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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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446 lines
13 KiB
C
446 lines
13 KiB
C
/* GStreamer
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* Copyright (C) <2007> Nokia Corporation (contact <stefan.kost@nokia.com>)
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* <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 version 2 as published by the Free Software Foundation.
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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 <gst/base/gstbitreader.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <string.h>
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#include "gstrtpmp4adepay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug);
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#define GST_CAT_DEFAULT (rtpmp4adepay_debug)
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static GstStaticPadTemplate gst_rtp_mp4a_depay_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 ("audio/mpeg,"
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"mpegversion = (int) 4," "framed = (boolean) true, "
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"stream-format = (string) raw")
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);
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static GstStaticPadTemplate gst_rtp_mp4a_depay_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 ("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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* "profile-level-id=[1,MAX]"
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* "config="
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*/
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)
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);
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#define gst_rtp_mp4a_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpMP4ADepay, gst_rtp_mp4a_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static void gst_rtp_mp4a_depay_finalize (GObject * object);
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static gboolean gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload,
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GstBuffer * buf);
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static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement *
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element, GstStateChange transition);
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static void
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gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gobject_class->finalize = gst_rtp_mp4a_depay_finalize;
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gstelement_class->change_state = gst_rtp_mp4a_depay_change_state;
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gstrtpbasedepayload_class->process = gst_rtp_mp4a_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_mp4a_depay_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_mp4a_depay_sink_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP MPEG4 audio depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts MPEG4 audio from RTP packets (RFC 3016)",
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"Nokia Corporation (contact <stefan.kost@nokia.com>), "
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"Wim Taymans <wim.taymans@gmail.com>");
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GST_DEBUG_CATEGORY_INIT (rtpmp4adepay_debug, "rtpmp4adepay", 0,
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"MPEG4 audio RTP Depayloader");
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}
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static void
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gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay)
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{
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rtpmp4adepay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_mp4a_depay_finalize (GObject * object)
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{
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GstRtpMP4ADepay *rtpmp4adepay;
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rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
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g_object_unref (rtpmp4adepay->adapter);
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rtpmp4adepay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000,
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44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350
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};
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static gboolean
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gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstRtpMP4ADepay *rtpmp4adepay;
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GstCaps *srccaps;
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const gchar *str;
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gint clock_rate;
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gint object_type;
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gint channels = 2; /* default */
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gboolean res;
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rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = 90000; /* default */
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depayload->clock_rate = clock_rate;
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if (!gst_structure_get_int (structure, "object", &object_type))
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object_type = 2; /* AAC LC default */
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srccaps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, 4,
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"framed", G_TYPE_BOOLEAN, TRUE, "channels", G_TYPE_INT, channels,
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"stream-format", G_TYPE_STRING, "raw", NULL);
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if ((str = gst_structure_get_string (structure, "config"))) {
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GValue v = { 0 };
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g_value_init (&v, GST_TYPE_BUFFER);
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if (gst_value_deserialize (&v, str)) {
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GstBuffer *buffer;
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GstMapInfo map;
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guint8 *data;
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gsize size;
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gint i;
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guint32 rate = 0;
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guint8 obj_type = 0, sr_idx = 0, channels = 0;
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GstBitReader br;
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buffer = gst_value_get_buffer (&v);
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gst_buffer_ref (buffer);
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g_value_unset (&v);
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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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GST_WARNING_OBJECT (depayload, "config too short (%d < 2)",
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(gint) size);
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goto bad_config;
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}
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/* Parse 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 == rtpmp4adepay->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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* We only require audioMuxVersion == 0;
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*
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* The remaining bit of the second byte and the rest of the bits are used
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* for audioSpecificConfig which we need to set in codec_info.
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*/
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if ((data[0] & 0x80) != 0x00) {
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GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1");
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goto bad_config;
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}
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rtpmp4adepay->numSubFrames = (data[0] & 0x3F);
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GST_LOG_OBJECT (rtpmp4adepay, "numSubFrames %d",
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rtpmp4adepay->numSubFrames);
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/* shift rest of string 15 bits down */
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size -= 2;
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for (i = 0; i < size; i++) {
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data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
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}
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gst_bit_reader_init (&br, data, size);
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/* any object type is fine, we need to copy it to the profile-level-id field. */
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if (!gst_bit_reader_get_bits_uint8 (&br, &obj_type, 5))
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goto bad_config;
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if (obj_type == 0) {
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GST_WARNING_OBJECT (depayload, "invalid object type 0");
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goto bad_config;
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}
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if (!gst_bit_reader_get_bits_uint8 (&br, &sr_idx, 4))
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goto bad_config;
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if (sr_idx >= G_N_ELEMENTS (aac_sample_rates) && sr_idx != 15) {
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GST_WARNING_OBJECT (depayload, "invalid sample rate index %d", sr_idx);
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goto bad_config;
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}
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GST_LOG_OBJECT (rtpmp4adepay, "sample rate index %u", sr_idx);
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if (!gst_bit_reader_get_bits_uint8 (&br, &channels, 4))
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goto bad_config;
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if (channels > 7) {
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GST_WARNING_OBJECT (depayload, "invalid channels %u", (guint) channels);
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goto bad_config;
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}
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/* rtp rate depends on sampling rate of the audio */
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if (sr_idx == 15) {
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/* index of 15 means we get the rate in the next 24 bits */
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if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
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goto bad_config;
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} else if (sr_idx >= G_N_ELEMENTS (aac_sample_rates)) {
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goto bad_config;
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} else {
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/* else use the rate from the table */
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rate = aac_sample_rates[sr_idx];
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}
