/* 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. */ /** * SECTION:gstbasertpaudiopayload * @short_description: Base class for audio RTP payloader * * * * Provides a base class for audio RTP payloaders for frame or sample based * audio codecs (constant bitrate) * * * This class derives from GstBaseRTPPayload. It can be used for payloading * audio codecs. It will only work with constant bitrate codecs. It supports * both frame based and sample based codecs. It takes care of packing up the * audio data into RTP packets and filling up the headers accordingly. The * payloading is done based on the maximum MTU (mtu) and the maximum time per * packet (max-ptime). The general idea is to divide large data buffers into * smaller RTP packets. The RTP packet size is the minimum of either the MTU, * max-ptime (if set) or available data. The RTP packet size is always larger or * equal to min-ptime (if set). If min-ptime is not set, any residual data is * sent in a last RTP packet. In the case of frame based codecs, the resulting * RTP packets always contain full frames. * * Usage * * To use this base class, your child element needs to call either * gst_base_rtp_audio_payload_set_frame_based() or * gst_base_rtp_audio_payload_set_sample_based(). This is usually done in the * element's _init() function. Then, the child element must call either * gst_base_rtp_audio_payload_set_frame_options() or * gst_base_rtp_audio_payload_set_sample_options(). Since GstBaseRTPAudioPayload * derives from GstBaseRTPPayload, the child element must set any variables or * call/override any functions required by that base class. The child element * does not need to override any other functions specific to * GstBaseRTPAudioPayload. * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include "gstbasertpaudiopayload.h" GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug); #define GST_CAT_DEFAULT (basertpaudiopayload_debug) typedef enum { AUDIO_CODEC_TYPE_NONE, AUDIO_CODEC_TYPE_FRAME_BASED, AUDIO_CODEC_TYPE_SAMPLE_BASED } AudioCodecType; struct _GstBaseRTPAudioPayloadPrivate { AudioCodecType type; GstAdapter *adapter; guint64 min_ptime; }; #define GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE(o) \ (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_BASE_RTP_AUDIO_PAYLOAD, \ GstBaseRTPAudioPayloadPrivate)) static void gst_base_rtp_audio_payload_finalize (GObject * object); static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); static GstFlowReturn gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer); static GstFlowReturn gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer); static GstStateChangeReturn gst_base_rtp_payload_audio_change_state (GstElement * element, GstStateChange transition); static gboolean gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event); GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); static void gst_base_rtp_audio_payload_base_init (gpointer klass) { } static void gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; g_type_class_add_private (klass, sizeof (GstBaseRTPAudioPayloadPrivate)); gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_finalize); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_change_state); gstbasertppayload_class->handle_buffer = GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_handle_buffer); gstbasertppayload_class->handle_event = GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_handle_event); GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0, "base audio RTP payloader"); } static void gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * basertpaudiopayload, GstBaseRTPAudioPayloadClass * klass) { basertpaudiopayload->priv = GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (basertpaudiopayload); basertpaudiopayload->base_ts = 0; basertpaudiopayload->priv->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; basertpaudiopayload->priv->adapter = gst_adapter_new (); } static void gst_base_rtp_audio_payload_finalize (GObject * object) { GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object); g_object_unref (basertpaudiopayload->priv->adapter); GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object)); } /** * gst_base_rtp_audio_payload_set_frame_based: * @basertpaudiopayload: a pointer to the element. * * Tells #GstBaseRTPAudioPayload that the child element is for a frame based * audio codec * */ void gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload * basertpaudiopayload) { g_return_if_fail (basertpaudiopayload != NULL); g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE); basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_FRAME_BASED; } /** * gst_base_rtp_audio_payload_set_sample_based: * @basertpaudiopayload: a pointer to