* <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>
*/
static gboolean gst_mpeg_audio_parse_start (GstBaseParse * parse);
static gboolean gst_mpeg_audio_parse_stop (GstBaseParse * parse);
-static gboolean gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
- GstBaseParseFrame * frame, guint * size, gint * skipsize);
-static GstFlowReturn gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse,
- GstBaseParseFrame * frame);
+static GstFlowReturn gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
+ GstBaseParseFrame * frame, gint * skipsize);
static GstFlowReturn gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
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,
+ GstCaps * filter);
+
+static void gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse *
+ mp3parse, GstBuffer * buf);
#define gst_mpeg_audio_parse_parent_class parent_class
G_DEFINE_TYPE (GstMpegAudioParse, gst_mpeg_audio_parse, GST_TYPE_BASE_PARSE);
parse_class->start = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_start);
parse_class->stop = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_stop);
- parse_class->check_valid_frame =
- GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_check_valid_frame);
- parse_class->parse_frame =
- GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_parse_frame);
+ parse_class->handle_frame =
+ GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_handle_frame);
parse_class->pre_push_frame =
GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_frame);
parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert);
+ parse_class->get_sink_caps =
+ GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_get_sink_caps);
/* register tags */
#define GST_TAG_CRC "has-crc"
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;
guint32 header, int bpf, gboolean at_eos, gint * valid)
{
guint32 next_header;
- guint8 *data;
- gsize available;
+ GstMapInfo map;
gboolean res = TRUE;
int frames_found = 1;
int offset = bpf;
- data = gst_buffer_map (buf, &available, NULL, GST_MAP_READ);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
while (frames_found < MIN_RESYNC_FRAMES) {
/* Check if we have enough data for all these frames, plus the next
frame header. */
- if (available < offset + 4) {
+ if (map.size < offset + 4) {
if (at_eos) {
/* Running out of data at EOS is fine; just accept it */
*valid = TRUE;
}
}
- next_header = GST_READ_UINT32_BE (data + offset);
+ next_header = GST_READ_UINT32_BE (map.data + offset);
GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X, bpf=%d",
offset, (unsigned int) header, (unsigned int) next_header, bpf);
(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++;
}
*valid = TRUE;
cleanup:
- gst_buffer_unmap (buf, data, available);
+ gst_buffer_unmap (buf, &map);
return res;
}
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_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
- GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
+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)
{
GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
GstBuffer *buf = frame->buffer;
gboolean lost_sync, draining, valid, caps_change;
guint32 header;
guint bitrate, layer, rate, channels, version, mode, crc;
- guint8 *data;
- gsize bufsize;
+ GstMapInfo map;
gboolean res = FALSE;
- data = gst_buffer_map (buf, &bufsize, NULL, GST_MAP_READ);
- if (G_UNLIKELY (bufsize < 6))
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ if (G_UNLIKELY (map.size < 6)) {
+ *skipsize = 1;
goto cleanup;
+ }
- gst_byte_reader_init (&reader, data, bufsize);
+ gst_byte_reader_init (&reader, map.data, map.size);
off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffe00000, 0xffe00000,
- 0, bufsize);
+ 0, map.size);
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 = bufsize - 3;
+ *skipsize = map.size - 3;
goto cleanup;
}
}
/* make sure the values in the frame header look sane */
- header = GST_READ_UINT32_BE (data);
+ header = GST_READ_UINT32_BE (map.data);
if (!gst_mpeg_audio_parse_head_check (mp3parse, header)) {
*skipsize = 1;
goto cleanup;
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,
/* restore default minimum */
gst_base_parse_set_min_frame_size (parse, MIN_FRAME_SIZE);
- *framesize = bpf;
res = TRUE;
+ /* metadata handling */
+ if (G_UNLIKELY (caps_change)) {
+ GstCaps *caps = gst_caps_new_simple ("audio/mpeg",
+ "mpegversion", G_TYPE_INT, 1,
+ "mpegaudioversion", G_TYPE_INT, version,
+ "layer", G_TYPE_INT, layer,
+ "rate", G_TYPE_INT, rate,
+ "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
+ gst_caps_unref (caps);
+
+ mp3parse->rate = rate;
+ mp3parse->channels = channels;
+ mp3parse->layer = layer;
+ mp3parse->version = version;
+
+ /* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
+ if (mp3parse->layer == 1)
+ mp3parse->spf = 384;
+ else if (mp3parse->layer == 2)
+ mp3parse->spf = 1152;
+ else if (mp3parse->version == 1) {
+ mp3parse->spf = 1152;
+ } else {
+ /* MPEG-2 or "2.5" */
+ mp3parse->spf = 576;
+ }
+
+ /* lead_in:
+ * We start pushing 9 frames earlier (29 frames for MPEG2) than
+ * segment start to be able to decode the first frame we want.
+ * 9 (29) frames are the theoretical maximum of frames that contain
+ * data for the current frame (bit reservoir).
