/* GStreamer * Copyright (C) <2007> Nokia Corporation * Copyright (C) <2007> Collabora Ltd * @author: Olivier Crete <olivier.crete@collabora.co.uk> * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /* * This payloader assumes that the data will ALWAYS come as zero or more * 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence. * Any other buffer format won't work */ #ifdef HAVE_CONFIG_H #include <config.h> #endif #include <string.h> #include <gst/rtp/gstrtpbuffer.h> #include <gst/base/gstadapter.h> #include <gst/audio/audio.h> #include "gstrtpg729pay.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug); #define GST_CAT_DEFAULT (rtpg729pay_debug) #define G729_FRAME_SIZE 10 #define G729B_CN_FRAME_SIZE 2 #define G729_FRAME_DURATION (10 * GST_MSECOND) #define G729_FRAME_DURATION_MS (10) static gboolean gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf); static GstStateChangeReturn gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition); static GstStaticPadTemplate gst_rtp_g729_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/G729, " /* according to RFC 3555 */ "channels = (int) 1, " "rate = (int) 8000") ); static GstStaticPadTemplate gst_rtp_g729_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_G729_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"; " "application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"") ); #define gst_rtp_g729_pay_parent_class parent_class G_DEFINE_TYPE (GstRTPG729Pay, gst_rtp_g729_pay, GST_TYPE_RTP_BASE_PAYLOAD); static void gst_rtp_g729_pay_finalize (GObject * object) { GstRTPG729Pay *pay = GST_RTP_G729_PAY (object); g_object_unref (pay->adapter); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass); GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0, "G.729 RTP Payloader"); gobject_class->finalize = gst_rtp_g729_pay_finalize; gstelement_class->change_state = gst_rtp_g729_pay_change_state; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_g729_pay_sink_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_g729_pay_src_template); gst_element_class_set_static_metadata (gstelement_class, "RTP G.729 payloader", "Codec/Payloader/Network/RTP", "Packetize G.729 audio into RTP packets", "Olivier Crete <olivier.crete@collabora.co.uk>"); payload_class->set_caps = gst_rtp_g729_pay_set_caps; payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer; } static void gst_rtp_g729_pay_init (GstRTPG729Pay * pay) { GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay); payload->pt = GST_RTP_PAYLOAD_G729; pay->adapter = gst_adapter_new (); } static void gst_rtp_g729_pay_reset (GstRTPG729Pay * pay) { gst_adapter_clear (pay->adapter); pay->discont = FALSE; pay->next_rtp_time = 0; pay->first_ts = GST_CLOCK_TIME_NONE; pay->first_rtp_time = 0; } static gboolean gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps) { gboolean res; gst_rtp_base_payload_set_options (payload, "audio", payload->pt != GST_RTP_PAYLOAD_G729, "G729", 8000); res = gst_rtp_base_payload_set_outcaps (payload, NULL); return res; } static GstFlowReturn gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay, GstBuffer * buf) { GstRTPBasePayload *basepayload; GstClockTime duration; guint frames; GstBuffer *outbuf; GstFlowReturn ret; GstRTPBuffer rtp = { NULL }; guint payload_len = gst_buffer_get_size (buf); basepayload = GST_RTP_BASE_PAYLOAD (rtpg729pay); GST_DEBUG_OBJECT (rtpg729pay, "Pushing %d bytes ts %" GST_TIME_FORMAT, payload_len, GST_TIME_ARGS (rtpg729pay->next_ts)); /* create buffer to hold the payload */ outbuf = gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (rtpg729pay), 0, 0, 0); gst_rtp_buffer_map (outbuf, GST_MAP_READWRITE, &rtp); /* set metadata */ frames = (payload_len / G729_FRAME_SIZE) + ((payload_len % G729_FRAME_SIZE) >> 1); duration = frames * G729_FRAME_DURATION; GST_BUFFER_PTS (outbuf) = rtpg729pay->next_ts; GST_BUFFER_DURATION (outbuf) = duration; GST_BUFFER_OFFSET (outbuf) = rtpg729pay->next_rtp_time; rtpg729pay->next_ts += duration; rtpg729pay->next_rtp_time += frames * 80; if (G_UNLIKELY (rtpg729pay->discont)) { GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit"); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); gst_rtp_buffer_set_marker (&rtp, TRUE); rtpg729pay->discont = FALSE; } gst_rtp_buffer_unmap (&rtp); /* append payload */ gst_rtp_copy_audio_meta (basepayload, outbuf, buf); outbuf = gst_buffer_append (outbuf, buf); ret = gst_rtp_base_payload_push (basepayload, outbuf); return ret; } static void gst_rtp_g729_pay_recalc_rtp_time (GstRTPG729Pay * rtpg729pay, GstClockTime time) { if (GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts) && GST_CLOCK_TIME_IS_VALID (time) && time >= rtpg729pay->first_ts) { GstClockTime diff; guint32 rtpdiff; diff = time - rtpg729pay->first_ts; rtpdiff = (diff / GST_MSECOND) * 