* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch filesrc location=test.mp3 ! mpegaudioparse ! mad ! autoaudiosink
+ * gst-launch-1.0 filesrc location=test.mp3 ! mpegaudioparse ! mad ! autoaudiosink
* ]|
* </refsect2>
*/
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",
+ gst_element_class_set_static_metadata (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>,"
mp3parse->sent_codec_tag = FALSE;
mp3parse->last_posted_crc = CRC_UNKNOWN;
mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN;
+ mp3parse->freerate = 0;
mp3parse->hdr_bitrate = 0;
bitrate = (header >> 12) & 0xF;
bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
- /* The caller has ensured we have a valid header, so bitrate can't be
- zero here. */
- g_assert (bitrate != 0);
+ if (!bitrate) {
+ GST_LOG_OBJECT (mp3parse, "using freeform bitrate");
+ bitrate = mp3parse->freerate;
+ }
samplerate = (header >> 10) & 0x3;
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
- padding = (header >> 9) & 0x1;
+ /* force 0 length if 0 bitrate */
+ padding = (bitrate > 0) ? (header >> 9) & 0x1 : 0;
mode = (header >> 6) & 0x3;
channels = (mode == 3) ? 1 : 2;
(guint) next_header & HDRMASK, bpf);
*valid = FALSE;
goto cleanup;
- } else if ((((next_header >> 12) & 0xf) == 0) ||
- (((next_header >> 12) & 0xf) == 0xf)) {
+ } else if (((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)");
bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
NULL, NULL, NULL, NULL, NULL, NULL, NULL);
+ /* if no bitrate, and no freeform rate known, then fail */
+ if (G_UNLIKELY (!bpf)) {
+ GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate 0)");
+ *valid = FALSE;
+ return TRUE;
+ }
+
offset += bpf;
frames_found++;
}
return FALSE;
}
/* if it's an invalid bitrate */
- if (((head >> 12) & 0xf) == 0x0) {
- GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx."
- "Free format files are not supported yet", (head >> 12) & 0xf);
- return FALSE;
- }
if (((head >> 12) & 0xf) == 0xf) {
GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
return FALSE;
return TRUE;
}
+/* Determines possible freeform frame rate/size by looking for next
+ * header with valid bitrate (0 or otherwise valid) (and sufficiently
+ * matching current header).
+ *
+ * Returns TRUE if we've found such one, and *rate then contains rate
+ * (or *rate contains 0 if decided no freeframe size could be determined).
+ * If not enough data, returns FALSE.
+ */
+static gboolean
+gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstMapInfo * map,
+ guint32 header, gboolean at_eos, gint * _rate)
+{
+ guint32 next_header;
+ const guint8 *data;
+ guint available;
+ int offset = 4;
+ gulong samplerate, rate, layer, padding;
+ gboolean valid;
+ gint lsf, mpg25;
+
+ available = map->size;
+ data = map->data;
+
+ *_rate = 0;
+
+ /* pick apart header again partially */
+ if (header & (1 << 20)) {
+ lsf = (header & (1 << 19)) ? 0 : 1;
+ mpg25 = 0;
+ } else {
+ lsf = 1;
+ mpg25 = 1;
+ }
+ layer = 4 - ((header >> 17) & 0x3);
+ samplerate = (header >> 10) & 0x3;
+ samplerate = mp3types_freqs[lsf + mpg25][samplerate];
+ padding = (header >> 9) & 0x1;
+
+ for (; offset < available; ++offset) {
+ /* Check if we have enough data for all these frames, plus the next
+ frame header. */
+ if (available < offset + 4) {
+ if (at_eos) {
+ /* Running out of data; failed to determine size */
+ return TRUE;
+ } else {
+ return FALSE;
+ }
+ }
+
+ valid = FALSE;
+ next_header = GST_READ_UINT32_BE (data + offset);
+ if ((next_header & 0xFFE00000) != 0xFFE00000)
+ goto next;
+
+ GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X",
+ offset, (unsigned int) header, (unsigned int) next_header);
+
+ if ((next_header & HDRMASK) != (header & HDRMASK)) {
