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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Remove the pt restrictions for all the depayloaders that have an encoding-name. We can use this to autoplug decoders. Remove the encoding-name for all the payloaders with a fixed payload type. We now either have an encoding-name or a pt in the sinkpad caps of a depayloader. See https://bugzilla.gnome.org/show_bug.cgi?id=639292
814 lines
25 KiB
C
814 lines
25 KiB
C
/* GStreamer
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* Copyright (C) <2010> Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
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* Copyright (C) <2010> Nokia Corporation
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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 <gst/rtp/gstrtpbuffer.h>
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#include <stdio.h>
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#include <string.h>
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#include "gstrtpmparobustdepay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpmparobustdepay_debug);
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#define GST_CAT_DEFAULT (rtpmparobustdepay_debug)
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static GstStaticPadTemplate gst_rtp_mpa_robust_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, " "mpegversion = (int) 1")
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);
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static GstStaticPadTemplate gst_rtp_mpa_robust_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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"clock-rate = (int) 90000, "
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"encoding-name = (string) \"MPA-ROBUST\" " "; "
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/* draft versions appear still in use out there */
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) [1, MAX], "
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"encoding-name = (string) { \"X-MP3-DRAFT-00\", \"X-MP3-DRAFT-01\", "
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" \"X-MP3-DRAFT-02\", \"X-MP3-DRAFT-03\", \"X-MP3-DRAFT-04\", "
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" \"X-MP3-DRAFT-05\", \"X-MP3-DRAFT-06\" }")
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);
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typedef struct _GstADUFrame
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{
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guint32 header;
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gint size;
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gint side_info;
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gint data_size;
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gint layer;
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gint backpointer;
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GstBuffer *buffer;
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} GstADUFrame;
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#define gst_rtp_mpa_robust_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpMPARobustDepay, gst_rtp_mpa_robust_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static GstStateChangeReturn gst_rtp_mpa_robust_change_state (GstElement *
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element, GstStateChange transition);
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static gboolean gst_rtp_mpa_robust_depay_setcaps (GstRTPBaseDepayload *
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depayload, GstCaps * caps);
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static GstBuffer *gst_rtp_mpa_robust_depay_process (GstRTPBaseDepayload *
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depayload, GstBuffer * buf);
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static void
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gst_rtp_mpa_robust_depay_finalize (GObject * object)
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{
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GstRtpMPARobustDepay *rtpmpadepay;
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rtpmpadepay = (GstRtpMPARobustDepay *) object;
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g_object_unref (rtpmpadepay->adapter);
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g_queue_free (rtpmpadepay->adu_frames);
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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_mpa_robust_depay_class_init (GstRtpMPARobustDepayClass * 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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GST_DEBUG_CATEGORY_INIT (rtpmparobustdepay_debug, "rtpmparobustdepay", 0,
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"Robust MPEG Audio RTP Depayloader");
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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_mpa_robust_depay_finalize;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_mpa_robust_change_state);
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_mpa_robust_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_mpa_robust_depay_sink_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP MPEG audio depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts MPEG audio from RTP packets (RFC 5219)",
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"Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
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gstrtpbasedepayload_class->set_caps = gst_rtp_mpa_robust_depay_setcaps;
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gstrtpbasedepayload_class->process = gst_rtp_mpa_robust_depay_process;
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}
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static void
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gst_rtp_mpa_robust_depay_init (GstRtpMPARobustDepay * rtpmpadepay)
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{
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rtpmpadepay->adapter = gst_adapter_new ();
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rtpmpadepay->adu_frames = g_queue_new ();
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}
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static gboolean
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gst_rtp_mpa_robust_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps)
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{
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GstRtpMPARobustDepay *rtpmpadepay;
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GstStructure *structure;
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GstCaps *outcaps;
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gint clock_rate, draft;
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gboolean res;
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const gchar *encoding;
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rtpmpadepay = GST_RTP_MPA_ROBUST_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;
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depayload->clock_rate = clock_rate;
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rtpmpadepay->has_descriptor = TRUE;
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if ((encoding = gst_structure_get_string (structure, "encoding-name"))) {
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if (sscanf (encoding, "X-MP3-DRAFT-%d", &draft) && (draft == 0))
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rtpmpadepay->has_descriptor = FALSE;
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}
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outcaps =
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gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 1, NULL);
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res = gst_pad_set_caps (depayload->srcpad, outcaps);
