diff options
| -rw-r--r-- | gst/rtp/Makefile.am | 4 | ||||
| -rw-r--r-- | gst/rtp/gstrtp.c | 8 | ||||
| -rw-r--r-- | gst/rtp/gstrtpg722depay.c | 260 | ||||
| -rw-r--r-- | gst/rtp/gstrtpg722depay.h | 64 | ||||
| -rw-r--r-- | gst/rtp/gstrtpg722pay.c | 209 | ||||
| -rw-r--r-- | gst/rtp/gstrtpg722pay.h | 61 |
6 files changed, 606 insertions, 0 deletions
diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am index b6aefe6c4..5a300b33c 100644 --- a/gst/rtp/Makefile.am +++ b/gst/rtp/Makefile.am @@ -23,6 +23,8 @@ libgstrtp_la_SOURCES = \ gstrtppcmudepay.c \ gstrtppcmupay.c \ gstrtppcmapay.c \ + gstrtpg722depay.c \ + gstrtpg722pay.c \ gstrtpg723depay.c \ gstrtpg723pay.c \ gstrtpg726pay.c \ @@ -106,6 +108,8 @@ noinst_HEADERS = \ gstrtppcmudepay.h \ gstrtppcmupay.h \ gstrtppcmapay.h \ + gstrtpg722depay.h \ + gstrtpg722pay.h \ gstrtpg723depay.h \ gstrtpg723pay.h \ gstrtpg726depay.h \ diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index 852906efa..339ff9747 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -35,6 +35,8 @@ #include "gstrtppcmapay.h" #include "gstrtppcmadepay.h" #include "gstrtppcmudepay.h" +#include "gstrtpg722depay.h" +#include "gstrtpg722pay.h" #include "gstrtpg723depay.h" #include "gstrtpg723pay.h" #include "gstrtpg726depay.h" @@ -119,6 +121,12 @@ plugin_init (GstPlugin * plugin) if (!gst_rtp_ilbc_depay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_g722_depay_plugin_init (plugin)) + return FALSE; + + if (!gst_rtp_g722_pay_plugin_init (plugin)) + return FALSE; + if (!gst_rtp_g723_depay_plugin_init (plugin)) return FALSE; diff --git a/gst/rtp/gstrtpg722depay.c b/gst/rtp/gstrtpg722depay.c new file mode 100644 index 000000000..ff479c183 --- /dev/null +++ b/gst/rtp/gstrtpg722depay.c @@ -0,0 +1,260 @@ +/* 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <string.h> +#include <stdlib.h> + +#include <gst/audio/audio.h> +#include <gst/audio/multichannel.h> + +#include "gstrtpg722depay.h" +#include "gstrtpchannels.h" + +GST_DEBUG_CATEGORY_STATIC (rtpg722depay_debug); +#define GST_CAT_DEFAULT (rtpg722depay_debug) + +static GstStaticPadTemplate gst_rtp_g722_depay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/G722, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") + ); + +static GstStaticPadTemplate gst_rtp_g722_depay_sink_template = + GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("application/x-rtp, " + "media = (string) \"audio\", " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " + "clock-rate = (int) 8000, " + /* "channels = (int) [1, MAX]" */ + /* "channel-order = (string) ANY" */ + "encoding-name = (string) \"G722\";" + "application/x-rtp, " + "media = (string) \"audio\", " + "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", " + "clock-rate = (int) [ 1, MAX ]" + /* "channels = (int) [1, MAX]" */ + /* "emphasis = (string) ANY" */ + /* "channel-order = (string) ANY" */ + ) + ); + +GST_BOILERPLATE (GstRtpG722Depay, gst_rtp_g722_depay, GstBaseRTPDepayload, + GST_TYPE_BASE_RTP_DEPAYLOAD); + +static gboolean gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload, + GstCaps * caps); +static GstBuffer *gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload, + GstBuffer * buf); + +static void +gst_rtp_g722_depay_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_g722_depay_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_g722_depay_sink_template)); + + gst_element_class_set_details_simple (element_class, "RTP audio depayloader", + "Codec/Depayloader/Network", + "Extracts G722 audio from RTP packets", + "Wim Taymans <wim.taymans@gmail.com>"); +} + +static void +gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass) +{ + GstBaseRTPDepayloadClass *gstbasertpdepayload_class; + + gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; + + gstbasertpdepayload_class->set_caps = gst_rtp_g722_depay_setcaps; + gstbasertpdepayload_class->process = gst_rtp_g722_depay_process; + + GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0, + "G722 RTP Depayloader"); +} + +static void +gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay, + GstRtpG722DepayClass * klass) +{ + /* needed because of GST_BOILERPLATE */ +} + +static gint +gst_rtp_g722_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_g722_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) +{ + GstStructure *structure; + GstRtpG722Depay *rtpg722depay; + gint clock_rate, payload, samplerate; + gint channels; + GstCaps *srccaps; + gboolean res; + const gchar *channel_order; + const GstRTPChannelOrder *order; + + rtpg722depay = GST_RTP_G722_DEPAY (depayload); + + structure = gst_caps_get_structure (caps, 0); + + payload = 96; + gst_structure_get_int (structure, "payload", &payload); + switch (payload) { + case GST_RTP_PAYLOAD_G722: + channels = 1; + clock_rate = 8000; + samplerate = 16000; + break; + default: + /* no fixed mapping, we need clock-rate */ + channels = 0; + clock_rate = 0; + samplerate = 0; + break; + } + + /* caps can overwrite defaults */ + clock_rate = + gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate); + if (clock_rate == 0) + goto no_clockrate; + + if (clock_rate == 8000) + samplerate = 16000; + + if (samplerate == 0) + samplerate = clock_rate; + + channels = + gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels); + if (channels == 0) { + channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels); + if (channels == 0) { + /* channels defaults to 1 otherwise */ + channels = 1; + } + } + + depayload->clock_rate = clock_rate; + rtpg722depay->rate = samplerate; + rtpg722depay->channels = channels; + + srccaps = gst_caps_new_simple ("audio/G722", + "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL); + + /* add channel positions */ + channel_order = gst_structure_get_string (structure, "channel-order"); + + order = gst_rtp_channels_get_by_order (channels, channel_order); + if (order) { + gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), + order->pos); + } else { + GstAudioChannelPosition *pos; + + GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE, + (NULL), ("Unknown channel order '%s' for %d channels", + GST_STR_NULL (channel_order), channels)); + /* create default NONE layout */ + pos = gst_rtp_channels_create_default (channels); + gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos); + g_free (pos); + } + + 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_g722_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) +{ + GstRtpG722Depay *rtpg722depay; + GstBuffer *outbuf; + gint payload_len; + gboolean marker; + + rtpg722depay = GST_RTP_G722_DEPAY (depayload); + + payload_len = gst_rtp_buffer_get_payload_len (buf); + + if (payload_len <= 0) + goto empty_packet; + + GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len); + + outbuf = gst_rtp_buffer_get_payload_buffer (buf); + marker = gst_rtp_buffer_get_marker (buf); + + if (marker) { + /* mark talk spurt with DISCONT */ + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); + } + + return outbuf; + + /* ERRORS */ +empty_packet: + { + GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE, + ("Empty Payload."), (NULL)); + return NULL; + } +} + +gboolean +gst_rtp_g722_depay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpg722depay", + GST_RANK_MARGINAL, GST_TYPE_RTP_G722_DEPAY); +} diff --git a/gst/rtp/gstrtpg722depay.h b/gst/rtp/gstrtpg722depay.h new file mode 100644 index 000000000..2f6b9800e --- /dev/null +++ b/gst/rtp/gstrtpg722depay.h @@ -0,0 +1,64 @@ +/* GStreamer + * Copyright (C) <2010> 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_RTP_G722_DEPAY_H__ +#define __GST_RTP_G722_DEPAY_H__ + +#include <gst/gst.h> +#include <gst/rtp/gstbasertpdepayload.h> + +G_BEGIN_DECLS + +/* Standard macros for defining types for this element. */ +#define GST_TYPE_RTP_G722_DEPAY \ + (gst_rtp_g722_depay_get_type()) +#define GST_RTP_G722_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G722_DEPAY,GstRtpG722Depay)) +#define GST_RTP_G722_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G722_DEPAY,GstRtpG722DepayClass)) +#define GST_IS_RTP_G722_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G722_DEPAY)) +#define GST_IS_RTP_G722_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G722_DEPAY)) + +typedef struct _GstRtpG722Depay GstRtpG722Depay; +typedef struct _GstRtpG722DepayClass GstRtpG722DepayClass; + +/* Definition of structure storing data for this element. */ +struct _GstRtpG722Depay +{ + GstBaseRTPDepayload depayload; + + guint rate; + guint channels; +}; + +/* Standard definition defining a class for this element. */ +struct _GstRtpG722DepayClass +{ + GstBaseRTPDepayloadClass parent_class; +}; + +GType gst_rtp_g722_depay_get_type (void); + +gboolean gst_rtp_g722_depay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_G722_DEPAY_H__ */ diff --git a/gst/rtp/gstrtpg722pay.c b/gst/rtp/gstrtpg722pay.c new file mode 100644 index 000000000..9788fa51a --- /dev/null +++ b/gst/rtp/gstrtpg722pay.c @@ -0,0 +1,209 @@ +/* GStreamer + * Copyright (C) <2010> 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include <string.h> + +#include <gst/audio/audio.h> +#include <gst/audio/multichannel.h> +#include <gst/rtp/gstrtpbuffer.h> + +#include "gstrtpg722pay.h" +#include "gstrtpchannels.h" + +GST_DEBUG_CATEGORY_STATIC (rtpg722pay_debug); +#define GST_CAT_DEFAULT (rtpg722pay_debug) + +static GstStaticPadTemplate gst_rtp_g722_pay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/G722, " "rate = (int) 16000, " "channels = (int) 1") + ); + +static GstStaticPadTemplate gst_rtp_g722_pay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("application/x-rtp, " + "media = (string) \"audio\", " + "encoding-name = (string) \"G722\", " + "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", " + "clock-rate = (int) 8000") + ); + +static gboolean gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, + GstCaps * caps); +static GstCaps *gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload, + GstPad * pad); + +GST_BOILERPLATE (GstRtpG722Pay, gst_rtp_g722_pay, GstBaseRTPAudioPayload, + GST_TYPE_BASE_RTP_AUDIO_PAYLOAD); + +static void +gst_rtp_g722_pay_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_g722_pay_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_g722_pay_sink_template)); + + gst_element_class_set_details_simple (element_class, "RTP audio payloader", + "Codec/Payloader/Network", + "Payload-encode Raw audio into RTP packets (RFC 3551)", + "Wim Taymans <wim.taymans@gmail.com>"); +} + +static void +gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass) +{ + GstBaseRTPPayloadClass *gstbasertppayload_class; + + gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + + gstbasertppayload_class->set_caps = gst_rtp_g722_pay_setcaps; + gstbasertppayload_class->get_caps = gst_rtp_g722_pay_getcaps; + + GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0, + "G722 RTP Payloader"); +} + +static void +gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay, GstRtpG722PayClass * klass) +{ + GstBaseRTPAudioPayload *basertpaudiopayload; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg722pay); + + /* tell basertpaudiopayload that this is a sample based codec */ + gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload); +} + +static gboolean +gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps) +{ + GstRtpG722Pay *rtpg722pay; + GstStructure *structure; + gint rate, channels, clock_rate; + gboolean res; + gchar *params; + GstAudioChannelPosition *pos; + const GstRTPChannelOrder *order; + GstBaseRTPAudioPayload *basertpaudiopayload; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); + rtpg722pay = GST_RTP_G722_PAY (basepayload); + + structure = gst_caps_get_structure (caps, 0); + + /* first parse input caps */ + if (!gst_structure_get_int (structure, "rate", &rate)) + goto no_rate; + + if (!gst_structure_get_int (structure, "channels", &channels)) + goto no_channels; + + /* get the channel order */ + pos = gst_audio_get_channel_positions (structure); + if (pos) + order = gst_rtp_channels_get_by_pos (channels, pos); + else + order = NULL; + + if (rate == 16000) + clock_rate = 8000; + + gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722", + clock_rate); + params = g_strdup_printf ("%d", channels); + + if (!order && channels > 2) { + GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE, + (NULL), ("Unknown channel order for %d channels", channels)); + } + + if (order && order->name) { + res = gst_basertppayload_set_outcaps (basepayload, + "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, + channels, "channel-order", G_TYPE_STRING, order->name, NULL); + } else { + res = gst_basertppayload_set_outcaps (basepayload, + "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, + channels, NULL); + } + + g_free (params); + g_free (pos); + + rtpg722pay->rate = rate; + rtpg722pay->channels = channels; + + /* octet-per-sample is 1 * channels for G722 */ + gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload, + 4 * rtpg722pay->channels); + + return res; + + /* ERRORS */ +no_rate: + { + GST_DEBUG_OBJECT (rtpg722pay, "no rate given"); + return FALSE; + } +no_channels: + { + GST_DEBUG_OBJECT (rtpg722pay, "no channels given"); + return FALSE; + } +} + +static GstCaps * +gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad) +{ + GstCaps *otherpadcaps; + GstCaps *caps; + + otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad); + caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); + + if (otherpadcaps) { + if (!gst_caps_is_empty (otherpadcaps)) { + GstStructure *structure; + + structure = gst_caps_get_structure (otherpadcaps, 0); + + gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL); + gst_caps_set_simple (caps, "rate", G_TYPE_INT, 16000, NULL); + } + gst_caps_unref (otherpadcaps); + } + return caps; +} + +gboolean +gst_rtp_g722_pay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpg722pay", + GST_RANK_NONE, GST_TYPE_RTP_G722_PAY); +} diff --git a/gst/rtp/gstrtpg722pay.h b/gst/rtp/gstrtpg722pay.h new file mode 100644 index 000000000..6712204d0 --- /dev/null +++ b/gst/rtp/gstrtpg722pay.h @@ -0,0 +1,61 @@ +/* GStreamer + * Copyright (C) <2010> 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_RTP_G722_PAY_H__ +#define __GST_RTP_G722_PAY_H__ + +#include <gst/gst.h> +#include <gst/rtp/gstbasertpaudiopayload.h> + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_G722_PAY \ + (gst_rtp_g722_pay_get_type()) +#define GST_RTP_G722_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G722_PAY,GstRtpG722Pay)) +#define GST_RTP_G722_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G722_PAY,GstRtpG722PayClass)) +#define GST_IS_RTP_G722_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G722_PAY)) +#define GST_IS_RTP_G722_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G722_PAY)) + +typedef struct _GstRtpG722Pay GstRtpG722Pay; +typedef struct _GstRtpG722PayClass GstRtpG722PayClass; + +struct _GstRtpG722Pay +{ + GstBaseRTPAudioPayload payload; + + gint rate; + gint channels; +}; + +struct _GstRtpG722PayClass +{ + GstBaseRTPAudioPayloadClass parent_class; +}; + +GType gst_rtp_g722_pay_get_type (void); + +gboolean gst_rtp_g722_pay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_G722_PAY_H__ */ |