;
+}
+
+/*
+ * gst_wavparse_adtl_chunk:
+ * @wav GstWavParse object
+ * @data holder for data
+ * @size holder for data size
+ *
+ * Parse adtl from @data.
+ *
+ * Returns: %TRUE when adtl chunk is available
+ */
+static gboolean
+gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
+{
+ guint32 ltag, lsize, offset = 0;
+
+ while (size >= 8) {
+ ltag = GST_READ_UINT32_LE (data + offset);
+ lsize = GST_READ_UINT32_LE (data + offset + 4);
+
+ if (lsize + 8 > size) {
+ GST_WARNING_OBJECT (wav, "Invalid adtl size: %u + 8 > %u", lsize, size);
+ return FALSE;
+ }
+
+ switch (ltag) {
+ case GST_RIFF_TAG_labl:
+ gst_wavparse_labl_chunk (wav, data + offset, size);
+ break;
+ case GST_RIFF_TAG_note:
+ gst_wavparse_note_chunk (wav, data + offset, size);
+ break;
+ default:
+ GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT,
+ GST_FOURCC_ARGS (ltag));
+ GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize);
+ break;
+ }
+ offset += 8 + GST_ROUND_UP_2 (lsize);
+ size -= 8 + GST_ROUND_UP_2 (lsize);
+ }
+
+ return TRUE;
+}
+
+static GstTagList *
+gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
+{
+ GstTagList *tags = NULL;
+ GstTocEntry *entry = NULL;
+
+ entry = gst_toc_find_entry (toc, id);
+ if (entry != NULL) {
+ tags = gst_toc_entry_get_tags (entry);
+ if (tags == NULL) {
+ tags = gst_tag_list_new_empty ();
+ gst_toc_entry_set_tags (entry, tags);
+ }
+ }
+
+ return tags;
+}
+
+/*
+ * gst_wavparse_create_toc:
+ * @wav GstWavParse object
+ *
+ * Create TOC from wav->cues and wav->labls.
+ */
+static gboolean
+gst_wavparse_create_toc (GstWavParse * wav)
+{
+ gint64 start, stop;
+ gchar *id;
+ GList *list;
+ GstWavParseCue *cue;
+ GstWavParseLabl *labl;
+ GstWavParseNote *note;
+ GstTagList *tags;
+ GstToc *toc;
+ GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
+
+ GST_OBJECT_LOCK (wav);
+ if (wav->toc) {
+ GST_OBJECT_UNLOCK (wav);
+ GST_WARNING_OBJECT (wav, "found another TOC");
+ return FALSE;
+ }
+
+ if (!wav->cues) {
+ GST_OBJECT_UNLOCK (wav);
+ return TRUE;
+ }
+
+ /* FIXME: send CURRENT scope toc too */
+ toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
+
+ /* add cue edition */
+ entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
+ gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
+ gst_toc_append_entry (toc, entry);
+
+ /* add tracks in cue edition */
+ list = wav->cues;
+ while (list) {
+ cue = list->data;
+ prev_subentry = cur_subentry;
+ /* previous track stop time = current track start time */
+ if (prev_subentry != NULL) {
+ gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
+ stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
+ gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
+ }
+ id = g_strdup_printf ("%08x", cue->id);
+ cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
+ g_free (id);
+ start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
+ stop = wav->duration;
+ gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
+ gst_toc_entry_append_sub_entry (entry, cur_subentry);
+ list = g_list_next (list);
+ }
+
+ /* add tags in tracks */
+ list = wav->labls;
+ while (list) {
+ labl = list->data;
+ id = g_strdup_printf ("%08x", labl->cue_point_id);
+ tags = gst_wavparse_get_tags_toc_entry (toc, id);
+ g_free (id);
+ if (tags != NULL) {
+ gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
+ NULL);
+ }
+ list = g_list_next (list);
+ }
+ list = wav->notes;
+ while (list) {
+ note = list->data;
+ id = g_strdup_printf ("%08x", note->cue_point_id);
+ tags = gst_wavparse_get_tags_toc_entry (toc, id);
+ g_free (id);
+ if (tags != NULL) {
+ gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
+ note->text, NULL);
+ }
+ list = g_list_next (list);
+ }
+
+ /* send data as TOC */
+ wav->toc = toc;
+
+ /* send TOC event */
+ if (wav->toc) {
+ GST_OBJECT_UNLOCK (wav);
+ gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
+ }
+
+ return TRUE;
}
+#define MAX_BUFFER_SIZE 4096
+
static gboolean
-gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
- guint32 size)
+parse_ds64 (GstWavParse * wav, GstBuffer * buf)
{
- guint flush;
+ GstMapInfo map;
+ guint32 dataSizeLow, dataSizeHigh;
+ guint32 sampleCountLow, sampleCountHigh;
- if (wav->streaming) {
- if (!gst_wavparse_peek_chunk (wav, &tag, &size))
- return FALSE;
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ dataSizeLow = GST_READ_UINT32_LE (map.data + 2 * 4);
+ dataSizeHigh = GST_READ_UINT32_LE (map.data + 3 * 4);
+ sampleCountLow = GST_READ_UINT32_LE (map.data + 4 * 4);
+ sampleCountHigh = GST_READ_UINT32_LE (map.data + 5 * 4);
+ gst_buffer_unmap (buf, &map);
+ if (dataSizeHigh != 0xFFFFFFFF && dataSizeLow != 0xFFFFFFFF) {
