From a956a6ceb2deb87cc1361aee1d6626449f46dab2 Mon Sep 17 00:00:00 2001 From: David Holroyd Date: Mon, 9 Sep 2013 11:16:40 +0200 Subject: [PATCH] rtp: add L24 pay and depayloader Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734 --- gst/rtp/Makefile.am | 2 + gst/rtp/gstrtp.c | 8 + gst/rtp/gstrtpL24depay.c | 277 ++++++++++++++++++++++++++ gst/rtp/gstrtpL24depay.h | 67 +++++++ gst/rtp/gstrtpL24pay.c | 244 +++++++++++++++++++++++ gst/rtp/gstrtpL24pay.h | 63 ++++++ tests/check/elements/rtp-payloading.c | 20 ++ 7 files changed, 681 insertions(+) create mode 100644 gst/rtp/gstrtpL24depay.c create mode 100644 gst/rtp/gstrtpL24depay.h create mode 100644 gst/rtp/gstrtpL24pay.c create mode 100644 gst/rtp/gstrtpL24pay.h diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am index 6c1d6b8ec3..1d5b19e0fa 100644 --- a/gst/rtp/Makefile.am +++ b/gst/rtp/Makefile.am @@ -50,6 +50,8 @@ libgstrtp_la_SOURCES = \ gstrtpjpegpay.c \ gstrtpL16depay.c \ gstrtpL16pay.c \ + gstrtpL24depay.c \ + gstrtpL24pay.c \ gstasteriskh263.c \ gstrtpmp1sdepay.c \ gstrtpmp2tdepay.c \ diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index d317156c16..893dafdef9 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -68,6 +68,8 @@ #include "gstrtpjpegpay.h" #include "gstrtpL16depay.h" #include "gstrtpL16pay.h" +#include "gstrtpL24depay.h" +#include "gstrtpL24pay.h" #include "gstasteriskh263.h" #include "gstrtpmp1sdepay.h" #include "gstrtpmp2tdepay.h" @@ -236,6 +238,12 @@ plugin_init (GstPlugin * plugin) if (!gst_rtp_L16_depay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_L24_pay_plugin_init (plugin)) + return FALSE; + + if (!gst_rtp_L24_depay_plugin_init (plugin)) + return FALSE; + if (!gst_asteriskh263_plugin_init (plugin)) return FALSE; diff --git a/gst/rtp/gstrtpL24depay.c b/gst/rtp/gstrtpL24depay.c new file mode 100644 index 0000000000..ba4cc94b60 --- /dev/null +++ b/gst/rtp/gstrtpL24depay.c @@ -0,0 +1,277 @@ +/* GStreamer + * Copyright (C) <2007> Wim Taymans + * + * 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-rtpL24depay + * @see_also: rtpL24pay + * + * Extract raw audio from RTP packets according to RFC 3190, section 4. + * For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt + * + * + * Example pipeline + * |[ + * gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink + * ]| This example pipeline will depayload an RTP raw audio stream. Refer to + * the rtpL24pay example to create the RTP stream. + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include + +#include + +#include "gstrtpL24depay.h" +#include "gstrtpchannels.h" + +GST_DEBUG_CATEGORY_STATIC (rtpL24depay_debug); +#define GST_CAT_DEFAULT (rtpL24depay_debug) + +static GstStaticPadTemplate gst_rtp_L24_depay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) S24BE, " + "layout = (string) interleaved, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") + ); + +static GstStaticPadTemplate gst_rtp_L24_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) \"L24\";" + "application/x-rtp, " + "media = (string) \"audio\", " + "payload = (int) { " GST_RTP_PAYLOAD_DYNAMIC_STRING " }," + "clock-rate = (int) [ 1, MAX ]" + /* "channels = (int) [1, MAX]" */ + /* "emphasis = (string) ANY" */ + /* "channel-order = (string) ANY" */ + ) + ); + +#define gst_rtp_L24_depay_parent_class parent_class +G_DEFINE_TYPE (GstRtpL24Depay, gst_rtp_L24_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); + +static gboolean gst_rtp_L24_depay_setcaps (GstRTPBaseDepayload * depayload, + GstCaps * caps); +static GstBuffer *gst_rtp_L24_depay_process (GstRTPBaseDepayload * depayload, + GstBuffer * buf); + +static void +gst_rtp_L24_depay_class_init (GstRtpL24DepayClass * klass) +{ + GstElementClass *gstelement_class; + GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; + + gstelement_class = (GstElementClass *) klass; + gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass; + + gstrtpbasedepayload_class->set_caps = gst_rtp_L24_depay_setcaps; + gstrtpbasedepayload_class->process = gst_rtp_L24_depay_process; + + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_L24_depay_src_template)); + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_L24_depay_sink_template)); + + gst_element_class_set_static_metadata (gstelement_class, + "RTP audio depayloader", "Codec/Depayloader/Network/RTP", + "Extracts raw 24-bit audio from RTP packets", + "Zeeshan Ali ," "Wim Taymans ," + "David Holroyd "); + + GST_DEBUG_CATEGORY_INIT (rtpL24depay_debug, "rtpL24depay", 0, + "Raw Audio RTP Depayloader"); +} + +static void +gst_rtp_L24_depay_init (GstRtpL24Depay * rtpL24depay) +{ +} + +static gint +gst_rtp_L24_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_L24_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) +{ + GstStructure *structure; + GstRtpL24Depay *rtpL24depay; + gint clock_rate, payload; + gint channels; + GstCaps *srccaps; + gboolean res; + const gchar *channel_order; + const GstRTPChannelOrder *order; + GstAudioInfo *info; + + rtpL24depay = GST_RTP_L24_DEPAY (depayload); + + structure = gst_caps_get_structure (caps, 0); + + payload = 96; + gst_structure_get_int (structure, "payload", &payload); + /* no fixed mapping, we need clock-rate */ + channels = 0; + clock_rate = 0; + + /* caps can overwrite defaults */ + clock_rate = + gst_rtp_L24_depay_parse_int (structure, "clock-rate", clock_rate); + if (clock_rate == 0) + goto no_clockrate; + + channels = + gst_rtp_L24_depay_parse_int (structure, "encoding-params", channels); + if (channels == 0) { + channels = gst_rtp_L24_depay_parse_int (structure, "channels", channels); + if (channels == 0) { + /* channels defaults to 1 otherwise */ + channels = 1; + } + } + + depayload->clock_rate = clock_rate; + + info = &rtpL24depay->info; + gst_audio_info_init (info); + info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S24BE); + 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); + rtpL24depay->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 (rtpL24depay, 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_L24_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf) +{ + GstRtpL24Depay *rtpL24depay; + GstBuffer *outbuf; + gint payload_len; + gboolean marker; + GstRTPBuffer rtp = { NULL }; + + rtpL24depay = GST_RTP_L24_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 (rtpL24depay, "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 (rtpL24depay->order && + !gst_audio_buffer_reorder_channels (outbuf, + rtpL24depay->info.finfo->format, rtpL24depay->info.channels, + rtpL24depay->info.position, rtpL24depay->order->pos)) { + goto reorder_failed; + } + + gst_rtp_buffer_unmap (&rtp); + + return outbuf; + + /* ERRORS */ +empty_packet: + { + GST_ELEMENT_WARNING (rtpL24depay, STREAM, DECODE, + ("Empty Payload."), (NULL)); + gst_rtp_buffer_unmap (&rtp); + return NULL; + } +reorder_failed: + { + GST_ELEMENT_ERROR (rtpL24depay, STREAM, DECODE, + ("Channel reordering failed."), (NULL)); + gst_rtp_buffer_unmap (&rtp); + return NULL; + } +} + +gboolean +gst_rtp_L24_depay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpL24depay", + GST_RANK_SECONDARY, GST_TYPE_RTP_L24_DEPAY); +} diff --git a/gst/rtp/gstrtpL24depay.h b/gst/rtp/gstrtpL24depay.h new file mode 100644 index 0000000000..c4e00e6f23 --- /dev/null +++ b/gst/rtp/gstrtpL24depay.h @@ -0,0 +1,67 @@ +/* GStreamer + * Copyright (C) <2007> Wim Taymans + * + * 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_L24_DEPAY_H__ +#define __GST_RTP_L24_DEPAY_H__ + +#include +#include +#include + +#include "gstrtpchannels.h" + +G_BEGIN_DECLS + +/* Standard macros for defining types for this element. */ +#define GST_TYPE_RTP_L24_DEPAY \ + (gst_rtp_L24_depay_get_type()) +#define GST_RTP_L24_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L24_DEPAY,GstRtpL24Depay)) +#define GST_RTP_L24_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L24_DEPAY,GstRtpL24DepayClass)) +#define GST_IS_RTP_L24_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L24_DEPAY)) +#define GST_IS_RTP_L24_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L24_DEPAY)) + +typedef