diff options
| -rw-r--r-- | ChangeLog | 7 | ||||
| m--------- | common | 0 | ||||
| -rw-r--r-- | gst/realmedia/rmdemux.c | 69 |
3 files changed, 68 insertions, 8 deletions
@@ -1,3 +1,10 @@ +2008-10-24 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream), + (gst_rmdemux_descramble_mp4a_audio), + (gst_rmdemux_handle_scrambled_packet): + Add suport for mpeg4 and aac audio. See #556714. + 2008-10-14 Michael Smith <msmith@songbirdnest.com> * gst/mpegaudioparse/gstmpegaudioparse.c: diff --git a/common b/common -Subproject 46eefd2f8474ee748864c59635be87b5a29317d +Subproject 2802bb17517a6cfbbb1be6da61ec19151be0750 diff --git a/gst/realmedia/rmdemux.c b/gst/realmedia/rmdemux.c index 6384b12bad..9dfd9e7553 100644 --- a/gst/realmedia/rmdemux.c +++ b/gst/realmedia/rmdemux.c @@ -1375,18 +1375,21 @@ gst_rmdemux_add_stream (GstRMDemux * rmdemux, GstRMDemuxStream * stream) stream->subpackets = NULL; break; - /* RealAudio 10 (AAC) */ - case GST_RM_AUD_RAAC: - codec_name = "Real Audio 10 (AAC)"; - version = 10; - break; - /* MPEG-4 based */ + case GST_RM_AUD_RAAC: case GST_RM_AUD_RACP: - /* FIXME: codec_name = */ + codec_name = "MPEG4 audio"; stream_caps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, - (int) 4, NULL); + (int) 4, "framed", G_TYPE_BOOLEAN, TRUE, NULL); + if (stream->extra_data_size > 0) { + /* strip off an unknown byte in the extra data */ + stream->extra_data_size--; + stream->extra_data++; + } + stream->needs_descrambling = TRUE; + stream->subpackets_needed = 1; + stream->subpackets = NULL; break; /* Sony ATRAC3 */ @@ -1979,6 +1982,52 @@ gst_rmdemux_descramble_dnet_audio (GstRMDemux * rmdemux, } static GstFlowReturn +gst_rmdemux_descramble_mp4a_audio (GstRMDemux * rmdemux, + GstRMDemuxStream * stream) +{ + GstFlowReturn res; + GstBuffer *buf, *outbuf; + guint frames, index, i; + guint8 *data; + guint size; + GstClockTime timestamp; + + res = GST_FLOW_OK; + + buf = g_ptr_array_index (stream->subpackets, 0); + g_ptr_array_index (stream->subpackets, 0) = NULL; + g_ptr_array_set_size (stream->subpackets, 0); + + data = GST_BUFFER_DATA (buf); + size = GST_BUFFER_SIZE (buf); + timestamp = GST_BUFFER_TIMESTAMP (buf); + + frames = (data[1] & 0xf0) >> 4; + index = 2 * frames + 2; + + for (i = 0; i < frames; i++) { + guint len = (data[i * 2 + 2] << 8) | data[i * 2 + 3]; + + outbuf = gst_buffer_create_sub (buf, index, len); + if (i == 0) + GST_BUFFER_TIMESTAMP (outbuf) = timestamp; + gst_buffer_set_caps (outbuf, GST_PAD_CAPS (stream->pad)); + + index += len; + + if (stream->discont) { + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); + stream->discont = FALSE; + } + res = gst_pad_push (stream->pad, outbuf); + if (res != GST_FLOW_OK) + break; + } + gst_buffer_unref (buf); + return res; +} + +static GstFlowReturn gst_rmdemux_handle_scrambled_packet (GstRMDemux * rmdemux, GstRMDemuxStream * stream, GstBuffer * buf, gboolean keyframe) { @@ -2008,6 +2057,10 @@ gst_rmdemux_handle_scrambled_packet (GstRMDemux * rmdemux, case GST_RM_AUD_COOK: ret = gst_rmdemux_descramble_cook_audio (rmdemux, stream); break; + case GST_RM_AUD_RAAC: + case GST_RM_AUD_RACP: + ret = gst_rmdemux_descramble_mp4a_audio (rmdemux, stream); + break; default: g_assert_not_reached (); } |