+ wav->datasize = ((guint64) dataSizeHigh << 32) | dataSizeLow;
}
- GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
- GST_FOURCC_ARGS (tag));
- flush = 8 + ((size + 1) & ~1);
- wav->offset += flush;
- if (wav->streaming) {
- gst_adapter_flush (wav->adapter, flush);
- } else {
- gst_buffer_unref (buf);
+ if (sampleCountHigh != 0xFFFFFFFF && sampleCountLow != 0xFFFFFFFF) {
+ wav->fact = ((guint64) sampleCountHigh << 32) | sampleCountLow;
}
+ GST_DEBUG_OBJECT (wav, "Got 'ds64' TAG, datasize : %" G_GINT64_FORMAT
+ " fact: %" G_GINT64_FORMAT, wav->datasize, wav->fact);
return TRUE;
}
-#define MAX_BUFFER_SIZE 4096
-
static GstFlowReturn
gst_wavparse_stream_headers (GstWavParse * wav)
{
gboolean gotdata = FALSE;
GstCaps *caps = NULL;
gchar *codec_name = NULL;
- GstEvent **event_p;
- GstFormat bformat;
gint64 upstream_size = 0;
+ GstStructure *s;
/* search for "_fmt" chunk, which should be first */
while (!wav->got_fmt) {
if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
- tag == GST_RIFF_TAG_IDVX) {
+ tag == GST_RIFF_TAG_id3 || tag == GST_RIFF_TAG_IDVX ||
+ tag == GST_BWF_TAG_iXML || tag == GST_BWF_TAG_qlty ||
+ tag == GST_BWF_TAG_mext || tag == GST_BWF_TAG_levl ||
+ tag == GST_BWF_TAG_link || tag == GST_BWF_TAG_axml ||
+ tag == GST_RIFF_TAG_Fake) {
GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
GST_FOURCC_ARGS (tag));
gst_buffer_unref (buf);
continue;
}
+ if (tag == GST_RS64_TAG_DS64) {
+ if (!parse_ds64 (wav, buf))
+ goto fail;
+ else
+ continue;
+ }
+
if (tag != GST_RIFF_TAG_fmt)
goto invalid_wav;
/* Note: gst_riff_create_audio_caps might need to fix values in
* the header header depending on the format, so call it first */
+ /* FIXME: Need to handle the channel reorder map */
caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
- NULL, &codec_name);
+ NULL, &codec_name, NULL);
if (extra)
gst_buffer_unref (extra);
if (!caps)
goto unknown_format;
+ /* If we got raw audio from upstream, we remove the codec_data field,
+ * which may have been added if the wav header included an extended
+ * chunk. We want to keep it for non raw audio.
+ */
+ s = gst_caps_get_structure (caps, 0);
+ if (s && gst_structure_has_name (s, "audio/x-raw")) {
+ gst_structure_remove_field (s, "codec_data");
+ }
+
/* do more sanity checks of header fields
* (these can be sanitized by gst_riff_create_audio_caps()
*/
wav->rate = header->rate;
wav->channels = header->channels;
wav->blockalign = header->blockalign;
- wav->depth = header->size;
+ wav->depth = header->bits_per_sample;
wav->av_bps = header->av_bps;
wav->vbr = FALSE;
{
/* Note: workaround for mp2/mp3 embedded in wav, that relies on the
* bitrate inside the mpeg stream */
- GST_INFO ("resetting bps from %d to 0 for mp2/3", wav->av_bps);
+ GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
wav->bps = 0;
break;
}
case GST_RIFF_WAVE_FORMAT_PCM:
- if (wav->blockalign > wav->channels * (guint) ceil (wav->depth / 8.0))
+ if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
goto invalid_blockalign;
/* fall through */
default:
wav->got_fmt = TRUE;
if (codec_name) {
- wav->tags = gst_tag_list_new ();
+ wav->tags = gst_tag_list_new_empty ();
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, codec_name, NULL);
}
- bformat = GST_FORMAT_BYTES;
- gst_pad_query_peer_duration (wav->sinkpad, &bformat, &upstream_size);
+ gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
/* loop headers until we get data */
if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
goto exit;
} else {
+ GstMapInfo map;
+
+ buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
&buf)) != GST_FLOW_OK)
goto header_read_error;
- tag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
- size = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 4);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ tag = GST_READ_UINT32_LE (map.data);
+ size = GST_READ_UINT32_LE (map.data + 4);
+ gst_buffer_unmap (buf, &map);
}
GST_INFO_OBJECT (wav,
- "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT,
- GST_FOURCC_ARGS (tag), wav->offset);
+ "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT ", size %"
+ G_GUINT32_FORMAT, GST_FOURCC_ARGS (tag), wav->offset, size);
+
+ /* Maximum valid size is INT_MAX */
+ if (size & 0x80000000) {
+ GST_WARNING_OBJECT (wav, "Invalid size, clipping to 0x7fffffff");
+ size = 0x7fffffff;
+ }
+
+ /* Clip to upstream size if known */
+ if (wav->datasize > 0 && size + wav->offset > wav->datasize) {
+ GST_WARNING_OBJECT (wav, "Clipping chunk size to file size");
+ size = wav->datasize - wav->offset;
+ }
/* wav is a st00pid format, we don't know for sure where data starts.