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gst_caps_unref (outcaps);
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return res;
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}
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/* thanks again go to mp3parse ... */
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static const guint mp3types_bitrates[2][3][16] = {
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{
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{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
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{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
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{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
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},
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{
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{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
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},
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};
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static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
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{22050, 24000, 16000},
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{11025, 12000, 8000}
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};
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static inline guint
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mp3_type_frame_length_from_header (GstElement * mp3parse, guint32 header,
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guint * put_version, guint * put_layer, guint * put_channels,
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guint * put_bitrate, guint * put_samplerate, guint * put_mode,
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guint * put_crc)
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{
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guint length;
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gulong mode, samplerate, bitrate, layer, channels, padding, crc;
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gulong version;
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gint lsf, mpg25;
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if (header & (1 << 20)) {
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lsf = (header & (1 << 19)) ? 0 : 1;
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mpg25 = 0;
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} else {
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lsf = 1;
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mpg25 = 1;
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}
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version = 1 + lsf + mpg25;
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layer = 4 - ((header >> 17) & 0x3);
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crc = (header >> 16) & 0x1;
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bitrate = (header >> 12) & 0xF;
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bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
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/* The caller has ensured we have a valid header, so bitrate can't be
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zero here. */
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if (bitrate == 0) {
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GST_DEBUG_OBJECT (mp3parse, "invalid bitrate");
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return 0;
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}
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samplerate = (header >> 10) & 0x3;
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samplerate = mp3types_freqs[lsf + mpg25][samplerate];
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padding = (header >> 9) & 0x1;
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mode = (header >> 6) & 0x3;
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channels = (mode == 3) ? 1 : 2;
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switch (layer) {
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case 1:
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length = 4 * ((bitrate * 12) / samplerate + padding);
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break;
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case 2:
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length = (bitrate * 144) / samplerate + padding;
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break;
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default:
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case 3:
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length = (bitrate * 144) / (samplerate << lsf) + padding;
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break;
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}
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GST_LOG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes", length);
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GST_LOG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, "
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"layer = %lu, channels = %lu, mode = %lu", samplerate, bitrate, version,
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layer, channels, mode);
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if (put_version)
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*put_version = version;
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if (put_layer)
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*put_layer = layer;
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if (put_channels)
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*put_channels = channels;
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if (put_bitrate)
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*put_bitrate = bitrate;
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if (put_samplerate)
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*put_samplerate = samplerate;
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if (put_mode)
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*put_mode = mode;
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if (put_crc)
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*put_crc = crc;
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GST_LOG_OBJECT (mp3parse, "size = %u", length);
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return length;
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}
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/* generate empty/silent/dummy frame that mimics @frame,
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* except for rate, where maximum possible is selected */
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static GstADUFrame *
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gst_rtp_mpa_robust_depay_generate_dummy_frame (GstRtpMPARobustDepay *
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rtpmpadepay, GstADUFrame * frame)
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{
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GstADUFrame *dummy;
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GstMapInfo map;
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dummy = g_slice_dup (GstADUFrame, frame);
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/* go for maximum bitrate */
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dummy->header = (frame->header & ~(0xf << 12)) | (0xe << 12);
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dummy->size =
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mp3_type_frame_length_from_header (GST_ELEMENT_CAST (rtpmpadepay),
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dummy->header, NULL, NULL, NULL, NULL, NULL, NULL, NULL);
