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gst/realmedia/rmdemux.c: Descramble cook audio streams before sending them to the decoder. Fixes #347292.
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
parent
6e31592841
commit
1b4d54debc
2 changed files with 176 additions and 64 deletions
13
ChangeLog
13
ChangeLog
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@ -1,3 +1,16 @@
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2006-07-27 Tim-Philipp Müller <tim at centricular dot net>
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* gst/realmedia/rmdemux.c: (gst_rmdemux_class_init),
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(gst_rmdemux_init), (gst_rmdemux_chain), (gst_rmdemux_add_stream),
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(gst_rmdemux_parse_mdpr), (gst_rmdemux_parse_data),
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(gst_rmdemux_stream_clear_cached_subpackets),
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(gst_rmdemux_descramble_cook_audio),
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(gst_rmdemux_descramble_dnet_audio),
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(gst_rmdemux_handle_scrambled_packet), (gst_rmdemux_parse_packet):
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Descramble cook audio streams before sending them to the
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decoder. Fixes #347292.
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Also miscellaneous clean-ups and log-level changes.
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2006-07-26 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* ext/lame/gstlame.c: (gst_lame_setup):
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@ -5,6 +5,7 @@
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* Copyright (C) <2005> Owen Fraser-Green <owen@discobabe.net>
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* Copyright (C) <2005> Michael Smith <fluendo.com>
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* Copyright (C) <2006> Wim Taymans <wim@fluendo.com>
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* Copyright (C) <2006> Tim-Philipp Müller <tim centricular net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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@ -69,6 +70,10 @@ struct _GstRMDemuxStream
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guint16 version;
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guint32 extra_data_size; /* codec_data_length */
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guint8 *extra_data; /* extras */
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gboolean needs_descrambling;
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guint subpackets_needed; /* subpackets needed for descrambling */
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GPtrArray *subpackets; /* array containing subpacket GstBuffers */
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};
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struct _GstRMDemuxIndex
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@ -84,16 +89,6 @@ static GstElementDetails gst_rmdemux_details = {
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"David Schleef <ds@schleef.org>"
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};
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enum
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{
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LAST_SIGNAL
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};
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enum
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{
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ARG_0
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};
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static GstStaticPadTemplate gst_rmdemux_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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@ -207,7 +202,7 @@ gst_rmdemux_class_init (GstRMDemuxClass * klass)
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parent_class = g_type_class_peek_parent (klass);
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gstelement_class->change_state = gst_rmdemux_change_state;
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rmdemux_change_state);
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GST_DEBUG_CATEGORY_INIT (rmdemux_debug, "rmdemux",
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0, "Demuxer for Realmedia streams");
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@ -232,8 +227,7 @@ static void
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gst_rmdemux_init (GstRMDemux * rmdemux)
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{
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rmdemux->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_rmdemux_sink_template), "sink");
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gst_pad_new_from_static_template (&gst_rmdemux_sink_template, "sink");
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gst_pad_set_event_function (rmdemux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_rmdemux_sink_event));
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gst_pad_set_chain_function (rmdemux->sinkpad,
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@ -1117,7 +1111,7 @@ gst_rmdemux_chain (GstPad * pad, GstBuffer * buffer)
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data = gst_adapter_peek (rmdemux->adapter, 2);
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version = RMDEMUX_GUINT16_GET (data);
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GST_DEBUG_OBJECT (rmdemux, "Data packet with version=%d", version);
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GST_LOG_OBJECT (rmdemux, "Data packet with version=%d", version);
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if (version == 0 || version == 1) {
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guint16 length;
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@ -1288,6 +1282,9 @@ gst_rmdemux_add_stream (GstRMDemux * rmdemux, GstRMDemuxStream * stream)
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stream->caps =
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gst_caps_new_simple ("audio/x-ac3", "rate", G_TYPE_INT,
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(int) stream->rate, NULL);
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stream->needs_descrambling = TRUE;
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stream->subpackets_needed = 1;
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stream->subpackets = NULL;
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break;
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/* RealAudio 10 (AAC) */
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@ -1314,6 +1311,9 @@ gst_rmdemux_add_stream (GstRMDemux * rmdemux, GstRMDemuxStream * stream)
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case GST_RM_AUD_COOK:
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codec_name = "Real Audio G2 (Cook)";
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version = 8;
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stream->needs_descrambling = TRUE;
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stream->subpackets_needed = stream->height;
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stream->subpackets = NULL;
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break;
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/* RALF is lossless */
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@ -1596,9 +1596,11 @@ gst_rmdemux_parse_mdpr (GstRMDemux * rmdemux, const void *data, int length)
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stream->version = RMDEMUX_GUINT16_GET (data + offset + 4);
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stream->flavor = RMDEMUX_GUINT16_GET (data + offset + 22);
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stream->packet_size = RMDEMUX_GUINT32_GET (data + offset + 24);
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/* stream->frame_size = RMDEMUX_GUINT32_GET (data + offset + 42); */
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stream->leaf_size = RMDEMUX_GUINT16_GET (data + offset + 44);
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stream->height = RMDEMUX_GUINT16_GET (data + offset + 40);
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GST_INFO ("stream version = %u", stream->version);
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switch (stream->version) {
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case 4:
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stream->rate = RMDEMUX_GUINT16_GET (data + offset + 48);
