diff options
Diffstat (limited to 'gst-libs/gst/rtp/gstbasertpaudiopayload.c')
| -rw-r--r-- | gst-libs/gst/rtp/gstbasertpaudiopayload.c | 968 |
1 files changed, 0 insertions, 968 deletions
diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c deleted file mode 100644 index dddbf498..00000000 --- a/gst-libs/gst/rtp/gstbasertpaudiopayload.c +++ /dev/null @@ -1,968 +0,0 @@ -/* GStreamer - * Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@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. - */ - -/** - * SECTION:gstbasertpaudiopayload - * @short_description: Base class for audio RTP payloader - * - * <refsect2> - * <para> - * Provides a base class for audio RTP payloaders for frame or sample based - * audio codecs (constant bitrate) - * </para> - * <para> - * 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. - * </para> - * <title>Usage</title> - * <para> - * 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(), - * gst_base_rtp_audio_payload_set_sample_options() or - * gst_base_rtp_audio_payload_set_samplebits_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. - * </para> - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include <stdlib.h> -#include <string.h> -#include <gst/rtp/gstrtpbuffer.h> -#include <gst/base/gstadapter.h> - -#include "gstbasertpaudiopayload.h" - -GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug); -#define GST_CAT_DEFAULT (basertpaudiopayload_debug) - -#define DEFAULT_BUFFER_LIST FALSE - -enum -{ - PROP_0, - PROP_BUFFER_LIST, - PROP_LAST -}; - -/* function to convert bytes to a time */ -typedef GstClockTime (*GetBytesToTimeFunc) (GstBaseRTPAudioPayload * payload, - guint64 bytes); -/* function to convert bytes to a RTP time */ -typedef guint32 (*GetBytesToRTPTimeFunc) (GstBaseRTPAudioPayload * payload, - guint64 bytes); -/* function to convert time to bytes */ -typedef guint64 (*GetTimeToBytesFunc) (GstBaseRTPAudioPayload * payload, - GstClockTime time); - -struct _GstBaseRTPAudioPayloadPrivate -{ - GetBytesToTimeFunc bytes_to_time; - GetBytesToRTPTimeFunc bytes_to_rtptime; - GetTimeToBytesFunc time_to_bytes; - - GstAdapter *adapter; - guint fragment_size; - GstClockTime frame_duration_ns; - gboolean discont; - guint64 offset; - GstClockTime last_timestamp; - guint32 last_rtptime; - guint align; - - guint cached_mtu; - guint cached_min_ptime; - guint cached_max_ptime; - guint cached_ptime; - guint cached_min_length; - guint cached_max_length; - - gboolean buffer_list; -}; - - -#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 void gst_base_rtp_audio_payload_set_property (GObject * object, - guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_base_rtp_audio_payload_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec); - -/* bytes to time functions */ -static GstClockTime -gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload * - payload, guint64 bytes); -static GstClockTime -gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload * - payload, guint64 bytes); - -/* bytes to RTP time functions */ -static guint32 -gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload * - payload, guint64 bytes); -static guint32 -gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload * - payload, guint64 bytes); - -/* time to bytes functions */ -static guint64 -gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload * - payload, GstClockTime time); -static guint64 -gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload * - payload, GstClockTime time); - -static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload - * payload, 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_base_rtp_audio_payload_finalize; - gobject_class->set_property = gst_base_rtp_audio_payload_set_property; - gobject_class->get_property = gst_base_rtp_audio_payload_get_property; - - g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST, - g_param_spec_boolean ("buffer-list", "Buffer List", - "Use Buffer Lists", - DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - 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 * payload, - GstBaseRTPAudioPayloadClass * klass) -{ - payload->priv = GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (payload); - - /* these need to be set by child object if frame based */ - payload->frame_size = 0; - payload->frame_duration = 0; - - /* these need to be set by child object if sample based */ - payload->sample_size = 0; - - payload->priv->adapter = gst_adapter_new (); - - payload->priv->buffer_list = DEFAULT_BUFFER_LIST; -} - -static void -gst_base_rtp_audio_payload_finalize (GObject * object) -{ - GstBaseRTPAudioPayload *payload; - - payload = GST_BASE_RTP_AUDIO_PAYLOAD (object); - - g_object_unref (payload->priv->adapter); - - GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object)); -} - -static void -gst_base_rtp_audio_payload_set_property (GObject * object, - guint prop_id, const GValue * value, GParamSpec * pspec) -{ - GstBaseRTPAudioPayload *payload; - - payload = GST_BASE_RTP_AUDIO_PAYLOAD (object); - - switch (prop_id) { - case PROP_BUFFER_LIST: - payload->priv->buffer_list = g_value_get_boolean (value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_base_rtp_audio_payload_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec) -{ - GstBaseRTPAudioPayload *payload; - - payload = GST_BASE_RTP_AUDIO_PAYLOAD (object); - - switch (prop_id) { - case PROP_BUFFER_LIST: - g_value_set_boolean (value, payload->priv->buffer_list); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -/** - * 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->time_to_bytes == NULL); - g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL); - g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL); - - basertpaudiopayload->priv->bytes_to_time = - gst_base_rtp_audio_payload_frame_bytes_to_time; - basertpaudiopayload->priv->bytes_to_rtptime = - gst_base_rtp_audio_payload_frame_bytes_to_rtptime; - basertpaudiopayload->priv->time_to_bytes = - gst_base_rtp_audio_payload_frame_time_to_bytes; -} - -/** - * 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->time_to_bytes == NULL); - g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL); - g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL); - - basertpaudiopayload->priv->bytes_to_time = - gst_base_rtp_audio_payload_sample_bytes_to_time; - basertpaudiopayload->priv->bytes_to_rtptime = - gst_base_rtp_audio_payload_sample_bytes_to_rtptime; - basertpaudiopayload->priv->time_to_bytes = - gst_base_rtp_audio_payload_sample_time_to_bytes; -} - -/** - * 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) -{ - GstBaseRTPAudioPayloadPrivate *priv; - - g_return_if_fail (basertpaudiopayload != NULL); - - priv = basertpaudiopayload->priv; - - basertpaudiopayload->frame_duration = frame_duration; - priv->frame_duration_ns = frame_duration * GST_MSECOND; - basertpaudiopayload->frame_size = frame_size; - priv->align = frame_size; - - gst_adapter_clear (priv->adapter); - - GST_DEBUG_OBJECT (basertpaudiopayload, "frame set to %d ms and size %d", - frame_duration, frame_size); -} - -/** - * 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); - - /* sample_size is in bits internally */ - gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload, - sample_size * 8); -} - -/** - * gst_base_rtp_audio_payload_set_samplebits_options: - * @basertpaudiopayload: a pointer to the element. - * @sample_size: Size per sample in bits. - * - * Sets the options for sample based audio codecs. - * - * Since: 0.10.18 - */ -void -gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload - * basertpaudiopayload, gint sample_size) -{ - guint fragment_size; - GstBaseRTPAudioPayloadPrivate *priv; - - g_return_if_fail (basertpaudiopayload != NULL); - - priv = basertpaudiopayload->priv; - - basertpaudiopayload->sample_size = sample_size; - - /* sample_size is in bits and is converted into multiple bytes */ - fragment_size = sample_size; - while ((fragment_size % 8) != 0) - fragment_size += fragment_size; - priv->fragment_size = fragment_size / 8; - priv->align = priv->fragment_size; - - gst_adapter_clear (priv->adapter); - - GST_DEBUG_OBJECT (basertpaudiopayload, - "Samplebits set to sample size %d bits", sample_size); -} - -static void -gst_base_rtp_audio_payload_set_meta (GstBaseRTPAudioPayload * payload, - GstBuffer * buffer, guint payload_len, GstClockTime timestamp) -{ - GstBaseRTPPayload *basepayload; - GstBaseRTPAudioPayloadPrivate *priv; - - basepayload = GST_BASE_RTP_PAYLOAD_CAST (payload); - priv = payload->priv; - - /* set payload type */ - gst_rtp_buffer_set_payload_type (buffer, basepayload->pt); - /* set marker bit for disconts */ - if (priv->discont) { - GST_DEBUG_OBJECT (payload, "Setting marker and DISCONT"); - gst_rtp_buffer_set_marker (buffer, TRUE); - GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT); - priv->discont = FALSE; - } - GST_BUFFER_TIMESTAMP (buffer) = timestamp; - - /* get the offset in RTP time */ - GST_BUFFER_OFFSET (buffer) = priv->bytes_to_rtptime (payload, priv->offset); - - priv->offset += payload_len; - - /* remember the last rtptime/timestamp pair. We will use this to realign our - * RTP