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dummy->data_size = dummy->size - 4 - dummy->side_info;
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dummy->backpointer = 0;
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dummy->buffer = gst_buffer_new_and_alloc (dummy->side_info + 4);
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gst_buffer_map (dummy->buffer, &map, GST_MAP_WRITE);
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memset (map.data, 0, map.size);
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GST_WRITE_UINT32_BE (map.data, dummy->header);
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gst_buffer_unmap (dummy->buffer, &map);
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GST_BUFFER_TIMESTAMP (dummy->buffer) = GST_BUFFER_TIMESTAMP (frame->buffer);
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return dummy;
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}
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/* validates and parses @buf, and queues for further transformation if valid,
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* otherwise discards @buf
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* Takes ownership of @buf. */
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static gboolean
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gst_rtp_mpa_robust_depay_queue_frame (GstRtpMPARobustDepay * rtpmpadepay,
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GstBuffer * buf)
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{
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GstADUFrame *frame = NULL;
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guint version, layer, channels, size;
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guint crc;
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GstMapInfo map;
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g_return_val_if_fail (buf != NULL, FALSE);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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if (map.size < 6)
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goto corrupt_frame;
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frame = g_slice_new0 (GstADUFrame);
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frame->header = GST_READ_UINT32_BE (map.data);
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size = mp3_type_frame_length_from_header (GST_ELEMENT_CAST (rtpmpadepay),
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frame->header, &version, &layer, &channels, NULL, NULL, NULL, &crc);
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if (!size)
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goto corrupt_frame;
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frame->size = size;
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frame->layer = layer;
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if (version == 1 && channels == 2)
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frame->side_info = 32;
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else if ((version == 1 && channels == 1) || (version >= 2 && channels == 2))
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frame->side_info = 17;
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else if (version >= 2 && channels == 1)
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frame->side_info = 9;
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else {
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g_assert_not_reached ();
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goto corrupt_frame;
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}
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/* backpointer */
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if (layer == 3) {
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frame->backpointer = GST_READ_UINT16_BE (map.data + 4);
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frame->backpointer >>= 7;
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GST_LOG_OBJECT (rtpmpadepay, "backpointer: %d", frame->backpointer);
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}
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if (!crc)
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frame->side_info += 2;
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GST_LOG_OBJECT (rtpmpadepay, "side info: %d", frame->side_info);
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frame->data_size = frame->size - 4 - frame->side_info;
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/* some size validation checks */
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if (4 + frame->side_info > map.size)
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goto corrupt_frame;
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/* ADU data would then extend past MP3 frame,
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* even using past byte reservoir */
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if (-frame->backpointer + (gint) (map.size) > frame->size)
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goto corrupt_frame;
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gst_buffer_unmap (buf, &map);
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/* ok, take buffer and queue */
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frame->buffer = buf;
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g_queue_push_tail (rtpmpadepay->adu_frames, frame);
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return TRUE;
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/* ERRORS */
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corrupt_frame:
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{
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GST_DEBUG_OBJECT (rtpmpadepay, "frame is corrupt");
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gst_buffer_unmap (buf, &map);
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gst_buffer_unref (buf);
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if (frame)
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g_slice_free (GstADUFrame, frame);
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return FALSE;
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}
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}
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static inline void
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gst_rtp_mpa_robust_depay_free_frame (GstADUFrame * frame)
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{
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if (frame->buffer)
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gst_buffer_unref (frame->buffer);
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g_slice_free (GstADUFrame, frame);
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}
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static inline void
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gst_rtp_mpa_robust_depay_dequeue_frame (GstRtpMPARobustDepay * rtpmpadepay)
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{
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GstADUFrame *head;
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GST_LOG_OBJECT (rtpmpadepay, "dequeueing ADU frame");
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if (rtpmpadepay->adu_frames->head == rtpmpadepay->cur_adu_frame)
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rtpmpadepay->cur_adu_frame = NULL;
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head = g_queue_pop_head (rtpmpadepay->adu_frames);
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g_assert (head->buffer);
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gst_rtp_mpa_robust_depay_free_frame (head);
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return;
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}
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/* returns TRUE if at least one new ADU frame was enqueued for MP3 conversion.