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@ -1721,9 +1723,8 @@ gst_rmdemux_parse_data (GstRMDemux * rmdemux, const void *data, int length)
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rmdemux->n_chunks = RMDEMUX_GUINT32_GET (data);
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rmdemux->data_offset = RMDEMUX_GUINT32_GET (data + 4);
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rmdemux->chunk_index = 0;
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GST_DEBUG_OBJECT (rmdemux,
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"Data chunk found with %d packets (next data at %p)", rmdemux->n_chunks,
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rmdemux->data_offset);
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GST_DEBUG_OBJECT (rmdemux, "Data chunk found with %d packets "
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"(next data at 0x%08x)", rmdemux->n_chunks, rmdemux->data_offset);
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}
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static void
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@ -1779,35 +1780,6 @@ gst_rmdemux_parse_cont (GstRMDemux * rmdemux, const void *data, int length)
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}
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}
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static void
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gst_rmdemux_fill_audio_packet (GstRMDemux * rmdemux, GstBuffer * buf,
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GstRMDemuxStream * stream, const void *in_data, guint size)
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{
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switch (stream->fourcc) {
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case GST_RM_AUD_DNET:{
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guint8 *data, *end;
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data = (guint8 *) GST_BUFFER_DATA (buf);
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end = (guint8 *) GST_BUFFER_DATA (buf) + GST_BUFFER_SIZE (buf);
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while (data < (end - 1)) {
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*((guint16 *) data) = GUINT16_SWAP_LE_BE (*((guint16 *) in_data));
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data += 2;
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in_data += 2;
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}
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break;
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}
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case GST_RM_AUD_28_8:
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case GST_RM_AUD_COOK:
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/* FIXME: might need to descramble packet */
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/* fallthrough for now */
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default:{
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/* nothing to do, just do a copy */
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memcpy (GST_BUFFER_DATA (buf), in_data, size);
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break;
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}
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}
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}
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static GstFlowReturn
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gst_rmdemux_combine_flows (GstRMDemux * rmdemux, GstRMDemuxStream * stream,
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GstFlowReturn ret)
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@ -1841,6 +1813,126 @@ done:
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return ret;
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}
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static void
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gst_rmdemux_stream_clear_cached_subpackets (GstRMDemux * rmdemux,
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GstRMDemuxStream * stream)
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{
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if (stream->subpackets == NULL || stream->subpackets->len == 0)
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return;
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GST_DEBUG_OBJECT (rmdemux, "discarding %u previously collected subpackets",
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stream->subpackets->len);
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g_ptr_array_foreach (stream->subpackets, (GFunc) gst_mini_object_unref, NULL);
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g_ptr_array_set_size (stream->subpackets, 0);
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}
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static GstFlowReturn
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gst_rmdemux_descramble_cook_audio (GstRMDemux * rmdemux,
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GstRMDemuxStream * stream)
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{
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GstFlowReturn ret;
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GstBuffer *outbuf;
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guint packet_size = stream->packet_size;
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guint height = stream->subpackets->len;
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guint leaf_size = stream->leaf_size;
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guint p, x;
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g_assert (stream->height == height);
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GST_LOG ("packet_size = %u, leaf_size = %u, height= %u", packet_size,
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leaf_size, height);
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ret = gst_pad_alloc_buffer_and_set_caps (stream->pad,
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GST_BUFFER_OFFSET_NONE, height * packet_size, stream->caps, &outbuf);
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if (ret != GST_FLOW_OK)
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goto done;
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for (p = 0; p < height; ++p) {
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GstBuffer *b = g_ptr_array_index (stream->subpackets, p);
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guint8 *b_data = GST_BUFFER_DATA (b);
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if (p == 0)
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GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (b);
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for (x = 0; x < packet_size / leaf_size; ++x) {
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guint idx;
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idx = height * x + ((height + 1) / 2) * (p % 2) + (p / 2);
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/* GST_LOG ("%3u => %3u", (height * p) + x, idx); */
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memcpy (GST_BUFFER_DATA (outbuf) + leaf_size * idx, b_data, leaf_size);
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b_data += leaf_size;
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}
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}
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ret = gst_pad_push (stream->pad, outbuf);
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done:
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gst_rmdemux_stream_clear_cached_subpackets (rmdemux, stream);
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return ret;
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}
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static GstFlowReturn
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gst_rmdemux_descramble_dnet_audio (GstRMDemux * rmdemux,
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GstRMDemuxStream * stream)
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{
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GstBuffer *buf;
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guint16 *data, *end;
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buf = g_ptr_array_index (stream->subpackets, 0);
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g_ptr_array_index (stream->subpackets, 0) = NULL;
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g_ptr_array_set_size (stream->subpackets, 0);