struct _GstRtpL24Depay GstRtpL24Depay; +typedef struct _GstRtpL24DepayClass GstRtpL24DepayClass; + +/* Definition of structure storing data for this element. */ +struct _GstRtpL24Depay +{ + GstRTPBaseDepayload depayload; + + GstAudioInfo info; + const GstRTPChannelOrder *order; +}; + +/* Standard definition defining a class for this element. */ +struct _GstRtpL24DepayClass +{ + GstRTPBaseDepayloadClass parent_class; +}; + +GType gst_rtp_L24_depay_get_type (void); + +gboolean gst_rtp_L24_depay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_L24_DEPAY_H__ */ diff --git a/gst/rtp/gstrtpL24pay.c b/gst/rtp/gstrtpL24pay.c new file mode 100644 index 0000000000..d2612e9c3d --- /dev/null +++ b/gst/rtp/gstrtpL24pay.c @@ -0,0 +1,244 @@ +/* GStreamer + * Copyright (C) <2007> Wim Taymans + * + * 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-rtpL24pay + * @see_also: rtpL24depay + * + * Payload raw 24-bit audio into RTP packets according to RFC 3190, section 4. + * For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt + * + * + * Example pipeline + * |[ + * gst-launch -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink + * ]| This example pipeline will payload raw audio. Refer to + * the rtpL24depay example to depayload and play the RTP stream. + * + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include + +#include +#include + +#include "gstrtpL24pay.h" +#include "gstrtpchannels.h" + +GST_DEBUG_CATEGORY_STATIC (rtpL24pay_debug); +#define GST_CAT_DEFAULT (rtpL24pay_debug) + +static GstStaticPadTemplate gst_rtp_L24_pay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) S24BE, " + "layout = (string) interleaved, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") + ); + +static GstStaticPadTemplate gst_rtp_L24_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) [ 96, 127 ], " + "clock-rate = (int) [ 1, MAX ], " + "encoding-name = (string) \"L24\", " "channels = (int) [ 1, MAX ];") + ); + +static gboolean gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload, + GstCaps * caps); +static GstCaps *gst_rtp_L24_pay_getcaps (GstRTPBasePayload * rtppayload, + GstPad * pad, GstCaps * filter); +static GstFlowReturn +gst_rtp_L24_pay_handle_buffer (GstRTPBasePayload * basepayload, + GstBuffer * buffer); + +#define gst_rtp_L24_pay_parent_class parent_class +G_DEFINE_TYPE (GstRtpL24Pay, gst_rtp_L24_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD); + +static void +gst_rtp_L24_pay_class_init (GstRtpL24PayClass * klass) +{ + GstElementClass *gstelement_class; + GstRTPBasePayloadClass *gstrtpbasepayload_class; + + gstelement_class = (GstElementClass *) klass; + gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; + + gstrtpbasepayload_class->set_caps = gst_rtp_L24_pay_setcaps; + gstrtpbasepayload_class->get_caps = gst_rtp_L24_pay_getcaps; + gstrtpbasepayload_class->handle_buffer = gst_rtp_L24_pay_handle_buffer; + + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_L24_pay_src_template)); + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_rtp_L24_pay_sink_template)); + + gst_element_class_set_static_metadata (gstelement_class, + "RTP audio payloader", "Codec/Payloader/Network/RTP", + "Payload-encode Raw 24-bit audio into RTP packets (RFC 3190)", + "Wim Taymans ," + "David Holroyd "); + + GST_DEBUG_CATEGORY_INIT (rtpL24pay_debug, "rtpL24pay", 0, + "L24 RTP Payloader"); +} + +static void +gst_rtp_L24_pay_init (GstRtpL24Pay * rtpL24pay) +{ + GstRTPBaseAudioPayload *rtpbaseaudiopayload; + + rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL24pay); + + /* tell rtpbaseaudiopayload that this is a sample based codec */ + gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); +} + +static gboolean +gst_rtp_L24_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps) +{ + GstRtpL24Pay *rtpL24pay; + gboolean res; + gchar *params; + GstAudioInfo *info; + const GstRTPChannelOrder *order; + GstRTPBaseAudioPayload *rtpbaseaudiopayload; + + rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload); + rtpL24pay = GST_RTP_L24_PAY (basepayload); + + info = &rtpL24pay->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); + rtpL24pay->order = order; + + gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L24", + info->rate); + params = g_strdup_printf ("%d", info->channels); + + if (!order && info->channels > 2) { + GST_ELEMENT_WARNING (rtpL24pay, 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 3 * channels for L24 */ + gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, + 3 * info->channels); + + return res; + + /* ERRORS */ +invalid_caps: + { + GST_DEBUG_OBJECT (rtpL24pay, "invalid caps"); + return FALSE; + } +} + +static GstCaps * +gst_rtp_L24_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); + } + + if (gst_structure_get_int (structure, "clock-rate", &rate)) { + gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL); + } + + } + 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_L24_pay_handle_buffer (GstRTPBasePayload * basepayload, + GstBuffer * buffer) +{ + GstRtpL24Pay *rtpL24pay; + + rtpL24pay = GST_RTP_L24_PAY (basepayload); + buffer = gst_buffer_make_writable (buffer); + + if (rtpL24pay->order && + !gst_audio_buffer_reorder_channels (buffer, rtpL24pay->info.finfo->format, + rtpL24pay->info.channels, rtpL24pay->info.position, + rtpL24pay->order->pos)) { + return GST_FLOW_ERROR; + } + + return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload, + buffer); +} + +gboolean +gst_rtp_L24_pay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpL24pay", + GST_RANK_SECONDARY, GST_TYPE_RTP_L24_PAY); +} diff --git a/gst/rtp/gstrtpL24pay.h b/gst/rtp/gstrtpL24pay.h new file mode 100644 index 0000000000..47395ad164 --- /dev/null +++ b/gst/rtp/gstrtpL24pay.h @@ -0,0 +1,63 @@ +/* GStreamer + * Copyright (C) <2005> Wim Taymans + * + * 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_L24_PAY_H__ +#define __GST_RTP_L24_PAY_H__ + +#include +#include + +#include "gstrtpchannels.h" + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_L24_PAY \ + (gst_rtp_L24_pay_get_type()) +#define GST_RTP_L24_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L24_PAY,GstRtpL24Pay)) +#define GST_RTP_L24_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L24_PAY,GstRtpL24PayClass)) +#define GST_IS_RTP_L24_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L24_PAY)) +#define GST_IS_RTP_L24_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L24_PAY)) + +typedef struct _GstRtpL24Pay GstRtpL24Pay; +typedef struct _GstRtpL24PayClass GstRtpL24PayClass; + +struct _GstRtpL24Pay +{ + GstRTPBaseAudioPayload payload; + + GstAudioInfo info; + const GstRTPChannelOrder *order; +}; + +struct _GstRtpL24PayClass +{ + GstRTPBaseAudioPayloadClass parent_class; +}; + +GType gst_rtp_L24_pay_get_type (void); + +gboolean gst_rtp_L24_pay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_L24_PAY_H__ */ diff --git a/tests/check/elements/rtp-payloading.c b/tests/check/elements/rtp-payloading.c index e5c1c8d8fd..32d785fdaf 100644 --- a/tests/check/elements/rtp-payloading.c +++ b/tests/check/elements/rtp-payloading.c @@ -650,6 +650,25 @@ GST_START_TEST (rtp_L16) "rtpL16pay", "rtpL16depay", 0, 0, FALSE); } +GST_END_TEST; + +static const guint8 rtp_L24_frame_data[] = + { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +}; + +static int rtp_L24_frame_data_size = 24; + +static int rtp_L24_frame_count = 1; + +GST_START_TEST (rtp_L24) +{ + rtp_pipeline_test (rtp_L24_frame_data, rtp_L24_frame_data_size, + rtp_L24_frame_count, + "audio/x-raw,format=S24BE,rate=1,channels=1,layout=(string)interleaved", + "rtpL24pay", "rtpL24depay", 0, 0, FALSE); +} + GST_END_TEST; static const guint8 rtp_mp2t_frame_data[] = { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, @@ -917,6 +936,7 @@ rtp_payloading_suite (void) tcase_add_test (tc_chain, rtp_h264_list_gt_mtu); tcase_add_test (tc_chain, rtp_h264_list_gt_mtu_avc); tcase_add_test (tc_chain, rtp_L16); + tcase_add_test (tc_chain, rtp_L24); tcase_add_test (tc_chain, rtp_mp2t); tcase_add_test (tc_chain, rtp_mp4v); tcase_add_test (tc_chain, rtp_mp4v_list);