/* GStreamer * Copyright (C) <2009> 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 <stdlib.h> #include <string.h> #include <gst/rtp/gstrtpbuffer.h> #include "gstrtpceltpay.h" GST_DEBUG_CATEGORY_STATIC (rtpceltpay_debug); #define GST_CAT_DEFAULT (rtpceltpay_debug) static GstStaticPadTemplate gst_rtp_celt_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-celt, " "rate = (int) [ 32000, 64000 ], " "channels = (int) [1, 2], " "frame-size = (int) [ 64, 512 ]") ); static GstStaticPadTemplate gst_rtp_celt_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) [ 32000, 48000 ], " "encoding-name = (string) \"CELT\"") ); static void gst_rtp_celt_pay_finalize (GObject * object); static GstStateChangeReturn gst_rtp_celt_pay_change_state (GstElement * element, GstStateChange transition); static gboolean gst_rtp_celt_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstCaps *gst_rtp_celt_pay_getcaps (GstBaseRTPPayload * payload, GstPad * pad); static GstFlowReturn gst_rtp_celt_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); GST_BOILERPLATE (GstRtpCELTPay, gst_rtp_celt_pay, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); static void gst_rtp_celt_pay_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_celt_pay_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_celt_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP CELT payloader", "Codec/Payloader/Network", "Payload-encodes CELT audio into a RTP packet", "Wim Taymans <wim.taymans@gmail.com>"); GST_DEBUG_CATEGORY_INIT (rtpceltpay_debug, "rtpceltpay", 0, "CELT RTP Payloader"); } static void gst_rtp_celt_pay_class_init (GstRtpCELTPayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; gobject_class->finalize = gst_rtp_celt_pay_finalize; gstelement_class->change_state = gst_rtp_celt_pay_change_state; gstbasertppayload_class->set_caps = gst_rtp_celt_pay_setcaps; gstbasertppayload_class->get_caps = gst_rtp_celt_pay_getcaps; gstbasertppayload_class->handle_buffer = gst_rtp_celt_pay_handle_buffer; } static void gst_rtp_celt_pay_init (GstRtpCELTPay * rtpceltpay, GstRtpCELTPayClass * klass) { rtpceltpay->queue = g_queue_new (); } static void gst_rtp_celt_pay_finalize (GObject * object) { GstRtpCELTPay *rtpceltpay; rtpceltpay = GST_RTP_CELT_PAY (object); g_queue_free (rtpceltpay->queue); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_rtp_celt_pay_clear_queued (GstRtpCELTPay * rtpceltpay) { GstBuffer *buf; while ((buf = g_queue_pop_head (rtpceltpay->queue))) gst_buffer_unref (buf); rtpceltpay->bytes = 0; rtpceltpay->sbytes = 0; rtpceltpay->qduration = 0; } static void gst_rtp_celt_pay_add_queued (GstRtpCELTPay * rtpceltpay, GstBuffer * buffer, guint ssize, guint size, GstClockTime duration) { g_queue_push_tail (rtpceltpay->queue, buffer); rtpceltpay->sbytes += ssize; rtpceltpay->bytes += size; /* only add durations when we have a valid previous duration */ if (rtpceltpay->qduration != -1) { if (duration != -1) /* only add valid durations */ rtpceltpay->qduration += duration; else /* if we add a buffer without valid duration, our total queued duration * becomes unknown */ rtpceltpay->qduration = -1; } } static gboolean gst_rtp_celt_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { /* don't configure yet, we wait for the ident packet */ return TRUE; } static GstCaps * gst_rtp_celt_pay_getcaps (GstBaseRTPPayload * payload, GstPad * pad) { GstCaps *otherpadcaps; GstCaps *caps; const gchar *params; otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad); caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); if (otherpadcaps) { if (!gst_caps_is_empty (otherpadcaps)) { GstStructure *ps = gst_caps_get_structure (otherpadcaps, 0); GstStructure *s = gst_caps_get_structure (caps, 0); gint clock_rate = 0, frame_size = 0, channels = 1; if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) { gst_structure_fixate_field_nearest_int (s, "rate", clock_rate); } if ((params = gst_structure_get_string (ps, "frame-size"))) frame_size = atoi (params); if (frame_size) gst_structure_set (s, "frame-size", G_TYPE_INT, frame_size, NULL); if ((params = gst_structure_get_string (ps, "encoding-params"))) channels = atoi (params); gst_structure_fixate_field_nearest_int (s, "channels", channels); GST_DEBUG_OBJECT (payload, "clock-rate=%d frame-size=%d channels=%d", clock_rate, frame_size, channels); } gst_caps_unref (otherpadcaps); } return caps; } static gboolean gst_rtp_celt_pay_parse_ident (GstRtpCELTPay * rtpceltpay, const guint8 * data, guint size) { guint32 version, header_size, rate, nb_channels, frame_size, overlap; guint32 bytes_per_packet; GstBaseRTPPayload *payload; gchar *cstr, *fsstr; gboolean res; /* we need the header string (8), the version string (20), the version * and the header length. */ if (size < 36) goto too_small; if (!g_str_has_prefix ((const gchar *) data, "CELT ")) goto wrong_header; /* skip header and version string */ data += 28; version = GST_READ_UINT32_LE (data); GST_DEBUG_OBJECT (rtpceltpay, "version %08x", version); #if 0 if (version != 1) goto wrong_version; #endif data += 4; /* ensure sizes */ header_size = GST_READ_UINT32_LE (data); if (header_size < 56) goto header_too_small; if (size < header_size) goto payload_too_small; data += 4; rate = GST_READ_UINT32_LE (data); data += 4; nb_channels = GST_READ_UINT32_LE (data); data += 4; frame_size = GST_READ_UINT32_LE (data); data += 4; overlap = GST_READ_UINT32_LE (data); data += 4; bytes_per_packet = GST_READ_UINT32_LE (data); GST_DEBUG_OBJECT (rtpceltpay, "rate %d, nb_channels %d, frame_size %d", rate, nb_channels, frame_size); GST_DEBUG_OBJECT (rtpceltpay, "overlap %d, bytes_per_packet %d", overlap, bytes_per_packet); payload = GST_BASE_RTP_PAYLOAD (rtpceltpay); gst_basertppayload_set_options (payload, "audio", FALSE, "CELT", rate); cstr = g_strdup_printf ("%d", nb_channels); fsstr = g_strdup_printf ("%d", frame_size); res = gst_basertppayload_set_outcaps (payload, "encoding-params", G_TYPE_STRING, cstr, "frame-size", G_TYPE_STRING, fsstr, NULL); g_free (cstr); g_free (fsstr); return res; /* ERRORS */ too_small: { GST_DEBUG_OBJECT (rtpceltpay, "ident packet too small, need at least 32 bytes"); return FALSE; } wrong_header: { GST_DEBUG_OBJECT (rtpceltpay, "ident packet does not start with \"CELT \""); return FALSE; } #if 0 wrong_version: { GST_DEBUG_OBJECT (rtpceltpay, "can only handle version 1, have version %d", version); return FALSE; } #endif header_too_small: { GST_DEBUG_OBJECT (rtpceltpay, "header size too small, need at least 80 bytes, " "got only %d", header_size); return FALSE; } payload_too_small: { GST_DEBUG_OBJECT (rtpceltpay, "payload too small, need at least %d bytes, got only %d", header_size, size); return FALSE; } } static GstFlowReturn gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay) { GstFlowReturn ret; GstBuffer *buf, *outbuf; guint8 *payload, *spayload; guint payload_len; GstClockTime duration; payload_len = rtpceltpay->bytes + rtpceltpay->sbytes; duration = rtpceltpay->qduration; GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT, payload_len, GST_TIME_ARGS (rtpceltpay->qduration)); /* get a big enough packet for the sizes + payloads */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); GST_BUFFER_DURATION (outbuf) = duration; /* point to the payload for size headers and data */ spayload = gst_rtp_buffer_get_payload (outbuf); payload = spayload + rtpceltpay->sbytes; while ((buf = g_queue_pop_head (rtpceltpay->queue))) { guint size; /* copy first timestamp to output */ if (GST_BUFFER_TIMESTAMP (outbuf) == -1) GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf); /* write the size to the header */ size = GST_BUFFER_SIZE (buf); while (size > 0xff) { *spayload++ = 0xff; size -= 0xff; } *spayload++ = size; size = GST_BUFFER_SIZE (buf); /* copy payload */ memcpy (payload, GST_BUFFER_DATA (buf), size); payload += size; gst_buffer_unref (buf); } /* we consumed it all */ rtpceltpay->bytes = 0; rtpceltpay->sbytes = 0; rtpceltpay->qduration = 0; ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpceltpay), outbuf); return ret; } static GstFlowReturn gst_rtp_celt_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstFlowReturn ret; GstRtpCELTPay *rtpceltpay; guint size, payload_len; guint8 *data; GstClockTime duration, packet_dur; guint i, ssize, packet_len; rtpceltpay = GST_RTP_CELT_PAY (basepayload); ret = GST_FLOW_OK; size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); switch (rtpceltpay->packet) { case 0: /* ident packet. We need to parse the headers to construct the RTP * properties. */ if (!gst_rtp_celt_pay_parse_ident (rtpceltpay, data, size)) goto parse_error; goto done; case 1: /* comment packet, we ignore it */ goto done; default: /* other packets go in the payload */ break; } duration = GST_BUFFER_DURATION (buffer); GST_LOG_OBJECT (rtpceltpay, "got buffer of duration %" GST_TIME_FORMAT ", size %u", GST_TIME_ARGS (duration), size); /* calculate the size of the size field and the payload */ ssize = 1; for (i = size; i > 0xff; i -= 0xff) ssize++; GST_DEBUG_OBJECT (rtpceltpay, "bytes for size %u", ssize); /* calculate what the new size and duration would be of the packet */ payload_len = ssize + size + rtpceltpay->bytes + rtpceltpay->sbytes; if (rtpceltpay->qduration != -1 && duration != -1) packet_dur = rtpceltpay->qduration + duration; else packet_dur = 0; packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0); if (gst_basertppayload_is_filled (basepayload, packet_len, packet_dur)) { /* size or duration would overflow the packet, flush the queued data */ ret = gst_rtp_celt_pay_flush_queued (rtpceltpay); } /* queue the packet */ gst_rtp_celt_pay_add_queued (rtpceltpay, buffer, ssize, size, duration); done: rtpceltpay->packet++; return ret; /* ERRORS */ parse_error: { GST_ELEMENT_ERROR (rtpceltpay, STREAM, DECODE, (NULL), ("Error parsing first identification packet.")); return GST_FLOW_ERROR; } } static GstStateChangeReturn gst_rtp_celt_pay_change_state (GstElement * element, GstStateChange transition) { GstRtpCELTPay *rtpceltpay; GstStateChangeReturn ret; rtpceltpay = GST_RTP_CELT_PAY (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: rtpceltpay->packet = 0; break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_rtp_celt_pay_clear_queued (rtpceltpay); break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return ret; } gboolean gst_rtp_celt_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpceltpay", GST_RANK_NONE, GST_TYPE_RTP_CELT_PAY); }