8; rtpg729pay->next_rtp_time = rtpg729pay->first_rtp_time + rtpdiff; GST_DEBUG_OBJECT (rtpg729pay, "elapsed time %" GST_TIME_FORMAT ", rtp %" G_GUINT32_FORMAT ", " "new offset %" G_GUINT32_FORMAT, GST_TIME_ARGS (diff), rtpdiff, rtpg729pay->next_rtp_time); } } static GstFlowReturn gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf) { GstFlowReturn ret = GST_FLOW_OK; GstRTPG729Pay *rtpg729pay = GST_RTP_G729_PAY (payload); GstAdapter *adapter = NULL; guint payload_len; guint available; guint maxptime_octets = G_MAXUINT; guint minptime_octets = 0; guint min_payload_len; guint max_payload_len; gsize size; GstClockTime timestamp; size = gst_buffer_get_size (buf); if (size % G729_FRAME_SIZE != 0 && size % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE) goto invalid_size; /* max number of bytes based on given ptime, has to be multiple of * frame_duration */ if (payload->max_ptime != -1) { guint ptime_ms = payload->max_ptime / GST_MSECOND; maxptime_octets = G729_FRAME_SIZE * (int) (ptime_ms / G729_FRAME_DURATION_MS); if (maxptime_octets < G729_FRAME_SIZE) { GST_WARNING_OBJECT (payload, "Given ptime %" G_GINT64_FORMAT " is smaller than minimum %d ns, overwriting to minimum", payload->max_ptime, G729_FRAME_DURATION_MS); maxptime_octets = G729_FRAME_SIZE; } } max_payload_len = MIN ( /* MTU max */ (int) (gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (payload), 0, 0) / G729_FRAME_SIZE) * G729_FRAME_SIZE, /* ptime max */ maxptime_octets); /* min number of bytes based on a given ptime, has to be a multiple of frame duration */ { guint64 min_ptime = payload->min_ptime; min_ptime = min_ptime / GST_MSECOND; minptime_octets = G729_FRAME_SIZE * (int) (min_ptime / G729_FRAME_DURATION_MS); } min_payload_len = MAX (minptime_octets, G729_FRAME_SIZE); if (min_payload_len > max_payload_len) { min_payload_len = max_payload_len; } /* If the ptime is specified in the caps, tried to adhere to it exactly */ if (payload->ptime) { guint64 ptime = payload->ptime / GST_MSECOND; guint ptime_in_bytes = G729_FRAME_SIZE * (guint) (ptime / G729_FRAME_DURATION_MS); /* clip to computed min and max lengths */ ptime_in_bytes = MAX (min_payload_len, ptime_in_bytes); ptime_in_bytes = MIN (max_payload_len, ptime_in_bytes); min_payload_len = max_payload_len = ptime_in_bytes; } GST_LOG_OBJECT (payload, "Calculated min_payload_len %u and max_payload_len %u", min_payload_len, max_payload_len); adapter = rtpg729pay->adapter; available = gst_adapter_available (adapter); timestamp = GST_BUFFER_PTS (buf); /* resync rtp time on discont or a discontinuous cn packet */ if (GST_BUFFER_IS_DISCONT (buf)) { /* flush remainder */ if (available > 0) { gst_rtp_g729_pay_push (rtpg729pay, gst_adapter_take_buffer_fast (adapter, available)); available = 0; } rtpg729pay->discont = TRUE; gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp); } if (size < G729_FRAME_SIZE) gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp); if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts))) { rtpg729pay->first_ts = timestamp; rtpg729pay->first_rtp_time = rtpg729pay->next_rtp_time; } /* let's reset the base timestamp when the adapter is empty */ if (available == 0) rtpg729pay->next_ts = timestamp; if (available == 0 && size >= min_payload_len && size <= max_payload_len) { ret = gst_rtp_g729_pay_push (rtpg729pay, buf); return ret; } gst_adapter_push (adapter, buf); available = gst_adapter_available (adapter); /* as long as we have full frames */ /* this loop will push all available buffers till the last frame */ while (available >= min_payload_len || available % G729_FRAME_SIZE == G729B_CN_FRAME_SIZE) { /* We send as much as we can */ if (available <= max_payload_len) { payload_len = available; } else { payload_len = MIN (max_payload_len, (available / G729_FRAME_SIZE) * G729_FRAME_SIZE); } ret = gst_rtp_g729_pay_push (rtpg729pay, gst_adapter_take_buffer_fast (adapter, payload_len)); available -= payload_len; } return ret; /* ERRORS */ invalid_size: { GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE, ("Invalid input buffer size"), ("Invalid buffer size, should be a multiple of" " G729_FRAME_SIZE(10) with an optional G729B_CN_FRAME_SIZE(2)" " added to it, but it is %" G_GSIZE_FORMAT, size)); gst_buffer_unref (buf); return GST_FLOW_ERROR; } } static GstStateChangeReturn gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; /* handle upwards state changes here */ switch (transition) { default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); /* handle downwards state changes */ switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_rtp_g729_pay_reset (GST_RTP_G729_PAY (element)); break; default: break; } return ret; } gboolean gst_rtp_g729_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg729pay", GST_RANK_SECONDARY, GST_TYPE_RTP_G729_PAY); }