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authorRonald S. Bultje <rbultje@ronald.bitfreak.net>2005-03-09 12:06:56 +0000
committerRonald S. Bultje <rbultje@ronald.bitfreak.net>2005-03-09 12:06:56 +0000
commit1c37934861f2dd7c3ca38fe770734e2b3b30cfdb (patch)
treee35e4d8254b1708b6a469ff2ee6c71c3de83f1b5
parentdb69a0a5961f3b044d643ff476c97a92407e2b89 (diff)
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configure.ac: Fix FAAD detection problems against FAAD-CVS.
Original commit message from CVS: * configure.ac: Fix FAAD detection problems against FAAD-CVS. * ext/faad/gstfaad.c: (gst_faad_class_init), (gst_faad_chanpos_to_gst), (gst_faad_srcconnect), (gst_faad_sync), (gst_faad_chain): Fix FAAD channel positions for mono/stereo against FAAD CVS. Implement raw stream sync support for AAC+ radio support. Embed info structure in our function to prevent unneeded excessive allocations. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_populate), (gst_ogg_demux_push): Only set first/last positions when we search for them. Fixes invalid length reporting for some video files. * gst/playback/gstdecodebin.c: (remove_element_chain): Always remove only our own kids. * gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak): Fix ESDS atom finding bug. * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Implement frame-finding (similar to MP3) to support AAC+ radio.
-rw-r--r--ChangeLog22
-rw-r--r--configure.ac5
-rw-r--r--ext/faad/gstfaad.c158
-rw-r--r--gst/qtdemux/qtdemux.c9
4 files changed, 153 insertions, 41 deletions
diff --git a/ChangeLog b/ChangeLog
index 872421ed..2f326f14 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,25 @@
+2005-03-09 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+
+ * configure.ac:
+ Fix FAAD detection problems against FAAD-CVS.
+ * ext/faad/gstfaad.c: (gst_faad_class_init),
+ (gst_faad_chanpos_to_gst), (gst_faad_srcconnect), (gst_faad_sync),
+ (gst_faad_chain):
+ Fix FAAD channel positions for mono/stereo against FAAD CVS.
+ Implement raw stream sync support for AAC+ radio support. Embed
+ info structure in our function to prevent unneeded excessive
+ allocations.
+ * ext/ogg/gstoggdemux.c: (gst_ogg_pad_populate),
+ (gst_ogg_demux_push):
+ Only set first/last positions when we search for them. Fixes
+ invalid length reporting for some video files.
+ * gst/playback/gstdecodebin.c: (remove_element_chain):
+ Always remove only our own kids.
+ * gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak):
+ Fix ESDS atom finding bug.
+ * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
+ Implement frame-finding (similar to MP3) to support AAC+ radio.
+
2005-03-09 Jan Schmidt <thaytan@mad.scientist.com>
* gst/videoscale/gstvideoscale.c:
diff --git a/configure.ac b/configure.ac
index 9214de1d..f32dce0b 100644
--- a/configure.ac
+++ b/configure.ac
@@ -1033,7 +1033,10 @@ dnl **** Free AAC Decoder (FAAD) ****
translit(dnm, m, l) AM_CONDITIONAL(USE_FAAD, true)
GST_CHECK_FEATURE(FAAD, [AAC decoder plug-in], faad, [
HAVE_FAAD="yes"
- GST_CHECK_LIBHEADER(FAAD, faad, faacDecOpen, -lm, faad.h, FAAD_LIBS="-lfaad -lm", HAVE_FAAD="no")
+ FAAD_LIBS="-lfaad -lm"
+ GST_CHECK_LIBHEADER(FAAD, faad, faacDecOpen, -lm, faad.h, , [