timestamp after a buffer discont */ - priv->last_rtptime = GST_BUFFER_OFFSET (buffer); - priv->last_timestamp = timestamp; -} - -/** - * gst_base_rtp_audio_payload_push: - * @baseaudiopayload: 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 */ - payload = gst_rtp_buffer_get_payload (outbuf); - memcpy (payload, data, payload_len); - - /* set metadata */ - gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, - timestamp); - - ret = gst_basertppayload_push (basepayload, outbuf); - - return ret; -} - -static GstFlowReturn -gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload * - baseaudiopayload, GstBuffer * buffer) -{ - GstBaseRTPPayload *basepayload; - GstBaseRTPAudioPayloadPrivate *priv; - GstBuffer *outbuf; - GstClockTime timestamp; - guint8 *payload; - guint payload_len; - GstFlowReturn ret; - - priv = baseaudiopayload->priv; - basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload); - - payload_len = GST_BUFFER_SIZE (buffer); - timestamp = GST_BUFFER_TIMESTAMP (buffer); - - GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT, - payload_len, GST_TIME_ARGS (timestamp)); - - if (priv->buffer_list) { - /* create just the RTP header buffer */ - outbuf = gst_rtp_buffer_new_allocate (0, 0, 0); - } else { - /* create buffer to hold the payload */ - outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); - } - - /* set metadata */ - gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, - timestamp); - - if (priv->buffer_list) { - GstBufferList *list; - GstBufferListIterator *it; - - list = gst_buffer_list_new (); - it = gst_buffer_list_iterate (list); - - /* add both buffers to the buffer list */ - gst_buffer_list_iterator_add_group (it); - gst_buffer_list_iterator_add (it, outbuf); - gst_buffer_list_iterator_add (it, buffer); - - gst_buffer_list_iterator_free (it); - - GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list); - ret = gst_basertppayload_push_list (basepayload, list); - } else { - /* copy payload */ - payload = gst_rtp_buffer_get_payload (outbuf); - memcpy (payload, GST_BUFFER_DATA (buffer), payload_len); - gst_buffer_unref (buffer); - - GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf); - ret = gst_basertppayload_push (basepayload, outbuf); - } - - return ret; -} - -/** - * gst_base_rtp_audio_payload_flush: - * @baseaudiopayload: a #GstBaseRTPPayload - * @payload_len: length of payload - * @timestamp: a #GstClockTime - * - * Create an RTP buffer and store @payload_len bytes of the adapter as the - * payload. Set the timestamp on the new buffer to @timestamp before pushing - * the buffer downstream. - * - * If @payload_len is -1, all pending bytes will be flushed. If @timestamp is - * -1, the timestamp will be calculated automatically. - * - * Returns: a #GstFlowReturn - * - * Since: 0.10.25 - */ -GstFlowReturn -gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload, - guint payload_len, GstClockTime timestamp) -{ - GstBaseRTPPayload *basepayload; - GstBaseRTPAudioPayloadPrivate *priv; - GstBuffer *outbuf; - guint8 *payload; - GstFlowReturn ret; - GstAdapter *adapter; - guint64 distance; - - priv = baseaudiopayload->priv; - adapter = priv->adapter; - - basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload); - - if (payload_len == -1) - payload_len = gst_adapter_available (adapter); - - /* nothing to do, just return */ - if (payload_len == 0) - return GST_FLOW_OK; - - if (timestamp == -1) { - /* calculate the timestamp */ - timestamp = gst_adapter_prev_timestamp (adapter, &distance); - - GST_LOG_OBJECT (baseaudiopayload, - "last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT, - GST_TIME_ARGS (timestamp), distance); - - if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) { - /* convert the number of bytes since the last timestamp to time and add to - * the last seen timestamp */ - timestamp += priv->bytes_to_time (baseaudiopayload, distance); - } - } - - GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT, - payload_len, GST_TIME_ARGS (timestamp)); - - if (priv->buffer_list && gst_adapter_available_fast (adapter) >= payload_len) { - GstBuffer *buffer; - /* we can quickly take a buffer out of the adapter without having to copy - * anything. */ - buffer = gst_adapter_take_buffer (adapter, payload_len); - - ret = gst_base_rtp_audio_payload_push_buffer (baseaudiopayload, buffer); - } else { - /* create buffer to hold the payload */ - outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); - - /* copy payload */ - payload = gst_rtp_buffer_get_payload (outbuf); - gst_adapter_copy (adapter, payload, 