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* Takes ownership of @buf. */
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static gboolean
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gst_rtp_mpa_robust_depay_deinterleave (GstRtpMPARobustDepay * rtpmpadepay,
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GstBuffer * buf)
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{
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gboolean ret = FALSE;
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GstMapInfo map;
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guint val, iindex, icc;
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gst_buffer_map (buf, &map, GST_MAP_READ);
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val = GST_READ_UINT16_BE (map.data) >> 5;
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gst_buffer_unmap (buf, &map);
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iindex = val >> 3;
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icc = val & 0x7;
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GST_LOG_OBJECT (rtpmpadepay, "sync: 0x%x, index: %u, cycle count: %u",
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val, iindex, icc);
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/* basic case; no interleaving ever seen */
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if (val == 0x7ff && rtpmpadepay->last_icc < 0) {
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ret = gst_rtp_mpa_robust_depay_queue_frame (rtpmpadepay, buf);
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} else {
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if (G_UNLIKELY (rtpmpadepay->last_icc < 0)) {
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rtpmpadepay->last_icc = icc;
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rtpmpadepay->last_ii = iindex;
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}
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if (icc != rtpmpadepay->last_icc || iindex == rtpmpadepay->last_ii) {
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gint i;
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for (i = 0; i < 256; ++i) {
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if (rtpmpadepay->deinter[i] != NULL) {
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ret |= gst_rtp_mpa_robust_depay_queue_frame (rtpmpadepay,
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rtpmpadepay->deinter[i]);
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rtpmpadepay->deinter[i] = NULL;
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}
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}
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}
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/* rewrite buffer sync header */
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gst_buffer_map (buf, &map, GST_MAP_READWRITE);
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val = GST_READ_UINT16_BE (map.data);
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val = (0x7ff << 5) | val;
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GST_WRITE_UINT16_BE (map.data, val);
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gst_buffer_unmap (buf, &map);
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/* store and keep track of last indices */
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rtpmpadepay->last_icc = icc;
|
|
rtpmpadepay->last_ii = iindex;
|
|
rtpmpadepay->deinter[iindex] = buf;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* Head ADU frame corresponds to mp3_frame (i.e. in header in side-info) that
|
|
* is currently being written
|
|
* cur_adu_frame refers to ADU frame whose data should be bytewritten next
|
|
* (possibly starting from offset rather than start 0) (and is typicall tail
|
|
* at time of last push round).
