mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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419 lines
13 KiB
C
419 lines
13 KiB
C
/* GStreamer
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* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) 2005 Edgard Lima <edgard.lima@indt.org.br>
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* Copyright (C) 2005 Nokia Corporation <kai.vehmanen@nokia.com>
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* Copyright (C) 2007,2008 Axis Communications <dev-gstreamer@axis.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 "gstrtpg726pay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpg726pay_debug);
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#define GST_CAT_DEFAULT (rtpg726pay_debug)
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#define DEFAULT_FORCE_AAL2 TRUE
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enum
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{
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PROP_0,
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PROP_FORCE_AAL2,
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PROP_LAST
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};
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static GstStaticPadTemplate gst_rtp_g726_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/x-adpcm, "
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"channels = (int) 1, "
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"rate = (int) 8000, "
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"bitrate = (int) { 16000, 24000, 32000, 40000 }, "
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"layout = (string) \"g726\"")
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);
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static GstStaticPadTemplate gst_rtp_g726_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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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\", "
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" \"AAL2-G726-16\", \"AAL2-G726-24\", \"AAL2-G726-32\", \"AAL2-G726-40\" } ")
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);
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static void gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static gboolean gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_g726_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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GST_BOILERPLATE (GstRtpG726Pay, gst_rtp_g726_pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtp_g726_pay_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_rtp_g726_pay_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_g726_pay_src_template));
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gst_element_class_set_details_simple (element_class, "RTP G.726 payloader",
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"Codec/Payloader/Network/RTP",
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"Payload-encodes G.726 audio into a RTP packet",
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"Axis Communications <dev-gstreamer@axis.com>");
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}
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static void
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gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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gobject_class->set_property = gst_rtp_g726_pay_set_property;
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gobject_class->get_property = gst_rtp_g726_pay_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FORCE_AAL2,
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g_param_spec_boolean ("force-aal2", "Force AAL2",
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"Force AAL2 encoding for compatibility with bad depayloaders",
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DEFAULT_FORCE_AAL2, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstbasertppayload_class->set_caps = gst_rtp_g726_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_g726_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpg726pay_debug, "rtpg726pay", 0,
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"G.726 RTP Payloader");
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}
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static void
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gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay, GstRtpG726PayClass * klass)
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{
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg726pay);
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GST_BASE_RTP_PAYLOAD (rtpg726pay)->clock_rate = 8000;
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rtpg726pay->force_aal2 = DEFAULT_FORCE_AAL2;
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/* sample based codec */
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gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
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}
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static gboolean
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gst_rtp_g726_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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gchar *encoding_name;
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GstStructure *structure;
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GstBaseRTPAudioPayload *basertpaudiopayload;
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GstRtpG726Pay *pay;
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GstCaps *peercaps;
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gboolean res;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload);
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pay = GST_RTP_G726_PAY (payload);
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "bitrate", &pay->bitrate))
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pay->bitrate = 32000;
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GST_DEBUG_OBJECT (payload, "using bitrate %d", pay->bitrate);
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pay->aal2 = FALSE;
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/* first see what we can do with the bitrate */
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switch (pay->bitrate) {
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case 16000:
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encoding_name = g_strdup ("G726-16");
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gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
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2);
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break;
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case 24000:
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encoding_name = g_strdup ("G726-24");
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gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
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3);
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break;
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case 32000:
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encoding_name = g_strdup ("G726-32");
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gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
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4);
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break;
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case 40000:
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encoding_name = g_strdup ("G726-40");
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gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
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5);
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break;
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default:
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goto invalid_bitrate;
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}
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GST_DEBUG_OBJECT (payload, "selected base encoding %s", encoding_name);
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/* now see if we need to produce AAL2 or not */
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peercaps = gst_pad_peer_get_caps (payload->srcpad);
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if (peercaps) {
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GstCaps *filter, *intersect;
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gchar *capsstr;
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GST_DEBUG_OBJECT (payload, "have peercaps %" GST_PTR_FORMAT, peercaps);
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capsstr = g_strdup_printf ("application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) %s; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) AAL2-%s", encoding_name, encoding_name);
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filter = gst_caps_from_string (capsstr);
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g_free (capsstr);
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/* intersect to filter */
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intersect = gst_caps_intersect (peercaps, filter);
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gst_caps_unref (peercaps);
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GST_DEBUG_OBJECT (payload, "intersected to %" GST_PTR_FORMAT, intersect);
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if (!intersect)
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goto no_format;
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if (gst_caps_is_empty (intersect)) {
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gst_caps_unref (intersect);
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goto no_format;
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}
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structure = gst_caps_get_structure (intersect, 0);
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/* now see what encoding name we settled on, we need to dup because the
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* string goes away when we unref the intersection below. */
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encoding_name =
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g_strdup (gst_structure_get_string (structure, "encoding-name"));
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/* if we managed to negotiate to AAL2, we definatly are going to do AAL2
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* encoding. Else we only encode AAL2 when explicitly set by the
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* property. */
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if (g_str_has_prefix (encoding_name, "AAL2-"))
