From: <wt...@ke...> - 2008-09-26 13:56:17
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CVS Root: /cvs/gstreamer Module: gst-plugins-good Changes by: wtay Date: Fri Sep 26 2008 13:56:02 UTC Log message: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_get_type), (gst_rtp_mp4a_pay_base_init), (gst_rtp_mp4a_pay_class_init), (gst_rtp_mp4a_pay_init), (gst_rtp_mp4a_pay_finalize), (gst_rtp_mp4a_pay_parse_audio_config), (gst_rtp_mp4a_pay_new_caps), (gst_rtp_mp4a_pay_setcaps), (gst_rtp_mp4a_pay_handle_buffer), (gst_rtp_mp4a_pay_change_state), (gst_rtp_mp4a_pay_plugin_init): * gst/rtp/gstrtpmp4apay.h: Added MP4A-LATM payloader to match the depayloader. Modified files: . : ChangeLog gst/rtp : Makefile.am gstrtp.c Added files: gst/rtp : gstrtpmp4apay.c gstrtpmp4apay.h Links: http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/ChangeLog.diff?r1=1.3743&r2=1.3744 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/Makefile.am.diff?r1=1.50&r2=1.51 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtp.c.diff?r1=1.45&r2=1.46 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtpmp4apay.c?rev=1.1&content-type=text/vnd.viewcvs-markup http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtp/gstrtpmp4apay.h?rev=1.1&content-type=text/vnd.viewcvs-markup ====Begin Diffs==== Index: ChangeLog =================================================================== RCS file: /cvs/gstreamer/gst-plugins-good/ChangeLog,v retrieving revision 1.3743 retrieving revision 1.3744 diff -u -d -r1.3743 -r1.3744 --- ChangeLog 25 Sep 2008 15:11:14 -0000 1.3743 +++ ChangeLog 26 Sep 2008 13:55:46 -0000 1.3744 @@ -1,3 +1,16 @@ +2008-09-26 Wim Taymans <wim...@co...> + + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: (plugin_init): + * gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_get_type), + (gst_rtp_mp4a_pay_base_init), (gst_rtp_mp4a_pay_class_init), + (gst_rtp_mp4a_pay_init), (gst_rtp_mp4a_pay_finalize), + (gst_rtp_mp4a_pay_parse_audio_config), (gst_rtp_mp4a_pay_new_caps), + (gst_rtp_mp4a_pay_setcaps), (gst_rtp_mp4a_pay_handle_buffer), + (gst_rtp_mp4a_pay_change_state), (gst_rtp_mp4a_pay_plugin_init): + * gst/rtp/gstrtpmp4apay.h: + Added MP4A-LATM payloader to match the depayloader. 2008-09-25 Wim Taymans <wim...@co...> * gst/videomixer/videomixer.c: (gst_videomixer_fill_queues), Index: Makefile.am RCS file: /cvs/gstreamer/gst-plugins-good/gst/rtp/Makefile.am,v retrieving revision 1.50 retrieving revision 1.51 diff -u -d -r1.50 -r1.51 --- Makefile.am 5 Aug 2008 13:54:17 -0000 1.50 +++ Makefile.am 26 Sep 2008 13:55:48 -0000 1.51 @@ -42,6 +42,7 @@ gstrtpmp4gdepay.c \ gstrtpmp4gpay.c \ gstrtpmp4adepay.c \ + gstrtpmp4apay.c \ gstrtpspeexdepay.c \ gstrtpspeexpay.c \ gstrtpsv3vdepay.c \ @@ -103,7 +104,8 @@ gstrtpmp4vpay.h \ gstrtpmp4gdepay.h \ gstrtpmp4gpay.h \ - gstrtpmp4adepay.h \ + gstrtpmp4adepay.h \ + gstrtpmp4apay.h \ gstrtpdepay.h \ gstasteriskh263.h \ gstrtpspeexdepay.h \ Index: gstrtp.c RCS file: /cvs/gstreamer/gst-plugins-good/gst/rtp/gstrtp.c,v retrieving revision 1.45 retrieving revision 1.46 diff -u -d -r1.45 -r1.46 --- gstrtp.c 5 Aug 2008 13:54:18 -0000 1.45 +++ gstrtp.c 26 Sep 2008 13:55:48 -0000 1.46 @@ -58,6 +58,7 @@ #include "gstrtpmp4vdepay.h" #include "gstrtpmp4vpay.h" #include "gstrtpmp4adepay.h" +#include "gstrtpmp4apay.h" #include "gstrtpmp4gdepay.h" #include "gstrtpmp4gpay.h" #include "gstrtpspeexpay.h" @@ -181,6 +182,9 @@ if (!gst_rtp_mp4v_depay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_mp4a_pay_plugin_init (plugin)) + return FALSE; if (!gst_rtp_mp4a_depay_plugin_init (plugin)) --- NEW FILE: gstrtpmp4apay.c --- /* GStreamer * Copyright (C) <2008> Wim Taymans <wim...