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263 lines
8.2 KiB
C
263 lines
8.2 KiB
C
/* GStreamer
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* Copyright (C) <2007> 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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/**
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* SECTION:element-rtpL16pay
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* @title: rtpL16pay
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* @see_also: rtpL16depay
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*
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* Payload raw audio into RTP packets according to RFC 3551.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
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*
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* ## Example pipeline
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* |[
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* gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink
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* ]| This example pipeline will payload raw audio. Refer to
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* the rtpL16depay example to depayload and play the RTP stream.
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*
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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 "gstrtpL16pay.h"
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#include "gstrtpchannels.h"
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GST_DEBUG_CATEGORY_STATIC (rtpL16pay_debug);
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#define GST_CAT_DEFAULT (rtpL16pay_debug)
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static GstStaticPadTemplate gst_rtp_L16_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-raw, "
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"format = (string) S16BE, "
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"layout = (string) interleaved, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
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);
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static GstStaticPadTemplate gst_rtp_L16_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) [ 96, 127 ], "
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"clock-rate = (int) [ 1, MAX ], "
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"encoding-name = (string) \"L16\", "
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"channels = (int) [ 1, MAX ];"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"encoding-name = (string) \"L16\", "
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"payload = (int) " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
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"clock-rate = (int) 44100;"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"encoding-name = (string) \"L16\", "
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"payload = (int) " GST_RTP_PAYLOAD_L16_MONO_STRING ", "
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"clock-rate = (int) 44100")
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);
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static gboolean gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload,
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GstCaps * caps);
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static GstCaps *gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload,
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GstPad * pad, GstCaps * filter);
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static GstFlowReturn
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gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer);
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#define gst_rtp_L16_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpL16Pay, gst_rtp_L16_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
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static void
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gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gstrtpbasepayload_class->set_caps = gst_rtp_L16_pay_setcaps;
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gstrtpbasepayload_class->get_caps = gst_rtp_L16_pay_getcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_L16_pay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_L16_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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GST_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0,
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"L16 RTP Payloader");
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}
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static void
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gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay)
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{
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GstRTPBaseAudioPayload *rtpbaseaudiopayload;
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rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL16pay);
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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_L16_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
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{
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GstRtpL16Pay *rtpL16pay;
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gboolean res;
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gchar *params;
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GstAudioInfo *info;
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const GstRTPChannelOrder *order;
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GstRTPBaseAudioPayload *rtpbaseaudiopayload;
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rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
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rtpL16pay = GST_RTP_L16_PAY (basepayload);
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info = &rtpL16pay->info;
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gst_audio_info_init (info);
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if (!gst_audio_info_from_caps (info, caps))
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goto invalid_caps;
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order = gst_rtp_channels_get_by_pos (info->channels, info->position);
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rtpL16pay->order = order;
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gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L16",
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info->rate);
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params = g_strdup_printf ("%d", info->channels);
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if (!order && info->channels > 2) {
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GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE,
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(NULL), ("Unknown channel order for %d channels", info->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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info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
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} else {
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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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info->channels, NULL);
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}
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g_free (params);
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/* octet-per-sample is 2 * channels for L16 */
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gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
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2 * info->channels);
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return res;
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/* ERRORS */
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invalid_caps:
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{
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GST_DEBUG_OBJECT (rtpL16pay, "invalid caps");
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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_L16_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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caps = gst_pad_get_pad_template_caps (pad);
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otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
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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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gint channels;
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gint pt;
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gint rate;
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structure = gst_caps_get_structure (otherpadcaps, 0);
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caps = gst_caps_make_writable (caps);
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if (gst_structure_get_int (structure, "channels", &channels)) {
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
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} else if (gst_structure_get_int (structure, "payload", &pt)) {
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if (pt == GST_RTP_PAYLOAD_L16_STEREO)
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
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else if (pt == GST_RTP_PAYLOAD_L16_MONO)
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
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}
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if (gst_structure_get_int (structure, "clock-rate", &rate)) {
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gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
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} else if (gst_structure_get_int (structure, "payload", &pt)) {
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if (pt == GST_RTP_PAYLOAD_L16_STEREO || pt == GST_RTP_PAYLOAD_L16_MONO)
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gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL);
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}
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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 *tcaps = caps;
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caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (tcaps);
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}
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return caps;
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}
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static GstFlowReturn
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gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpL16Pay *rtpL16pay;
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rtpL16pay = GST_RTP_L16_PAY (basepayload);
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buffer = gst_buffer_make_writable (buffer);
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if (rtpL16pay->order &&
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!gst_audio_buffer_reorder_channels (buffer, rtpL16pay->info.finfo->format,
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rtpL16pay->info.channels, rtpL16pay->info.position,
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rtpL16pay->order->pos)) {
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return GST_FLOW_ERROR;
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}
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return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
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buffer);
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}
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gboolean
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gst_rtp_L16_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpL16pay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_L16_PAY);
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}
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