mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 13:48:55 +00:00
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>
465 lines
13 KiB
C
465 lines
13 KiB
C
/* GStreamer
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* Copyright (C) <2008> 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/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include "gstrtpmp4apay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpmp4apay_debug);
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#define GST_CAT_DEFAULT (rtpmp4apay_debug)
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/* FIXME: add framed=(boolean)true once our encoders have this field set
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* on their output caps */
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static GstStaticPadTemplate gst_rtp_mp4a_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/mpeg, mpegversion=(int)4, "
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"stream-format=(string)raw")
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);
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static GstStaticPadTemplate gst_rtp_mp4a_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) [1, MAX ], "
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"encoding-name = (string) \"MP4A-LATM\""
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/* All optional parameters
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*
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* "cpresent = (string) \"0\""
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* "config="
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*/
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)
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);
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static void gst_rtp_mp4a_pay_finalize (GObject * object);
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static gboolean gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload *
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payload, GstBuffer * buffer);
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#define gst_rtp_mp4a_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpMP4APay, gst_rtp_mp4a_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gobject_class->finalize = gst_rtp_mp4a_pay_finalize;
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gstrtpbasepayload_class->set_caps = gst_rtp_mp4a_pay_setcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_mp4a_pay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_mp4a_pay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP MPEG4 audio payloader", "Codec/Payloader/Network/RTP",
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"Payload MPEG4 audio as RTP packets (RFC 3016)",
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"Wim Taymans <wim.taymans@gmail.com>");
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GST_DEBUG_CATEGORY_INIT (rtpmp4apay_debug, "rtpmp4apay", 0,
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"MP4A-LATM RTP Payloader");
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}
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static void
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gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay)
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{
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rtpmp4apay->rate = 90000;
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rtpmp4apay->profile = g_strdup ("1");
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}
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static void
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gst_rtp_mp4a_pay_finalize (GObject * object)
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{
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GstRtpMP4APay *rtpmp4apay;
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rtpmp4apay = GST_RTP_MP4A_PAY (object);
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g_free (rtpmp4apay->params);
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rtpmp4apay->params = NULL;
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if (rtpmp4apay->config)
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gst_buffer_unref (rtpmp4apay->config);
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rtpmp4apay->config = NULL;
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g_free (rtpmp4apay->profile);
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rtpmp4apay->profile = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static const unsigned int sampling_table[16] = {
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96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
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16000, 12000, 11025, 8000, 7350, 0, 0, 0
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};
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static gboolean
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gst_rtp_mp4a_pay_parse_audio_config (GstRtpMP4APay * rtpmp4apay,
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GstBuffer * buffer)
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{
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GstMapInfo map;
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guint8 *data;
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gsize size;
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guint8 objectType;
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guint8 samplingIdx;
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guint8 channelCfg;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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data = map.data;
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size = map.size;
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if (size < 2)
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goto too_short;
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/* any object type is fine, we need to copy it to the profile-level-id field. */
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objectType = (data[0] & 0xf8) >> 3;
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if (objectType == 0)
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goto invalid_object;
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samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
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/* only fixed values for now */
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if (samplingIdx > 12 && samplingIdx != 15)
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goto wrong_freq;
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channelCfg = ((data[1] & 0x78) >> 3);
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if (channelCfg > 7)
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goto wrong_channels;
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/* rtp rate depends on sampling rate of the audio */
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if (samplingIdx == 15) {
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if (size < 5)
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goto too_short;
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/* index of 15 means we get the rate in the next 24 bits */
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rtpmp4apay->rate = ((data[1] & 0x7f) << 17) |
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((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
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} else {
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/* else use the rate from the table */
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rtpmp4apay->rate = sampling_table[samplingIdx];
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}
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/* extra rtp params contain the number of channels */
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g_free (rtpmp4apay->params);
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rtpmp4apay->params = g_strdup_printf ("%d", channelCfg);
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/* audio stream type */
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rtpmp4apay->streamtype = "5";
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/* profile */
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g_free (rtpmp4apay->profile);
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rtpmp4apay->profile = g_strdup_printf ("%d", objectType);
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GST_DEBUG_OBJECT (rtpmp4apay,
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"objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
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samplingIdx, rtpmp4apay->rate, channelCfg);
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gst_buffer_unmap (buffer, &map);
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return TRUE;
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/* ERROR */
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too_short:
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{
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GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
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(NULL),
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("config string too short, expected 2 bytes, got %" G_GSIZE_FORMAT,
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size));
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gst_buffer_unmap (buffer, &map);
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return FALSE;
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}
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invalid_object:
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{
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GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
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(NULL), ("invalid object type 0"));
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gst_buffer_unmap (buffer, &map);
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return FALSE;
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}
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wrong_freq:
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{
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GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
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(NULL), ("unsupported frequency index %d", samplingIdx));
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gst_buffer_unmap (buffer, &map);
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return FALSE;
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}
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wrong_channels:
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{
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GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
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(NULL), ("unsupported number of channels %d, must < 8", channelCfg));
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gst_buffer_unmap (buffer, &map);
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return FALSE;
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}
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}
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static gboolean
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gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
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{
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gchar *config;
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GValue v = { 0 };
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gboolean res;
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g_value_init (&v, GST_TYPE_BUFFER);
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gst_value_set_buffer (&v, rtpmp4apay->config);
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config = gst_value_serialize (&v);
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res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4apay),
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"cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
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g_value_unset (&v);
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g_free (config);
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return res;
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}
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static gboolean
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gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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GstRtpMP4APay *rtpmp4apay;
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GstStructure *structure;
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const GValue *codec_data;
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gboolean res, framed = TRUE;
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const gchar *stream_format;