0, payload_len); - gst_adapter_flush (adapter, payload_len); - - /* set metadata */ - gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, - timestamp); - - ret = gst_basertppayload_push (basepayload, outbuf); - } - - return ret; -} - -#define ALIGN_DOWN(val,len) ((val) - ((val) % (len))) - -/* calculate the min and max length of a packet. This depends on the configured - * mtu and min/max_ptime values. We cache those so that we don't have to redo - * all the calculations */ -static gboolean -gst_base_rtp_audio_payload_get_lengths (GstBaseRTPPayload * - basepayload, guint * min_payload_len, guint * max_payload_len, - guint * align) -{ - GstBaseRTPAudioPayload *payload; - GstBaseRTPAudioPayloadPrivate *priv; - guint max_mtu, mtu; - guint maxptime_octets; - guint minptime_octets; - - payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload); - priv = payload->priv; - - if (priv->align == 0) - return FALSE; - - *align = priv->align; - - mtu = GST_BASE_RTP_PAYLOAD_MTU (payload); - - /* check cached values */ - if (G_LIKELY (priv->cached_mtu == mtu - && priv->cached_ptime == basepayload->abidata.ABI.ptime - && priv->cached_max_ptime == basepayload->max_ptime - && priv->cached_min_ptime == basepayload->min_ptime)) { - /* if nothing changed, return cached values */ - *min_payload_len = priv->cached_min_length; - *max_payload_len = priv->cached_max_length; - return TRUE; - } - - /* ptime max */ - if (basepayload->max_ptime != -1) { - maxptime_octets = priv->time_to_bytes (payload, basepayload->max_ptime); - } else { - maxptime_octets = G_MAXUINT; - } - /* MTU max */ - max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, 0); - /* round down to alignment */ - max_mtu = ALIGN_DOWN (max_mtu, *align); - - /* combine max ptime and max payload length */ - *max_payload_len = MIN (max_mtu, maxptime_octets); - - /* min number of bytes based on a given ptime */ - minptime_octets = priv->time_to_bytes (payload, basepayload->min_ptime); - /* must be at least one frame size */ - *min_payload_len = MAX (minptime_octets, *align); - - 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 (basepayload->abidata.ABI.ptime) { - guint ptime_in_bytes = priv->time_to_bytes (payload, - basepayload->abidata.ABI.ptime); - - /* 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; - } - - /* cache values */ - priv->cached_mtu = mtu; - priv->cached_ptime = basepayload->abidata.ABI.ptime; - priv->cached_min_ptime = basepayload->min_ptime; - priv->cached_max_ptime = basepayload->max_ptime; - priv->cached_min_length = *min_payload_len; - priv->cached_max_length = *max_payload_len; - - return TRUE; -} - -/* frame conversions functions */ -static GstClockTime -gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload * - payload, guint64 bytes) -{ - return (bytes / payload->frame_size) * (payload->priv->frame_duration_ns); -} - -static guint32 -gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload * - payload, guint64 bytes) -{ - guint64 time; - - time = (bytes / payload->frame_size) * (payload->priv->frame_duration_ns); - - return gst_util_uint64_scale_int (time, - GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND); -} - -static guint64 -gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload * - payload, GstClockTime time) -{ - return gst_util_uint64_scale (time, payload->frame_size, - payload->priv->frame_duration_ns); -} - -/* sample conversion functions */ -static GstClockTime -gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload * - payload, guint64 bytes) -{ - guint64 rtptime; - - /* avoid division when we can */ - if (G_LIKELY (payload->sample_size != 8)) - rtptime = gst_util_uint64_scale_int (bytes, 8, payload->sample_size); - else - rtptime = bytes; - - return gst_util_uint64_scale_int (rtptime, GST_SECOND, - GST_BASE_RTP_PAYLOAD (payload)->clock_rate); -} - -static guint32 -gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload * - payload, guint64 bytes) -{ - /* avoid division when we can */ - if (G_LIKELY (payload->sample_size != 8)) - return gst_util_uint64_scale_int (bytes, 8, payload->sample_size); - else - return bytes; -} - -static guint64 -gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload * - payload, guint64 time) -{ - guint64 samples; - - samples = gst_util_uint64_scale_int (time, - GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND); - - /* avoid multiplication when we can */ - if (G_LIKELY (payload->sample_size != 8)) - return gst_util_uint64_scale_int (samples, payload->sample_size, 8); - else - return samples; -} - -static GstFlowReturn -gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * - basepayload, GstBuffer * buffer) -{ - GstBaseRTPAudioPayload *payload; - GstBaseRTPAudioPayloadPrivate *priv; - guint payload_len; - GstFlowReturn ret; - guint available; - guint min_payload_len; - guint max_payload_len; - guint align; - guint size; - gboolean discont; - - ret = GST_FLOW_OK; - - payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload); - priv = payload->priv; - - discont = GST_BUFFER_IS_DISCONT (buffer); - if (discont) { - GstClockTime timestamp; - - GST_DEBUG_OBJECT (payload, "Got DISCONT"); - /* flush everything out of the adapter, mark DISCONT */ - ret = gst_base_rtp_audio_payload_flush (payload, -1, -1); - priv->discont = TRUE; - - timestamp = GST_BUFFER_TIMESTAMP (buffer); - - /* get the distance between the timestamp gap and produce the same gap in - * the RTP timestamps */ - if (priv->last_timestamp != -1 && timestamp != -1) { - /* we had a last timestamp, compare it to the new timestamp and update the - * offset counter for RTP timestamps. The effect is that we will produce - * output buffers containing the same RTP timestamp gap as the gap - * between the GST timestamps. */ - if (timestamp > priv->last_timestamp) { - GstClockTime diff; - guint64 bytes; - /* we're only going to apply a positive gap, otherwise we let the marker - * bit do its thing. simply convert to bytes and add the the current - * offset */ - diff = timestamp - priv->last_timestamp; - bytes = priv->time_to_bytes (payload, diff); - priv->offset += bytes; - - GST_DEBUG_OBJECT (payload, - "elapsed time %" GST_TIME_FORMAT ", bytes %" G_GUINT64_FORMAT - ", new offset %" G_GUINT64_FORMAT, GST_TIME_ARGS (diff), bytes, - priv->offset); - } - } - } - - if (!gst_base_rtp_audio_payload_get_lengths (basepayload, &min_payload_len, - &max_payload_len, &align)) - goto config_error; - - GST_DEBUG_OBJECT (payload, - "Calculated min_payload_len %u and max_payload_len %u", - min_payload_len, max_payload_len); - - size = GST_BUFFER_SIZE (buffer); - - /* shortcut, we don't need to use the adapter when the packet can be pushed - * through directly. */ - available = gst_adapter_available (priv->adapter); - - GST_DEBUG_OBJECT (payload, "got buffer size %u, available %u", - size, available); - - if (available == 0 && (size >= min_payload_len && size <= max_payload_len)) { - /* If buffer fits on an RTP packet, let's just push it through - * this will check against max_ptime and max_mtu */ - GST_DEBUG_OBJECT (payload, "Fast packet push"); - ret = gst_base_rtp_audio_payload_push_buffer (payload, buffer); - } else { - /* push the buffer in the adapter */ - gst_adapter_push (priv->adapter, buffer); - available += size; - - GST_DEBUG_OBJECT (payload, "available now %u", available); - - /* as long as we have full frames */ - while (available >= min_payload_len) { - /* get multiple of alignment */ - payload_len = MIN (max_payload_len, available); - payload_len = ALIGN_DOWN (payload_len, align); - - /* and flush out the bytes from the adapter, automatically set the - * timestamp. */ - ret = gst_base_rtp_audio_payload_flush (payload, payload_len, -1); - - available -= payload_len; - GST_DEBUG_OBJECT (payload, "available after push %u", available); - } - } - return ret; - - /* ERRORS */ -config_error: - { - GST_ELEMENT_ERROR (payload, STREAM, NOT_IMPLEMENTED, (NULL), - ("subclass did not configure us properly")); - gst_buffer_unref (buffer); - return GST_FLOW_ERROR; - } -} - -static GstStateChangeReturn -gst_base_rtp_payload_audio_change_state (GstElement * element, - GstStateChange transition) -{ - GstBaseRTPAudioPayload *basertppayload; - GstStateChangeReturn ret; - - basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element); - - switch (transition) { - case GST_STATE_CHANGE_READY_TO_PAUSED: - basertppayload->priv->cached_mtu = -1; - basertppayload->priv->last_rtptime = -1; - basertppayload->priv->last_timestamp = -1; - break; - default: - break; - } - - ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PAUSED_TO_READY: - 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 *payload; - gboolean res = FALSE; - - payload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - /* flush remaining bytes in the adapter */ - gst_base_rtp_audio_payload_flush (payload, -1, -1); - break; - case GST_EVENT_FLUSH_STOP: - gst_adapter_clear (payload->priv->adapter); - break; - default: - break; - } - - gst_object_unref (payload); - - /* return FALSE to let parent handle the remainder of the event */ - 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; -} |