+ GST_CHECK_LIBHEADER(FAAD, faad, NeAACDecOpen, -lm, faad.h, , HAVE_FAAD="no")
+ ])
if test $HAVE_FAAD = "yes"; then
AC_MSG_CHECKING([Checking for FAAD >= 2])
AC_TRY_RUN([
diff --git a/ext/faad/gstfaad.c b/ext/faad/gstfaad.c
index 14fda859..e08ebcda 100644
--- a/ext/faad/gstfaad.c
+++ b/ext/faad/gstfaad.c
@@ -27,6 +27,9 @@
#include "gstfaad.h"
+GST_DEBUG_CATEGORY_STATIC (faad_debug);
+#define GST_CAT_DEFAULT faad_debug
+
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
@@ -142,6 +145,8 @@ gst_faad_class_init (GstFaadClass * klass)
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gstelement_class->change_state = gst_faad_change_state;
+
+ GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "faad MPEG-AAC decoding");
}
static void
@@ -263,6 +268,16 @@ gst_faad_chanpos_to_gst (guchar * fpos, guint num)
case LFE_CHANNEL:
pos[n] = GST_AUDIO_CHANNEL_POSITION_LFE;
break;
+ case UNKNOWN_CHANNEL:
+ if (num == 1) {
+ pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+ return pos;
+ } else if (num == 2) {
+ pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ return pos;
+ }
+ /* fall-through */
default:
GST_WARNING ("Unsupported FAAD channel position 0x%x encountered",
fpos[n]);
@@ -464,7 +479,7 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
}
/* Another internal checkup. */
- if (faad->need_channel_setup) {
+ if (faad->need_channel_setup && 0) {
GstAudioChannelPosition *pos;
guchar *fpos;
guint i;
@@ -542,6 +557,68 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
return GST_PAD_LINK_REFUSED;
}
+/*
+ * Find syncpoint in ADTS/ADIF stream. Doesn't work for raw,
+ * packetized streams. Be careful when calling.
+ * Returns FALSE on no-sync, fills offset/length if one/two
+ * syncpoints are found, only returns TRUE when it finds two
+ * subsequent syncpoints (similar to mp3 typefinding in
+ * gst/typefind/) for ADTS because 12 bits isn't very reliable.
+ */
+
+static gboolean
+gst_faad_sync (GstBuffer * buf, guint * off)
+{
+ guint8 *data = GST_BUFFER_DATA (buf);
+ guint size = GST_BUFFER_SIZE (buf), n;
+ gint snc;
+
+ GST_DEBUG ("Finding syncpoint");
+
+ /* FIXME: for no-sync, we go over the same data for every new buffer.
+ * We should save the information somewhere. */
+ for (n = 0; n < size - 3; n++) {
+ snc = GST_READ_UINT16_BE (&data[n]);
+ if ((snc & 0xfff6) == 0xfff0) {
+ /* we have an ADTS syncpoint. Parse length and find
+ * next syncpoint. */
+ guint len;
+
+ GST_DEBUG ("Found one ADTS syncpoint at offset 0x%x, tracing next...", n);
+
+ if (size - n < 5) {
+ GST_DEBUG ("Not enough data to parse ADTS header");
+ return FALSE;
+ }
+
+ *off = n;
+ len = ((data[n + 3] & 0x03) << 11) |
+ (data[n + 4] << 3) | ((data[n + 5] & 0xe0) >> 5);
+ if (n + len + 2 >= size) {
+ GST_DEBUG ("Next frame is not within reach");
+ return FALSE;
+ }
+
+ snc = GST_READ_UINT16_BE (&data[n + len]);
+ if ((snc & 0xfff6) == 0xfff0) {
+ GST_DEBUG ("Found ADTS syncpoint at offset 0x%x (framelen %u)", n, len);
+ return TRUE;
+ }
+
+ GST_DEBUG ("No next frame found... (should be at 0x%x)", n + len);
+ } else if (!memcmp (&data[n], "ADIF", 4)) {
+ /* we have an ADIF syncpoint. 4 bytes is enough. */
+ *off = n;