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rtpmp4adepay->frame_len = 1024;
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switch (obj_type) {
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case 1:
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case 2:
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case 3:
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case 4:
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case 6:
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case 7:
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{
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guint8 frameLenFlag = 0;
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if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
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if (frameLenFlag)
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rtpmp4adepay->frame_len = 960;
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break;
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}
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default:
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break;
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}
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/* ignore remaining bit, we're only interested in full bytes */
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gst_buffer_resize (buffer, 0, size);
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gst_buffer_unmap (buffer, &map);
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data = NULL;
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gst_caps_set_simple (srccaps,
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"channels", G_TYPE_INT, (gint) channels,
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"rate", G_TYPE_INT, (gint) rate,
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"codec_data", GST_TYPE_BUFFER, buffer, NULL);
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bad_config:
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if (data)
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gst_buffer_unmap (buffer, &map);
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gst_buffer_unref (buffer);
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} else {
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g_warning ("cannot convert config to buffer");
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}
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}
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res = gst_pad_set_caps (depayload->srcpad, srccaps);
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gst_caps_unref (srccaps);
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return res;
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}
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static GstBuffer *
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gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
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{
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GstRtpMP4ADepay *rtpmp4adepay;
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GstBuffer *outbuf;
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GstRTPBuffer rtp = { NULL };
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GstMapInfo map;
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rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
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/* flush remaining data on discont */
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if (GST_BUFFER_IS_DISCONT (buf)) {
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gst_adapter_clear (rtpmp4adepay->adapter);
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}
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gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
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outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
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outbuf = gst_buffer_make_writable (outbuf);
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GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
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gst_adapter_push (rtpmp4adepay->adapter, outbuf);
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/* RTP marker bit indicates the last packet of the AudioMuxElement => create
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* and push a buffer */
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if (gst_rtp_buffer_get_marker (&rtp)) {
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guint avail;
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guint i;
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guint8 *data;
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guint pos;
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GstClockTime timestamp;
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avail = gst_adapter_available (rtpmp4adepay->adapter);
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timestamp = gst_adapter_prev_pts (rtpmp4adepay->adapter, NULL);
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GST_LOG_OBJECT (rtpmp4adepay, "have marker and %u available", avail);
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outbuf = gst_adapter_take_buffer (rtpmp4adepay->adapter, avail);
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gst_buffer_map (outbuf, &map, GST_MAP_READ);
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data = map.data;
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/* position in data we are at */
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pos = 0;
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/* looping through the number of sub-frames in the audio payload */
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for (i = 0; i <= rtpmp4adepay->numSubFrames; i++) {
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/* determine payload length and set buffer data pointer accordingly */
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guint skip;
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guint data_len;
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GstBuffer *tmp = NULL;
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/* each subframe starts with a variable length encoding */
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data_len = 0;
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for (skip = 0; skip < avail; skip++) {
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data_len += data[skip];
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if (data[skip] != 0xff)
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break;
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}
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skip++;
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/* this can not be possible, we have not enough data or the length
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* decoding failed because we ran out of data. */
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if (skip + data_len > avail)
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goto wrong_size;
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GST_LOG_OBJECT (rtpmp4adepay,
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"subframe %u, header len %u, data len %u, left %u", i, skip, data_len,
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avail);
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/* take data out, skip the header */
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pos += skip;
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tmp = gst_buffer_copy_region (outbuf, GST_BUFFER_COPY_MEMORY, pos,
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data_len);
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/* skip data too */
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skip += data_len;
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pos += data_len;
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/* update our pointers whith what we consumed */
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data += skip;
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avail -= skip;
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GST_BUFFER_TIMESTAMP (tmp) = timestamp;
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gst_rtp_base_depayload_push (depayload, tmp);
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/* shift ts for next buffers */
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if (rtpmp4adepay->frame_len && timestamp != -1
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&& depayload->clock_rate != 0) {
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timestamp +=
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gst_util_uint64_scale_int (rtpmp4adepay->frame_len, GST_SECOND,
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depayload->clock_rate);
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}
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}
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/* just a check that lengths match */
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if (avail) {
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GST_ELEMENT_WARNING (depayload, STREAM, DECODE,
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("Packet invalid"), ("Not all payload consumed: "
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"possible wrongly encoded packet."));
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}
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gst_buffer_unmap (outbuf, &map);
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gst_buffer_unref (outbuf);
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}
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gst_rtp_buffer_unmap (&rtp);
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return NULL;
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/* ERRORS */
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wrong_size:
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{
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GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
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("Packet did not validate"), ("wrong packet size"));
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gst_buffer_unmap (outbuf, &map);
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gst_buffer_unref (outbuf);
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gst_rtp_buffer_unmap (&rtp);
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return NULL;
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}
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}
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static GstStateChangeReturn
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gst_rtp_mp4a_depay_change_state (GstElement * element,
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GstStateChange transition)
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{
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GstRtpMP4ADepay *rtpmp4adepay;
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GstStateChangeReturn ret;
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rtpmp4adepay = GST_RTP_MP4A_DEPAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_adapter_clear (rtpmp4adepay->adapter);
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rtpmp4adepay->frame_len = 0;
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rtpmp4adepay->numSubFrames = 0;
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break;
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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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switch (transition) {
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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_mp4a_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpmp4adepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_DEPAY);
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}
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