* So we have to go bit by bit until we find the 'data' header
*/
switch (tag) {
case GST_RIFF_TAG_data:{
- GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
+ guint64 size64;
+
+ GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
+ size64 = size;
+ if (wav->ignore_length) {
+ GST_DEBUG_OBJECT (wav, "Ignoring length");
+ size64 = 0;
+ }
if (wav->streaming) {
gst_adapter_flush (wav->adapter, 8);
gotdata = TRUE;
}
wav->offset += 8;
wav->datastart = wav->offset;
+ /* use size from ds64 chunk if available */
+ if (size64 == -1 && wav->datasize > 0) {
+ GST_DEBUG_OBJECT (wav, "Using ds64 datasize");
+ size64 = wav->datasize;
+ }
/* If size is zero, then the data chunk probably actually extends to
the end of the file */
- if (size == 0 && upstream_size) {
- size = upstream_size - wav->datastart;
+ if (size64 == 0 && upstream_size) {
+ size64 = upstream_size - wav->datastart;
}
/* Or the file might be truncated */
else if (upstream_size) {
- size = MIN (size, (upstream_size - wav->datastart));
+ size64 = MIN (size64, (upstream_size - wav->datastart));
}
- wav->datasize = (guint64) size;
- wav->dataleft = (guint64) size;
- wav->end_offset = size + wav->datastart;
+ wav->datasize = size64;
+ wav->dataleft = size64;
+ wav->end_offset = size64 + wav->datastart;
if (!wav->streaming) {
/* We will continue parsing tags 'till end */
- wav->offset += size;
+ wav->offset += size64;
}
- GST_DEBUG_OBJECT (wav, "datasize = %d", size);
+ GST_DEBUG_OBJECT (wav, "datasize = %" G_GUINT64_FORMAT, size64);
break;
}
case GST_RIFF_TAG_fact:{
- if (wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
+ if (wav->fact == 0 &&
+ wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
const guint data_size = 4;
/* need more data */
goto exit;
}
- GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk",
+ GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
data_size, size);
break;
}
goto exit;
}
gst_adapter_flush (wav->adapter, 8);
- data = gst_adapter_peek (wav->adapter, data_size);
+ data = gst_adapter_map (wav->adapter, data_size);
wav->fact = GST_READ_UINT32_LE (data);
+ gst_adapter_unmap (wav->adapter);
gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
} else {
gst_buffer_unref (buf);
+ buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
data_size, &buf)) != GST_FLOW_OK)
goto header_read_error;
- wav->fact = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf));
+ gst_buffer_extract (buf, 0, &wav->fact, 4);
+ wav->fact = GUINT32_FROM_LE (wav->fact);
gst_buffer_unref (buf);
}
- GST_DEBUG_OBJECT (wav, "have fact %u", wav->fact);
+ GST_DEBUG_OBJECT (wav, "have fact %" G_GUINT64_FORMAT, wav->fact);
wav->offset += 8 + GST_ROUND_UP_2 (size);
break;
} else {
case GST_RIFF_TAG_acid:{
const gst_riff_acid *acid = NULL;
const guint data_size = sizeof (gst_riff_acid);
+ gfloat tempo;
GST_INFO_OBJECT (wav, "Have acid chunk");
if (size < data_size) {
/* need more data */
goto exit;
}
- GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk",
+ GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
data_size, size);
break;
}
goto exit;
}
gst_adapter_flush (wav->adapter, 8);
- acid = (const gst_riff_acid *) gst_adapter_peek (wav->adapter,
+ acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
data_size);
+ tempo = acid->tempo;
+ gst_adapter_unmap (wav->adapter);
} else {
+ GstMapInfo map;
gst_buffer_unref (buf);
+ buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
size, &buf)) != GST_FLOW_OK)
goto header_read_error;
- acid = (const gst_riff_acid *) GST_BUFFER_DATA (buf);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ acid = (const gst_riff_acid *) map.data;
+ tempo = acid->tempo;
+ gst_buffer_unmap (buf, &map);
}
/* send data as tags */
if (!wav->tags)
- wav->tags = gst_tag_list_new ();
+ wav->tags = gst_tag_list_new_empty ();
gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
- GST_TAG_BEATS_PER_MINUTE, acid->tempo, NULL);
+ GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
size = GST_ROUND_UP_2 (size);
if (wav->streaming) {
if (gst_adapter_available (wav->adapter) < 12) {
goto exit;
}
- data = gst_adapter_peek (wav->adapter, 12);
+ data = gst_adapter_map (wav->adapter, 12);
ltag = GST_READ_UINT32_LE (data + 8);
+ gst_adapter_unmap (wav->adapter);
} else {
gst_buffer_unref (buf);
+ buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
&buf)) != GST_FLOW_OK)
goto header_read_error;
- ltag = GST_READ_UINT32_LE (GST_BUFFER_DATA (buf) + 8);
+ gst_buffer_extract (buf, 8, <ag, 4);
+ ltag = GUINT32_FROM_LE (ltag);
}
switch (ltag) {
case GST_RIFF_LIST_INFO:{
} else {
wav->offset += 12;
gst_buffer_unref (buf);
+ buf = NULL;
if (data_size > 0) {
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset,
wav->tags =
gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
if (old)
- gst_tag_list_free (old);
- gst_tag_list_free (new);
+ gst_tag_list_unref (old);
+ gst_tag_list_unref (new);
}
gst_buffer_unref (buf);
wav->offset += GST_ROUND_UP_2 (data_size);
}
break;
}
+ case GST_RIFF_LIST_adtl:{
+ const gint data_size = size - 4;
+
+ GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
+ if (wav->streaming) {
+ const guint8 *data = NULL;
+
+ gst_adapter_flush (wav->adapter, 12);
+ wav->offset += 12;
+ data = gst_adapter_map (wav->adapter, data_size);
+ gst_wavparse_adtl_chunk (wav, data, data_size);
+ gst_adapter_unmap (wav->adapter);
+ } else {
+ GstMapInfo map;
+
+ gst_buffer_unref (buf);
+ buf = NULL;
+ wav->offset += 12;
+ if ((res =
+ gst_pad_pull_range (wav->sinkpad, wav->offset,
+ data_size, &buf)) != GST_FLOW_OK)
+ goto header_read_error;
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
+ data_size);
+ gst_buffer_unmap (buf, &map);
+ }
+ wav->offset += GST_ROUND_UP_2 (data_size);
