/* 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, 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 "gstrtpg729pay.h" /* TODO: fix gstrtpbuffer.h */ #undef GST_RTP_PAYLOAD_G729 #define GST_RTP_PAYLOAD_G729 18 #undef GST_RTP_PAYLOAD_G729_STRING #define GST_RTP_PAYLOAD_G729_STRING "18" #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 (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf); static const GstElementDetails gst_rtp_g729_pay_details = GST_ELEMENT_DETAILS ("RTP G.729 payloader", "Codec/Payloader/Network", "Packetize G.729 audio into RTP packets", "Olivier Crete <olivier.crete@collabora.co.uk>"); 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\"") ); static void gst_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass); GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, GstBaseRTPAudioPayload, GST_TYPE_BASE_RTP_AUDIO_PAYLOAD); static void gst_rtp_g729_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_g729_pay_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_g729_pay_src_template)); gst_element_class_set_details (element_class, &gst_rtp_g729_pay_details); } static void gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass) { GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass); 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, GstRTPG729PayClass * klass) { GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay); GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay); payload->pt = GST_RTP_PAYLOAD_G729; gst_basertppayload_set_options (payload, "audio", FALSE, "G729", 8000); gst_base_rtp_audio_payload_set_frame_based (audiopayload); gst_base_rtp_audio_payload_set_frame_options (audiopayload, G729_FRAME_DURATION_MS, G729_FRAME_SIZE); } static gboolean gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps) { gboolean res; GstStructure *structure; gint pt; structure = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (structure, "payload", &pt)) pt = GST_RTP_PAYLOAD_G729; payload->pt = pt; payload->dynamic = pt != GST_RTP_PAYLOAD_G729; res = gst_basertppayload_set_outcaps (payload, NULL); return res; } static GstFlowReturn gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf) { GstFlowReturn ret = GST_FLOW_OK; GstBaseRTPAudioPayload *basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload); GstAdapter *adapter = NULL; guint payload_len; const guint8 *data = NULL; guint available; guint maxptime_octets = G_MAXUINT; guint minptime_octets = 0; guint min_payload_len; guint max_payload_len; gboolean use_adapter = FALSE; available = GST_BUFFER_SIZE (buf); if (available % G729_FRAME_SIZE != 0 && available % 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 / 1000000; maxptime_octets = G729_FRAME_SIZE * (int) (ptime_ms / G729_FRAME_DURATION_MS); if (maxptime_octets < G729_FRAME_SIZE) { GST_WARNING_OBJECT (basertpaudiopayload, "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_BASE_RTP_PAYLOAD_MTU (basertpaudiopayload), 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; g_object_get (G_OBJECT (payload), "min-ptime", &min_ptime, NULL); min_ptime = min_ptime / 1000000; 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; } GST_DEBUG_OBJECT (basertpaudiopayload, "Calculated min_payload_len %u and max_payload_len %u", min_payload_len, max_payload_len); adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload); if (adapter && gst_adapter_available (adapter)) { /* If there is always data in the adapter, we have to use it */ gst_adapter_push (adapter, buf); available = gst_adapter_available (adapter); use_adapter = TRUE; } else { /* let's set the base timestamp */ basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf); /* If buffer fits on an RTP packet, let's just push it through */ /* this will check against max_ptime and max_mtu */ if (GST_BUFFER_SIZE (buf) >= min_payload_len && GST_BUFFER_SIZE (buf) <= max_payload_len) { ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), GST_BUFFER_TIMESTAMP (buf)); gst_buffer_unref (buf); return ret; } available = GST_BUFFER_SIZE (buf); data = (guint8 *) GST_BUFFER_DATA (buf); } /* 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) { guint num; /* 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); } if (use_adapter) { data = gst_adapter_peek (adapter, payload_len); } ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len, basertpaudiopayload->base_ts); num = payload_len / G729_FRAME_SIZE; basertpaudiopayload->base_ts += G729_FRAME_DURATION * num; if (use_adapter) { gst_adapter_flush (adapter, payload_len); available = gst_adapter_available (adapter); } else { available -= payload_len; data += payload_len; } } if (!use_adapter) { if (available != 0 && adapter) { GstBuffer *buf2; buf2 = gst_buffer_create_sub (buf, GST_BUFFER_SIZE (buf) - available, available); gst_adapter_push (adapter, buf2); } else { gst_buffer_unref (buf); } } if (adapter) { g_object_unref (adapter); } 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 %u", available)); gst_buffer_unref (buf); return GST_FLOW_ERROR; } } gboolean gst_rtp_g729_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg729pay", GST_RANK_NONE, GST_TYPE_RTP_G729_PAY); }