+ GST_DEBUG ("Found ADIF syncpoint at offset 0x%x", n);
+ return TRUE;
+ }
+ }
+
+ GST_DEBUG ("Found no syncpoint");
+
+ return FALSE;
+}
+
static void
gst_faad_chain (GstPad * pad, GstData * data)
{
@@ -550,10 +627,11 @@ gst_faad_chain (GstPad * pad, GstData * data)
guchar *input_data;
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
GstBuffer *buf, *outbuf;
- faacDecFrameInfo *info;
+ faacDecFrameInfo info;
guint64 next_ts;
void *out;
gboolean run_loop = TRUE;
+ guint sync_off;
if (GST_IS_EVENT (data)) {
GstEvent *event = GST_EVENT (data);
@@ -573,8 +651,6 @@ gst_faad_chain (GstPad * pad, GstData * data)
}
}
- info = g_new0 (faacDecFrameInfo, 1);
-
/* buffer + remaining data */
buf = GST_BUFFER (data);
next_ts = GST_BUFFER_TIMESTAMP (buf);
@@ -582,6 +658,16 @@ gst_faad_chain (GstPad * pad, GstData * data)
buf = gst_buffer_join (faad->tempbuf, buf);
faad->tempbuf = NULL;
}
+ input_data = GST_BUFFER_DATA (buf);
+ input_size = GST_BUFFER_SIZE (buf);
+ if (!faad->packetised) {
+ if (!gst_faad_sync (buf, &sync_off))
+ goto next;
+ else {
+ input_data += sync_off;
+ input_size -= sync_off;
+ }
+ }
/* init if not already done during capsnego */
if (!faad->init) {
@@ -589,29 +675,28 @@ gst_faad_chain (GstPad * pad, GstData * data)
guchar channels;
glong init_res;
- init_res = faacDecInit (faad->handle,
- GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), &samplerate, &channels);
+ init_res = faacDecInit (faad->handle, input_data, input_size,
+ &samplerate, &channels);
if (init_res < 0) {
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Failed to init decoder from stream"));
return;
}
- skip_bytes = init_res;
+ skip_bytes = 0; //init_res;
faad->init = TRUE;
/* store for renegotiation later on */
/* FIXME: that's moot, info will get zeroed in DecDecode() */
- info->samplerate = samplerate;
- info->channels = channels;
+ info.samplerate = samplerate;
+ info.channels = channels;
} else {
- info->samplerate = 0;
- info->channels = 0;
+ info.samplerate = 0;
+ info.channels = 0;
}
/* decode cycle */
- input_data = GST_BUFFER_DATA (buf);
- input_size = GST_BUFFER_SIZE (buf);
- info->bytesconsumed = input_size - skip_bytes;
+ info.bytesconsumed = input_size - skip_bytes;
+ info.error = 0;
if (!faad->packetised) {
/* We must check that ourselves for raw stream */
@@ -624,37 +709,36 @@ gst_faad_chain (GstPad * pad, GstData * data)
/* Only one packet per buffer, no matter how much is really consumed */
run_loop = FALSE;
} else {
- if (input_size < FAAD_MIN_STREAMSIZE || info->bytesconsumed <= 0) {
+ if (input_size < FAAD_MIN_STREAMSIZE || info.bytesconsumed <= 0) {
break;
}
}
- out = faacDecDecode (faad->handle, info, input_data + skip_bytes,
+ out = faacDecDecode (faad->handle, &info, input_data + skip_bytes,
input_size - skip_bytes);
- if (info->error) {
- GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
- ("Failed to decode buffer: %s",
- faacDecGetErrorMessage (info->error)));
+ if (info.error) {
+ GST_ERROR_OBJECT (faad, "Failed to decode buffer: %s",
+ faacDecGetErrorMessage (info.error));
break;
}
- if (info->bytesconsumed > input_size)
- info->bytesconsumed = input_size;
- input_size -= info->bytesconsumed;
- input_data += info->bytesconsumed;