|
|
* If at start, position where it should start writing depends on (data) sizes
|
|
* of previous mp3 frames (corresponding to foregoing ADU frames) kept in size,
|
|
* and its backpointer */
|
|
static GstFlowReturn
|
|
gst_rtp_mpa_robust_depay_push_mp3_frames (GstRtpMPARobustDepay * rtpmpadepay)
|
|
{
|
|
GstBuffer *buf;
|
|
GstADUFrame *frame, *head;
|
|
gint av;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
while (1) {
|
|
GstMapInfo map;
|
|
|
|
if (G_UNLIKELY (!rtpmpadepay->cur_adu_frame)) {
|
|
rtpmpadepay->cur_adu_frame = rtpmpadepay->adu_frames->head;
|
|
rtpmpadepay->offset = 0;
|
|
rtpmpadepay->size = 0;
|
|
}
|
|
|
|
if (G_UNLIKELY (!rtpmpadepay->cur_adu_frame))
|
|
break;
|
|
|
|
frame = (GstADUFrame *) rtpmpadepay->cur_adu_frame->data;
|
|
head = (GstADUFrame *) rtpmpadepay->adu_frames->head->data;
|
|
|
|
/* special case: non-layer III are sent straight through */
|
|
if (G_UNLIKELY (frame->layer != 3)) {
|
|
GST_DEBUG_OBJECT (rtpmpadepay, "layer %d frame, sending as-is",
|
|
frame->layer);
|
|
gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmpadepay),
|
|
frame->buffer);
|
|
frame->buffer = NULL;
|
|
/* and remove it from any further consideration */
|
|
g_slice_free (GstADUFrame, frame);
|
|
g_queue_delete_link (rtpmpadepay->adu_frames, rtpmpadepay->cur_adu_frame);
|
|
rtpmpadepay->cur_adu_frame = NULL;
|
|
continue;
|
|
}
|
|
|
|
if (rtpmpadepay->offset == gst_buffer_get_size (frame->buffer)) {
|
|
if (g_list_next (rtpmpadepay->cur_adu_frame)) {
|
|
rtpmpadepay->size += frame->data_size;
|
|
rtpmpadepay->cur_adu_frame = g_list_next (rtpmpadepay->cur_adu_frame);
|
|
frame = (GstADUFrame *) rtpmpadepay->cur_adu_frame->data;
|
|
rtpmpadepay->offset = 0;
|
|
GST_LOG_OBJECT (rtpmpadepay,
|
|
"moving to next ADU frame, size %d, side_info %d, backpointer %d",
|
|
frame->size, frame->side_info, frame->backpointer);
|
|
/* layer I and II packets have no bitreservoir and must be sent as-is;
|
|
* so flush any pending frame */
|
|
if (G_UNLIKELY (frame->layer != 3 && rtpmpadepay->mp3_frame))
|
|
goto flush;
|
|
} else {
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (G_UNLIKELY (!rtpmpadepay->mp3_frame)) {
|
|
GST_LOG_OBJECT (rtpmpadepay,
|
|
"setting up new MP3 frame of size %d, side_info %d",
|
|
head->size, head->side_info);
|
|
rtpmpadepay->mp3_frame = gst_byte_writer_new_with_size (head->size, TRUE);
|
|
/* 0-fill possible gaps */
|
|
gst_byte_writer_fill_unchecked (rtpmpadepay->mp3_frame, 0, head->size);
|
|
gst_byte_writer_set_pos (rtpmpadepay->mp3_frame, 0);
|
|
/* bytewriter corresponds to head frame,
|
|
* i.e. the header and the side info must match */
|
|
g_assert (4 + head->side_info <= head->size);
|
|
gst_buffer_map (head->buffer, &map, GST_MAP_READ);
|
|
gst_byte_writer_put_data_unchecked (rtpmpadepay->mp3_frame,
|
|
map.data, 4 + head->side_info);
|
|
gst_buffer_unmap (head->buffer, &map);
|
|
}
|
|
|
|
buf = frame->buffer;
|
|
av = gst_byte_writer_get_remaining (rtpmpadepay->mp3_frame);
|
|
GST_LOG_OBJECT (rtpmpadepay, "current mp3 frame remaining: %d", av);
|
|
GST_LOG_OBJECT (rtpmpadepay, "accumulated ADU frame data_size: %d",
|
|
rtpmpadepay->size);
|
|
|
|
if (rtpmpadepay->offset) {
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
/* no need to position, simply append */
|
|
g_assert (map.size > rtpmpadepay->offset);
|
|
av = MIN (av, map.size - rtpmpadepay->offset);
|
|
GST_LOG_OBJECT (rtpmpadepay,
|
|
"appending %d bytes from ADU frame at offset %d", av,
|
|
rtpmpadepay->offset);
|
|
gst_byte_writer_put_data_unchecked (rtpmpadepay->mp3_frame,
|
|
map.data + rtpmpadepay->offset, av);
|
|
rtpmpadepay->offset += av;
|
|
gst_buffer_unmap (buf, &map);
|
|
} else {
|
|
gint pos, tpos;
|
|
|
|
/* position writing according to ADU frame backpointer */
|
|
pos = gst_byte_writer_get_pos (rtpmpadepay->mp3_frame);
|
|
