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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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rtp: add BroadcomVoice payloader
This commit is contained in:
Wim Taymans
parent
dd82612340
commit
02476fb5a3
4 changed files with 294 additions and 0 deletions
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@ -6,6 +6,7 @@ libgstrtp_la_SOURCES = \
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gstrtpchannels.c \
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gstrtpdepay.c \
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gstrtpac3depay.c \
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gstrtpbvpay.c \
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gstrtpceltdepay.c \
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gstrtpceltpay.c \
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gstrtpdvdepay.c \
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@ -85,6 +86,7 @@ noinst_HEADERS = \
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gstrtpL16depay.h \
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gstrtpL16pay.h \
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gstrtpac3depay.h \
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gstrtpbvpay.h \
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gstrtpceltpay.h \
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gstrtpceltdepay.h \
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gstrtpdvdepay.h \
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@ -23,6 +23,7 @@
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#include "gstrtpdepay.h"
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#include "gstrtpac3depay.h"
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#include "gstrtpbvpay.h"
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#include "gstrtpceltdepay.h"
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#include "gstrtpceltpay.h"
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#include "gstrtpdvdepay.h"
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@ -89,6 +90,9 @@ plugin_init (GstPlugin * plugin)
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if (!gst_rtp_ac3_depay_plugin_init (plugin))
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return FALSE;
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if (!gst_rtp_bv_pay_plugin_init (plugin))
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return FALSE;
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if (!gst_rtp_celt_depay_plugin_init (plugin))
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return FALSE;
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230
gst/rtp/gstrtpbvpay.c
Normal file
230
gst/rtp/gstrtpbvpay.c
Normal file
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@ -0,0 +1,230 @@
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/* GStreamer
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* Copyright (C) <2009> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpbvpay.h"
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/* elementfactory information */
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static GstElementDetails gst_rtpbvpay_details = {
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"RTP BV Payloader",
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"Codec/Payloader/Network",
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"Packetize BroadcomVoice audio streams into RTP packets",
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"Wim Taymans <wim.taymans@collabora.co.uk>"
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};
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GST_DEBUG_CATEGORY_STATIC (rtpbvpay_debug);
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#define GST_CAT_DEFAULT (rtpbvpay_debug)
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static GstStaticPadTemplate gst_rtpbvpay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-bv, " "mode = (int) {16, 32}")
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);
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static GstStaticPadTemplate gst_rtpbvpay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) \"BV16\";"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 16000, " "encoding-name = (string) \"BV32\"")
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);
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static GstCaps *gst_rtpbvpay_sink_getcaps (GstBaseRTPPayload * payload,
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GstPad * pad);
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static gboolean gst_rtpbvpay_sink_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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GST_BOILERPLATE (GstRTPBVPay, gst_rtpbvpay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtpbvpay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtpbvpay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtpbvpay_src_template));
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gst_element_class_set_details (element_class, &gst_rtpbvpay_details);
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}
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static void
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gst_rtpbvpay_class_init (GstRTPBVPayClass * klass)
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{
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
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gstbasertppayload_class->set_caps = gst_rtpbvpay_sink_setcaps;
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gstbasertppayload_class->get_caps = gst_rtpbvpay_sink_getcaps;
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GST_DEBUG_CATEGORY_INIT (rtpbvpay_debug, "rtpbvpay", 0,
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"BroadcomVoice audio RTP payloader");
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}
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static void
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gst_rtpbvpay_init (GstRTPBVPay * rtpbvpay, GstRTPBVPayClass * klass)
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{
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GstBaseRTPPayload *basertppayload;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertppayload = GST_BASE_RTP_PAYLOAD (rtpbvpay);
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpbvpay);
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rtpbvpay->mode = -1;
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/* tell basertpaudiopayload that this is a frame based codec */
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gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
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}
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static gboolean
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gst_rtpbvpay_sink_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
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{
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GstRTPBVPay *rtpbvpay;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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gint mode;
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GstStructure *structure;
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const char *payload_name;
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rtpbvpay = GST_RTP_BV_PAY (basertppayload);
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
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structure = gst_caps_get_structure (caps, 0);
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payload_name = gst_structure_get_name (structure);
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if (g_ascii_strcasecmp ("audio/x-bv", payload_name))
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goto wrong_caps;
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if (!gst_structure_get_int (structure, "mode", &mode))
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goto no_mode;
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if (mode != 16 && mode != 32)
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goto wrong_mode;
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if (mode == 16) {
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gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV16",
