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gst/realmedia/rmdemux.c: Descramble cook audio streams before sending them to the decoder. Fixes #347292.

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Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_class_init),
(gst_rmdemux_init), (gst_rmdemux_chain), (gst_rmdemux_add_stream),
(gst_rmdemux_parse_mdpr), (gst_rmdemux_parse_data),
(gst_rmdemux_stream_clear_cached_subpackets),
(gst_rmdemux_descramble_cook_audio),
(gst_rmdemux_descramble_dnet_audio),
(gst_rmdemux_handle_scrambled_packet), (gst_rmdemux_parse_packet):
Descramble cook audio streams before sending them to the
decoder. Fixes #347292.
Also miscellaneous clean-ups and log-level changes.
This commit is contained in:
Tim-Philipp Müller 2006-07-27 20:34:25 +00:00
parent 6e31592841
commit 1b4d54debc
2 changed files with 176 additions and 64 deletions
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13
ChangeLog
View file

@ -1,3 +1,16 @@
2006-07-27 Tim-Philipp Müller <tim at centricular dot net>
* gst/realmedia/rmdemux.c: (gst_rmdemux_class_init),
(gst_rmdemux_init), (gst_rmdemux_chain), (gst_rmdemux_add_stream),
(gst_rmdemux_parse_mdpr), (gst_rmdemux_parse_data),
(gst_rmdemux_stream_clear_cached_subpackets),
(gst_rmdemux_descramble_cook_audio),
(gst_rmdemux_descramble_dnet_audio),
(gst_rmdemux_handle_scrambled_packet), (gst_rmdemux_parse_packet):
Descramble cook audio streams before sending them to the
decoder. Fixes #347292.
Also miscellaneous clean-ups and log-level changes.
2006-07-26 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/lame/gstlame.c: (gst_lame_setup):

227
gst/realmedia/rmdemux.c
View file

@ -5,6 +5,7 @@
* Copyright (C) <2005> Owen Fraser-Green <owen@discobabe.net>
* Copyright (C) <2005> Michael Smith <fluendo.com>
* Copyright (C) <2006> Wim Taymans <wim@fluendo.com>
* Copyright (C) <2006> Tim-Philipp Müller <tim centricular net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@ -69,6 +70,10 @@ struct _GstRMDemuxStream
guint16 version;
guint32 extra_data_size; /* codec_data_length */
guint8 *extra_data; /* extras */
gboolean needs_descrambling;
guint subpackets_needed; /* subpackets needed for descrambling */
GPtrArray *subpackets; /* array containing subpacket GstBuffers */
};
struct _GstRMDemuxIndex
@ -84,16 +89,6 @@ static GstElementDetails gst_rmdemux_details = {
"David Schleef <ds@schleef.org>"
};
enum
{
LAST_SIGNAL
};
enum
{
ARG_0
};
static GstStaticPadTemplate gst_rmdemux_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
@ -207,7 +202,7 @@ gst_rmdemux_class_init (GstRMDemuxClass * klass)
parent_class = g_type_class_peek_parent (klass);
gstelement_class->change_state = gst_rmdemux_change_state;
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rmdemux_change_state);
GST_DEBUG_CATEGORY_INIT (rmdemux_debug, "rmdemux",
0, "Demuxer for Realmedia streams");
@ -232,8 +227,7 @@ static void
gst_rmdemux_init (GstRMDemux * rmdemux)
{
rmdemux->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_rmdemux_sink_template), "sink");
gst_pad_new_from_static_template (&gst_rmdemux_sink_template, "sink");
gst_pad_set_event_function (rmdemux->sinkpad,
GST_DEBUG_FUNCPTR (gst_rmdemux_sink_event));
gst_pad_set_chain_function (rmdemux->sinkpad,
@ -1117,7 +1111,7 @@ gst_rmdemux_chain (GstPad * pad, GstBuffer * buffer)
data = gst_adapter_peek (rmdemux->adapter, 2);
version = RMDEMUX_GUINT16_GET (data);
GST_DEBUG_OBJECT (rmdemux, "Data packet with version=%d", version);
GST_LOG_OBJECT (rmdemux, "Data packet with version=%d", version);
if (version == 0 || version == 1) {
guint16 length;
@ -1288,6 +1282,9 @@ gst_rmdemux_add_stream (GstRMDemux * rmdemux, GstRMDemuxStream * stream)
stream->caps =
gst_caps_new_simple ("audio/x-ac3", "rate", G_TYPE_INT,
(int) stream->rate, NULL);
stream->needs_descrambling = TRUE;
stream->subpackets_needed = 1;
stream->subpackets = NULL;