+ break;
+ }
default:
- GST_INFO_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
+ GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (ltag));
if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
/* need more data */
}
break;
}
+ case GST_RIFF_TAG_cue:{
+ const guint data_size = size;
+
+ GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
+ if (wav->streaming) {
+ const guint8 *data = NULL;
+
+ if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
+ goto exit;
+ }
+ gst_adapter_flush (wav->adapter, 8);
+ wav->offset += 8;
+ data = gst_adapter_map (wav->adapter, data_size);
+ if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
+ goto header_read_error;
+ }
+ gst_adapter_unmap (wav->adapter);
+ } else {
+ GstMapInfo map;
+
+ wav->offset += 8;
+ gst_buffer_unref (buf);
+ buf = NULL;
+ if ((res =
+ gst_pad_pull_range (wav->sinkpad, wav->offset,
+ data_size, &buf)) != GST_FLOW_OK)
+ goto header_read_error;
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
+ data_size)) {
+ goto header_read_error;
+ }
+ gst_buffer_unmap (buf, &map);
+ }
+ size = GST_ROUND_UP_2 (size);
+ if (wav->streaming) {
+ gst_adapter_flush (wav->adapter, size);
+ } else {
+ gst_buffer_unref (buf);
+ }
+ size = GST_ROUND_UP_2 (size);
+ wav->offset += size;
+ break;
+ }
+ case GST_RIFF_TAG_smpl:{
+ const gint data_size = size;
+
+ GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size);
+ if (wav->streaming) {
+ const guint8 *data = NULL;
+
+ if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
+ goto exit;
+ }
+ gst_adapter_flush (wav->adapter, 8);
+ wav->offset += 8;
+ data = gst_adapter_map (wav->adapter, data_size);
+ if (!gst_wavparse_smpl_chunk (wav, data, data_size)) {
+ goto header_read_error;
+ }
+ gst_adapter_unmap (wav->adapter);
+ } else {
+ GstMapInfo map;
+
+ wav->offset += 8;
+ gst_buffer_unref (buf);
+ buf = NULL;
+ if ((res =
+ gst_pad_pull_range (wav->sinkpad, wav->offset,
+ data_size, &buf)) != GST_FLOW_OK)
+ goto header_read_error;
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data,
+ data_size)) {
+ goto header_read_error;
+ }
+ gst_buffer_unmap (buf, &map);
+ }
+ size = GST_ROUND_UP_2 (size);
+ if (wav->streaming) {
+ gst_adapter_flush (wav->adapter, size);
+ } else {
+ gst_buffer_unref (buf);
+ }
+ size = GST_ROUND_UP_2 (size);
+ wav->offset += size;
+ break;
+ }
default:
+ GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT,
+ GST_FOURCC_ARGS (tag));
if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
/* need more data */
goto exit;
wav->bps =
(guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
(guint64) wav->fact);
- GST_INFO_OBJECT (wav, "calculated bps : %d, enabling VBR", wav->bps);
+ GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
#endif
wav->vbr = TRUE;
}
if (gst_wavparse_calculate_duration (wav)) {
gst_segment_init (&wav->segment, GST_FORMAT_TIME);
- gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, wav->duration);
+ if (!wav->ignore_length)
+ wav->segment.duration = wav->duration;
+ if (!wav->toc)
+ gst_wavparse_create_toc (wav);
} else {
/* no bitrate, let downstream peer do the math, we'll feed it bytes. */
gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
- gst_segment_set_duration (&wav->segment, GST_FORMAT_BYTES, wav->datasize);
+ if (!wav->ignore_length)
+ wav->segment.duration = wav->datasize;
}
/* now we have all the info to perform a pending seek if any, if no
* the right newsegment event downstream. */
gst_wavparse_perform_seek (wav, wav->seek_event);
/* remove pending event */
- event_p = &wav->seek_event;
- gst_event_replace (event_p, NULL);
+ gst_event_replace (&wav->seek_event, NULL);
/* we just started, we are discont */
wav->discont = TRUE;
* that is, buffers not too small either size or time wise
* so we do not end up with too many of them */
/* var abuse */
- upstream_size = 0;
- gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size);
- wav->max_buf_size = upstream_size;
+ if (gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size))
+ wav->max_buf_size = upstream_size;
+ else
+ wav->max_buf_size = 0;
wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
if (wav->blockalign > 0)
wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
- GST_DEBUG_OBJECT (wav, "max buffer size %d", wav->max_buf_size);
+ GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
return GST_FLOW_OK;
/* ERROR */
exit:
{
- if (codec_name)
- g_free (codec_name);
- if (header)
- g_free (header);
+ g_free (codec_name);
+ g_free (header);
if (caps)
gst_caps_unref (caps);
return res;
{
GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
("Stream claims blockalign = %u, which is more than %u - invalid data",
- wav->blockalign, wav->channels * (guint) ceil (wav->depth / 8.0)));
+ wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
goto fail;
}
invalid_bps:
unknown_format:
{
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
- ("No caps found for format 0x%x, %d channels, %d Hz",
+ ("No caps found for format 0x%x, %u channels, %u Hz",
wav->format, wav->channels, wav->rate));
goto fail;
}
{
GstWavParse *wav = GST_WAVPARSE (element);
gboolean res = FALSE;
- GstEvent **event_p;
GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
} else {
GST_DEBUG_OBJECT (wav, "queuing seek for later");
- event_p = &wav->seek_event;
- gst_event_replace (event_p, event);
+ gst_event_replace (&wav->seek_event, event);
/* we always return true */
res = TRUE;
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_has_name (s, "audio/x-dts"))
return FALSE;
- if (prob >= GST_TYPE_FIND_LIKELY)
+ /* typefind behavior for DTS:
+ * MAXIMUM: multiple frame syncs detected, certainly DTS
+ * LIKELY: single frame sync at offset 0. Maybe DTS?
