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Diffstat (limited to 'gst/rtp/gstrtpL8depay.c')
-rw-r--r-- | gst/rtp/gstrtpL8depay.c | 267 |
1 files changed, 267 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpL8depay.c b/gst/rtp/gstrtpL8depay.c new file mode 100644 index 000000000..5b9520a8d --- /dev/null +++ b/gst/rtp/gstrtpL8depay.c @@ -0,0 +1,267 @@ +/* GStreamer + * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com> + * Copyright (C) <2015> GE Intelligent Platforms Embedded Systems, Inc. + * + * 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-rtpL8depay + * @see_also: rtpL8pay + * + * Extract raw audio from RTP packets according to RFC 3551. + * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt + * + * <refsect2> + * <title>Example pipeline</title> + * |[ + * gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L8, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL8depay ! pulsesink + * ]| This example pipeline will depayload an RTP raw audio stream. Refer to + * the rtpL8pay example to create the RTP stream. + * </refsect2> + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <string.h> +#include <stdlib.h> + +#include <gst/audio/audio.h> + +#include "gstrtpL8depay.h" +#include "gstrtpchannels.h" + +GST_DEBUG_CATEGORY_STATIC (rtpL8depay_debug); +#define GST_CAT_DEFAULT (rtpL8depay_debug) + +static GstStaticPadTemplate gst_rtp_L8_depay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) U8, " + "layout = (string) interleaved, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") + ); + +static GstStaticPadTemplate gst_rtp_L8_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) [ 1, MAX ], " + /* "channels = (int) [1, MAX]" */ + /* "emphasis = (string) ANY" */ + /* "channel-order = (string) ANY" */ + "encoding-name = (string) L8;") + ); + +#define gst_rtp_L8_depay_parent_class parent_class +G_DEFINE_TYPE (GstRtpL8Depay, gst_rtp_L8_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); + +static gboolean gst_rtp_L8_depay_setcaps (GstRTPBaseDepayload * depayload, + GstCaps * caps); +static GstBuffer *gst_rtp_L8_depay_process (GstRTPBaseDepayload * depayload, + GstBuffer * buf); + +static void +gst_rtp_L8_depay_class_init (GstRtpL8DepayClass * klass) +{ + GstElementClass *gstelement_class; + GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; + + gstelement_class = (GstElementClass *) klass; + gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass; + + gstrtpbasedepayload_class->set_caps = gst_rtp_L8_depay_setcaps; + gstrtpbasedepayload_class->process = gst_rtp_L8_depay_process; + + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_L8_depay_src_template)); + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_L8_depay_sink_template)); + + gst_element_class_set_static_metadata (gstelement_class, + "RTP audio depayloader", "Codec/Depayloader/Network/RTP", + "Extracts raw audio from RTP packets", + "Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>, " + "GE Intelligent Platforms Embedded Systems, Inc."); + + GST_DEBUG_CATEGORY_INIT (rtpL8depay_debug, "rtpL8depay", 0, + "Raw Audio RTP Depayloader"); +} + +static void +gst_rtp_L8_depay_init (GstRtpL8Depay * rtpL8depay) +{ +} + +static gint +gst_rtp_L8_depay_parse_int (GstStructure * structure, const gchar * field, + gint def) +{ + const gchar *str; + gint res; + + if ((str = gst_structure_get_string (structure, field))) + return atoi (str); + + if (gst_structure_get_int (structure, field, &res)) + return res; + + return def; +} + +static gboolean +gst_rtp_L8_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) +{ + GstStructure *structure; + GstRtpL8Depay *rtpL8depay; + gint clock_rate; + gint channels; + GstCaps *srccaps; + gboolean res; + const gchar *channel_order; + const GstRTPChannelOrder *order; + GstAudioInfo *info; + + rtpL8depay = GST_RTP_L8_DEPAY (depayload); + + structure = gst_caps_get_structure (caps, 0); + + /* no fixed mapping, we need clock-rate */ + channels = 0; + clock_rate = 0; + + /* caps can overwrite defaults */ + clock_rate = gst_rtp_L8_depay_parse_int (structure, "clock-rate", clock_rate); + if (clock_rate == 0) + goto no_clockrate; + + channels = + gst_rtp_L8_depay_parse_int (structure, "encoding-params", channels); + if (channels == 0) { + channels = gst_rtp_L8_depay_parse_int (structure, "channels", channels); + if (channels == 0) { + /* channels defaults to 1 otherwise */ + channels = 1; + } + } + + depayload->clock_rate = clock_rate; + + info = &rtpL8depay->info; + gst_audio_info_init (info); + info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_U8); + info->rate = clock_rate; + info->channels = channels; + info->bpf = (info->finfo->width / 8) * channels; + + /* add channel positions */ + channel_order = gst_structure_get_string (structure, "channel-order"); + + order = gst_rtp_channels_get_by_order (channels, channel_order); + rtpL8depay->order = order; + if (order) { + memcpy (info->position, order->pos, + sizeof (GstAudioChannelPosition) * channels); + gst_audio_channel_positions_to_valid_order (info->position, info->channels); + } else { + GST_ELEMENT_WARNING (rtpL8depay, STREAM, DECODE, + (NULL), ("Unknown channel order '%s' for %d channels", + GST_STR_NULL (channel_order), channels)); + /* create default NONE layout */ + gst_rtp_channels_create_default (channels, info->position); + } + + srccaps = gst_audio_info_to_caps (info); + res = gst_pad_set_caps (depayload->srcpad, srccaps); + gst_caps_unref (srccaps); + + return res; + + /* ERRORS */ +no_clockrate: + { + GST_ERROR_OBJECT (depayload, "no clock-rate specified"); + return FALSE; + } +} + +static GstBuffer * +gst_rtp_L8_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf) +{ + GstRtpL8Depay *rtpL8depay; + GstBuffer *outbuf; + gint payload_len; + gboolean marker; + GstRTPBuffer rtp = { NULL }; + + rtpL8depay = GST_RTP_L8_DEPAY (depayload); + + gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp); + payload_len = gst_rtp_buffer_get_payload_len (&rtp); + + if (payload_len <= 0) + goto empty_packet; + + GST_DEBUG_OBJECT (rtpL8depay, "got payload of %d bytes", payload_len); + + outbuf = gst_rtp_buffer_get_payload_buffer (&rtp); + marker = gst_rtp_buffer_get_marker (&rtp); + + if (marker) { + /* mark talk spurt with RESYNC */ + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC); + } + + outbuf = gst_buffer_make_writable (outbuf); + if (rtpL8depay->order && + !gst_audio_buffer_reorder_channels (outbuf, + rtpL8depay->info.finfo->format, rtpL8depay->info.channels, + rtpL8depay->info.position, rtpL8depay->order->pos)) { + goto reorder_failed; + } + + gst_rtp_buffer_unmap (&rtp); + + return outbuf; + + /* ERRORS */ +empty_packet: + { + GST_ELEMENT_WARNING (rtpL8depay, STREAM, DECODE, + ("Empty Payload."), (NULL)); + gst_rtp_buffer_unmap (&rtp); + return NULL; + } +reorder_failed: + { + GST_ELEMENT_ERROR (rtpL8depay, STREAM, DECODE, + ("Channel reordering failed."), (NULL)); + gst_rtp_buffer_unmap (&rtp); + return NULL; + } +} + +gboolean +gst_rtp_L8_depay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpL8depay", + GST_RANK_SECONDARY, GST_TYPE_RTP_L8_DEPAY); +} |