the element. * * Tells #GstBaseRTPAudioPayload that the child element is for a sample based * audio codec * */ void gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload * basertpaudiopayload) { g_return_if_fail (basertpaudiopayload != NULL); g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE); basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_SAMPLE_BASED; } /** * gst_base_rtp_audio_payload_set_frame_options: * @basertpaudiopayload: a pointer to the element. * @frame_duration: The duraction of an audio frame in milliseconds. * @frame_size: The size of an audio frame in bytes. * * Sets the options for frame based audio codecs. * */ void gst_base_rtp_audio_payload_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; if (basertpaudiopayload->priv->adapter) { gst_adapter_clear (basertpaudiopayload->priv->adapter); } } /** * gst_base_rtp_audio_payload_set_sample_options: * @basertpaudiopayload: a pointer to the element. * @sample_size: Size per sample in bytes. * * Sets the options for sample based audio codecs. * */ void gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload * basertpaudiopayload, gint sample_size) { g_return_if_fail (basertpaudiopayload != NULL); basertpaudiopayload->sample_size = sample_size; if (basertpaudiopayload->priv->adapter) { gst_adapter_clear (basertpaudiopayload->priv->adapter); } } static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstFlowReturn ret; GstBaseRTPAudioPayload *basertpaudiopayload; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); ret = GST_FLOW_ERROR; if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_FRAME_BASED) { ret = gst_base_rtp_audio_payload_handle_frame_based_buffer (basepayload, buffer); } else if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) { ret = gst_base_rtp_audio_payload_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_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstBaseRTPAudioPayload *basertpaudiopayload; guint payload_len; const guint8 *data = NULL; GstFlowReturn ret; guint available; gint frame_size, frame_duration; guint maxptime_octets = G_MAXUINT; guint minptime_octets = 0; guint min_payload_len; guint max_payload_len; gboolean use_adapter = FALSE; guint minptime_ms; ret = GST_FLOW_OK; 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; /* 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; } } max_payload_len = 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); /* min number of bytes based on a given ptime, has to be a multiple of frame duration */ minptime_ms = basepayload->min_ptime / 1000000; minptime_octets = frame_size * (int) (minptime_ms / frame_duration); min_payload_len = MAX (minptime_octets, 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); if (basertpaudiopayload->priv->adapter && gst_adapter_available (basertpaudiopayload->priv->adapter)) { /* If there is always data in the adapter, we have to use it */ gst_adapter_push (basertpaudiopayload->priv->adapter, buffer); available = gst_adapter_available (basertpaudiopayload->priv->adapter); use_adapter = TRUE; } else { /* let's set the base timestamp */ basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer); /* 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 (buffer) >= min_payload_len && GST_BUFFER_SIZE (buffer) <= max_payload_len) { ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer)); gst_buffer_unref (buffer); return ret; } available = GST_BUFFER_SIZE (buffer); data = (guint8 *) GST_BUFFER_DATA (buffer); } /* as long as we have full frames */ while (available >= min_payload_len) { gfloat ts_inc; /* We send as much as we can */ payload_len = MIN (max_payload_len, (available / frame_size) * frame_size); if (use_adapter) { data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len); } ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len, basertpaudiopayload->base_ts); ts_inc = (payload_len * frame_duration) / frame_size; ts_inc = ts_inc * GST_MSECOND; basertpaudiopayload->base_ts += gst_gdouble_to_guint64 (ts_inc); if (use_adapter) { gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len); available = gst_adapter_available (basertpaudiopayload->priv->adapter); } else { available -= payload_len; data += payload_len; } } if (!use_adapter) { if (available != 0 && basertpaudiopayload->priv->adapter) { GstBuffer *buf; buf = gst_buffer_create_sub (buffer, GST_BUFFER_SIZE (buffer) - available, available); gst_adapter_push (basertpaudiopayload->priv->adapter, buf); } else { gst_buffer_unref (buffer); } } return ret; } static GstFlowReturn gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstBaseRTPAudioPayload *basertpaudiopayload; guint payload_len; const guint8 *data = NULL; GstFlowReturn ret; guint available; guint maxptime_octets = G_MAXUINT; guint minptime_octets = 0; guint min_payload_len; guint max_payload_len; gboolean use_adapter = FALSE; guint sample_size; ret = GST_FLOW_OK; 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; /* 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); } max_payload_len = MIN ( /* MTU max */ gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU (basertpaudiopayload), 0, 0), /* ptime max */ maxptime_octets); /* min number of bytes based on a given ptime, has to be a multiple of sample rate */ minptime_octets = basepayload->min_ptime * basepayload->clock_rate / (sample_size * GST_SECOND); min_payload_len = MAX (minptime_octets, sample_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); if (basertpaudiopayload->priv->adapter && gst_adapter_available (basertpaudiopayload->priv->adapter)) { /* If there is always data in the adapter, we have to use it */ gst_adapter_push (basertpaudiopayload->priv->adapter, buffer); available = gst_adapter_available (basertpaudiopayload->priv->adapter); use_adapter = TRUE; } else { /* let's set the base timestamp */ basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer); /* 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 (buffer) >= min_payload_len && GST_BUFFER_SIZE (buffer) <= max_payload_len) { ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer)); gst_buffer_unref (buffer); return ret; } available = GST_BUFFER_SIZE (buffer); data = (guint8 *) GST_BUFFER_DATA (buffer); } while (available >= min_payload_len) { gfloat num, datarate; payload_len = MIN (max_payload_len, (available / sample_size) * sample_size); if (use_adapter) { data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len); } ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len, basertpaudiopayload->base_ts); num = payload_len; datarate = (sample_size * basepayload->clock_rate); basertpaudiopayload->base_ts += /* payload_len (bytes) * nsecs/sec / datarate (bytes*sec) */ gst_gdouble_to_guint64 (num / datarate * GST_SECOND); GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT, GST_TIME_ARGS (basertpaudiopayload->base_ts)); if (use_adapter) { gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len); available = gst_adapter_available (basertpaudiopayload->priv->adapter); } else { available -= payload_len; data += payload_len; } } if (!use_adapter) { if (available != 0 && basertpaudiopayload->priv->adapter) { GstBuffer *buf; buf = gst_buffer_create_sub (buffer, GST_BUFFER_SIZE (buffer) - available, available); gst_adapter_push (basertpaudiopayload->priv->adapter, buf); } else { gst_buffer_unref (buffer); } } return ret; } /** * gst_base_rtp_audio_payload_push: * @basepayload: a #GstBaseRTPPayload * @data: data to set as payload * @payload_len: length of payload * @timestamp: a #GstClockTime * * Create an RTP buffer and store @payload_len bytes of @data as the * payload. Set the timestamp on the new buffer to @timestamp before pushing * the buffer downstream. * * Returns: a #GstFlowReturn * * Since: 0.10.13 */ GstFlowReturn gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload, const guint8 * data, guint payload_len, GstClockTime timestamp) { GstBaseRTPPayload *basepayload; GstBuffer *outbuf; guint8 *payload; GstFlowReturn ret; basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT, payload_len, GST_TIME_ARGS (timestamp)); /* 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; } static GstStateChangeReturn gst_base_rtp_payload_audio_change_state (GstElement * element, GstStateChange transition) { GstBaseRTPAudioPayload *basertppayload; GstStateChangeReturn ret; basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element); ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: if (basertppayload->priv->adapter) { gst_adapter_clear (basertppayload->priv->adapter); } break; default: break; } return ret; } static gboolean gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event) { GstBaseRTPAudioPayload *basertpaudiopayload; gboolean res = TRUE; basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: if (basertpaudiopayload->priv->adapter) { gst_adapter_clear (basertpaudiopayload->priv->adapter); } break; case GST_EVENT_FLUSH_STOP: if (basertpaudiopayload->priv->adapter) { gst_adapter_clear (basertpaudiopayload->priv->adapter); } break; default: break; } gst_object_unref (basertpaudiopayload); return res; } /** * gst_base_rtp_audio_payload_get_adapter: * @basertpaudiopayload: a #GstBaseRTPAudioPayload * * Gets the internal adapter used by the depayloader. * * Returns: a #GstAdapter. * * Since: 0.10.13 */ GstAdapter * gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload * basertpaudiopayload) { GstAdapter *adapter; if ((adapter = basertpaudiopayload->priv->adapter)) g_object_ref (adapter); return adapter; }