mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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This is required to add RTP header extensions from the meta automatically. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/674>
398 lines
12 KiB
C
398 lines
12 KiB
C
/* GStreamer
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* Copyright (C) <2007> Nokia Corporation
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* Copyright (C) <2007> Collabora Ltd
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* @author: Olivier Crete <olivier.crete@collabora.co.uk>
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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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* This payloader assumes that the data will ALWAYS come as zero or more
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* 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
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* Any other buffer format won't work
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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/rtp/gstrtpbuffer.h>
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#include <gst/base/gstadapter.h>
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#include <gst/audio/audio.h>
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#include "gstrtpg729pay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug);
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#define GST_CAT_DEFAULT (rtpg729pay_debug)
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#define G729_FRAME_SIZE 10
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#define G729B_CN_FRAME_SIZE 2
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#define G729_FRAME_DURATION (10 * GST_MSECOND)
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#define G729_FRAME_DURATION_MS (10)
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static gboolean
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gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps);
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static GstFlowReturn
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gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf);
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static GstStateChangeReturn
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gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition);
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static GstStaticPadTemplate gst_rtp_g729_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/G729, " /* according to RFC 3555 */
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"channels = (int) 1, " "rate = (int) 8000")
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);
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static GstStaticPadTemplate gst_rtp_g729_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_G729_STRING ", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) \"G729\"; "
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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) 8000, " "encoding-name = (string) \"G729\"")
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);
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#define gst_rtp_g729_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRTPG729Pay, gst_rtp_g729_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_g729_pay_finalize (GObject * object)
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{
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GstRTPG729Pay *pay = GST_RTP_G729_PAY (object);
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g_object_unref (pay->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
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"G.729 RTP Payloader");
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gobject_class->finalize = gst_rtp_g729_pay_finalize;
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gstelement_class->change_state = gst_rtp_g729_pay_change_state;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_g729_pay_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_g729_pay_src_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP G.729 payloader", "Codec/Payloader/Network/RTP",
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"Packetize G.729 audio into RTP packets",
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"Olivier Crete <olivier.crete@collabora.co.uk>");
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payload_class->set_caps = gst_rtp_g729_pay_set_caps;
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payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer;
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}
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static void
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gst_rtp_g729_pay_init (GstRTPG729Pay * pay)
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{
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GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
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payload->pt = GST_RTP_PAYLOAD_G729;
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pay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_g729_pay_reset (GstRTPG729Pay * pay)
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{
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gst_adapter_clear (pay->adapter);
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pay->discont = FALSE;
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pay->next_rtp_time = 0;
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pay->first_ts = GST_CLOCK_TIME_NONE;
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pay->first_rtp_time = 0;
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}
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static gboolean
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gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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gboolean res;
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gst_rtp_base_payload_set_options (payload, "audio",
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payload->pt != GST_RTP_PAYLOAD_G729, "G729", 8000);
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res = gst_rtp_base_payload_set_outcaps (payload, NULL);
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return res;
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}
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static GstFlowReturn
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gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay, GstBuffer * buf)
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{
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GstRTPBasePayload *basepayload;
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GstClockTime duration;
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guint frames;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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GstRTPBuffer rtp = { NULL };
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guint payload_len = gst_buffer_get_size (buf);
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basepayload = GST_RTP_BASE_PAYLOAD (rtpg729pay);
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GST_DEBUG_OBJECT (rtpg729pay, "Pushing %d bytes ts %" GST_TIME_FORMAT,
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payload_len, GST_TIME_ARGS (rtpg729pay->next_ts));
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/* create buffer to hold the payload */
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outbuf =
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gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
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(rtpg729pay), 0, 0, 0);
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gst_rtp_buffer_map (outbuf, GST_MAP_READWRITE, &rtp);
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/* set metadata */
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frames =
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(payload_len / G729_FRAME_SIZE) + ((payload_len % G729_FRAME_SIZE) >> 1);
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duration = frames * G729_FRAME_DURATION;
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GST_BUFFER_PTS (outbuf) = rtpg729pay->next_ts;
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GST_BUFFER_DURATION (outbuf) = duration;
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GST_BUFFER_OFFSET (outbuf) = rtpg729pay->next_rtp_time;
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rtpg729pay->next_ts += duration;
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rtpg729pay->next_rtp_time += frames * 80;
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if (G_UNLIKELY (rtpg729pay->discont)) {
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GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
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gst_rtp_buffer_set_marker (&rtp, TRUE);
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rtpg729pay->discont = FALSE;
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}
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gst_rtp_buffer_unmap (&rtp);
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/* append payload */
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gst_rtp_copy_audio_meta (basepayload, outbuf, buf);
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outbuf = gst_buffer_append (outbuf, buf);
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ret = gst_rtp_base_payload_push (basepayload, outbuf);
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return ret;
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}
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static void
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gst_rtp_g729_pay_recalc_rtp_time (GstRTPG729Pay * rtpg729pay, GstClockTime time)
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{
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if (GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts)
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&& GST_CLOCK_TIME_IS_VALID (time) && time >= rtpg729pay->first_ts) {
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GstClockTime diff;
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guint32 rtpdiff;
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diff = time - rtpg729pay->first_ts;
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rtpdiff = (diff / GST_MSECOND) * 8;
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rtpg729pay->next_rtp_time = rtpg729pay->first_rtp_time + rtpdiff;
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GST_DEBUG_OBJECT (rtpg729pay,
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"elapsed time %" GST_TIME_FORMAT ", rtp %" G_GUINT32_FORMAT ", "
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"new offset %" G_GUINT32_FORMAT, GST_TIME_ARGS (diff), rtpdiff,
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rtpg729pay->next_rtp_time);
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}
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}
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static GstFlowReturn
