diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index b2a6706e17..f9a823feda 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -70,6 +70,7 @@ #include "gstrtph264pay.h" #include "gstrtph265depay.h" #include "gstrtph265pay.h" +#include "gstrtpisacpay.h" #include "gstrtpj2kdepay.h" #include "gstrtpj2kpay.h" #include "gstrtpjpegdepay.h" @@ -393,6 +394,9 @@ plugin_init (GstPlugin * plugin) if (!gst_rtp_stream_depay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_isac_pay_plugin_init (plugin)) + return FALSE; + if (!gst_element_register (plugin, "rtpredenc", GST_RANK_NONE, GST_TYPE_RTP_RED_ENC)) return FALSE; diff --git a/gst/rtp/gstrtpisacpay.c b/gst/rtp/gstrtpisacpay.c new file mode 100644 index 0000000000..750d9759e5 --- /dev/null +++ b/gst/rtp/gstrtpisacpay.c @@ -0,0 +1,187 @@ +/* GStreamer + * Copyright (C) 2020 Collabora Ltd. + * Author: Guillaume Desmottes , Collabora Ltd. + * + * 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-isacpay + * @title: isacpay + * @short_description: iSAC RTP Payloader + * + * Since: 1.20 + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include + +#include "gstrtpisacpay.h" +#include "gstrtputils.h" + +GST_DEBUG_CATEGORY_STATIC (rtpisacpay_debug); +#define GST_CAT_DEFAULT (rtpisacpay_debug) + +static GstStaticPadTemplate gst_rtp_isac_pay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/isac, " + "rate = (int) { 16000, 32000 }, " "channels = (int) 1") + ); + +static GstStaticPadTemplate gst_rtp_isac_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) { 16000, 32000 }, " + "encoding-name = (string) \"ISAC\", " + "encoding-params = (string) \"1\"") + ); + +struct _GstRtpIsacPay +{ + /*< private > */ + GstRTPBasePayload parent; +}; + +#define gst_rtp_isac_pay_parent_class parent_class +G_DEFINE_TYPE (GstRtpIsacPay, gst_rtp_isac_pay, GST_TYPE_RTP_BASE_PAYLOAD); + +static GstCaps * +gst_rtp_isac_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad, + GstCaps * filter) +{ + GstCaps *otherpadcaps; + GstCaps *caps; + + otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad); + caps = gst_pad_get_pad_template_caps (pad); + + if (otherpadcaps) { + if (!gst_caps_is_empty (otherpadcaps)) { + GstStructure *ps; + GstStructure *s; + const GValue *v; + + ps = gst_caps_get_structure (otherpadcaps, 0); + caps = gst_caps_make_writable (caps); + s = gst_caps_get_structure (caps, 0); + + v = gst_structure_get_value (ps, "clock-rate"); + if (v) + gst_structure_set_value (s, "rate", v); + } + gst_caps_unref (otherpadcaps); + } + + if (filter) { + GstCaps *tcaps = caps; + + caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST); + gst_caps_unref (tcaps); + } + + GST_DEBUG_OBJECT (payload, "%" GST_PTR_FORMAT, caps); + + return caps; +} + +static gboolean +gst_rtp_isac_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) +{ + GstStructure *s; + gint rate; + + GST_DEBUG_OBJECT (payload, "%" GST_PTR_FORMAT, caps); + + s = gst_caps_get_structure (caps, 0); + if (!gst_structure_get_int (s, "rate", &rate)) { + GST_ERROR_OBJECT (payload, "Missing 'rate' in caps"); + return FALSE; + } + + gst_rtp_base_payload_set_options (payload, "audio", TRUE, "ISAC", rate); + + return gst_rtp_base_payload_set_outcaps (payload, NULL); +} + +static GstFlowReturn +gst_rtp_isac_pay_handle_buffer (GstRTPBasePayload * basepayload, + GstBuffer * buffer) +{ + GstBuffer *outbuf; + GstClockTime pts, dts, duration; + + pts = GST_BUFFER_PTS (buffer); + dts = GST_BUFFER_DTS (buffer); + duration = GST_BUFFER_DURATION (buffer); + + outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0); + + gst_rtp_copy_audio_meta (basepayload, outbuf, buffer); + + outbuf = gst_buffer_append (outbuf, buffer); + + GST_BUFFER_PTS (outbuf) = pts; + GST_BUFFER_DTS (outbuf) = dts; + GST_BUFFER_DURATION (outbuf) = duration; + + return gst_rtp_base_payload_push (basepayload, outbuf); +} + +static void +gst_rtp_isac_pay_class_init (GstRtpIsacPayClass * klass) +{ + GstElementClass *gstelement_class = (GstElementClass *) klass; + GstRTPBasePayloadClass *payload_class = (GstRTPBasePayloadClass *) klass; + + payload_class->get_caps = gst_rtp_isac_pay_getcaps; + payload_class->set_caps = gst_rtp_isac_pay_setcaps; + payload_class->handle_buffer = gst_rtp_isac_pay_handle_buffer; + + gst_element_class_add_static_pad_template (gstelement_class, + &gst_rtp_isac_pay_sink_template); + gst_element_class_add_static_pad_template (gstelement_class, + &gst_rtp_isac_pay_src_template); + + gst_element_class_set_static_metadata (gstelement_class, + "RTP iSAC payloader", "Codec/Payloader/Network/RTP", + "Payload-encodes iSAC audio into a RTP packet", + "Guillaume Desmottes "); + + GST_DEBUG_CATEGORY_INIT (rtpisacpay_debug, "rtpisacpay", 0, + "iSAC RTP Payloader"); +} + +static void +gst_rtp_isac_pay_init (GstRtpIsacPay * rtpisacpay) +{ +} + +gboolean +gst_rtp_isac_pay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpisacpay", + GST_RANK_SECONDARY, GST_TYPE_RTP_ISAC_PAY); +} diff --git a/gst/rtp/gstrtpisacpay.h b/gst/rtp/gstrtpisacpay.h new file mode 100644 index 0000000000..764fae455d --- /dev/null +++ b/gst/rtp/gstrtpisacpay.h @@ -0,0 +1,33 @@ +/* GStreamer + * Copyright (C) 2020 Collabora Ltd. + * Author: Guillaume Desmottes , Collabora Ltd. + * + * 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 + */ + + +#ifndef __GST_RTP_ISAC_PAY_H__ +#define __GST_RTP_ISAC_PAY_H__ + +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_ISAC_PAY gst_rtp_isac_pay_get_type () + +G_DECLARE_FINAL_TYPE(GstRtpIsacPay, gst_rtp_isac_pay, GST, RTP_ISAC_PAY, GstRTPBasePayload); + +gboolean gst_rtp_isac_pay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_ISAC_PAY_H__ */ diff --git a/gst/rtp/meson.build b/gst/rtp/meson.build index d57a195aeb..f0825fac45 100644 --- a/gst/rtp/meson.build +++ b/gst/rtp/meson.build @@ -102,6 +102,7 @@ rtp_sources = [ 'rtpstorage.c', 'rtpstoragestream.c', 'gstrtpstorage.c', + 'gstrtpisacpay.c', ] rtp_args = [