+ if (info.bytesconsumed > input_size)
+ info.bytesconsumed = input_size;
+ input_size -= info.bytesconsumed;
+ input_data += info.bytesconsumed;
- if (out && info->samples > 0) {
+ if (out && info.samples > 0) {
gboolean fmt_change = FALSE;
/* see if we need to renegotiate */
- if (info->samplerate != faad->samplerate ||
- info->channels != faad->channels || !faad->channel_positions) {
+ if (info.samplerate != faad->samplerate ||
+ info.channels != faad->channels || !faad->channel_positions) {
fmt_change = TRUE;
} else {
gint i;
- for (i = 0; i < info->channels; i++) {
- if (info->channel_position[i] != faad->channel_positions[i])
+ for (i = 0; i < info.channels; i++) {
+ if (info.channel_position[i] != faad->channel_positions[i])
fmt_change = TRUE;
}
}
@@ -663,13 +747,12 @@ gst_faad_chain (GstPad * pad, GstData * data)
GstPadLinkReturn ret;
/* store new negotiation information */
- faad->samplerate = info->samplerate;
- faad->channels = info->channels;
+ faad->samplerate = info.samplerate;
+ faad->channels = info.channels;
if (faad->channel_positions)
g_free (faad->channel_positions);
faad->channel_positions = g_new (guint8, faad->channels);
- memcpy (faad->channel_positions, info->channel_position,
- faad->channels);
+ memcpy (faad->channel_positions, info.channel_position, faad->channels);
/* and negotiate */
ret = gst_pad_renegotiate (faad->srcpad);
@@ -680,13 +763,13 @@ gst_faad_chain (GstPad * pad, GstData * data)
}
/* play decoded data */
- if (info->samples > 0) {
- outbuf = gst_buffer_new_and_alloc (info->samples * faad->bps);
+ if (info.samples > 0) {
+ outbuf = gst_buffer_new_and_alloc (info.samples * faad->bps);
/* ugh */
memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
GST_BUFFER_TIMESTAMP (outbuf) = next_ts;
GST_BUFFER_DURATION (outbuf) =
- (guint64) GST_SECOND *info->samples / faad->samplerate;
+ (guint64) GST_SECOND *info.samples / faad->samplerate;
if (GST_CLOCK_TIME_IS_VALID (next_ts)) {
next_ts += GST_BUFFER_DURATION (outbuf);
}
@@ -695,6 +778,7 @@ gst_faad_chain (GstPad * pad, GstData * data)
}
}
+next:
/* Keep the leftovers in raw stream */
if (input_size > 0 && !faad->packetised) {
if (input_size < GST_BUFFER_SIZE (buf)) {
@@ -707,8 +791,6 @@ gst_faad_chain (GstPad * pad, GstData * data)
}
gst_buffer_unref (buf);
-
- g_free (info);
}
static GstElementStateReturn
diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c
index a65c9c68..72949962 100644
--- a/gst/qtdemux/qtdemux.c
+++ b/gst/qtdemux/qtdemux.c
@@ -1240,9 +1240,10 @@ qtdemux_parse (GstQTDemux * qtdemux, GNode * node, void *buffer, int length)
guint32 version;
version = QTDEMUX_GUINT32_GET (buffer + 16);
- if (version == 0x00010000) {
- buf = buffer + 0x34;
+ if (version == 0x00010000 || 1) {
+ buf = buffer + 0x24;
end = buffer + length;
+
while (buf < end) {
GNode *child;
@@ -2139,9 +2140,13 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak)
wave = NULL;
if (mp4a)
wave = qtdemux_tree_get_child_by_type (mp4a, FOURCC_wave);
+
esds = NULL;
if (wave)
esds = qtdemux_tree_get_child_by_type (wave, FOURCC_esds);
+ else if (mp4a)
+ esds = qtdemux_tree_get_child_by_type (mp4a, FOURCC_esds);
+
if (esds) {
gst_qtdemux_handle_esds (qtdemux, stream, esds);
#if 0