/* GStreamer * Copyright (C) <2008> Wim Taymans <wim.taymans@gmail.com> * * 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 MPEG4 audio payloader", "Codec/Payloader/Network", "Payload MPEG4 audio as RTP packets (RFC 3016)", "Wim Taymans <wim.taymans@gmail.com>"); /* FIXME: add framed=(boolean)true once our encoders have this field set * on their output caps */ 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_PAD_ALWAYS, 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 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, NULL, NULL, 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_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mp4a_pay_sink_template)); gst_element_class_set_details (element_class, &gst_rtp_mp4apay_details); } static void gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass) { GObjectClass *gobject_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->finalize = gst_rtp_mp4a_pay_finalize; 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"); } static void gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay) { rtpmp4apay->rate = 90000; rtpmp4apay->profile = g_strdup ("1"); } static void 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 */ g_free (rtpmp4apay->params); rtpmp4apay->params = g_strdup_printf ("%d", channelCfg); /* audio stream type */ rtpmp4apay->streamtype = "5"; /* profile */ g_free (rtpmp4apay->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: { GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT, (NULL), ("invalid object type 0")); return FALSE; } wrong_freq: { GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED, (NULL), ("unsupported frequency index %d", samplingIdx)); return FALSE; } wrong_channels: { GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED, (NULL), ("unsupported number of channels %d, must < 8", channelCfg)); return FALSE; } } static gboolean gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay) { gchar *config; GValue v = { 0 }; gboolean res; g_value_init (&v, GST_TYPE_BUFFER); gst_value_set_buffer (&v, rtpmp4apay->config); config = gst_value_serialize (&v); res = 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); return res; } static gboolean gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { GstRtpMP4APay *rtpmp4apay; GstStructure *structure; const GValue *codec_data; gboolean res, framed = TRUE; 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; 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; } } if (gst_structure_get_boolean (structure, "framed", &framed) && !framed) { GST_WARNING_OBJECT (payload, "Need framed AAC data as input!"); } gst_basertppayload_set_options (payload, "audio", TRUE, "MP4A-LATM", rtpmp4apay->rate); res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay); return res; /* ERRORS */ config_failed: { GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config"); return FALSE; } } /* we expect buffers as exactly one complete AU */ static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstRtpMP4APay *rtpmp4apay; GstFlowReturn ret; GstBuffer *outbuf; guint count, mtu, size; guint8 *data; gboolean fragmented; GstClockTime timestamp; ret = GST_FLOW_OK; rtpmp4apay = GST_RTP_MP4A_PAY (basepayload); size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); timestamp = GST_BUFFER_TIMESTAMP (buffer); 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); if (!fragmented) { /* first packet write the header */ count = size; while (count >= 0xff) { *payload++ = 0xff; payload_len--; count -= 0xff; } *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); /* copy incomming timestamp (if any) to outgoing buffers */ GST_BUFFER_TIMESTAMP (outbuf) = timestamp; ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4apay), outbuf); fragmented = TRUE; } gst_buffer_unref (buffer); return ret; } gboolean gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp4apay", GST_RANK_NONE, GST_TYPE_RTP_MP4A_PAY); }