+ * POSSIBLE: single frame sync, not at offset 0. Highly unlikely
+ * to be DTS. */
+ if (prob > GST_TYPE_FIND_LIKELY)
return TRUE;
- /* DTS at non-0 offsets and without second sync may yield POSSIBLE .. */
- if (prob < GST_TYPE_FIND_POSSIBLE)
+ if (prob <= GST_TYPE_FIND_POSSIBLE)
return FALSE;
- /* .. in which case we want at least a valid-looking rate and channels */
+ /* for maybe, check for at least a valid-looking rate and channels */
if (!gst_structure_has_field (s, "channels"))
return FALSE;
/* and for extra assurance we could also check the rate from the DTS frame
return gst_structure_has_field (s, "rate");
}
+static GstTagList *
+gst_wavparse_get_upstream_tags (GstWavParse * wav, GstTagScope scope)
+{
+ GstTagList *tags = NULL;
+ GstEvent *ev;
+ gint i;
+
+ i = 0;
+ while ((ev = gst_pad_get_sticky_event (wav->sinkpad, GST_EVENT_TAG, i++))) {
+ gst_event_parse_tag (ev, &tags);
+ if (tags != NULL && gst_tag_list_get_scope (tags) == scope) {
+ tags = gst_tag_list_copy (tags);
+ gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
+ gst_event_unref (ev);
+ break;
+ }
+ tags = NULL;
+ gst_event_unref (ev);
+ }
+ return tags;
+}
+
static void
gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
{
GstStructure *s;
+ GstTagList *tags, *utags;
GST_DEBUG_OBJECT (wav, "adding src pad");
- if (wav->caps) {
- s = gst_caps_get_structure (wav->caps, 0);
- if (s && gst_structure_has_name (s, "audio/x-raw-int") && buf != NULL) {
- GstTypeFindProbability prob;
- GstCaps *tf_caps;
-
- tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
- if (tf_caps != NULL) {
- GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
- if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
- GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
- gst_caps_unref (wav->caps);
- wav->caps = tf_caps;
-
- gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
- GST_TAG_AUDIO_CODEC, "dts", NULL);
- } else {
- GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
- "marked as raw PCM audio, but ignoring for now", tf_caps);
- gst_caps_unref (tf_caps);
- }
+ g_assert (wav->caps != NULL);
+
+ s = gst_caps_get_structure (wav->caps, 0);
+ if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
+ GstTypeFindProbability prob;
+ GstCaps *tf_caps;
+
+ tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
+ if (tf_caps != NULL) {
+ GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
+ if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
+ GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
+ gst_caps_unref (wav->caps);
+ wav->caps = tf_caps;
+
+ gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
+ GST_TAG_AUDIO_CODEC, "dts", NULL);
+ } else {
+ GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
+ "marked as raw PCM audio, but ignoring for now", tf_caps);
+ gst_caps_unref (tf_caps);
}
}
}
- gst_wavparse_create_sourcepad (wav);
- gst_pad_set_active (wav->srcpad, TRUE);
gst_pad_set_caps (wav->srcpad, wav->caps);
gst_caps_replace (&wav->caps, NULL);
- gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad);
- gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
-
- if (wav->close_segment) {
- GST_DEBUG_OBJECT (wav, "Send close segment event on newpad");
- gst_pad_push_event (wav->srcpad, wav->close_segment);
- wav->close_segment = NULL;
- }
if (wav->start_segment) {
GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
gst_pad_push_event (wav->srcpad, wav->start_segment);
wav->start_segment = NULL;
}
- if (wav->tags) {
- gst_element_found_tags_for_pad (GST_ELEMENT_CAST (wav), wav->srcpad,
- wav->tags);
+ /* upstream tags, e.g. from id3/ape tag before the wav file; assume for now
+ * that there'll be only one scope/type of tag list from upstream, if any */
+ utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_GLOBAL);
+ if (utags == NULL)
+ utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_STREAM);
+
+ /* if there's a tag upstream it's probably been added to override the
+ * tags from inside the wav header, so keep upstream tags if in doubt */
+ tags = gst_tag_list_merge (utags, wav->tags, GST_TAG_MERGE_KEEP);
+
+ if (wav->tags != NULL) {
+ gst_tag_list_unref (wav->tags);
wav->tags = NULL;
}
+
+ if (utags != NULL)
+ gst_tag_list_unref (utags);
+
+ /* send tags downstream, if any */
+ if (tags != NULL)
+ gst_pad_push_event (wav->srcpad, gst_event_new_tag (tags));
}
static GstFlowReturn
* amounts of data regardless of the playback rate */
desired =
MIN (gst_guint64_to_gdouble (wav->dataleft),
- wav->max_buf_size * wav->segment.abs_rate);
+ wav->max_buf_size * ABS (wav->segment.rate));
if (desired >= wav->blockalign && wav->blockalign > 0)
desired -= (desired % wav->blockalign);
if (G_UNLIKELY (extra)) {
extra = wav->bytes_per_sample - extra;
if (extra <= avail) {
- GST_DEBUG_OBJECT (wav, "flushing %d bytes to sample boundary", extra);
+ GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
gst_adapter_flush (wav->adapter, extra);
wav->offset += extra;
wav->dataleft -= extra;
goto iterate_adapter;
} else {
- GST_DEBUG_OBJECT (wav, "flushing %d bytes", avail);
+ GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
gst_adapter_clear (wav->adapter);
wav->offset += avail;
wav->dataleft -= avail;
}
if (avail < desired) {
- GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
+ GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
return GST_FLOW_OK;
}
goto pull_error;
/* we may get a short buffer at the end of the file */
- if (GST_BUFFER_SIZE (buf) < desired) {
- GST_LOG_OBJECT (wav, "Got only %u bytes of data", GST_BUFFER_SIZE (buf));
- if (GST_BUFFER_SIZE (buf) >= wav->blockalign) {
- buf = gst_buffer_make_metadata_writable (buf);
- GST_BUFFER_SIZE (buf) -= (GST_BUFFER_SIZE (buf) % wav->blockalign);
+ if (gst_buffer_get_size (buf) < desired) {
+ gsize size = gst_buffer_get_size (buf);
+
+ GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
+ if (size >= wav->blockalign) {
+ if (wav->blockalign > 0) {
+ buf = gst_buffer_make_writable (buf);
+ gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
+ }
} else {
gst_buffer_unref (buf);
goto found_eos;
}
}
- obtained = GST_BUFFER_SIZE (buf);
+ obtained = gst_buffer_get_size (buf);
/* our positions in bytes */
pos = wav->offset - wav->datastart;
