mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
Do not allocate payload size outbuf if appending payload buffer.
The commit 137672ff18
attached payload
to the output buffer but forgot to remove payload allocation. That
effectively doubled payload size and add zero'ed or random bytes.
Makes the following pipeline work again:
gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay ! rtpgsmdepay ! gsmdec ! autoaudiosink
https://bugzilla.gnome.org/show_bug.cgi?id=784616
181 lines
5.3 KiB
C
181 lines
5.3 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2005> Zeeshan Ali <zeenix@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 <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include "gstrtpgsmpay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpgsmpay_debug);
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#define GST_CAT_DEFAULT (rtpgsmpay_debug)
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static GstStaticPadTemplate gst_rtp_gsm_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-gsm, " "rate = (int) 8000, " "channels = (int) 1")
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);
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static GstStaticPadTemplate gst_rtp_gsm_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_GSM_STRING ", "
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"clock-rate = (int) 8000, " "encoding-name = (string) \"GSM\"; "
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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) \"GSM\"")
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);
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static gboolean gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_gsm_pay_handle_buffer (GstRTPBasePayload * payload,
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GstBuffer * buffer);
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#define gst_rtp_gsm_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRTPGSMPay, gst_rtp_gsm_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_gsm_pay_class_init (GstRTPGSMPayClass * 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 (rtpgsmpay_debug, "rtpgsmpay", 0,
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"GSM Audio 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_gsm_pay_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_gsm_pay_src_template);
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gst_element_class_set_static_metadata (gstelement_class, "RTP GSM payloader",
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"Codec/Payloader/Network/RTP",
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"Payload-encodes GSM audio into a RTP packet",
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"Zeeshan Ali <zeenix@gmail.com>");
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gstrtpbasepayload_class->set_caps = gst_rtp_gsm_pay_setcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_gsm_pay_handle_buffer;
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}
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static void
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gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay)
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{
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GST_RTP_BASE_PAYLOAD (rtpgsmpay)->clock_rate = 8000;
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GST_RTP_BASE_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM;
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}
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static gboolean
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gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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const char *stname;
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GstStructure *structure;
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gboolean res;
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structure = gst_caps_get_structure (caps, 0);
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stname = gst_structure_get_name (structure);
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if (strcmp ("audio/x-gsm", stname))
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goto invalid_type;
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gst_rtp_base_payload_set_options (payload, "audio",
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payload->pt != GST_RTP_PAYLOAD_GSM, "GSM", 8000);
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res = gst_rtp_base_payload_set_outcaps (payload, NULL);
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return res;
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/* ERRORS */
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invalid_type:
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{
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GST_WARNING_OBJECT (payload, "invalid media type received");
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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_gsm_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstRTPGSMPay *rtpgsmpay;
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guint payload_len;
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GstBuffer *outbuf;
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GstClockTime timestamp, duration;
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GstFlowReturn ret;
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rtpgsmpay = GST_RTP_GSM_PAY (basepayload);
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timestamp = GST_BUFFER_PTS (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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/* FIXME, only one GSM frame per RTP packet for now */
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payload_len = gst_buffer_get_size (buffer);
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/* FIXME, just error out for now */
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if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))
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goto too_big;
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outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
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/* copy timestamp and duration */
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GST_BUFFER_PTS (outbuf) = timestamp;
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GST_BUFFER_DURATION (outbuf) = duration;
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gst_rtp_copy_audio_meta (rtpgsmpay, outbuf, buffer);
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/* append payload */
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outbuf = gst_buffer_append (outbuf, buffer);
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GST_DEBUG ("gst_rtp_gsm_pay_chain: pushing buffer of size %" G_GSIZE_FORMAT,
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gst_buffer_get_size (outbuf));
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ret = gst_rtp_base_payload_push (basepayload, outbuf);
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return ret;
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/* ERRORS */
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too_big:
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{
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GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL),
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("payload_len %u > mtu %u", payload_len,
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GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay)));
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return GST_FLOW_ERROR;
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}
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}
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gboolean
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gst_rtp_gsm_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpgsmpay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_PAY);
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}
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