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rtpmp4apay = GST_RTP_MP4A_PAY (payload);
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structure = gst_caps_get_structure (caps, 0);
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/* this is already handled by the template caps, but it is better
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* to leave here to have meaningful warning messages when linking
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* fails */
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stream_format = gst_structure_get_string (structure, "stream-format");
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if (stream_format) {
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if (strcmp (stream_format, "raw") != 0) {
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GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format must be 'raw', "
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"%s is not supported", stream_format);
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return FALSE;
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}
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} else {
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GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format not specified, "
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"assuming 'raw'");
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}
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codec_data = gst_structure_get_value (structure, "codec_data");
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if (codec_data) {
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GST_LOG_OBJECT (rtpmp4apay, "got codec_data");
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if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
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GstBuffer *buffer, *cbuffer;
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GstMapInfo map;
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GstMapInfo cmap;
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guint i;
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buffer = gst_value_get_buffer (codec_data);
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GST_LOG_OBJECT (rtpmp4apay, "configuring codec_data");
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/* parse buffer */
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res = gst_rtp_mp4a_pay_parse_audio_config (rtpmp4apay, buffer);
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if (!res)
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goto config_failed;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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/* make the StreamMuxConfig, we need 15 bits for the header */
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cbuffer = gst_buffer_new_and_alloc (map.size + 2);
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gst_buffer_map (cbuffer, &cmap, GST_MAP_WRITE);
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memset (cmap.data, 0, map.size + 2);
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/* Create StreamMuxConfig according to ISO/IEC 14496-3:
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*
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* audioMuxVersion == 0 (1 bit)
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* allStreamsSameTimeFraming == 1 (1 bit)
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* numSubFrames == numSubFrames (6 bits)
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* numProgram == 0 (4 bits)
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* numLayer == 0 (3 bits)
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*/
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cmap.data[0] = 0x40;
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cmap.data[1] = 0x00;
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/* append the config bits, shifting them 1 bit left */
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for (i = 0; i < map.size; i++) {
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cmap.data[i + 1] |= ((map.data[i] & 0x80) >> 7);
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cmap.data[i + 2] |= ((map.data[i] & 0x7f) << 1);
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}
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gst_buffer_unmap (cbuffer, &cmap);
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gst_buffer_unmap (buffer, &map);
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/* now we can configure the buffer */
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if (rtpmp4apay->config)
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gst_buffer_unref (rtpmp4apay->config);
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rtpmp4apay->config = cbuffer;
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}
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}
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if (gst_structure_get_boolean (structure, "framed", &framed) && !framed) {
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GST_WARNING_OBJECT (payload, "Need framed AAC data as input!");
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}
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gst_rtp_base_payload_set_options (payload, "audio", TRUE, "MP4A-LATM",
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rtpmp4apay->rate);
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res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
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return res;
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/* ERRORS */
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config_failed:
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{
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GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config");
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return FALSE;
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}
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}
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#define RTP_HEADER_LEN 12
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/* we expect buffers as exactly one complete AU
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*/
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static GstFlowReturn
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gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpMP4APay *rtpmp4apay;
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GstFlowReturn ret;
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GstBufferList *list;
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guint mtu;
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guint offset;
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gsize size;
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gboolean fragmented;
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GstClockTime timestamp;
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ret = GST_FLOW_OK;
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rtpmp4apay = GST_RTP_MP4A_PAY (basepayload);
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offset = 0;
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size = gst_buffer_get_size (buffer);
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timestamp = GST_BUFFER_PTS (buffer);
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fragmented = FALSE;
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mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4apay);
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list = gst_buffer_list_new_sized (size / (mtu - RTP_HEADER_LEN) + 1);
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while (size > 0) {
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guint towrite;
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GstBuffer *outbuf;
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guint payload_len;
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guint packet_len;
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guint header_len;
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GstBuffer *paybuf;
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GstRTPBuffer rtp = { NULL };
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header_len = 0;
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if (!fragmented) {
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guint count;
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/* first packet calculate space for the packet including the header */
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count = size;
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while (count >= 0xff) {
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header_len++;
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count -= 0xff;
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}
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header_len++;
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}
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packet_len = gst_rtp_buffer_calc_packet_len (header_len + size, 0, 0);
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towrite = MIN (packet_len, mtu);
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payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
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payload_len -= header_len;
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GST_DEBUG_OBJECT (rtpmp4apay,
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"avail %" G_GSIZE_FORMAT
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", header_len %d, packet_len %d, payload_len %d", size, header_len,
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packet_len, payload_len);
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/* create buffer to hold the payload. */
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outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload,
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header_len, 0, 0);
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/* copy payload */
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gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
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if (!fragmented) {
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guint8 *payload = gst_rtp_buffer_get_payload (&rtp);
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guint count;
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/* first packet write the header */
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count = size;
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while (count >= 0xff) {
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*payload++ = 0xff;
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count -= 0xff;
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}
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*payload++ = count;
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}
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/* marker only if the packet is complete */
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gst_rtp_buffer_set_marker (&rtp, size == payload_len);
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gst_rtp_buffer_unmap (&rtp);
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/* create a new buf to hold the payload */
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paybuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL,
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offset, payload_len);
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/* join memory parts */
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gst_rtp_copy_audio_meta (rtpmp4apay, outbuf, paybuf);
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outbuf = gst_buffer_append (outbuf, paybuf);
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gst_buffer_list_add (list, outbuf);
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offset += payload_len;
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size -= payload_len;
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/* copy incoming timestamp (if any) to outgoing buffers */
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GST_BUFFER_PTS (outbuf) = timestamp;
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fragmented = TRUE;
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}
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ret =
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gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmp4apay), list);
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gst_buffer_unref (buffer);
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return ret;
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}
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gboolean
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gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpmp4apay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_PAY);
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}
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