static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse,
GstFormat src_format, gint64 src_value,
GstFormat dest_format, gint64 * dest_value);
-static GstCaps *gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse);
+static GstCaps *gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse,
+ GstCaps * filter);
-GST_BOILERPLATE (GstMpegAudioParse, gst_mpeg_audio_parse, GstBaseParse,
- GST_TYPE_BASE_PARSE);
+#define gst_mpeg_audio_parse_parent_class parent_class
+G_DEFINE_TYPE (GstMpegAudioParse, gst_mpeg_audio_parse, GST_TYPE_BASE_PARSE);
#define GST_TYPE_MPEG_AUDIO_CHANNEL_MODE \
(gst_mpeg_audio_channel_mode_get_type())
}
static void
-gst_mpeg_audio_parse_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_template));
-
- gst_element_class_set_details_simple (element_class, "MPEG1 Audio Parser",
- "Codec/Parser/Audio",
- "Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
- "Jan Schmidt <thaytan@mad.scientist.com>,"
- "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
-}
-
-static void
gst_mpeg_audio_parse_class_init (GstMpegAudioParseClass * klass)
{
GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (mpeg_audio_parse_debug, "mpegaudioparse", 0,
"channel mode", "MPEG audio channel mode", NULL);
g_type_class_ref (GST_TYPE_MPEG_AUDIO_CHANNEL_MODE);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_template));
+
+ gst_element_class_set_details_simple (element_class, "MPEG1 Audio Parser",
+ "Codec/Parser/Audio",
+ "Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
+ "Jan Schmidt <thaytan@mad.scientist.com>,"
+ "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
}
static void
}
static void
-gst_mpeg_audio_parse_init (GstMpegAudioParse * mp3parse,
- GstMpegAudioParseClass * klass)
+gst_mpeg_audio_parse_init (GstMpegAudioParse * mp3parse)
{
gst_mpeg_audio_parse_reset (mp3parse);
}
guint32 header, int bpf, gboolean at_eos, gint * valid)
{
guint32 next_header;
- const guint8 *data;
- guint available;
+ guint8 *data;
+ gsize available;
+ gboolean res = TRUE;
int frames_found = 1;
int offset = bpf;
- available = GST_BUFFER_SIZE (buf);
- data = GST_BUFFER_DATA (buf);
+ data = gst_buffer_map (buf, &available, NULL, GST_MAP_READ);
while (frames_found < MIN_RESYNC_FRAMES) {
/* Check if we have enough data for all these frames, plus the next
if (at_eos) {
/* Running out of data at EOS is fine; just accept it */
*valid = TRUE;
- return TRUE;
+ goto cleanup;
} else {
*valid = offset + 4;
- return FALSE;
+ res = FALSE;
+ goto cleanup;
}
}
(guint) header, (guint) header & HDRMASK, (guint) next_header,
(guint) next_header & HDRMASK, bpf);
*valid = FALSE;
- return TRUE;
+ goto cleanup;
} else if ((((next_header >> 12) & 0xf) == 0) ||
(((next_header >> 12) & 0xf) == 0xf)) {
/* The essential parts were the same, but the bitrate held an
invalid value - also reject */
GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
*valid = FALSE;
- return TRUE;
+ goto cleanup;
}
bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
}
*valid = TRUE;
- return TRUE;
+
+cleanup:
+ gst_buffer_unmap (buf, data, available);
+ return res;
}
static gboolean
{
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
GstBuffer *buf = frame->buffer;
- GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
+ GstByteReader reader;
gint off, bpf;
gboolean lost_sync, draining, valid, caps_change;
guint32 header;
guint bitrate, layer, rate, channels, version, mode, crc;
