#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
+
#include <string.h>
#include <math.h>
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 ("sink",
GST_PAD_SINK,
G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
DEBUG_INIT);
+typedef struct
+{
+ /* 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;
+
static void
gst_wavparse_class_init (GstWavParseClass * 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.
+ *
+ * Since: 0.10.36
+ */
+ 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;
GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
gst_element_class_add_pad_template (gstelement_class, src_template);
- gst_element_class_set_details_simple (gstelement_class, "WAV audio demuxer",
+ 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>");
if (wav->tags)
gst_tag_list_free (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;
gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
}
-/* 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
/* 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);
+ /* FIXME: Need to handle the channel reorder map */
+ caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL, NULL);
g_free (header);
if (caps == NULL)
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);
+ GST_DEBUG ("frequency %u, channels %u", wav->rate, wav->channels);
return TRUE;
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);
+ GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %u", tag,
+ (const gchar *) &tag, length);
switch (tag) {
case GST_RIFF_TAG_LIST:
}
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);
+ *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
return TRUE;
}
if (!event || wav->state != GST_WAVPARSE_DATA) {
if (wav->start_segment)
gst_event_unref (wav->start_segment);
- // TODO
-/* 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 */
/* and start the streaming task again */
if (!wav->streaming) {
gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
- wav->sinkpad);
+ wav->sinkpad, NULL);
}
GST_PAD_STREAM_UNLOCK (wav->sinkpad);
*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_int_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 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 = (gchar *) g_new0 (gchar *, size - 4 + 1);
+ memcpy (labl->text, data + 4, size - 4);
+
+ wav->labls = g_list_append (wav->labls, labl);
+
+ 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);
+ switch (ltag) {
+ case GST_RIFF_TAG_labl:
+ gst_wavparse_labl_chunk (wav, data + offset, size);
+ default:
+ break;
+ }
+ offset += 8 + GST_ROUND_UP_2 (lsize);
+ size -= 8 + GST_ROUND_UP_2 (lsize);
+ }
+
+ return TRUE;
+}
+
+/*
+ * 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;
+ 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;
+ }
+
+ toc = gst_toc_new ();
+
+ /* 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 chapters in cue edition */
+ list = g_list_first (wav->cues);
+ while (list != NULL) {
+ cue = list->data;
+ prev_subentry = cur_subentry;
+ /* previous chapter stop time = current chapter 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_CHAPTER, 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 chapters */
+ list = g_list_first (wav->labls);
+ while (list != NULL) {
+ labl = list->data;
+ id = g_strdup_printf ("%08x", labl->cue_point_id);
+ cur_subentry = gst_toc_find_entry (toc, id);
+ g_free (id);
+ if (cur_subentry != NULL) {
+ tags = gst_tag_list_new_empty ();
+ gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
+ NULL);
+ gst_toc_entry_set_tags (cur_subentry, tags);
+ }
+ 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 GstFlowReturn
/* 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);
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:
if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
goto exit;
} else {
- guint8 *data;
+ GstMapInfo map;
+ buf = NULL;
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
&buf)) != GST_FLOW_OK)
goto header_read_error;
- data = gst_buffer_map (buf, NULL, NULL, -1);
- tag = GST_READ_UINT32_LE (data);
- size = GST_READ_UINT32_LE (data + 4);
- gst_buffer_unmap (buf, data, -1);
+ 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,
*/
switch (tag) {
case GST_RIFF_TAG_data:{
- GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
+ GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
+ if (wav->ignore_length) {
+ GST_DEBUG_OBJECT (wav, "Ignoring length");
+ size = 0;
+ }
if (wav->streaming) {
gst_adapter_flush (wav->adapter, 8);
gotdata = TRUE;
/* We will continue parsing tags 'till end */
wav->offset += size;
}
- GST_DEBUG_OBJECT (wav, "datasize = %d", size);
+ GST_DEBUG_OBJECT (wav, "datasize = %u", size);
break;
}
case GST_RIFF_TAG_fact:{
/* 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;
}
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)
/* 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;
}
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_map (buf, NULL, NULL,
- GST_MAP_READ);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ acid = (const gst_riff_acid *) map.data;
tempo = acid->tempo;
- gst_buffer_unmap (buf, (guint8 *) acid, -1);
+ gst_buffer_unmap (buf, &map);
}
/* send data as tags */
if (!wav->tags)
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)
} else {
wav->offset += 12;
gst_buffer_unref (buf);
+ buf = NULL;
if (data_size > 0) {
if ((res =
gst_pad_pull_range (wav->sinkpad, wav->offset,
}
break;
}
+ case GST_RIFF_LIST_adtl:{
+ const gint data_size = size;
+
+ GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
+ if (wav->streaming) {
+ const guint8 *data = NULL;
+
+ gst_adapter_flush (wav->adapter, 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;
+ if ((res =
+ gst_pad_pull_range (wav->sinkpad, wav->offset + 12,
+ 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);
+ }
+ }
default:
GST_INFO_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (ltag));
}
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;
+ }
default:
if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
/* need more data */
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);
- wav->segment.duration = 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);
- wav->segment.duration = 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
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;
{
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;
}
}
if (wav->tags) {
- gst_pad_push_event (wav->srcpad, gst_event_new_tag (wav->tags));
+ gst_pad_push_event (wav->srcpad, gst_event_new_tag ("GstParser",
+ wav->tags));
wav->tags = NULL;
}
}
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;
}
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 */
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
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,
GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
gst_pad_pause_task (pad);
- if (ret == GST_FLOW_UNEXPECTED) {
+ 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 */
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,
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 {
segment.stop = stop;
/* accept upstream's notion of segment and distribute along */
- segment.time = segment.start = segment.position;
+ 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);
"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;
}
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:
}
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;
if (active) {
/* if we have a scheduler we can start the task */
res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
- sinkpad);
+ sinkpad, NULL);
} else {
res = gst_pad_stop_task (sinkpad);
}
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