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pay->aal2 = TRUE;
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else
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pay->aal2 = pay->force_aal2;
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GST_DEBUG_OBJECT (payload, "final encoding %s, AAL2 %d", encoding_name,
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pay->aal2);
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gst_caps_unref (intersect);
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} else {
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/* downstream can do anything but we prefer the better supported non-AAL2 */
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pay->aal2 = pay->force_aal2;
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GST_DEBUG_OBJECT (payload, "no peer caps, AAL2 %d", pay->aal2);
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}
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gst_basertppayload_set_options (payload, "audio", TRUE, encoding_name, 8000);
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res = gst_basertppayload_set_outcaps (payload, NULL);
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g_free (encoding_name);
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return res;
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/* ERRORS */
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invalid_bitrate:
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{
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GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", pay->bitrate);
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return FALSE;
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}
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no_format:
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{
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GST_ERROR_OBJECT (payload, "could not negotiate format");
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_rtp_g726_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
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{
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GstFlowReturn res;
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GstRtpG726Pay *pay;
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pay = GST_RTP_G726_PAY (payload);
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if (!pay->aal2) {
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guint8 *data, tmp;
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guint len;
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/* for non AAL2, we need to reshuffle the bytes, we can do this in-place
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* when the buffer is writable. */
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buffer = gst_buffer_make_writable (buffer);
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data = GST_BUFFER_DATA (buffer);
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len = GST_BUFFER_SIZE (buffer);
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GST_LOG_OBJECT (pay, "packing %u bytes of data", len);
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/* we need to reshuffle the bytes, output is of the form:
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* A B C D .. with the number of bits depending on the bitrate. */
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switch (pay->bitrate) {
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case 16000:
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{
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/* 0
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* 0 1 2 3 4 5 6 7
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* +-+-+-+-+-+-+-+-+-
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* |D D|C C|B B|A A| ...
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* |0 1|0 1|0 1|0 1|
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* +-+-+-+-+-+-+-+-+-
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*/
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while (len > 0) {
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tmp = *data;
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*data++ = ((tmp & 0xc0) >> 6) |
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((tmp & 0x30) >> 2) | ((tmp & 0x0c) << 2) | ((tmp & 0x03) << 6);
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len--;
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}
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break;
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}
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case 24000:
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{
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/* 0 1 2
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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* |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...
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* |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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*/
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while (len > 2) {
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tmp = *data;
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*data++ = ((tmp & 0xc0) >> 6) |
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((tmp & 0x38) >> 1) | ((tmp & 0x07) << 5);
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tmp = *data;
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*data++ = ((tmp & 0x80) >> 7) |
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((tmp & 0x70) >> 3) | ((tmp & 0x0e) << 4) | ((tmp & 0x01) << 7);
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tmp = *data;
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*data++ = ((tmp & 0xe0) >> 5) |
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((tmp & 0x1c) >> 2) | ((tmp & 0x03) << 6);
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len -= 3;
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}
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break;
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}
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case 32000:
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{
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/* 0 1
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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* |B B B B|A A A A|D D D D|C C C C| ...
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* |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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*/
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while (len > 0) {
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tmp = *data;
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*data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
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len--;
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}
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break;
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}
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case 40000:
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{
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/* 0 1 2 3 4
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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* |B B B|A A A A A|D|C C C C C|B B|E E E E|D D D D|G G|F F F F F|E|H H H H H|G G G|
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* |2 3 4|0 1 2 3 4|4|0 1 2 3 4|0 1|1 2 3 4|0 1 2 3|3 4|0 1 2 3 4|0|0 1 2 3 4|0 1 2|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
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*/
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while (len > 4) {
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tmp = *data;
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*data++ = ((tmp & 0xe0) >> 5) | ((tmp & 0x1f) << 3);
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tmp = *data;
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*data++ = ((tmp & 0x80) >> 7) |
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((tmp & 0x7c) >> 2) | ((tmp & 0x03) << 6);
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tmp = *data;
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*data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
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tmp = *data;
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*data++ = ((tmp & 0xc0) >> 6) |
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((tmp & 0x3e) << 2) | ((tmp & 0x01) << 7);
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tmp = *data;
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*data++ = ((tmp & 0xf8) >> 3) | ((tmp & 0x07) << 5);
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len -= 5;
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}
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break;
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}
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}
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}
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res =
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GST_BASE_RTP_PAYLOAD_CLASS (parent_class)->handle_buffer (payload,
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buffer);
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return res;
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}
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static void
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gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpG726Pay *rtpg726pay;
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rtpg726pay = GST_RTP_G726_PAY (object);
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switch (prop_id) {
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case PROP_FORCE_AAL2:
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rtpg726pay->force_aal2 = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpG726Pay *rtpg726pay;
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rtpg726pay = GST_RTP_G726_PAY (object);
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switch (prop_id) {
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case PROP_FORCE_AAL2:
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g_value_set_boolean (value, rtpg726pay->force_aal2);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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gboolean
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gst_rtp_g726_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpg726pay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_G726_PAY);
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}
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