tpos = rtpmpadepay->size - frame->backpointer + 4 + head->side_info;
|
|
GST_LOG_OBJECT (rtpmpadepay, "current MP3 frame at position %d, "
|
|
"starting new ADU frame data at offset %d", pos, tpos);
|
|
if (tpos < pos) {
|
|
GstADUFrame *dummy;
|
|
|
|
/* try to insert as few frames as possible,
|
|
* so go for a reasonably large dummy frame size */
|
|
GST_LOG_OBJECT (rtpmpadepay,
|
|
"overlapping previous data; inserting dummy frame");
|
|
dummy =
|
|
gst_rtp_mpa_robust_depay_generate_dummy_frame (rtpmpadepay, frame);
|
|
g_queue_insert_before (rtpmpadepay->adu_frames,
|
|
rtpmpadepay->cur_adu_frame, dummy);
|
|
/* offset is known to be zero, so we can shift current one */
|
|
rtpmpadepay->cur_adu_frame = rtpmpadepay->cur_adu_frame->prev;
|
|
if (!rtpmpadepay->size) {
|
|
g_assert (rtpmpadepay->cur_adu_frame ==
|
|
rtpmpadepay->adu_frames->head);
|
|
GST_LOG_OBJECT (rtpmpadepay, "... which is new head frame");
|
|
gst_byte_writer_free (rtpmpadepay->mp3_frame);
|
|
rtpmpadepay->mp3_frame = NULL;
|
|
}
|
|
/* ... and continue adding that empty one immediately,
|
|
* and then see if that provided enough extra space */
|
|
continue;
|
|
} else if (tpos >= pos + av) {
|
|
/* ADU frame no longer needs current MP3 frame; move to its end */
|
|
GST_LOG_OBJECT (rtpmpadepay, "passed current MP3 frame");
|
|
gst_byte_writer_set_pos (rtpmpadepay->mp3_frame, pos + av);
|
|
} else {
|
|
/* position and append */
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
GST_LOG_OBJECT (rtpmpadepay, "adding to current MP3 frame");
|
|
gst_byte_writer_set_pos (rtpmpadepay->mp3_frame, tpos);
|
|
av -= (tpos - pos);
|
|
g_assert (map.size >= 4 + frame->side_info);
|
|
av = MIN (av, map.size - 4 - frame->side_info);
|
|
gst_byte_writer_put_data_unchecked (rtpmpadepay->mp3_frame,
|
|
map.data + 4 + frame->side_info, av);
|
|
rtpmpadepay->offset += av + 4 + frame->side_info;
|
|
gst_buffer_unmap (buf, &map);
|
|
}
|
|
}
|
|
|
|
/* if mp3 frame filled, send on its way */
|
|
if (gst_byte_writer_get_remaining (rtpmpadepay->mp3_frame) == 0) {
|
|
flush:
|
|
buf = gst_byte_writer_free_and_get_buffer (rtpmpadepay->mp3_frame);
|
|
rtpmpadepay->mp3_frame = NULL;
|
|
GST_BUFFER_TIMESTAMP (buf) = GST_BUFFER_TIMESTAMP (head->buffer);
|
|
/* no longer need head ADU frame header and side info */
|
|
/* NOTE maybe head == current, then size and offset go off a bit,
|
|
* but current gets reset to NULL, and then also offset and size */
|
|
rtpmpadepay->size -= head->data_size;
|
|
gst_rtp_mpa_robust_depay_dequeue_frame (rtpmpadepay);
|
|
/* send */
|
|
ret = gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmpadepay),
|
|
buf);
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* process ADU frame @buf through:
|
|
* - deinterleaving
|
|
* - converting to MP3 frames
|
|
* Takes ownership of @buf.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_mpa_robust_depay_submit_adu (GstRtpMPARobustDepay * rtpmpadepay,
|
|
GstBuffer * buf)
|
|
{
|
|
if (gst_rtp_mpa_robust_depay_deinterleave (rtpmpadepay, buf))
|
|
return gst_rtp_mpa_robust_depay_push_mp3_frames (rtpmpadepay);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_rtp_mpa_robust_depay_process (GstRTPBaseDepayload * depayload,
|
|
GstBuffer * buf)
|
|
{
|
|
GstRtpMPARobustDepay *rtpmpadepay;
|
|
gint payload_len, offset;
|
|
guint8 *payload;
|
|
gboolean cont, dtype;
|
|
guint av, size;
|
|
GstClockTime timestamp;
|
|
GstRTPBuffer rtp = { NULL };
|
|
|
|
rtpmpadepay = GST_RTP_MPA_ROBUST_DEPAY (depayload);
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (buf);
|
|
|
|
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
|
|
|
|
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
|
|
if (payload_len <= 1)
|
|
goto short_read;
|
|
|
|
payload = gst_rtp_buffer_get_payload (&rtp);
|
|
offset = 0;
|
|
GST_LOG_OBJECT (rtpmpadepay, "payload_len: %d", payload_len);
|
|
|
|
/* strip off descriptor