break;
/* RealAudio 10 (AAC) */
@ -1314,6 +1311,9 @@ gst_rmdemux_add_stream (GstRMDemux * rmdemux, GstRMDemuxStream * stream)
case GST_RM_AUD_COOK:
codec_name = "Real Audio G2 (Cook)";
version = 8;
stream->needs_descrambling = TRUE;
stream->subpackets_needed = stream->height;
stream->subpackets = NULL;
break;
/* RALF is lossless */
@ -1596,9 +1596,11 @@ gst_rmdemux_parse_mdpr (GstRMDemux * rmdemux, const void *data, int length)
stream->version = RMDEMUX_GUINT16_GET (data + offset + 4);
stream->flavor = RMDEMUX_GUINT16_GET (data + offset + 22);
stream->packet_size = RMDEMUX_GUINT32_GET (data + offset + 24);
/* stream->frame_size = RMDEMUX_GUINT32_GET (data + offset + 42); */
stream->leaf_size = RMDEMUX_GUINT16_GET (data + offset + 44);
stream->height = RMDEMUX_GUINT16_GET (data + offset + 40);
GST_INFO ("stream version = %u", stream->version);
switch (stream->version) {
case 4:
stream->rate = RMDEMUX_GUINT16_GET (data + offset + 48);
@ -1721,9 +1723,8 @@ gst_rmdemux_parse_data (GstRMDemux * rmdemux, const void *data, int length)
rmdemux->n_chunks = RMDEMUX_GUINT32_GET (data);
rmdemux->data_offset = RMDEMUX_GUINT32_GET (data + 4);
rmdemux->chunk_index = 0;
GST_DEBUG_OBJECT (rmdemux,
"Data chunk found with %d packets (next data at %p)", rmdemux->n_chunks,
rmdemux->data_offset);
GST_DEBUG_OBJECT (rmdemux, "Data chunk found with %d packets "
"(next data at 0x%08x)", rmdemux->n_chunks, rmdemux->data_offset);
}
static void
@ -1779,35 +1780,6 @@ gst_rmdemux_parse_cont (GstRMDemux * rmdemux, const void *data, int length)
}
}
static void
gst_rmdemux_fill_audio_packet (GstRMDemux * rmdemux, GstBuffer * buf,
GstRMDemuxStream * stream, const void *in_data, guint size)
{
switch (stream->fourcc) {
case GST_RM_AUD_DNET:{
guint8 *data, *end;
data = (guint8 *) GST_BUFFER_DATA (buf);
end = (guint8 *) GST_BUFFER_DATA (buf) + GST_BUFFER_SIZE (buf);
while (data < (end - 1)) {
*((guint16 *) data) = GUINT16_SWAP_LE_BE (*((guint16 *) in_data));
data += 2;
in_data += 2;
}
break;
}
case GST_RM_AUD_28_8:
case GST_RM_AUD_COOK:
/* FIXME: might need to descramble packet */
/* fallthrough for now */
default:{
/* nothing to do, just do a copy */
memcpy (GST_BUFFER_DATA (buf), in_data, size);
break;
}
}
}
static GstFlowReturn
gst_rmdemux_combine_flows (GstRMDemux * rmdemux, GstRMDemuxStream * stream,
GstFlowReturn ret)
@ -1841,6 +1813,126 @@ done:
return ret;
}
static void
gst_rmdemux_stream_clear_cached_subpackets (GstRMDemux * rmdemux,
GstRMDemuxStream * stream)
{
if (stream->subpackets == NULL || stream->subpackets->len == 0)
return;
GST_DEBUG_OBJECT (rmdemux, "discarding %u previously collected subpackets",
stream->subpackets->len);
g_ptr_array_foreach (stream->subpackets, (GFunc) gst_mini_object_unref, NULL);
g_ptr_array_set_size (stream->subpackets, 0);
}
static GstFlowReturn
gst_rmdemux_descramble_cook_audio (GstRMDemux * rmdemux,
GstRMDemuxStream * stream)
{
GstFlowReturn ret;
GstBuffer *outbuf;
guint packet_size = stream->packet_size;
guint height = stream->subpackets->len;
guint leaf_size = stream->leaf_size;
guint p, x;
g_assert (stream->height == height);
GST_LOG ("packet_size = %u, leaf_size = %u, height= %u", packet_size,
leaf_size, height);
ret = gst_pad_alloc_buffer_and_set_caps (stream->pad,
GST_BUFFER_OFFSET_NONE, height * packet_size, stream->caps, &outbuf);
if (ret != GST_FLOW_OK)
goto done;
for (p = 0; p < height; ++p) {
GstBuffer *b = g_ptr_array_index (stream->subpackets, p);
guint8 *b_data = GST_BUFFER_DATA (b);
if (p == 0)
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (b);
for (x = 0; x < packet_size / leaf_size; ++x) {
guint idx;
idx = height * x + ((height + 1) / 2) * (p % 2) + (p / 2);
/* GST_LOG ("%3u => %3u", (height * p) + x, idx); */
memcpy (GST_BUFFER_DATA (outbuf) + leaf_size * idx, b_data, leaf_size);
b_data += leaf_size;
}
}
ret = gst_pad_push (stream->pad, outbuf);
done:
gst_rmdemux_stream_clear_cached_subpackets (rmdemux, stream);
return ret;
}
static GstFlowReturn
gst_rmdemux_descramble_dnet_audio (GstRMDemux * rmdemux,
GstRMDemuxStream * stream)
{
GstBuffer *buf;
guint16 *data, *end;
buf = g_ptr_array_index (stream->subpackets, 0);
g_ptr_array_index (stream->subpackets, 0) = NULL;
g_ptr_array_set_size (stream->subpackets, 0);
/* descramble is a bit of a misnomer, it's just byte-order swapped AC3 .. */
data = (guint16 *) GST_BUFFER_DATA (buf);
end = (guint16 *) (GST_BUFFER_DATA (buf) + GST_BUFFER_SIZE (buf));
while (data < end) {
*data = GUINT16_SWAP_LE_BE (*data);
++data;
}
return gst_pad_push (stream->pad, buf);
}
static GstFlowReturn
gst_rmdemux_handle_scrambled_packet (GstRMDemux * rmdemux,
GstRMDemuxStream * stream, GstBuffer * buf, gboolean keyframe)
{
GstFlowReturn ret;
if (stream->subpackets == NULL)
stream->subpackets = g_ptr_array_new ();
GST_LOG ("Got subpacket %u/%u, len=%u, key=%d", stream->subpackets->len + 1,
stream->subpackets_needed, GST_BUFFER_SIZE (buf), keyframe);
if (keyframe && stream->subpackets->len > 0) {
gst_rmdemux_stream_clear_cached_subpackets (rmdemux, stream);
}
g_ptr_array_add (stream->subpackets, buf);
if (stream->subpackets->len < stream->subpackets_needed)
return GST_FLOW_OK;
g_assert (stream->subpackets->len >= 1);
switch (stream->fourcc) {
case GST_RM_AUD_DNET:
ret = gst_rmdemux_descramble_dnet_audio (rmdemux, stream);
break;
case GST_RM_AUD_COOK:
ret = gst_rmdemux_descramble_cook_audio (rmdemux, stream);
break;
default:
g_assert_not_reached ();
}
return ret;
}
static GstFlowReturn
gst_rmdemux_parse_packet (GstRMDemux * rmdemux, const void *data,
guint16 version, guint16 length)
@ -1851,6 +1943,7 @@ gst_rmdemux_parse_packet (GstRMDemux * rmdemux, const void *data,
guint16 packet_size;
GstFlowReturn cret, ret;
GstClockTime timestamp;
gboolean key = FALSE;
id = RMDEMUX_GUINT16_GET (data);
/* timestamp in Msec */
@ -1858,18 +1951,25 @@ gst_rmdemux_parse_packet (GstRMDemux * rmdemux, const void *data,
gst_segment_set_last_stop (&rmdemux->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (rmdemux,
"Parsing a packet for stream=%d, timestamp=%" GST_TIME_FORMAT
", version=%d", id, GST_TIME_ARGS (timestamp), version);
GST_LOG_OBJECT (rmdemux, "Parsing a packet for stream=%d, timestamp=%"
GST_TIME_FORMAT ", version=%d", id, GST_TIME_ARGS (timestamp), version);
/* TODO: We read 6 bytes previously; this is skipping over either 2 or 3
* bytes (version dependent) without even reading it. What are these for? */
/* skip stream_id and timestamp */
data += 2 + 4;
packet_size = length - (2 + 4);
/* skip other stuff */
if (version == 0) {
data += 8;
packet_size = length - 8;
/* uint8 packet_group */
/* uint8 flags */
key = ((GST_READ_UINT8 (data + 1) & 0x02) != 0);
data += 2;
packet_size -= 2;
} else {
data += 9;
packet_size = length - 9;
/* uint16 asm_rule */
/* uint8 asm_flags */
data += 3;
packet_size -= 3;
}
stream = gst_rmdemux_get_stream_by_id (rmdemux, id);
@ -1891,18 +1991,17 @@ gst_rmdemux_parse_packet (GstRMDemux * rmdemux, const void *data,
if (ret != GST_FLOW_OK)
goto beach;
if (stream->subtype == GST_RMDEMUX_STREAM_AUDIO) {
gst_rmdemux_fill_audio_packet (rmdemux, buffer, stream, data, packet_size);
} else {
memcpy (GST_BUFFER_DATA (buffer), (guint8 *) data, packet_size);
}
memcpy (GST_BUFFER_DATA (buffer), (guint8 *) data, packet_size);
GST_BUFFER_TIMESTAMP (buffer) = timestamp;
GST_LOG_OBJECT (rmdemux, "Pushing buffer of size %d to pad %s",
GST_BUFFER_SIZE (buffer), GST_PAD_NAME (stream->pad));
if (stream->needs_descrambling) {
ret = gst_rmdemux_handle_scrambled_packet (rmdemux, stream, buffer, key);
} else {
GST_LOG_OBJECT (rmdemux, "Pushing buffer of size %d to pad %s",
GST_BUFFER_SIZE (buffer), GST_PAD_NAME (stream->pad));
ret = gst_pad_push (stream->pad, buffer);
ret = gst_pad_push (stream->pad, buffer);
}
cret = gst_rmdemux_combine_flows (rmdemux, stream, ret);