+ guint8 *data;
+ gsize bufsize;
+ gboolean res = FALSE;
- if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < 6))
- return FALSE;
+ data = gst_buffer_map (buf, &bufsize, NULL, GST_MAP_READ);
+ if (G_UNLIKELY (bufsize < 6))
+ goto cleanup;
+
+ gst_byte_reader_init (&reader, data, bufsize);
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffe00000, 0xffe00000,
- 0, GST_BUFFER_SIZE (buf));
+ 0, bufsize);
GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
/* didn't find anything that looks like a sync word, skip */
if (off < 0) {
- *skipsize = GST_BUFFER_SIZE (buf) - 3;
- return FALSE;
+ *skipsize = bufsize - 3;
+ goto cleanup;
}
/* possible frame header, but not at offset 0? skip bytes before sync */
if (off > 0) {
*skipsize = off;
- return FALSE;
+ goto cleanup;
}
/* make sure the values in the frame header look sane */
- header = GST_READ_UINT32_BE (GST_BUFFER_DATA (buf));
+ header = GST_READ_UINT32_BE (data);
if (!gst_mpeg_audio_parse_head_check (mp3parse, header)) {
*skipsize = 1;
- return FALSE;
+ goto cleanup;
}
GST_LOG_OBJECT (parse, "got frame");
/* not enough data */
gst_base_parse_set_min_frame_size (parse, valid);
*skipsize = 0;
- return FALSE;
+ goto cleanup;
} else {
if (!valid) {
*skipsize = off + 2;
- return FALSE;
+ goto cleanup;
}
}
} else if (draining && lost_sync && caps_change && mp3parse->rate > 0) {
/* avoid caps jitter that we can't be sure of */
*skipsize = off + 2;
- return FALSE;
+ goto cleanup;
}
/* restore default minimum */
gst_base_parse_set_min_frame_size (parse, MIN_FRAME_SIZE);
*framesize = bpf;
- return TRUE;
+ res = TRUE;
+
+cleanup:
+ gst_buffer_unmap (buf, data, bufsize);
+ return res;
}
static void
gint offset_xing, offset_vbri;
guint64 avail;
gint64 upstream_total_bytes = 0;
- GstFormat fmt = GST_FORMAT_BYTES;
guint32 read_id_xing = 0, read_id_vbri = 0;
- const guint8 *data;
+ guint8 *data, *origdata;
+ gsize bufsize;
guint bitrate;
if (mp3parse->sent_codec_tag)
offset_vbri += 4;
/* Check if we have enough data to read the Xing header */
- avail = GST_BUFFER_SIZE (buf);
- data = GST_BUFFER_DATA (buf);
+ origdata = data = gst_buffer_map (buf, &bufsize, NULL, GST_MAP_READ);
+ avail = bufsize;
if (avail >= offset_xing + 4) {
read_id_xing = GST_READ_UINT32_BE (data + offset_xing);
}
/* obtain real upstream total bytes */
- fmt = GST_FORMAT_BYTES;
- if (!gst_pad_query_peer_duration (GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE
- (mp3parse)), &fmt, &upstream_total_bytes))
+ if (!gst_pad_peer_query_duration (GST_BASE_PARSE_SINK_PAD (mp3parse),
+ GST_FORMAT_BYTES, &upstream_total_bytes))
upstream_total_bytes = 0;
if (read_id_xing == xing_id || read_id_xing == info_id) {
if (avail < bytes_needed) {
GST_DEBUG_OBJECT (mp3parse,
"Not enough data to read Xing header (need %d)", bytes_needed);
- return;
+ goto cleanup;
}
GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
if (avail < offset_vbri + 26) {
GST_DEBUG_OBJECT (mp3parse,
"Not enough data to read VBRI header (need %d)", offset_vbri + 26);
- return;
+ goto cleanup;
}
GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header");
if (GST_READ_UINT16_BE (data) != 0x0001) {
GST_WARNING_OBJECT (mp3parse,
"Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data));
- return;
+ goto cleanup;
}
data += 2;
GST_DEBUG_OBJECT (mp3parse,
"Not enough data to read VBRI header (need %d)",
offset_vbri + 26 + nseek_points * seek_bytes);