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8000);
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basertppayload->clock_rate = 8000;
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} else {
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gst_basertppayload_set_options (basertppayload, "audio", TRUE, "BV32",
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16000);
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basertppayload->clock_rate = 16000;
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}
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/* set options for this frame based audio codec */
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gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload,
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mode, mode == 16 ? 10 : 20);
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if (mode != rtpbvpay->mode && rtpbvpay->mode != -1)
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goto mode_changed;
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rtpbvpay->mode = mode;
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return TRUE;
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/* ERRORS */
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wrong_caps:
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{
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GST_ERROR_OBJECT (rtpbvpay, "expected audio/x-bv, received %s",
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payload_name);
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return FALSE;
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}
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no_mode:
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{
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GST_ERROR_OBJECT (rtpbvpay, "did not receive a mode");
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return FALSE;
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}
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wrong_mode:
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{
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GST_ERROR_OBJECT (rtpbvpay, "mode must be 16 or 32, received %d", mode);
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return FALSE;
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}
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mode_changed:
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{
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GST_ERROR_OBJECT (rtpbvpay, "Mode has changed from %d to %d! "
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"Mode cannot change while streaming", rtpbvpay->mode, mode);
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return FALSE;
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}
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}
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/* we return the padtemplate caps with the mode field fixated to a value if we
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* can */
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static GstCaps *
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gst_rtpbvpay_sink_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
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{
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GstCaps *otherpadcaps;
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GstCaps *caps;
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otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
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caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
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if (otherpadcaps) {
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if (!gst_caps_is_empty (otherpadcaps)) {
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GstStructure *structure;
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const gchar *mode_str;
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gint mode;
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structure = gst_caps_get_structure (otherpadcaps, 0);
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/* construct mode, if we can */
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mode_str = gst_structure_get_string (structure, "encoding-name");
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if (mode_str) {
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if (!strcmp (mode_str, "BV16"))
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mode = 16;
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else if (!strcmp (mode_str, "BV32"))
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mode = 32;
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else
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mode = -1;
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if (mode == 16 || mode == 32) {
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_set (structure, "mode", G_TYPE_INT, mode, NULL);
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}
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}
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}
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gst_caps_unref (otherpadcaps);
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}
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return caps;
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}
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gboolean
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gst_rtp_bv_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpbvpay",
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GST_RANK_NONE, GST_TYPE_RTP_BV_PAY);
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}
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58
gst/rtp/gstrtpbvpay.h
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58
gst/rtp/gstrtpbvpay.h
Normal file
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@ -0,0 +1,58 @@
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/* GStreamer
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* Copyright (C) <2009> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_RTP_BV_PAY_H__
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#define __GST_RTP_BV_PAY_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstbasertpaudiopayload.h>
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G_BEGIN_DECLS
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#define GST_TYPE_RTP_BV_PAY \
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(gst_rtpbvpay_get_type())
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#define GST_RTP_BV_PAY(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BV_PAY,GstRTPBVPay))
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#define GST_RTP_BV_PAY_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BV_PAY,GstRTPBVPayClass))
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#define GST_IS_RTP_BV_PAY(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BV_PAY))
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#define GST_IS_RTP_BV_PAY_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BV_PAY))
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typedef struct _GstRTPBVPay GstRTPBVPay;
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typedef struct _GstRTPBVPayClass GstRTPBVPayClass;
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struct _GstRTPBVPay
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{
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GstBaseRTPAudioPayload audiopayload;
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gint mode;
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};
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struct _GstRTPBVPayClass
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{
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GstBaseRTPAudioPayloadClass parent_class;
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};
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gboolean gst_rtp_bv_pay_plugin_init (GstPlugin * plugin);
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G_END_DECLS
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#endif /* __GST_RTP_BV_PAY_H__ */
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