mpegaudioparse: support parsing freeform bitrate stream

This commit is contained in:
Mark Nauwelaerts 2012-02-21 18:42:31 +01:00
parent 1fe69911a4
commit 0d5b5d839a
2 changed files with 156 additions and 14 deletions

View file

@ -200,6 +200,7 @@ gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse)
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;
@ -307,14 +308,16 @@ mp3_type_frame_length_from_header (GstMpegAudioParse * mp3parse, guint32 header,
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;
@ -419,8 +422,7 @@ gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
(guint) next_header & HDRMASK, bpf);
*valid = FALSE;
return TRUE;
} 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)");
@ -431,6 +433,13 @@ gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
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++;
}
@ -461,11 +470,6 @@ gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse,
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;
@ -486,6 +490,115 @@ gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse,
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, GstBuffer * buf,
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 = GST_BUFFER_SIZE (buf);
data = GST_BUFFER_DATA (buf);
*_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 %d", 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 gboolean
gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
@ -527,9 +640,14 @@ gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
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)
@ -537,8 +655,30 @@ gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
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, buf, header, draining, &valid)) {
/* not enough data */
*framesize = G_MAXUINT;
*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,

View file

@ -60,6 +60,8 @@ struct _GstMpegAudioParse {
/* samples per frame */
gint spf;
gint freerate;
gboolean sent_codec_tag;
guint last_posted_bitrate;
gint last_posted_crc, last_crc;