- return;
+ goto cleanup;
}
- data = GST_BUFFER_DATA (buf);
+ data = origdata;
data += offset_vbri + 26;
/* VBRI seek table: frame/seek_frames -> byte */
bitrate = 0;
gst_base_parse_set_average_bitrate (GST_BASE_PARSE (mp3parse), bitrate);
+
+cleanup:
+ gst_buffer_unmap (buf, origdata, bufsize);
}
static GstFlowReturn
{
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
GstBuffer *buf = frame->buffer;
+ guint8 *data;
+ gsize bufsize;
guint bitrate, layer, rate, channels, version, mode, crc;
- g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= 4, GST_FLOW_ERROR);
+ data = gst_buffer_map (buf, &bufsize, NULL, GST_MAP_READ);
+ if (G_UNLIKELY (bufsize < 4))
+ goto short_buffer;
if (!mp3_type_frame_length_from_header (mp3parse,
- GST_READ_UINT32_BE (GST_BUFFER_DATA (buf)),
+ GST_READ_UINT32_BE (data),
&version, &layer, &channels, &bitrate, &rate, &mode, &crc))
goto broken_header;
"layer", G_TYPE_INT, layer,
"rate", G_TYPE_INT, rate,
"channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
- gst_buffer_set_caps (buf, caps);
gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
gst_caps_unref (caps);
mp3parse->last_crc = crc;
mp3parse->last_mode = mode;
+ gst_buffer_unmap (buf, data, bufsize);
return GST_FLOW_OK;
/* ERRORS */
broken_header:
{
/* this really shouldn't ever happen */
+ gst_buffer_unmap (buf, data, bufsize);
GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL), (NULL));
return GST_FLOW_ERROR;
}
+
+short_buffer:
+ {
+ gst_buffer_unmap (buf, data, bufsize);
+ return GST_FLOW_ERROR;
+ }
}
static gboolean
codec = g_strdup_printf ("MPEG %d Audio, Layer %d",
mp3parse->version, mp3parse->layer);
}
- taglist = gst_tag_list_new ();
- gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
- GST_TAG_AUDIO_CODEC, codec, NULL);
+ taglist = gst_tag_list_new (GST_TAG_AUDIO_CODEC, codec, NULL);
if (mp3parse->hdr_bitrate > 0 && mp3parse->xing_bitrate == 0 &&
mp3parse->vbri_bitrate == 0) {
/* We don't have a VBR bitrate, so post the available bitrate as
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_NOMINAL_BITRATE, mp3parse->hdr_bitrate, NULL);
}
- gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
- GST_BASE_PARSE_SRC_PAD (mp3parse), taglist);
+ gst_pad_push_event (GST_BASE_PARSE_SRC_PAD (mp3parse),
+ gst_event_new_tag (taglist));
g_free (codec);
/* also signals the end of first-frame processing */
gboolean using_crc;
if (!taglist) {
- taglist = gst_tag_list_new ();
+ taglist = gst_tag_list_new_empty ();
}
mp3parse->last_posted_crc = mp3parse->last_crc;
if (mp3parse->last_posted_crc == CRC_PROTECTED) {
if (mp3parse->last_posted_channel_mode != mp3parse->last_mode) {
if (!taglist) {
- taglist = gst_tag_list_new ();
+ taglist = gst_tag_list_new_empty ();
}
mp3parse->last_posted_channel_mode = mp3parse->last_mode;
/* if the taglist exists, we need to send it */
if (taglist) {
- gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
- GST_BASE_PARSE_SRC_PAD (mp3parse), taglist);
+ gst_pad_push_event (GST_BASE_PARSE_SRC_PAD (mp3parse),
+ gst_event_new_tag (taglist));
}
/* usual clipping applies */
}
static GstCaps *
-gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse)
+gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
GstCaps *peercaps;
GstCaps *res;
+ /* FIXME: handle filter caps */
+
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
if (peercaps) {
guint i, n;
projects / platform / upstream / gstreamer.git / blobdiff