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gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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GstRTPG729Pay *rtpg729pay = GST_RTP_G729_PAY (payload);
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GstAdapter *adapter = NULL;
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guint payload_len;
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guint available;
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guint maxptime_octets = G_MAXUINT;
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guint minptime_octets = 0;
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guint min_payload_len;
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guint max_payload_len;
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gsize size;
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GstClockTime timestamp;
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size = gst_buffer_get_size (buf);
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if (size % G729_FRAME_SIZE != 0 &&
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size % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE)
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goto invalid_size;
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/* max number of bytes based on given ptime, has to be multiple of
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* frame_duration */
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if (payload->max_ptime != -1) {
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guint ptime_ms = payload->max_ptime / GST_MSECOND;
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maxptime_octets = G729_FRAME_SIZE *
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(int) (ptime_ms / G729_FRAME_DURATION_MS);
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if (maxptime_octets < G729_FRAME_SIZE) {
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GST_WARNING_OBJECT (payload, "Given ptime %" G_GINT64_FORMAT
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" is smaller than minimum %d ns, overwriting to minimum",
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payload->max_ptime, G729_FRAME_DURATION_MS);
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maxptime_octets = G729_FRAME_SIZE;
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}
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}
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max_payload_len = MIN (
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/* MTU max */
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(int) (gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU
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(payload), 0, 0) / G729_FRAME_SIZE)
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* G729_FRAME_SIZE,
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/* ptime max */
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maxptime_octets);
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/* min number of bytes based on a given ptime, has to be a multiple
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of frame duration */
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{
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guint64 min_ptime = payload->min_ptime;
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min_ptime = min_ptime / GST_MSECOND;
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minptime_octets = G729_FRAME_SIZE *
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(int) (min_ptime / G729_FRAME_DURATION_MS);
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}
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min_payload_len = MAX (minptime_octets, G729_FRAME_SIZE);
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if (min_payload_len > max_payload_len) {
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min_payload_len = max_payload_len;
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}
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/* If the ptime is specified in the caps, tried to adhere to it exactly */
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if (payload->ptime) {
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guint64 ptime = payload->ptime / GST_MSECOND;
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guint ptime_in_bytes = G729_FRAME_SIZE *
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(guint) (ptime / G729_FRAME_DURATION_MS);
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/* clip to computed min and max lengths */
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ptime_in_bytes = MAX (min_payload_len, ptime_in_bytes);
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ptime_in_bytes = MIN (max_payload_len, ptime_in_bytes);
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min_payload_len = max_payload_len = ptime_in_bytes;
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}
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GST_LOG_OBJECT (payload,
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"Calculated min_payload_len %u and max_payload_len %u",
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min_payload_len, max_payload_len);
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adapter = rtpg729pay->adapter;
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available = gst_adapter_available (adapter);
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timestamp = GST_BUFFER_PTS (buf);
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/* resync rtp time on discont or a discontinuous cn packet */
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if (GST_BUFFER_IS_DISCONT (buf)) {
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/* flush remainder */
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if (available > 0) {
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gst_rtp_g729_pay_push (rtpg729pay,
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gst_adapter_take_buffer_fast (adapter, available));
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available = 0;
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}
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rtpg729pay->discont = TRUE;
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gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
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}
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if (size < G729_FRAME_SIZE)
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gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
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if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts))) {
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rtpg729pay->first_ts = timestamp;
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rtpg729pay->first_rtp_time = rtpg729pay->next_rtp_time;
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}
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/* let's reset the base timestamp when the adapter is empty */
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if (available == 0)
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rtpg729pay->next_ts = timestamp;
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if (available == 0 && size >= min_payload_len && size <= max_payload_len) {
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ret = gst_rtp_g729_pay_push (rtpg729pay, buf);
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return ret;
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}
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gst_adapter_push (adapter, buf);
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available = gst_adapter_available (adapter);
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/* as long as we have full frames */
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/* this loop will push all available buffers till the last frame */
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while (available >= min_payload_len ||
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available % G729_FRAME_SIZE == G729B_CN_FRAME_SIZE) {
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/* We send as much as we can */
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if (available <= max_payload_len) {
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payload_len = available;
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} else {
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payload_len = MIN (max_payload_len,
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(available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
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}
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ret = gst_rtp_g729_pay_push (rtpg729pay,
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gst_adapter_take_buffer_fast (adapter, payload_len));
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available -= payload_len;
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}
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return ret;
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/* ERRORS */
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invalid_size:
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{
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GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
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("Invalid input buffer size"),
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("Invalid buffer size, should be a multiple of"
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" G729_FRAME_SIZE(10) with an optional G729B_CN_FRAME_SIZE(2)"
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" added to it, but it is %" G_GSIZE_FORMAT, size));
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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}
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}
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static GstStateChangeReturn
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gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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/* handle upwards state changes here */
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switch (transition) {
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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/* handle downwards state changes */
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_rtp_g729_pay_reset (GST_RTP_G729_PAY (element));
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break;
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default:
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break;
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}
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return ret;
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}
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gboolean
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gst_rtp_g729_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpg729pay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_G729_PAY);
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}
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