+ *
+ * lead_out:
+ * Some mp3 streams have an offset in the timestamps, for which we have to
+ * push the frame *after* the end position in order for the decoder to be
+ * able to decode everything up until the segment.stop position. */
+ gst_base_parse_set_frame_rate (parse, mp3parse->rate, mp3parse->spf,
+ (version == 1) ? 10 : 30, 2);
+ }
+
+ mp3parse->hdr_bitrate = bitrate;
+
+ /* For first frame; check for seek tables and output a codec tag */
+ gst_mpeg_audio_parse_handle_first_frame (mp3parse, buf);
+
+ /* store some frame info for later processing */
+ mp3parse->last_crc = crc;
+ mp3parse->last_mode = mode;
+
cleanup:
- gst_buffer_unmap (buf, data, bufsize);
- return res;
+ gst_buffer_unmap (buf, &map);
+
+ if (res && bpf <= map.size) {
+ return gst_base_parse_finish_frame (parse, frame, bpf);
+ }
+
+ return GST_FLOW_OK;
}
static void
guint64 avail;
gint64 upstream_total_bytes = 0;
guint32 read_id_xing = 0, read_id_vbri = 0;
- guint8 *data, *origdata;
- gsize bufsize;
+ GstMapInfo map;
+ guint8 *data;
guint bitrate;
if (mp3parse->sent_codec_tag)
offset_vbri += 4;
/* Check if we have enough data to read the Xing header */
- origdata = data = gst_buffer_map (buf, &bufsize, NULL, GST_MAP_READ);
- avail = bufsize;
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ data = map.data;
+ avail = map.size;
if (avail >= offset_xing + 4) {
read_id_xing = GST_READ_UINT32_BE (data + offset_xing);
}
/* obtain real upstream total bytes */
- if (!gst_pad_query_peer_duration (GST_BASE_PARSE_SINK_PAD (mp3parse),
+ if (!gst_pad_peer_query_duration (GST_BASE_PARSE_SINK_PAD (mp3parse),
GST_FORMAT_BYTES, &upstream_total_bytes))
upstream_total_bytes = 0;
GST_DEBUG_OBJECT (mp3parse, "Encoder delay %u, encoder padding %u",
encoder_delay, encoder_padding);
}
- }
-
- if (read_id_vbri == vbri_id) {
+ } else if (read_id_vbri == vbri_id) {
gint64 total_bytes, total_frames;
GstClockTime total_time;
guint16 nseek_points;
goto cleanup;
}
- data = origdata;
+ data = map.data;
data += offset_vbri + 26;
/* VBRI seek table: frame/seek_frames -> byte */
gst_base_parse_set_average_bitrate (GST_BASE_PARSE (mp3parse), bitrate);
cleanup:
- gst_buffer_unmap (buf, origdata, bufsize);
-}
-
-static GstFlowReturn
-gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse,
- GstBaseParseFrame * frame)
-{
- GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
- GstBuffer *buf = frame->buffer;
- guint8 *data;
- gsize bufsize;
- guint bitrate, layer, rate, channels, version, mode, crc;
-
- 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 (data),
- &version, &layer, &channels, &bitrate, &rate, &mode, &crc))
- goto broken_header;
-
- if (G_UNLIKELY (channels != mp3parse->channels || rate != mp3parse->rate ||
- layer != mp3parse->layer || version != mp3parse->version)) {
- GstCaps *caps = gst_caps_new_simple ("audio/mpeg",
- "mpegversion", G_TYPE_INT, 1,
- "mpegaudioversion", G_TYPE_INT, version,
- "layer", G_TYPE_INT, layer,
- "rate", G_TYPE_INT, rate,
- "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
- gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
- gst_caps_unref (caps);
-
- mp3parse->rate = rate;
- mp3parse->channels = channels;
- mp3parse->layer = layer;
- mp3parse->version = version;
-
- /* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
- if (mp3parse->layer == 1)
- mp3parse->spf = 384;
- else if (mp3parse->layer == 2)
- mp3parse->spf = 1152;
- else if (mp3parse->version == 1) {
- mp3parse->spf = 1152;
- } else {
- /* MPEG-2 or "2.5" */
- mp3parse->spf = 576;
- }
-
- /* lead_in:
- * We start pushing 9 frames earlier (29 frames for MPEG2) than
- * segment start to be able to decode the first frame we want.
- * 9 (29) frames are the theoretical maximum of frames that contain
- * data for the current frame (bit reservoir).
- *
- * lead_out:
- * Some mp3 streams have an offset in the timestamps, for which we have to
- * push the frame *after* the end position in order for the decoder to be
- * able to decode everything up until the segment.stop position. */
- gst_base_parse_set_frame_rate (parse, mp3parse->rate, mp3parse->spf,
- (version == 1) ? 10 : 30, 2);
- }
-
- mp3parse->hdr_bitrate = bitrate;
-
- /* For first frame; check for seek tables and output a codec tag */
- gst_mpeg_audio_parse_handle_first_frame (mp3parse, buf);
-
- /* store some frame info for later processing */
- 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;
- }
+ gst_buffer_unmap (buf, &map);
}
static gboolean
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 */
/* 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 */
return GST_FLOW_OK;
}
+
+static GstCaps *
+gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
+{
+ GstCaps *peercaps, *templ;
+ GstCaps *res;
+
+ 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;
+
+ /* Remove the parsed field */
+ peercaps = gst_caps_make_writable (peercaps);
+ n = gst_caps_get_size (peercaps);
+ for (i = 0; i < n; i++) {
+ GstStructure *s = gst_caps_get_structure (peercaps, i);
+
+ gst_structure_remove_field (s, "parsed");
+ }
+
+ 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 = 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