| /* GStreamer |
| * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public |
| * License along with this library; if not, write to the |
| * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| /** |
| * SECTION:element-rtpac3depay |
| * @see_also: rtpac3pay |
| * |
| * Extract AC3 audio from RTP packets according to RFC 4184. |
| * For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt |
| * |
| * <refsect2> |
| * <title>Example pipeline</title> |
| * |[ |
| * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)AC3, payload=(int)96' ! rtpac3depay ! a52dec ! pulsesink |
| * ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to |
| * the rtpac3pay example to create the RTP stream. |
| * </refsect2> |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| # include "config.h" |
| #endif |
| |
| #include <gst/rtp/gstrtpbuffer.h> |
| #include <gst/audio/audio.h> |
| |
| #include <string.h> |
| #include "gstrtpac3depay.h" |
| #include "gstrtputils.h" |
| |
| GST_DEBUG_CATEGORY_STATIC (rtpac3depay_debug); |
| #define GST_CAT_DEFAULT (rtpac3depay_debug) |
| |
| static GstStaticPadTemplate gst_rtp_ac3_depay_src_template = |
| GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/ac3") |
| ); |
| |
| static GstStaticPadTemplate gst_rtp_ac3_depay_sink_template = |
| GST_STATIC_PAD_TEMPLATE ("sink", |
| GST_PAD_SINK, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("application/x-rtp, " |
| "media = (string) \"audio\", " |
| "clock-rate = (int) { 32000, 44100, 48000 }, " |
| "encoding-name = (string) \"AC3\"") |
| ); |
| |
| G_DEFINE_TYPE (GstRtpAC3Depay, gst_rtp_ac3_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); |
| |
| static gboolean gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload, |
| GstCaps * caps); |
| static GstBuffer *gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload, |
| GstRTPBuffer * rtp); |
| |
| static void |
| gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass) |
| { |
| GstElementClass *gstelement_class; |
| GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; |
| |
| gstelement_class = (GstElementClass *) klass; |
| gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass; |
| |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_ac3_depay_src_template); |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_rtp_ac3_depay_sink_template); |
| |
| gst_element_class_set_static_metadata (gstelement_class, |
| "RTP AC3 depayloader", "Codec/Depayloader/Network/RTP", |
| "Extracts AC3 audio from RTP packets (RFC 4184)", |
| "Wim Taymans <wim.taymans@gmail.com>"); |
| |
| gstrtpbasedepayload_class->set_caps = gst_rtp_ac3_depay_setcaps; |
| gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_ac3_depay_process; |
| |
| GST_DEBUG_CATEGORY_INIT (rtpac3depay_debug, "rtpac3depay", 0, |
| "AC3 Audio RTP Depayloader"); |
| } |
| |
| static void |
| gst_rtp_ac3_depay_init (GstRtpAC3Depay * rtpac3depay) |
| { |
| /* needed because of G_DEFINE_TYPE */ |
| } |
| |
| static gboolean |
| gst_rtp_ac3_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) |
| { |
| GstStructure *structure; |
| gint clock_rate; |
| GstCaps *srccaps; |
| gboolean res; |
| |
| structure = gst_caps_get_structure (caps, 0); |
| |
| if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) |
| clock_rate = 90000; /* default */ |
| depayload->clock_rate = clock_rate; |
| |
| srccaps = gst_caps_new_empty_simple ("audio/ac3"); |
| res = gst_pad_set_caps (depayload->srcpad, srccaps); |
| gst_caps_unref (srccaps); |
| |
| return res; |
| } |
| |
| static GstBuffer * |
| gst_rtp_ac3_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp) |
| { |
| GstRtpAC3Depay *rtpac3depay; |
| GstBuffer *outbuf; |
| guint8 *payload; |
| guint16 FT, NF; |
| |
| rtpac3depay = GST_RTP_AC3_DEPAY (depayload); |
| |
| if (gst_rtp_buffer_get_payload_len (rtp) < 2) |
| goto empty_packet; |
| |
| payload = gst_rtp_buffer_get_payload (rtp); |
| |
| /* strip off header |
| * |
| * 0 1 |
| * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 |
| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| * | MBZ | FT| NF | |
| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| */ |
| FT = payload[0] & 0x3; |
| NF = payload[1]; |
| |
| GST_DEBUG_OBJECT (rtpac3depay, "FT: %d, NF: %d", FT, NF); |
| |
| /* We don't bother with fragmented packets yet */ |
| outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 2, -1); |
| |
| if (outbuf) { |
| gst_rtp_drop_non_audio_meta (rtpac3depay, outbuf); |
| GST_DEBUG_OBJECT (rtpac3depay, "pushing buffer of size %" G_GSIZE_FORMAT, |
| gst_buffer_get_size (outbuf)); |
| } |
| |
| return outbuf; |
| |
| /* ERRORS */ |
| empty_packet: |
| { |
| GST_ELEMENT_WARNING (rtpac3depay, STREAM, DECODE, |
| ("Empty Payload."), (NULL)); |
| return NULL; |
| } |
| } |
| |
| gboolean |
| gst_rtp_ac3_depay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpac3depay", |
| GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_DEPAY); |
| } |