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593615de46
The G722 payload only accepts G722 audio with channels=1, so it must specify the encoding-params=1 in its src caps, otherwise it causes issues with farstream which thinks it supports 2 channels G722 and when confronted with a remote that has G722/8000/2, it will negotiate it and error out with a not-negotiated when the caps don't intersect at runtime. https://bugzilla.gnome.org/show_bug.cgi?id=789878
237 lines
6.9 KiB
C
237 lines
6.9 KiB
C
/* GStreamer
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* Copyright (C) <2010> 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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 <string.h>
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#include <gst/audio/audio.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpg722pay.h"
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#include "gstrtpchannels.h"
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GST_DEBUG_CATEGORY_STATIC (rtpg722pay_debug);
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#define GST_CAT_DEFAULT (rtpg722pay_debug)
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static GstStaticPadTemplate gst_rtp_g722_pay_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/G722, " "rate = (int) 16000, " "channels = (int) 1")
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);
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static GstStaticPadTemplate gst_rtp_g722_pay_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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"encoding-name = (string) \"G722\", "
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"payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
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"encoding-params = (string) 1, "
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"clock-rate = (int) 8000; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"encoding-name = (string) \"G722\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"encoding-params = (string) 1, "
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"clock-rate = (int) 8000")
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);
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static gboolean gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload,
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GstCaps * caps);
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static GstCaps *gst_rtp_g722_pay_getcaps (GstRTPBasePayload * rtppayload,
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GstPad * pad, GstCaps * filter);
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#define gst_rtp_g722_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpG722Pay, gst_rtp_g722_pay,
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GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
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static void
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gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0,
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"G722 RTP Payloader");
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_g722_pay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_g722_pay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP audio payloader", "Codec/Payloader/Network/RTP",
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"Payload-encode Raw audio into RTP packets (RFC 3551)",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasepayload_class->set_caps = gst_rtp_g722_pay_setcaps;
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gstrtpbasepayload_class->get_caps = gst_rtp_g722_pay_getcaps;
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}
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static void
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gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay)
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{
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GstRTPBaseAudioPayload *rtpbaseaudiopayload;
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rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg722pay);
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GST_RTP_BASE_PAYLOAD (rtpg722pay)->pt = GST_RTP_PAYLOAD_G722;
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/* tell rtpbaseaudiopayload that this is a sample based codec */
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gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
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}
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static gboolean
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gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
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{
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GstRtpG722Pay *rtpg722pay;
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GstStructure *structure;
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gint rate, channels, clock_rate;
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gboolean res;
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gchar *params;
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#if 0
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GstAudioChannelPosition *pos;
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const GstRTPChannelOrder *order;
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#endif
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GstRTPBaseAudioPayload *rtpbaseaudiopayload;
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rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
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rtpg722pay = GST_RTP_G722_PAY (basepayload);
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structure = gst_caps_get_structure (caps, 0);
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/* first parse input caps */
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if (!gst_structure_get_int (structure, "rate", &rate))
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goto no_rate;
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if (!gst_structure_get_int (structure, "channels", &channels))
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goto no_channels;
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/* FIXME: Do something with the channel positions */
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#if 0
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/* get the channel order */
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pos = gst_audio_get_channel_positions (structure);
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if (pos)
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order = gst_rtp_channels_get_by_pos (channels, pos);
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else
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order = NULL;
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#endif
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/* Clock rate is always 8000 Hz for G722 according to
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* RFC 3551 although the sampling rate is 16000 Hz */
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clock_rate = 8000;
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gst_rtp_base_payload_set_options (basepayload, "audio",
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basepayload->pt != GST_RTP_PAYLOAD_G722, "G722", clock_rate);
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params = g_strdup_printf ("%d", channels);
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#if 0
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if (!order && channels > 2) {
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GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
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(NULL), ("Unknown channel order for %d channels", channels));
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}
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if (order && order->name) {
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res = gst_rtp_base_payload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
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channels, "channel-order", G_TYPE_STRING, order->name, NULL);
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} else {
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#endif
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res = gst_rtp_base_payload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
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channels, NULL);
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#if 0
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}
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#endif
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g_free (params);
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#if 0
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g_free (pos);
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#endif
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rtpg722pay->rate = rate;
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rtpg722pay->channels = channels;
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/* bits-per-sample is 4 * channels for G722, but as the RTP clock runs at
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* half speed (8 instead of 16 khz), pretend it's 8 bits per sample
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* channels. */
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gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
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8 * rtpg722pay->channels);
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return res;
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/* ERRORS */
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no_rate:
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{
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GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
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return FALSE;
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}
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no_channels:
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{
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GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
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return FALSE;
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}
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}
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static GstCaps *
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gst_rtp_g722_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
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GstCaps * filter)
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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_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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caps = gst_caps_make_writable (caps);
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
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gst_caps_set_simple (caps, "rate", G_TYPE_INT, 16000, NULL);
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}
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gst_caps_unref (otherpadcaps);
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}
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if (filter) {
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GstCaps *tmp;
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GST_DEBUG_OBJECT (rtppayload, "Intersect %" GST_PTR_FORMAT " and filter %"
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GST_PTR_FORMAT, caps, filter);
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tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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caps = tmp;
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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_g722_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpg722pay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_G722_PAY);
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}
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