From 02476fb5a3d9437ad49af4c7f302b076edfef236 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 11 Nov 2009 11:37:07 +0100 Subject: rtp: add BroadcomVoice payloader --- gst/rtp/Makefile.am | 2 + gst/rtp/gstrtp.c | 4 + gst/rtp/gstrtpbvpay.c | 230 ++++++++++++++++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpbvpay.h | 58 +++++++++++++ 4 files changed, 294 insertions(+) create mode 100644 gst/rtp/gstrtpbvpay.c create mode 100644 gst/rtp/gstrtpbvpay.h diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am index 1cfaa726..facc59ed 100644 --- a/gst/rtp/Makefile.am +++ b/gst/rtp/Makefile.am @@ -6,6 +6,7 @@ libgstrtp_la_SOURCES = \ gstrtpchannels.c \ gstrtpdepay.c \ gstrtpac3depay.c \ + gstrtpbvpay.c \ gstrtpceltdepay.c \ gstrtpceltpay.c \ gstrtpdvdepay.c \ @@ -85,6 +86,7 @@ noinst_HEADERS = \ gstrtpL16depay.h \ gstrtpL16pay.h \ gstrtpac3depay.h \ + gstrtpbvpay.h \ gstrtpceltpay.h \ gstrtpceltdepay.h \ gstrtpdvdepay.h \ diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index a7cd1ca1..d21efccf 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -23,6 +23,7 @@ #include "gstrtpdepay.h" #include "gstrtpac3depay.h" +#include "gstrtpbvpay.h" #include "gstrtpceltdepay.h" #include "gstrtpceltpay.h" #include "gstrtpdvdepay.h" @@ -89,6 +90,9 @@ plugin_init (GstPlugin * plugin) if (!gst_rtp_ac3_depay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_bv_pay_plugin_init (plugin)) + return FALSE; + if (!gst_rtp_celt_depay_plugin_init (plugin)) return FALSE; diff --git a/gst/rtp/gstrtpbvpay.c b/gst/rtp/gstrtpbvpay.c new file mode 100644 index 00000000..b30efb1a --- /dev/null +++ b/gst/rtp/gstrtpbvpay.c @@ -0,0 +1,230 @@ +/* GStreamer + * Copyright (C) <2009> 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include +#include + +#include +#include "gstrtpbvpay.h" + +/* elementfactory information */ +static GstElementDetails gst_rtpbvpay_details = { + "RTP BV Payloader", + "Codec/Payloader/Network", + "Packetize BroadcomVoice audio streams into RTP packets", + "Wim Taymans " +}; + +GST_DEBUG_CATEGORY_STATIC (rtpbvpay_debug); +#define GST_CAT_DEFAULT (rtpbvpay_debug) + +static GstStaticPadTemplate gst_rtpbvpay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-bv, " "mode = (int) {16, 32}") + ); + +static GstStaticPadTemplate gst_rtpbvpay_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) 8000, " + "encoding-name = (string) \"BV16\";" + "application/x-rtp, " + "media = (string) \"audio\", " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " + "clock-rate = (int) 16000, " "encoding-name = (string) \"BV32\"") + ); + + +static GstCaps *gst_rtpbvpay_sink_getcaps (GstBaseRTPPayload * payload, + GstPad * pad); +static gboolean gst_rtpbvpay_sink_setcaps (GstBaseRTPPayload * payload, + GstCaps * caps); + +GST_BOILERPLATE (GstRTPBVPay, gst_rtpbvpay, GstBaseRTPAudioPayload, + GST_TYPE_BASE_RTP_AUDIO_PAYLOAD); + +static void +gst_rtpbvpay_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_rtpbvpay_sink_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtpbvpay_src_template)); + gst_element_class_set_details (element_class, &gst_rtpbvpay_details); +} + +static void +gst_rtpbvpay_class_init (GstRTPBVPayClass * klass) +{ + GstBaseRTPPayloadClass *gstbasertppayload_class; + + gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + + parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD); + + gstbasertppayload_class->set_caps = gst_rtpbvpay_sink_setcaps; + gstbasertppayload_class->get_caps = gst_rtpbvpay_sink_getcaps; + + GST_DEBUG_CATEGORY_INIT (rtpbvpay_debug, "rtpbvpay", 0, + "BroadcomVoice audio RTP payloader"); +} + +static void +gst_rtpbvpay_init (GstRTPBVPay * rtpbvpay, GstRTPBVPayClass * klass) +{ + GstBaseRTPPayload *basertppayload; + GstBaseRTPAudioPayload *basertpaudiopayload; + + basertppayload = GST_BASE_RTP_PAYLOAD (rtpbvpay); + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpbvpay); + + rtpbvpay->mode = -1; + + /* tell basertpaudiopayload that this is a frame based codec */ + gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload); +} + +static gboolean +gst_rtpbvpay_sink_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps) +{ + GstRTPBVPay *rtpbvpay; + GstBaseRTPAudioPayload *basertpaudiopayload; + gint mode; + GstStructure *structure; + const char *payload_name; + + rtpbvpay = GST_RTP_BV_PAY (basertppayload); + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload); + + structure = gst_caps_get_structure (caps, 0); + + payload_name = gst_structure_get_name (structure); + if (g_ascii_strcasecmp ("audio/x-bv", payload_name)) + goto wrong_caps; + + if (!gst_structure_get_int (structure, "mode", &mode)) + goto no_mode; + + if (mode != 16 && mode != 32) + goto wrong_mode; + + if (mode == 16) { + gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV16", + 8000); + basertppayload->clock_rate = 8000; + } else { + gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV32", + 16000); + basertppayload->clock_rate = 16000; + } + + /* set options for this frame based audio codec */ + gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, + mode, mode == 16 ? 10 : 20); + + if (mode != rtpbvpay->mode && rtpbvpay->mode != -1) + goto mode_changed; + + rtpbvpay->mode = mode; + + return TRUE; + + /* ERRORS */ +wrong_caps: + { + GST_ERROR_OBJECT (rtpbvpay, "expected audio/x-bv, received %s", + payload_name); + return FALSE; + } +no_mode: + { + GST_ERROR_OBJECT (rtpbvpay, "did not receive a mode"); + return FALSE; + } +wrong_mode: + { + GST_ERROR_OBJECT (rtpbvpay, "mode must be 16 or 32, received %d", mode); + return FALSE; + } +mode_changed: + { + GST_ERROR_OBJECT (rtpbvpay, "Mode has changed from %d to %d! " + "Mode cannot change while streaming", rtpbvpay->mode, mode); + return FALSE; + } +} + +/* we return the padtemplate caps with the mode field fixated to a value if we + * can */ +static GstCaps * +gst_rtpbvpay_sink_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; + const gchar *mode_str; + gint mode; + + structure = gst_caps_get_structure (otherpadcaps, 0); + + /* construct mode, if we can */ + mode_str = gst_structure_get_string (structure, "encoding-name"); + if (mode_str) { + if (!strcmp (mode_str, "BV16")) + mode = 16; + else if (!strcmp (mode_str, "BV32")) + mode = 32; + else + mode = -1; + + if (mode == 16 || mode == 32) { + structure = gst_caps_get_structure (caps, 0); + gst_structure_set (structure, "mode", G_TYPE_INT, mode, NULL); + } + } + } + gst_caps_unref (otherpadcaps); + } + return caps; +} + +gboolean +gst_rtp_bv_pay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpbvpay", + GST_RANK_NONE, GST_TYPE_RTP_BV_PAY); +} diff --git a/gst/rtp/gstrtpbvpay.h b/gst/rtp/gstrtpbvpay.h new file mode 100644 index 00000000..ed7f2987 --- /dev/null +++ b/gst/rtp/gstrtpbvpay.h @@ -0,0 +1,58 @@ +/* GStreamer + * Copyright (C) <2009> 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_RTP_BV_PAY_H__ +#define __GST_RTP_BV_PAY_H__ + +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_BV_PAY \ + (gst_rtpbvpay_get_type()) +#define GST_RTP_BV_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BV_PAY,GstRTPBVPay)) +#define GST_RTP_BV_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BV_PAY,GstRTPBVPayClass)) +#define GST_IS_RTP_BV_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BV_PAY)) +#define GST_IS_RTP_BV_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BV_PAY)) + +typedef struct _GstRTPBVPay GstRTPBVPay; +typedef struct _GstRTPBVPayClass GstRTPBVPayClass; + +struct _GstRTPBVPay +{ + GstBaseRTPAudioPayload audiopayload; + + gint mode; +}; + +struct _GstRTPBVPayClass +{ + GstBaseRTPAudioPayloadClass parent_class; +}; + +gboolean gst_rtp_bv_pay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_BV_PAY_H__ */ -- cgit v1.2.3