nextpos = pos + obtained;
/* update offsets, does not overflow. */
+ buf = gst_buffer_make_writable (buf);
GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
/* this will also push the segment events */
gst_wavparse_add_src_pad (wav, buf);
} else {
- /* If we have a pending close/start segment, send it now. */
- if (G_UNLIKELY (wav->close_segment != NULL)) {
- gst_pad_push_event (wav->srcpad, wav->close_segment);
- wav->close_segment = NULL;
- }
+ /* If we have a pending start segment, send it now. */
if (G_UNLIKELY (wav->start_segment != NULL)) {
gst_pad_push_event (wav->srcpad, wav->start_segment);
wav->start_segment = NULL;
if (wav->bps > 0) {
/* and timestamps if we have a bitrate, be careful for overflows */
- timestamp = uint64_ceiling_scale (pos, GST_SECOND, (guint64) wav->bps);
+ timestamp =
+ gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
next_timestamp =
- uint64_ceiling_scale (nextpos, GST_SECOND, (guint64) wav->bps);
+ gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
duration = next_timestamp - timestamp;
/* update current running segment position */
if (G_LIKELY (next_timestamp >= wav->segment.start))
- gst_segment_set_last_stop (&wav->segment, GST_FORMAT_TIME,
- next_timestamp);
+ wav->segment.position = next_timestamp;
} else if (wav->fact) {
guint64 bps =
gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
/* and timestamps if we have a bitrate, be careful for overflows */
- timestamp = uint64_ceiling_scale (pos, GST_SECOND, bps);
- next_timestamp = uint64_ceiling_scale (nextpos, GST_SECOND, bps);
+ timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
+ next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
duration = next_timestamp - timestamp;
} else {
/* no bitrate, all we know is that the first sample has timestamp 0, all
duration = GST_CLOCK_TIME_NONE;
/* update current running segment position with byte offset */
if (G_LIKELY (nextpos >= wav->segment.start))
- gst_segment_set_last_stop (&wav->segment, GST_FORMAT_BYTES, nextpos);
+ wav->segment.position = nextpos;
}
if ((pos > 0) && wav->vbr) {
/* don't set timestamps for VBR files if it's not the first buffer */
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
- /* don't forget to set the caps on the buffer */
- gst_buffer_set_caps (buf, GST_PAD_CAPS (wav->srcpad));
-
GST_LOG_OBJECT (wav,
"Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
- ", size:%u", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
- GST_BUFFER_SIZE (buf));
+ ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
+ GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
found_eos:
{
GST_DEBUG_OBJECT (wav, "found EOS");
- return GST_FLOW_UNEXPECTED;
+ return GST_FLOW_EOS;
}
pull_error:
{
/* check if we got EOS */
- if (res == GST_FLOW_UNEXPECTED)
+ if (res == GST_FLOW_EOS)
goto found_eos;
GST_WARNING_OBJECT (wav,
{
GstFlowReturn ret;
GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
+ GstEvent *event;
+ gchar *stream_id;
GST_LOG_OBJECT (wav, "process data");
if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
goto pause;
+ stream_id =
+ gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL);
+ event = gst_event_new_stream_start (stream_id);
+ gst_event_set_group_id (event, gst_util_group_id_next ());
+ gst_pad_push_event (wav->srcpad, event);
+ g_free (stream_id);
+
wav->state = GST_WAVPARSE_HEADER;
/* fall-through */
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
- wav->segment_running = FALSE;
gst_pad_pause_task (pad);
- if (ret == GST_FLOW_UNEXPECTED) {
- /* add pad before we perform EOS */
- if (G_UNLIKELY (wav->first)) {
- wav->first = FALSE;
- gst_wavparse_add_src_pad (wav, NULL);
+ if (ret == GST_FLOW_EOS) {
+ /* handle end-of-stream/segment */
+ /* so align our position with the end of it, if there is one
+ * this ensures a subsequent will arrive at correct base/acc time */
+ if (wav->segment.format == GST_FORMAT_TIME) {
+ if (wav->segment.rate > 0.0 &&
+ GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
+ wav->segment.position = wav->segment.stop;
+ else if (wav->segment.rate < 0.0)
+ wav->segment.position = wav->segment.start;
}
+ if (wav->state == GST_WAVPARSE_START) {
+ GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
+ ("No valid input found before end of stream"));
+ gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
+ } else {
+ /* add pad before we perform EOS */
+ if (G_UNLIKELY (wav->first)) {
+ wav->first = FALSE;
+ gst_wavparse_add_src_pad (wav, NULL);
+ }
- if (wav->state == GST_WAVPARSE_START)
- GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE,
- ("No valid input found before end of stream"), (NULL));
-
- /* perform EOS logic */
- if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
- GstClockTime stop;
+ /* perform EOS logic */
+ if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
+ GstClockTime stop;
- if ((stop = wav->segment.stop) == -1)
- stop = wav->segment.duration;
+ if ((stop = wav->segment.stop) == -1)
+ stop = wav->segment.duration;
- gst_element_post_message (GST_ELEMENT_CAST (wav),
- gst_message_new_segment_done (GST_OBJECT_CAST (wav),
- wav->segment.format, stop));
- } else {
- if (wav->srcpad != NULL)
+ gst_element_post_message (GST_ELEMENT_CAST (wav),
+ gst_message_new_segment_done (GST_OBJECT_CAST (wav),
+ wav->segment.format, stop));
+ gst_pad_push_event (wav->srcpad,
+ gst_event_new_segment_done (wav->segment.format, stop));
+ } else {
gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
+ }
}
- } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_UNEXPECTED) {
+ } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_ERROR (wav, STREAM, FAILED,
(_("Internal data flow error.")),
("streaming task paused, reason %s (%d)", reason, ret));
- if (wav->srcpad != NULL)
- gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
+ gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
}
return;
}
}
static GstFlowReturn
-gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
+gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret;
- GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
+ GstWavParse *wav = GST_WAVPARSE (parent);
- GST_LOG_OBJECT (wav, "adapter_push %u bytes", GST_BUFFER_SIZE (buf));
+ GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
+ gst_buffer_get_size (buf));
gst_adapter_push (wav->adapter, buf);
guint av;
if ((av = gst_adapter_available (wav->adapter)) > 0) {
- wav->dataleft = av;
- wav->end_offset = wav->offset + av;
ret = gst_wavparse_stream_data (wav);
}
}
static gboolean
-gst_wavparse_sink_event (GstPad * pad, GstEvent * event)
+gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
- GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
+ GstWavParse *wav = GST_WAVPARSE (parent);
gboolean ret = TRUE;
GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_NEWSEGMENT:
+ case GST_EVENT_CAPS:
{
- GstFormat format;
- gdouble rate, arate;
- gint64 start, stop, time, offset = 0, end_offset = -1;
- gboolean update;
+ /* discard, we'll come up with proper src caps */
+ gst_event_unref (event);
+ break;
+ }
+ case GST_EVENT_SEGMENT:
+ {
+ gint64 start, stop, offset = 0, end_offset = -1;
GstSegment segment;
/* some debug output */
- gst_segment_init (&segment, GST_FORMAT_UNDEFINED);
- gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
- &start, &stop, &time);
- gst_segment_set_newsegment_full (&segment, update, rate, arate, format,
- start, stop, time);
- GST_DEBUG_OBJECT (wav,
- "received format %d newsegment %" GST_SEGMENT_FORMAT, format,
+ gst_event_copy_segment (event, &segment);
+ GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
&segment);
if (wav->state != GST_WAVPARSE_DATA) {
/* now we are either committed to TIME or BYTE format,
* and we only expect a BYTE segment, e.g. following a seek */
- if (format == GST_FORMAT_BYTES) {
+ if (segment.format == GST_FORMAT_BYTES) {
+ /* handle (un)signed issues */
+ start = segment.start;
+ stop = segment.stop;
if (start > 0) {
offset = start;
start -= wav->datastart;
if (bps) {
if (start >= 0)
start =
- uint64_ceiling_scale (start, GST_SECOND, (guint64) wav->bps);
+ gst_util_uint64_scale_ceil (start, GST_SECOND,
+ (guint64) wav->bps);
if (stop >= 0)
stop =
- uint64_ceiling_scale (stop, GST_SECOND, (guint64) wav->bps);
+ gst_util_uint64_scale_ceil (stop, GST_SECOND,
+ (guint64) wav->bps);
}
}
} else {
goto exit;
}
+ segment.start = start;
+ segment.stop = stop;
+
/* accept upstream's notion of segment and distribute along */
- gst_segment_set_newsegment_full (&wav->segment, update, rate, arate,
- wav->segment.format, start, stop, start);
+ segment.format = wav->segment.format;
+ segment.time = segment.position = segment.start;
+ segment.duration = wav->segment.duration;
+ segment.base = gst_segment_to_running_time (&wav->segment,
+ GST_FORMAT_TIME, wav->segment.position);
+
+ gst_segment_copy_into (&segment, &wav->segment);
+
/* also store the newsegment event for the streaming thread */
if (wav->start_segment)
gst_event_unref (wav->start_segment);
- wav->start_segment =
- gst_event_new_new_segment_full (update, rate, arate,
- wav->segment.format, start, stop, start);
- GST_DEBUG_OBJECT (wav, "Pushing newseg update %d, rate %g, "
- "applied rate %g, format %d, start %" G_GINT64_FORMAT ", "
- "stop %" G_GINT64_FORMAT, update, rate, arate, wav->segment.format,
- start, stop);
+ GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
+ wav->start_segment = gst_event_new_segment (&segment);
/* stream leftover data in current segment */
gst_wavparse_flush_data (wav);
/* and set up streaming thread for next one */
wav->offset = offset;
wav->end_offset = end_offset;
- if (wav->end_offset > 0) {
+
+ if (wav->datasize > 0 && (wav->end_offset == -1
+ || wav->end_offset > wav->datastart + wav->datasize))
+ wav->end_offset = wav->datastart + wav->datasize;
+
+ if (wav->end_offset != -1) {
wav->dataleft = wav->end_offset - wav->offset;
} else {
/* infinity; upstream will EOS when done */
break;
}
case GST_EVENT_EOS:
- /* add pad if needed so EOS is seen downstream */
- if (G_UNLIKELY (wav->first)) {
- wav->first = FALSE;
- gst_wavparse_add_src_pad (wav, NULL);
+ if (wav->state == GST_WAVPARSE_START) {
+ GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
+ ("No valid input found before end of stream"));
} else {
- /* stream leftover data in current segment */
- gst_wavparse_flush_data (wav);
+ /* add pad if needed so EOS is seen downstream */
+ if (G_UNLIKELY (wav->first)) {
+ wav->first = FALSE;
+ gst_wavparse_add_src_pad (wav, NULL);
+ } else {
+ /* stream leftover data in current segment */
+ gst_wavparse_flush_data (wav);
+ }
}
- if (wav->state == GST_WAVPARSE_START)
- GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE,
- ("No valid input found before end of stream"), (NULL));
-
/* fall-through */
case GST_EVENT_FLUSH_STOP:
+ {
+ GstClockTime dur;
+
gst_adapter_clear (wav->adapter);
wav->discont = TRUE;
+ dur = wav->segment.duration;
+ gst_segment_init (&wav->segment, wav->segment.format);
+ wav->segment.duration = dur;
/* fall-through */
+ }
default:
- ret = gst_pad_event_default (wav->sinkpad, event);
+ ret = gst_pad_event_default (wav->sinkpad, parent, event);
break;
}
static const GstFormat *
gst_wavparse_get_formats (GstPad * pad)
{
- static GstFormat formats[] = {
+ static const GstFormat formats[] = {
GST_FORMAT_TIME,
GST_FORMAT_BYTES,
GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
"src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
wavparse->offset);
if (wavparse->bps > 0)
- *dest_value = uint64_ceiling_scale (src_value, GST_SECOND,
+ *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
(guint64) wavparse->bps);
else if (wavparse->fact) {
- guint64 bps = uint64_ceiling_scale_int (wavparse->datasize,
+ guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
wavparse->rate, wavparse->fact);
- *dest_value = uint64_ceiling_scale_int (src_value, GST_SECOND, bps);
+ *dest_value =
+ gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
} else {
res = FALSE;
}
}
}
-static const GstQueryType *
-gst_wavparse_get_query_types (GstPad * pad)
-{
- static const GstQueryType types[] = {
- GST_QUERY_POSITION,
- GST_QUERY_DURATION,
- GST_QUERY_CONVERT,
- GST_QUERY_SEEKING,
- 0
- };
-
- return types;
-}
-
/* handle queries for location and length in requested format */
static gboolean
-gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
+gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
gboolean res = TRUE;
- GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (pad));
+ GstWavParse *wav = GST_WAVPARSE (parent);
/* only if we know */
if (wav->state != GST_WAVPARSE_DATA) {
- gst_object_unref (wav);
return FALSE;
}
GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
switch (format) {
- case GST_FORMAT_TIME:
- res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
- &format, &cur);
- break;
- default:
+ case GST_FORMAT_BYTES:
format = GST_FORMAT_BYTES;
cur = curb;
break;
+ default:
+ res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
+ &format, &cur);
+ break;
}
if (res)
gst_query_set_position (query, format, cur);
gint64 duration = 0;
GstFormat format;
+ if (wav->ignore_length) {
+ res = FALSE;
+ break;
+ }
+
gst_query_parse_duration (query, &format, NULL);
switch (format) {
- case GST_FORMAT_TIME:{
+ case GST_FORMAT_BYTES:{
+ format = GST_FORMAT_BYTES;
+ duration = wav->datasize;
+ break;
+ }
+ case GST_FORMAT_TIME:
if ((res = gst_wavparse_calculate_duration (wav))) {
duration = wav->duration;
}
break;
- }
default:
- format = GST_FORMAT_BYTES;
- duration = wav->datasize;
+ res = FALSE;
break;
}
- gst_query_set_duration (query, format, duration);
+ if (res)
+ gst_query_set_duration (query, format, duration);
break;
}
case GST_QUERY_CONVERT:
break;
}
default:
- res = gst_pad_query_default (pad, query);
+ res = gst_pad_query_default (pad, parent, query);
break;
}
- gst_object_unref (wav);
return res;
}
static gboolean
-gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event)
+gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
- GstWavParse *wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
+ GstWavParse *wavparse = GST_WAVPARSE (parent);
gboolean res = FALSE;
GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
}
gst_event_unref (event);
break;
+
+ case GST_EVENT_TOC_SELECT:
+ {
+ char *uid = NULL;
+ GstTocEntry *entry = NULL;
+ GstEvent *seek_event;
+ gint64 start_pos;
+
+ if (!wavparse->toc) {
+ GST_DEBUG_OBJECT (wavparse, "no TOC to select");
+ return FALSE;
+ } else {
+ gst_event_parse_toc_select (event, &uid);
+ if (uid != NULL) {
+ GST_OBJECT_LOCK (wavparse);
+ entry = gst_toc_find_entry (wavparse->toc, uid);
+ if (entry == NULL) {
+ GST_OBJECT_UNLOCK (wavparse);
+ GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
+ uid);
+ res = FALSE;
+ } else {
+ gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
+ GST_OBJECT_UNLOCK (wavparse);
+ seek_event = gst_event_new_seek (1.0,
+ GST_FORMAT_TIME,
+ GST_SEEK_FLAG_FLUSH,
+ GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
+ res = gst_wavparse_perform_seek (wavparse, seek_event);
+ gst_event_unref (seek_event);
+ }
+ g_free (uid);
+ } else {
+ GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
+ res = FALSE;
+ }
+ }
+ gst_event_unref (event);
+ break;
+ }
+
default:
res = gst_pad_push_event (wavparse->sinkpad, event);
break;
}
- gst_object_unref (wavparse);
return res;
}
static gboolean
-gst_wavparse_sink_activate (GstPad * sinkpad)
+gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
{
- GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
- gboolean res;
+ GstWavParse *wav = GST_WAVPARSE (parent);
+ GstQuery *query;
+ gboolean pull_mode;
if (wav->adapter) {
gst_adapter_clear (wav->adapter);
wav->adapter = NULL;
}
- if (gst_pad_check_pull_range (sinkpad)) {
- GST_DEBUG ("going to pull mode");
- wav->streaming = FALSE;
- res = gst_pad_activate_pull (sinkpad, TRUE);
- } else {
- GST_DEBUG ("going to push (streaming) mode");
+ query = gst_query_new_scheduling ();
+
+ if (!gst_pad_peer_query (sinkpad, query)) {
+ gst_query_unref (query);
+ goto activate_push;
+ }
+
+ pull_mode = gst_query_has_scheduling_mode_with_flags (query,
+ GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
+ gst_query_unref (query);
+
+ if (!pull_mode)
+ goto activate_push;
+
+ GST_DEBUG_OBJECT (sinkpad, "activating pull");
+ wav->streaming = FALSE;
+ return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
+
+activate_push:
+ {
+ GST_DEBUG_OBJECT (sinkpad, "activating push");
wav->streaming = TRUE;
wav->adapter = gst_adapter_new ();
- res = gst_pad_activate_push (sinkpad, TRUE);
+ return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
}
- gst_object_unref (wav);
- return res;
}
static gboolean
-gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active)
+gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
+ GstPadMode mode, gboolean active)
{
- GstWavParse *wav = GST_WAVPARSE (GST_OBJECT_PARENT (sinkpad));
+ gboolean res;
- if (active) {
- /* if we have a scheduler we can start the task */
- wav->segment_running = TRUE;
- return gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
- sinkpad);
- } else {
- wav->segment_running = FALSE;
- return gst_pad_stop_task (sinkpad);
+ switch (mode) {
+ case GST_PAD_MODE_PUSH:
+ res = TRUE;
+ break;
+ case GST_PAD_MODE_PULL:
+ if (active) {
+ /* if we have a scheduler we can start the task */
+ res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
+ sinkpad, NULL);
+ } else {
+ res = gst_pad_stop_task (sinkpad);
+ }
+ break;
+ default:
+ res = FALSE;
+ break;
}
-};
+ return res;
+}
static GstStateChangeReturn
gst_wavparse_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
- gst_wavparse_destroy_sourcepad (wav);
gst_wavparse_reset (wav);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
return ret;
}
+static void
+gst_wavparse_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstWavParse *self;
+
+ g_return_if_fail (GST_IS_WAVPARSE (object));
+ self = GST_WAVPARSE (object);
+
+ switch (prop_id) {
+ case PROP_IGNORE_LENGTH:
+ self->ignore_length = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
+ }
+
+}
+
+static void
+gst_wavparse_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstWavParse *self;
+
+ g_return_if_fail (GST_IS_WAVPARSE (object));
+ self = GST_WAVPARSE (object);
+
+ switch (prop_id) {
+ case PROP_IGNORE_LENGTH:
+ g_value_set_boolean (value, self->ignore_length);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
+ }
+}
+
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
- "wavparse",
+ wavparse,
"Parse a .wav file into raw audio",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
projects / platform / upstream / gst-plugins-good.git / blobdiff