@gm...> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include <string.h> #include <gst/rtp/gstrtpbuffer.h> #include "gstrtpmp4apay.h" GST_DEBUG_CATEGORY_STATIC (rtpmp4apay_debug); #define GST_CAT_DEFAULT (rtpmp4apay_debug) /* elementfactory information */ static const GstElementDetails gst_rtp_mp4apay_details = GST_ELEMENT_DETAILS ("RTP packet payloader", "Codec/Payloader/Network", "Payload MPEG4 audio as RTP packets (RFC 3016)", "Wim Taymans <wim...@gm...>"); static GstStaticPadTemplate gst_rtp_mp4a_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg," "mpegversion=(int) 4") ); static GstStaticPadTemplate gst_rtp_mp4a_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP4A-LATM\"" /* All optional parameters * * "cpresent = (string) \"0\"" * "config=" */ ) static void gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass); static void gst_rtp_mp4a_pay_base_init (GstRtpMP4APayClass * klass); static void gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay); static void gst_rtp_mp4a_pay_finalize (GObject * object); static gboolean gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstStateChangeReturn gst_rtp_mp4a_pay_change_state (GstElement * element, GstStateChange transition); static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); static GstBaseRTPPayloadClass *parent_class = NULL; static GType gst_rtp_mp4a_pay_get_type (void) { static GType rtpmp4apay_type = 0; if (!rtpmp4apay_type) { static const GTypeInfo rtpmp4apay_info = { sizeof (GstRtpMP4APayClass), (GBaseInitFunc) gst_rtp_mp4a_pay_base_init, NULL, (GClassInitFunc) gst_rtp_mp4a_pay_class_init, sizeof (GstRtpMP4APay), 0, (GInstanceInitFunc) gst_rtp_mp4a_pay_init, }; rtpmp4apay_type = g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4APay", &rtpmp4apay_info, 0); } return rtpmp4apay_type; } static void gst_rtp_mp4a_pay_base_init (GstRtpMP4APayClass * klass) GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mp4a_pay_src_template)); gst_static_pad_template_get (&gst_rtp_mp4a_pay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_mp4apay_details); gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass) GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->finalize = gst_rtp_mp4a_pay_finalize; gstelement_class->change_state = gst_rtp_mp4a_pay_change_state; gstbasertppayload_class->set_caps = gst_rtp_mp4a_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer; GST_DEBUG_CATEGORY_INIT (rtpmp4apay_debug, "rtpmp4apay", 0, "MP4A-LATM RTP Payloader"); gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay) rtpmp4apay->rate = 90000; rtpmp4apay->profile = g_strdup ("1"); gst_rtp_mp4a_pay_finalize (GObject * object) GstRtpMP4APay *rtpmp4apay; rtpmp4apay = GST_RTP_MP4A_PAY (object); g_free (rtpmp4apay->params); rtpmp4apay->params = NULL; if (rtpmp4apay->config) gst_buffer_unref (rtpmp4apay->config); rtpmp4apay->config = NULL; g_free (rtpmp4apay->profile); rtpmp4apay->profile = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); static unsigned sampling_table[16] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 0 }; static gboolean gst_rtp_mp4a_pay_parse_audio_config (GstRtpMP4APay * rtpmp4apay, GstBuffer * buffer) guint8 *data; guint size; guint8 objectType; guint8 samplingIdx; guint8 channelCfg; data = GST_BUFFER_DATA (buffer); size = GST_BUFFER_SIZE (buffer); if (size < 2) goto too_short; /* any object type is fine, we need to copy it to the profile-level-id field. */ objectType = (data[0] & 0xf8) >> 3; if (objectType == 0) goto invalid_object; samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7); /* only fixed values for now */ if (samplingIdx > 12 && samplingIdx != 15) goto wrong_freq; channelCfg = ((data[1] & 0x78) >> 3); if (channelCfg > 7) goto wrong_channels; /* rtp rate depends on sampling rate of the audio */ if (samplingIdx == 15) { if (size < 5) goto too_short; /* index of 15 means we get the rate in the next 24 bits */ rtpmp4apay->rate = ((data[1] & 0x7f) << 17) | ((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7); } else { /* else use the rate from the table */ rtpmp4apay->rate = sampling_table[samplingIdx]; /* extra rtp params contain the number of channels */ rtpmp4apay->params = g_strdup_printf ("%d", channelCfg); /* audio stream type */ rtpmp4apay->streamtype = "5"; /* profile */ rtpmp4apay->profile = g_strdup_printf ("%d", objectType); GST_DEBUG_OBJECT (rtpmp4apay, "objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType, samplingIdx, rtpmp4apay->rate, channelCfg); return TRUE; /* ERROR */ too_short: { GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT, (NULL), ("config string too short, expected 2 bytes, got %d", size)); return FALSE; invalid_object: (NULL), ("invalid object type 0")); wrong_freq: GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED, (NULL), ("unsupported frequency index %d", samplingIdx)); wrong_channels: (NULL), ("unsupported number of channels %d, must < 8", channelCfg)); gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay) gchar *config; GValue v = { 0 }; g_value_init (&v, GST_TYPE_BUFFER); gst_value_set_buffer (&v, rtpmp4apay->config); config = gst_value_serialize (&v); gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay), "cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL); g_value_unset (&v); g_free (config); gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) GstStructure *structure; const GValue *codec_data; rtpmp4apay = GST_RTP_MP4A_PAY (payload); structure = gst_caps_get_structure (caps, 0); codec_data = gst_structure_get_value (structure, "codec_data"); if (codec_data) { GST_LOG_OBJECT (rtpmp4apay, "got codec_data"); if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) { GstBuffer *buffer, *cbuffer; guint8 *config; guint8 *data; guint size, i; gboolean res; buffer = gst_value_get_buffer (codec_data); GST_LOG_OBJECT (rtpmp4apay, "configuring codec_data"); /* parse buffer */ res = gst_rtp_mp4a_pay_parse_audio_config (rtpmp4apay, buffer); if (!res) goto config_failed; size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); /* make the StreamMuxConfig, we need 15 bits for the header */ config = g_malloc0 (size + 2); /* Create StreamMuxConfig according to ISO/IEC 14496-3: * * audioMuxVersion == 0 (1 bit) * allStreamsSameTimeFraming == 1 (1 bit) * numSubFrames == numSubFrames (6 bits) * numProgram == 0 (4 bits) * numLayer == 0 (3 bits) */ config[0] = 0x40; config[1] = 0x00; /* append the config bits, shifting them 1 bit left */ for (i = 0; i < size; i++) { config[i + 1] |= ((data[i] & 0x80) >> 7); config[i + 2] |= ((data[i] & 0x7f) << 1); } cbuffer = gst_buffer_new (); GST_BUFFER_DATA (cbuffer) = config; GST_BUFFER_MALLOCDATA (cbuffer) = config; GST_BUFFER_SIZE (cbuffer) = size + 2; /* now we can configure the buffer */ if (rtpmp4apay->config) gst_buffer_unref (rtpmp4apay->config); rtpmp4apay->config = cbuffer; } gst_basertppayload_set_options (payload, "audio", TRUE, "MP4A-LATM", rtpmp4apay->rate); gst_rtp_mp4a_pay_new_caps (rtpmp4apay); /* ERRORS */ config_failed: GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config"); /* we expect buffers as exactly one complete AU static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstFlowReturn ret; GstBuffer *outbuf; guint count, mtu, size; gboolean fragmented; ret = GST_FLOW_OK; rtpmp4apay = GST_RTP_MP4A_PAY (basepayload); fragmented = FALSE; mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpmp4apay); while (size > 0) { guint towrite; guint8 *payload; guint payload_len; guint packet_len; /* this will be the total lenght of the packet */ packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0); if (!fragmented) { /* first packet calculate space for the packet including the header */ count = size; while (count >= 0xff) { packet_len++; count -= 0xff; packet_len++; /* fill one MTU or all available bytes */ towrite = MIN (packet_len, mtu); /* this is the payload length */ payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0); GST_DEBUG_OBJECT (rtpmp4apay, "avail %d, towrite %d, packet_len %d, payload_len %d", size, towrite, packet_len, payload_len); /* create buffer to hold the payload. */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); /* copy payload */ payload = gst_rtp_buffer_get_payload (outbuf); /* first packet write the header */ *payload++ = 0xff; payload_len--; *payload++ = count; payload_len--; /* copy data to payload */ memcpy (payload, data, payload_len); data += payload_len; size -= payload_len; /* marker only if the packet is complete */ gst_rtp_buffer_set_marker (outbuf, size == 0); ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4apay), outbuf); fragmented = TRUE; return ret; static GstStateChangeReturn gst_rtp_mp4a_pay_change_state (GstElement * element, GstStateChange transition) GstStateChangeReturn ret; rtpmp4apay = GST_RTP_MP4A_PAY (element); switch (transition) { default: break; ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); gboolean gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin) return gst_element_register (plugin, "rtpmp4apay", GST_RANK_NONE, GST_TYPE_RTP_MP4A_PAY); --- NEW FILE: gstrtpmp4apay.h --- * Copyright (C) <2005> Wim Taymans <wi...@fl...> #ifndef __GST_RTP_MP4A_PAY_H__ #define __GST_RTP_MP4A_PAY_H__ #include <gst/gst.h> #include <gst/rtp/gstbasertppayload.h> #include <gst/base/gstadapter.h> G_BEGIN_DECLS #define GST_TYPE_RTP_MP4A_PAY \ (gst_rtp_mp4a_pay_get_type()) #define GST_RTP_MP4A_PAY(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_MP4A_PAY,GstRtpMP4APay)) #define GST_RTP_MP4A_PAY_CLASS(klass) \ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_MP4A_PAY,GstRtpMP4APayClass)) #define GST_IS_RTP_MP4A_PAY(obj) \ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_MP4A_PAY)) #define GST_IS_RTP_MP4A_PAY_CLASS(klass) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_MP4A_PAY)) typedef struct _GstRtpMP4APay GstRtpMP4APay; typedef struct _GstRtpMP4APayClass GstRtpMP4APayClass; struct _GstRtpMP4APay GstBaseRTPPayload payload; gint rate; gchar *params; gchar *profile; const gchar *streamtype; GstBuffer *config; struct _GstRtpMP4APayClass GstBaseRTPPayloadClass parent_class; gboolean gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin); G_END_DECLS #endif /* __GST_RTP_MP4A_PAY_H__ */ |