From db07fc7e8507f55afb8e8c2e9e1b880550b4d76e Mon Sep 17 00:00:00 2001 From: Tim Allen Date: Fri, 15 May 2015 17:00:26 +0100 Subject: [PATCH] rtp: add L8 audio support --- gst/rtp/Makefile.am | 4 + gst/rtp/gstrtp.c | 8 ++ gst/rtp/gstrtpL8depay.c | 267 ++++++++++++++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpL8depay.h | 65 ++++++++++++ gst/rtp/gstrtpL8pay.c | 250 +++++++++++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpL8pay.h | 64 ++++++++++++ gst/rtp/meson.build | 2 + 7 files changed, 660 insertions(+) create mode 100644 gst/rtp/gstrtpL8depay.c create mode 100644 gst/rtp/gstrtpL8depay.h create mode 100644 gst/rtp/gstrtpL8pay.c create mode 100644 gst/rtp/gstrtpL8pay.h diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am index 66af79a..7b2afd5 100644 --- a/gst/rtp/Makefile.am +++ b/gst/rtp/Makefile.am @@ -54,6 +54,8 @@ libgstrtp_la_SOURCES = \ gstrtpjpegpay.c \ gstrtpklvdepay.c \ gstrtpklvpay.c \ + gstrtpL8depay.c \ + gstrtpL8pay.c \ gstrtpL16depay.c \ gstrtpL16pay.c \ gstrtpL24depay.c \ @@ -112,6 +114,8 @@ noinst_HEADERS = \ dboolhuff.h \ fnv1hash.h \ gstrtpchannels.h \ + gstrtpL8depay.h \ + gstrtpL8pay.h \ gstrtpL16depay.h \ gstrtpL16pay.h \ gstrtpL24depay.h \ diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index 186cf9c..8b0ef58 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -76,6 +76,8 @@ #include "gstrtpjpegpay.h" #include "gstrtpklvdepay.h" #include "gstrtpklvpay.h" +#include "gstrtpL8depay.h" +#include "gstrtpL8pay.h" #include "gstrtpL16depay.h" #include "gstrtpL16pay.h" #include "gstrtpL24depay.h" @@ -275,6 +277,12 @@ plugin_init (GstPlugin * plugin) if (!gst_rtp_klv_pay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_L8_pay_plugin_init (plugin)) + return FALSE; + + if (!gst_rtp_L8_depay_plugin_init (plugin)) + return FALSE; + if (!gst_rtp_L16_pay_plugin_init (plugin)) return FALSE; diff --git a/gst/rtp/gstrtpL8depay.c b/gst/rtp/gstrtpL8depay.c new file mode 100644 index 0000000..5b9520a --- /dev/null +++ b/gst/rtp/gstrtpL8depay.c @@ -0,0 +1,267 @@ +/* GStreamer + * Copyright (C) <2007> Wim Taymans + * 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 + * + * + * Example pipeline + * |[ + * 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. + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include + +#include + +#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 ," "Wim Taymans , " + "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); +} diff --git a/gst/rtp/gstrtpL8depay.h b/gst/rtp/gstrtpL8depay.h new file mode 100644 index 0000000..a2d9bec --- /dev/null +++ b/gst/rtp/gstrtpL8depay.h @@ -0,0 +1,65 @@ +/* GStreamer + * Copyright (C) <2007> Wim Taymans + * 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. + */ + +#ifndef __GST_RTP_L8_DEPAY_H__ +#define __GST_RTP_L8_DEPAY_H__ + +#include +#include +#include + +#include "gstrtpchannels.h" + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_L8_DEPAY \ + (gst_rtp_L8_depay_get_type()) +#define GST_RTP_L8_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L8_DEPAY,GstRtpL8Depay)) +#define GST_RTP_L8_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L8_DEPAY,GstRtpL8DepayClass)) +#define GST_IS_RTP_L8_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L8_DEPAY)) +#define GST_IS_RTP_L8_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L8_DEPAY)) + +typedef struct _GstRtpL8Depay GstRtpL8Depay; +typedef struct _GstRtpL8DepayClass GstRtpL8DepayClass; + +struct _GstRtpL8Depay +{ + GstRTPBaseDepayload depayload; + + GstAudioInfo info; + const GstRTPChannelOrder *order; +}; + +struct _GstRtpL8DepayClass +{ + GstRTPBaseDepayloadClass parent_class; +}; + +GType gst_rtp_L8_depay_get_type (void); + +gboolean gst_rtp_L8_depay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_L8_DEPAY_H__ */ diff --git a/gst/rtp/gstrtpL8pay.c b/gst/rtp/gstrtpL8pay.c new file mode 100644 index 0000000..86e7b22 --- /dev/null +++ b/gst/rtp/gstrtpL8pay.c @@ -0,0 +1,250 @@ +/* GStreamer + * Copyright (C) <2007> Wim Taymans + * 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-rtpL8pay + * @see_also: rtpL8depay + * + * Payload raw audio into RTP packets according to RFC 3551. + * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt + * + * + * Example pipeline + * |[ + * gst-launch -v audiotestsrc ! audioconvert ! rtpL8pay ! udpsink + * ]| This example pipeline will payload raw audio. Refer to + * the rtpL8depay example to depayload and play the RTP stream. + * + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include + +#include +#include + +#include "gstrtpL8pay.h" +#include "gstrtpchannels.h" + +GST_DEBUG_CATEGORY_STATIC (rtpL8pay_debug); +#define GST_CAT_DEFAULT (rtpL8pay_debug) + +static GstStaticPadTemplate gst_rtp_L8_pay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + 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_pay_src_template = + GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("application/x-rtp, " + "media = (string) audio, " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " + "clock-rate = (int) [ 1, MAX ], " + "encoding-name = (string) L8, " "channels = (int) [ 1, MAX ];") + ); + +static gboolean gst_rtp_L8_pay_setcaps (GstRTPBasePayload * basepayload, + GstCaps * caps); +static GstCaps *gst_rtp_L8_pay_getcaps (GstRTPBasePayload * rtppayload, + GstPad * pad, GstCaps * filter); +static GstFlowReturn +gst_rtp_L8_pay_handle_buffer (GstRTPBasePayload * basepayload, + GstBuffer * buffer); + +#define gst_rtp_L8_pay_parent_class parent_class +G_DEFINE_TYPE (GstRtpL8Pay, gst_rtp_L8_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD); + +static void +gst_rtp_L8_pay_class_init (GstRtpL8PayClass * klass) +{ + GstElementClass *gstelement_class; + GstRTPBasePayloadClass *gstrtpbasepayload_class; + + gstelement_class = (GstElementClass *) klass; + gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; + + gstrtpbasepayload_class->set_caps = gst_rtp_L8_pay_setcaps; + gstrtpbasepayload_class->get_caps = gst_rtp_L8_pay_getcaps; + gstrtpbasepayload_class->handle_buffer = gst_rtp_L8_pay_handle_buffer; + + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_L8_pay_src_template)); + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_L8_pay_sink_template)); + + gst_element_class_set_static_metadata (gstelement_class, + "RTP audio payloader", "Codec/Payloader/Network/RTP", + "Payload-encode Raw audio into RTP packets (RFC 3551)", + "Wim Taymans , " + "GE Intelligent Platforms Embedded Systems, Inc."); + + GST_DEBUG_CATEGORY_INIT (rtpL8pay_debug, "rtpL8pay", 0, "L8 RTP Payloader"); +} + +static void +gst_rtp_L8_pay_init (GstRtpL8Pay * rtpL8pay) +{ + GstRTPBaseAudioPayload *rtpbaseaudiopayload; + + rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL8pay); + + /* tell rtpbaseaudiopayload that this is a sample based codec */ + gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); +} + +static gboolean +gst_rtp_L8_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps) +{ + GstRtpL8Pay *rtpL8pay; + gboolean res; + gchar *params; + GstAudioInfo *info; + const GstRTPChannelOrder *order; + GstRTPBaseAudioPayload *rtpbaseaudiopayload; + + rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload); + rtpL8pay = GST_RTP_L8_PAY (basepayload); + + info = &rtpL8pay->info; + gst_audio_info_init (info); + if (!gst_audio_info_from_caps (info, caps)) + goto invalid_caps; + + order = gst_rtp_channels_get_by_pos (info->channels, info->position); + rtpL8pay->order = order; + + gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L8", + info->rate); + params = g_strdup_printf ("%d", info->channels); + + if (!order && info->channels > 2) { + GST_ELEMENT_WARNING (rtpL8pay, STREAM, DECODE, + (NULL), ("Unknown channel order for %d channels", info->channels)); + } + + if (order && order->name) { + res = gst_rtp_base_payload_set_outcaps (basepayload, + "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, + info->channels, "channel-order", G_TYPE_STRING, order->name, NULL); + } else { + res = gst_rtp_base_payload_set_outcaps (basepayload, + "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT, + info->channels, NULL); + } + + g_free (params); + + /* octet-per-sample is # channels for L8 */ + gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, + info->channels); + + return res; + + /* ERRORS */ +invalid_caps: + { + GST_DEBUG_OBJECT (rtpL8pay, "invalid caps"); + return FALSE; + } +} + +static GstCaps * +gst_rtp_L8_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad, + GstCaps * filter) +{ + GstCaps *otherpadcaps; + GstCaps *caps; + + caps = gst_pad_get_pad_template_caps (pad); + + otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad); + if (otherpadcaps) { + if (!gst_caps_is_empty (otherpadcaps)) { + GstStructure *structure; + gint channels; + gint rate; + + structure = gst_caps_get_structure (otherpadcaps, 0); + caps = gst_caps_make_writable (caps); + + if (gst_structure_get_int (structure, "channels", &channels)) { + gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL); + } else { + /* Support any number of channels, if not explicitly specified */ + gst_structure_remove_field (structure, "channels"); + } + + if (gst_structure_get_int (structure, "clock-rate", &rate)) { + gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL); + } else { + /* Support any rate, if not explicitly specified */ + gst_structure_remove_field (structure, "rate"); + } + + } + gst_caps_unref (otherpadcaps); + } + + if (filter) { + GstCaps *tcaps = caps; + + caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST); + gst_caps_unref (tcaps); + } + + return caps; +} + +static GstFlowReturn +gst_rtp_L8_pay_handle_buffer (GstRTPBasePayload * basepayload, + GstBuffer * buffer) +{ + GstRtpL8Pay *rtpL8pay; + + rtpL8pay = GST_RTP_L8_PAY (basepayload); + buffer = gst_buffer_make_writable (buffer); + + if (rtpL8pay->order && + !gst_audio_buffer_reorder_channels (buffer, rtpL8pay->info.finfo->format, + rtpL8pay->info.channels, rtpL8pay->info.position, + rtpL8pay->order->pos)) { + return GST_FLOW_ERROR; + } + + return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload, + buffer); +} + +gboolean +gst_rtp_L8_pay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpL8pay", + GST_RANK_SECONDARY, GST_TYPE_RTP_L8_PAY); +} diff --git a/gst/rtp/gstrtpL8pay.h b/gst/rtp/gstrtpL8pay.h new file mode 100644 index 0000000..183eb2f --- /dev/null +++ b/gst/rtp/gstrtpL8pay.h @@ -0,0 +1,64 @@ +/* GStreamer + * Copyright (C) <2005> Wim Taymans + * 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. + */ + +#ifndef __GST_RTP_L8_PAY_H__ +#define __GST_RTP_L8_PAY_H__ + +#include +#include + +#include "gstrtpchannels.h" + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_L8_PAY \ + (gst_rtp_L8_pay_get_type()) +#define GST_RTP_L8_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L8_PAY,GstRtpL8Pay)) +#define GST_RTP_L8_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L8_PAY,GstRtpL8PayClass)) +#define GST_IS_RTP_L8_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L8_PAY)) +#define GST_IS_RTP_L8_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L8_PAY)) + +typedef struct _GstRtpL8Pay GstRtpL8Pay; +typedef struct _GstRtpL8PayClass GstRtpL8PayClass; + +struct _GstRtpL8Pay +{ + GstRTPBaseAudioPayload payload; + + GstAudioInfo info; + const GstRTPChannelOrder *order; +}; + +struct _GstRtpL8PayClass +{ + GstRTPBaseAudioPayloadClass parent_class; +}; + +GType gst_rtp_L8_pay_get_type (void); + +gboolean gst_rtp_L8_pay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_L8_PAY_H__ */ diff --git a/gst/rtp/meson.build b/gst/rtp/meson.build index 4696633..6f24152 100644 --- a/gst/rtp/meson.build +++ b/gst/rtp/meson.build @@ -54,6 +54,8 @@ rtp_sources = [ 'gstrtpj2kpay.c', 'gstrtpjpegdepay.c', 'gstrtpjpegpay.c', + 'gstrtpL8depay.c', + 'gstrtpL8pay.c', 'gstrtpL16depay.c', 'gstrtpL16pay.c', 'gstrtpL24depay.c', -- 2.7.4