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/* descramble is a bit of a misnomer, it's just byte-order swapped AC3 .. */
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data = (guint16 *) GST_BUFFER_DATA (buf);
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end = (guint16 *) (GST_BUFFER_DATA (buf) + GST_BUFFER_SIZE (buf));
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while (data < end) {
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*data = GUINT16_SWAP_LE_BE (*data);
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++data;
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}
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return gst_pad_push (stream->pad, buf);
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}
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static GstFlowReturn
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gst_rmdemux_handle_scrambled_packet (GstRMDemux * rmdemux,
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GstRMDemuxStream * stream, GstBuffer * buf, gboolean keyframe)
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{
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GstFlowReturn ret;
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if (stream->subpackets == NULL)
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stream->subpackets = g_ptr_array_new ();
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GST_LOG ("Got subpacket %u/%u, len=%u, key=%d", stream->subpackets->len + 1,
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stream->subpackets_needed, GST_BUFFER_SIZE (buf), keyframe);
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if (keyframe && stream->subpackets->len > 0) {
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gst_rmdemux_stream_clear_cached_subpackets (rmdemux, stream);
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}
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g_ptr_array_add (stream->subpackets, buf);
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if (stream->subpackets->len < stream->subpackets_needed)
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return GST_FLOW_OK;
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g_assert (stream->subpackets->len >= 1);
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switch (stream->fourcc) {
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case GST_RM_AUD_DNET:
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ret = gst_rmdemux_descramble_dnet_audio (rmdemux, stream);
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break;
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case GST_RM_AUD_COOK:
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ret = gst_rmdemux_descramble_cook_audio (rmdemux, stream);
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break;
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default:
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g_assert_not_reached ();
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}
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return ret;
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}
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static GstFlowReturn
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gst_rmdemux_parse_packet (GstRMDemux * rmdemux, const void *data,
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guint16 version, guint16 length)
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@ -1851,6 +1943,7 @@ gst_rmdemux_parse_packet (GstRMDemux * rmdemux, const void *data,
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guint16 packet_size;
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GstFlowReturn cret, ret;
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GstClockTime timestamp;
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gboolean key = FALSE;
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id = RMDEMUX_GUINT16_GET (data);
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/* timestamp in Msec */
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@ -1858,18 +1951,25 @@ gst_rmdemux_parse_packet (GstRMDemux * rmdemux, const void *data,
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gst_segment_set_last_stop (&rmdemux->segment, GST_FORMAT_TIME, timestamp);
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GST_DEBUG_OBJECT (rmdemux,
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"Parsing a packet for stream=%d, timestamp=%" GST_TIME_FORMAT
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", version=%d", id, GST_TIME_ARGS (timestamp), version);
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GST_LOG_OBJECT (rmdemux, "Parsing a packet for stream=%d, timestamp=%"
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GST_TIME_FORMAT ", version=%d", id, GST_TIME_ARGS (timestamp), version);
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/* TODO: We read 6 bytes previously; this is skipping over either 2 or 3
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* bytes (version dependent) without even reading it. What are these for? */
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/* skip stream_id and timestamp */
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data += 2 + 4;
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packet_size = length - (2 + 4);
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/* skip other stuff */
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if (version == 0) {
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data += 8;
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packet_size = length - 8;
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/* uint8 packet_group */
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/* uint8 flags */
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key = ((GST_READ_UINT8 (data + 1) & 0x02) != 0);
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data += 2;
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packet_size -= 2;
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} else {
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data += 9;
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packet_size = length - 9;
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/* uint16 asm_rule */
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/* uint8 asm_flags */
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data += 3;
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packet_size -= 3;
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}
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stream = gst_rmdemux_get_stream_by_id (rmdemux, id);
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@ -1891,18 +1991,17 @@ gst_rmdemux_parse_packet (GstRMDemux * rmdemux, const void *data,
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if (ret != GST_FLOW_OK)
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goto beach;
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if (stream->subtype == GST_RMDEMUX_STREAM_AUDIO) {
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gst_rmdemux_fill_audio_packet (rmdemux, buffer, stream, data, packet_size);
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} else {
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memcpy (GST_BUFFER_DATA (buffer), (guint8 *) data, packet_size);
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}
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memcpy (GST_BUFFER_DATA (buffer), (guint8 *) data, packet_size);
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GST_BUFFER_TIMESTAMP (buffer) = timestamp;
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GST_LOG_OBJECT (rmdemux, "Pushing buffer of size %d to pad %s",
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GST_BUFFER_SIZE (buffer), GST_PAD_NAME (stream->pad));
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if (stream->needs_descrambling) {
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ret = gst_rmdemux_handle_scrambled_packet (rmdemux, stream, buffer, key);
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} else {
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GST_LOG_OBJECT (rmdemux, "Pushing buffer of size %d to pad %s",
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GST_BUFFER_SIZE (buffer), GST_PAD_NAME (stream->pad));
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ret = gst_pad_push (stream->pad, buffer);
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ret = gst_pad_push (stream->pad, buffer);
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}
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cret = gst_rmdemux_combine_flows (rmdemux, stream, ret);
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