|
|
*
|
|
* 0 1
|
|
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
|
|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
* |C|T| ADU size |
|
|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
*
|
|
* C: if 1, data is continuation
|
|
* T: if 1, size is 14 bits, otherwise 6 bits
|
|
* ADU size: size of following packet (not including descriptor)
|
|
*/
|
|
while (payload_len) {
|
|
if (G_LIKELY (rtpmpadepay->has_descriptor)) {
|
|
cont = ! !(payload[offset] & 0x80);
|
|
dtype = ! !(payload[offset] & 0x40);
|
|
if (dtype) {
|
|
size = (payload[offset] & 0x3f) << 8 | payload[offset + 1];
|
|
payload_len--;
|
|
offset++;
|
|
} else if (payload_len >= 2) {
|
|
size = (payload[offset] & 0x3f);
|
|
payload_len -= 2;
|
|
offset += 2;
|
|
} else {
|
|
goto short_read;
|
|
}
|
|
} else {
|
|
cont = FALSE;
|
|
dtype = -1;
|
|
size = payload_len;
|
|
}
|
|
|
|
GST_LOG_OBJECT (rtpmpadepay, "offset %d has cont: %d, dtype: %d, size: %d",
|
|
offset, cont, dtype, size);
|
|
|
|
buf = gst_rtp_buffer_get_payload_subbuffer (&rtp, offset,
|
|
MIN (size, payload_len));
|
|
|
|
if (cont) {
|
|
av = gst_adapter_available (rtpmpadepay->adapter);
|
|
if (G_UNLIKELY (!av)) {
|
|
GST_DEBUG_OBJECT (rtpmpadepay,
|
|
"discarding continuation fragment without prior fragment");
|
|
gst_buffer_unref (buf);
|
|
} else {
|
|
av += gst_buffer_get_size (buf);
|
|
gst_adapter_push (rtpmpadepay->adapter, buf);
|
|
if (av == size) {
|
|
timestamp = gst_adapter_prev_pts (rtpmpadepay->adapter, NULL);
|
|
buf = gst_adapter_take_buffer (rtpmpadepay->adapter, size);
|
|
GST_BUFFER_TIMESTAMP (buf) = timestamp;
|
|
gst_rtp_mpa_robust_depay_submit_adu (rtpmpadepay, buf);
|
|
} else if (av > size) {
|
|
GST_DEBUG_OBJECT (rtpmpadepay,
|
|
"assembled ADU size %d larger than expected %d; discarding",
|
|
av, size);
|
|
gst_adapter_clear (rtpmpadepay->adapter);
|
|
}
|
|
}
|
|
size = payload_len;
|
|
} else {
|
|
/* not continuation, first fragment or whole ADU */
|
|
if (payload_len == size) {
|
|
/* whole ADU */
|
|
GST_BUFFER_TIMESTAMP (buf) = timestamp;
|
|
gst_rtp_mpa_robust_depay_submit_adu (rtpmpadepay, buf);
|
|
} else if (payload_len < size) {
|
|
/* first fragment */
|
|
gst_adapter_push (rtpmpadepay->adapter, buf);
|
|
size = payload_len;
|
|
}
|
|
}
|
|
|
|
offset += size;
|
|
payload_len -= size;
|
|
|
|
/* timestamp applies to first payload, no idea for subsequent ones */
|
|
timestamp = GST_CLOCK_TIME_NONE;
|
|
}
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
return NULL;
|
|
|
|
/* ERRORS */
|
|
short_read:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpmpadepay, STREAM, DECODE,
|
|
(NULL), ("Packet contains invalid data"));
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_mpa_robust_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
GstRtpMPARobustDepay *rtpmpadepay;
|
|
|
|
rtpmpadepay = GST_RTP_MPA_ROBUST_DEPAY (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
rtpmpadepay->last_ii = -1;
|
|
rtpmpadepay->last_icc = -1;
|
|
rtpmpadepay->size = 0;
|
|
rtpmpadepay->offset = 0;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (ret != GST_STATE_CHANGE_SUCCESS)
|
|
return ret;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
{
|
|
gint i;
|
|
|
|
gst_adapter_clear (rtpmpadepay->adapter);
|
|
for (i = 0; i < G_N_ELEMENTS (rtpmpadepay->deinter); i++) {
|
|
gst_buffer_replace (&rtpmpadepay->deinter[i], NULL);
|
|
}
|
|
rtpmpadepay->cur_adu_frame = NULL;
|
|
g_queue_foreach (rtpmpadepay->adu_frames,
|
|
(GFunc) gst_rtp_mpa_robust_depay_free_frame, NULL);
|
|
g_queue_clear (rtpmpadepay->adu_frames);
|
|
if (rtpmpadepay->mp3_frame)
|
|
gst_byte_writer_free (rtpmpadepay->mp3_frame);
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_mpa_robust_depay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpmparobustdepay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_ROBUST_DEPAY);
|
|
}
|