/* GStreamer * Copyright (C) <2006> Philippe Khalaf * * 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 #include #include #include #include "gstbasertpaudiopayload.h" GST_DEBUG_CATEGORY (basertpaudiopayload_debug); #define GST_CAT_DEFAULT (basertpaudiopayload_debug) /* let us define a minimum of 10 ms for sample based codecs */ #define GST_RTP_MIN_PTIME_MS 10 static void gst_basertpaudiopayload_finalize (GObject * object); static GstFlowReturn gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data, guint payload_len, GstClockTime timestamp); static GstFlowReturn gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); static GstFlowReturn gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer); static GstFlowReturn gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer); GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_basertpaudiopayload, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); static void gst_basertpaudiopayload_base_init (gpointer klass) { } static void gst_basertpaudiopayload_class_init (GstBaseRTPAudioPayloadClass * 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_DEBUG_FUNCPTR (gst_basertpaudiopayload_finalize); parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD); gstbasertppayload_class->handle_buffer = GST_DEBUG_FUNCPTR (gst_basertpaudiopayload_handle_buffer); GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0, "base audio RTP payloader"); } static void gst_basertpaudiopayload_init (GstBaseRTPAudioPayload * basertpaudiopayload, GstBaseRTPAudioPayloadClass * klass) { basertpaudiopayload->adapter = gst_adapter_new (); basertpaudiopayload->adapter_base_ts = 0; basertpaudiopayload->type = AUDIO_CODEC_TYPE_NONE; /* these need to be set by child object if frame based */ basertpaudiopayload->frame_size = 0; basertpaudiopayload->frame_duration = 0; /* these need to be set by child object if sample based */ basertpaudiopayload->sample_size = 0; } static void gst_basertpaudiopayload_finalize (GObject * object) { GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object); g_object_unref (basertpaudiopayload->adapter); basertpaudiopayload->adapter = NULL; GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object)); } void gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload * basertpaudiopayload) { g_return_if_fail (basertpaudiopayload != NULL); if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE) { GST_ERROR_OBJECT (basertpaudiopayload, "Codec type already set! You should only set this once!"); } basertpaudiopayload->type = AUDIO_CODEC_TYPE_FRAME_BASED; } void gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload * basertpaudiopayload) { g_return_if_fail (basertpaudiopayload != NULL); if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE) { GST_ERROR_OBJECT (basertpaudiopayload, "Codec type already set! You should only set this once!"); } basertpaudiopayload->type = AUDIO_CODEC_TYPE_SAMPLE_BASED; } /* These are options that need to be set for frame based audio codecs */ void gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload * basertpaudiopayload, gint frame_duration, gint frame_size) { g_return_if_fail (basertpaudiopayload != NULL); basertpaudiopayload->frame_size = frame_size; basertpaudiopayload->frame_duration = frame_duration; } void gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload * basertpaudiopayload, gint sample_size) { g_return_if_fail (basertpaudiopayload != NULL); basertpaudiopayload->sample_size = sample_size; } static GstFlowReturn gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstFlowReturn ret; GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); ret = GST_FLOW_ERROR; if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_FRAME_BASED) { ret = gst_basertpaudiopayload_handle_frame_based_buffer (basepayload, buffer); } else if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) { ret = gst_basertpaudiopayload_handle_sample_based_buffer (basepayload, buffer); } else { GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set"); } return ret; } /* this assumes all frames have a constant duration and a constant size */ static GstFlowReturn gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstBaseRTPAudioPayload *basertpaudiopayload; guint payload_len; guint8 *data; GstFlowReturn ret; guint available; gint frame_size, frame_duration; guint maxptime_octets = G_MAXUINT; ret = GST_FLOW_ERROR; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); if (basertpaudiopayload->frame_size == 0 || basertpaudiopayload->frame_duration == 0) { GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set"); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } frame_size = basertpaudiopayload->frame_size; frame_duration = basertpaudiopayload->frame_duration; /* If buffer fits on an RTP packet, let's just push it through without using * the adapter */ /* this will check again max_ptime and max_mtu */ if (!gst_basertppayload_is_filled (basepayload, gst_rtp_buffer_calc_packet_len (GST_BUFFER_SIZE (buffer), 0, 0), GST_BUFFER_DURATION (buffer))) { ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer)); gst_buffer_unref (buffer); return ret; } /* TODO : would be nice if we had some property that told the payloader to put * just 1 frame per RTP packet, for the moment we can set the ptime to 0 or * something smaller or equal to a frame duration */ /* max number of bytes based on given ptime, has to be multiple of * frame_duration */ if (basepayload->max_ptime != -1) { guint ptime_ms = basepayload->max_ptime / 1000000; maxptime_octets = frame_size * (int) (ptime_ms / frame_duration); if (maxptime_octets == 0) { GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than minimum %d ms, overwriting to minimum", ptime_ms, frame_duration); maxptime_octets = frame_size; } } /* if the adapter is empty (should be), let's set the base timestamp */ if (gst_adapter_available (basertpaudiopayload->adapter) == 0) { basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP (buffer); } else { GST_ERROR_OBJECT (basertpaudiopayload, "Adapter should be empty but is not!"); return GST_FLOW_ERROR; } gst_adapter_push (basertpaudiopayload->adapter, buffer); available = gst_adapter_available (basertpaudiopayload->adapter); /* as long as we have full frames */ /* this loop will always empty the adapter till the last frame */ /* TODO Make it possible to set a minimum size per packet, this way the * algorithm doesn't empty the adapter if there is too little data left and * will wait until the next buffers to arrive */ while (available >= frame_size) { /* we need to see how many frames we can get based on maximum MTU, maximum * ptime and the number of bytes available in the adapter */ payload_len = MIN (MIN ( /* MTU max */ (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU (basertpaudiopayload), 0, 0) / frame_size) * frame_size, /* ptime max */ maxptime_octets), /* currently available */ floor (available / frame_size) * frame_size); data = (guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len); ret = gst_basertpaudiopayload_push (basepayload, data, payload_len, basertpaudiopayload->adapter_base_ts); gst_adapter_flush (basertpaudiopayload->adapter, payload_len); gfloat ts_inc = (payload_len * frame_duration) / frame_size; ts_inc = ts_inc * GST_MSECOND; basertpaudiopayload->adapter_base_ts += ts_inc; GST_DEBUG_OBJECT (basertpaudiopayload, "%f %f %d", ts_inc, ts_inc * GST_MSECOND, (payload_len * frame_duration) / frame_size); GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT, GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts)); available = gst_adapter_available (basertpaudiopayload->adapter); } /* adapter should be freed by now */ if (available != 0) { GST_ERROR_OBJECT (basertpaudiopayload, "Adapter should be empty but is not!"); return GST_FLOW_ERROR; } return ret; } static GstFlowReturn gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstBaseRTPAudioPayload *basertpaudiopayload; guint payload_len; guint8 *data; GstFlowReturn ret; guint available; guint maxptime_octets = G_MAXUINT; guint minptime_octets = 0; guint sample_size; ret = GST_FLOW_ERROR; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); if (basertpaudiopayload->sample_size == 0) { GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set"); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } sample_size = basertpaudiopayload->sample_size; /* If buffer fits on an RTP packet, let's just push it through without using * the adapter */ /* this will check again max_ptime and max_mtu */ if (!gst_basertppayload_is_filled (basepayload, gst_rtp_buffer_calc_packet_len (GST_BUFFER_SIZE (buffer), 0, 0), GST_BUFFER_DURATION (buffer))) { ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer)); gst_buffer_unref (buffer); return ret; } /* max number of bytes based on given ptime */ if (basepayload->max_ptime != -1) { maxptime_octets = basepayload->max_ptime * basepayload->clock_rate / (sample_size * GST_SECOND); minptime_octets = GST_RTP_MIN_PTIME_MS * basepayload->clock_rate / (sample_size * 1000); GST_DEBUG_OBJECT (basertpaudiopayload, "Calculated max_octects %u and min_octets %u", maxptime_octets, minptime_octets); if (maxptime_octets < minptime_octets) { GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than minimum %d, replacing by %d", maxptime_octets, minptime_octets, minptime_octets); maxptime_octets = minptime_octets; } } /* if the adapter is empty (should be), let's set the base timestamp */ if (gst_adapter_available (basertpaudiopayload->adapter) == 0) { basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP (buffer); GST_DEBUG_OBJECT (basertpaudiopayload, "Setting to %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); } gst_adapter_push (basertpaudiopayload->adapter, buffer); available = gst_adapter_available (basertpaudiopayload->adapter); /* as long as we have full frames */ /* this loop will always empty the adapter till the last frame */ /* TODO Make it possible to set a minimum size per packet, this way the * algorithm doesn't empty the adapter if there is too little data left and * will wait until the next buffers to arrive */ while (available >= minptime_octets) { /* we need to see how many frames we can get based on maximum MTU, maximum * ptime and the number of bytes available in the adapter */ payload_len = MIN (MIN ( /* MTU max */ gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU (basertpaudiopayload), 0, 0), /* ptime max */ maxptime_octets), /* currently available */ available); data = (guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len); GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT, GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts)); ret = gst_basertpaudiopayload_push (basepayload, data, payload_len, basertpaudiopayload->adapter_base_ts); gst_adapter_flush (basertpaudiopayload->adapter, payload_len); gfloat num = payload_len; gfloat datarate = (sample_size * basepayload->clock_rate); basertpaudiopayload->adapter_base_ts += /* payload_len (bytes) * nsecs/sec / datarate (bytes*sec) */ num / datarate * GST_SECOND; GST_DEBUG_OBJECT (basertpaudiopayload, "Calculating ts inc %f %f %f", num, datarate, num / datarate * GST_SECOND); GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT, GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts)); available = gst_adapter_available (basertpaudiopayload->adapter); } return ret; } static GstFlowReturn gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data, guint payload_len, GstClockTime timestamp) { GstBuffer *outbuf; guint8 *payload; GstFlowReturn ret; /* create buffer to hold the payload */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); /* copy payload */ gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt); payload = gst_rtp_buffer_get_payload (outbuf); memcpy (payload, data, payload_len); GST_BUFFER_TIMESTAMP (outbuf) = timestamp; ret = gst_basertppayload_push (basepayload, outbuf); return ret; }