+ /* If any of the unmasked bits don't match, then it's not valid */
+ GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
+ "(header=%08X (%08X), header2=%08X (%08X))",
+ (guint) header, (guint) header & HDRMASK, (guint) next_header,
+ (guint) next_header & HDRMASK);
+ goto next;
+ } else if (((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)");
+ goto next;
+ }
+
+ valid = TRUE;
+
+ next:
+ /* almost accept as free frame */
+ if (layer == 1) {
+ rate = samplerate * (offset - 4 * padding + 4) / 48000;
+ } else {
+ rate = samplerate * (offset - padding + 1) / (144 >> lsf) / 1000;
+ }
+
+ if (valid) {
+ GST_LOG_OBJECT (mp3parse, "calculated rate %lu", rate * 1000);
+ if (rate < 8 || (layer == 3 && rate > 640)) {
+ GST_DEBUG_OBJECT (mp3parse, "rate invalid");
+ if (rate < 8) {
+ /* maybe some hope */
+ continue;
+ } else {
+ GST_DEBUG_OBJECT (mp3parse, "aborting");
+ /* give up */
+ break;
+ }
+ }
+ *_rate = rate * 1000;
+ break;
+ } else {
+ /* avoid indefinite searching */
+ if (rate > 1000) {
+ GST_DEBUG_OBJECT (mp3parse, "exceeded sanity rate; aborting");
+ break;
+ }
+ }
+ }
+
+ return TRUE;
+}
+
static GstFlowReturn
gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize)
GST_LOG_OBJECT (parse, "got frame");
+ lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
+ draining = GST_BASE_PARSE_DRAINING (parse);
+
+ if (G_UNLIKELY (lost_sync))
+ mp3parse->freerate = 0;
+
bpf = mp3_type_frame_length_from_header (mp3parse, header,
&version, &layer, &channels, &bitrate, &rate, &mode, &crc);
- g_assert (bpf != 0);
if (channels != mp3parse->channels || rate != mp3parse->rate ||
layer != mp3parse->layer || version != mp3parse->version)
else
caps_change = FALSE;
- lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
- draining = GST_BASE_PARSE_DRAINING (parse);
+ /* maybe free format */
+ if (bpf == 0) {
+ GST_LOG_OBJECT (mp3parse, "possibly free format");
+ if (lost_sync || mp3parse->freerate == 0) {
+ GST_DEBUG_OBJECT (mp3parse, "finding free format rate");
+ if (!gst_mp3parse_find_freerate (mp3parse, &map, header, draining,
+ &valid)) {
+ /* not enough data */
+ gst_base_parse_set_min_frame_size (parse, valid);
+ *skipsize = 0;
+ return FALSE;
+ } else {
+ GST_DEBUG_OBJECT (parse, "determined freeform size %d", valid);
+ mp3parse->freerate = valid;
+ }
+ }
+ /* try again */
+ bpf = mp3_type_frame_length_from_header (mp3parse, header,
+ &version, &layer, &channels, &bitrate, &rate, &mode, &crc);
+ if (!bpf) {
+ /* did not come up with valid freeform length, reject after all */
+ *skipsize = 1;
+ return FALSE;
+ }
+ }
if (!draining && (lost_sync || caps_change)) {
if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, draining,
static GstCaps *
gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
{
- GstCaps *peercaps;
+ GstCaps *peercaps, *templ;
GstCaps *res;
- /* FIXME: handle filter caps */
-
+ templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
+
if (peercaps) {
guint i, n;
gst_structure_remove_field (s, "parsed");
}
- res =
- gst_caps_intersect_full (peercaps,
- gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
- GST_CAPS_INTERSECT_FIRST);
+ res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
+
+ /* Append the template caps because we still want to accept
+ * caps without any fields in the case upstream does not
+ * know anything.
+ */
+ gst_caps_append (res, templ);
} else {
- res =
- gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
- (parse)));
+ res = templ;
+ }
+
+ if (filter) {
+ GstCaps *intersection;
+
+ intersection =
+ gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
+ gst_caps_unref (res);
+ res = intersection;
}
return res;
projects / platform / upstream / gstreamer.git / blobdiff