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Avoid taking the global quark lock for every single buffer. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/295>
1024 lines
32 KiB
C
1024 lines
32 KiB
C
/* GStreamer
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* Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstrtpbaseaudiopayload
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* @title: GstRTPBaseAudioPayload
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* @short_description: Base class for audio RTP payloader
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*
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* Provides a base class for audio RTP payloaders for frame or sample based
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* audio codecs (constant bitrate)
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*
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* This class derives from GstRTPBasePayload. It can be used for payloading
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* audio codecs. It will only work with constant bitrate codecs. It supports
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* both frame based and sample based codecs. It takes care of packing up the
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* audio data into RTP packets and filling up the headers accordingly. The
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* payloading is done based on the maximum MTU (mtu) and the maximum time per
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* packet (max-ptime). The general idea is to divide large data buffers into
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* smaller RTP packets. The RTP packet size is the minimum of either the MTU,
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* max-ptime (if set) or available data. The RTP packet size is always larger or
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* equal to min-ptime (if set). If min-ptime is not set, any residual data is
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* sent in a last RTP packet. In the case of frame based codecs, the resulting
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* RTP packets always contain full frames.
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*
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* ## Usage
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*
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* To use this base class, your child element needs to call either
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* gst_rtp_base_audio_payload_set_frame_based() or
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* gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the
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* element's `_init()` function. Then, the child element must call either
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* gst_rtp_base_audio_payload_set_frame_options(),
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* gst_rtp_base_audio_payload_set_sample_options() or
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* gst_rtp_base_audio_payload_set_samplebits_options. Since
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* GstRTPBaseAudioPayload derives from GstRTPBasePayload, the child element
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* must set any variables or call/override any functions required by that base
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* class. The child element does not need to override any other functions
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* specific to GstRTPBaseAudioPayload.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/base/gstadapter.h>
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#include <gst/audio/audio.h>
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#include "gstrtpbaseaudiopayload.h"
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GST_DEBUG_CATEGORY_STATIC (rtpbaseaudiopayload_debug);
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#define GST_CAT_DEFAULT (rtpbaseaudiopayload_debug)
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#define DEFAULT_BUFFER_LIST FALSE
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enum
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{
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PROP_0,
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PROP_BUFFER_LIST,
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PROP_LAST
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};
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/* function to convert bytes to a time */
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typedef GstClockTime (*GetBytesToTimeFunc) (GstRTPBaseAudioPayload * payload,
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guint64 bytes);
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/* function to convert bytes to a RTP time */
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typedef guint32 (*GetBytesToRTPTimeFunc) (GstRTPBaseAudioPayload * payload,
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guint64 bytes);
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/* function to convert time to bytes */
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typedef guint64 (*GetTimeToBytesFunc) (GstRTPBaseAudioPayload * payload,
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GstClockTime time);
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struct _GstRTPBaseAudioPayloadPrivate
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{
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GetBytesToTimeFunc bytes_to_time;
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GetBytesToRTPTimeFunc bytes_to_rtptime;
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GetTimeToBytesFunc time_to_bytes;
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GstAdapter *adapter;
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guint fragment_size;
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GstClockTime frame_duration_ns;
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gboolean discont;
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guint64 offset;
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GstClockTime last_timestamp;
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guint32 last_rtptime;
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guint align;
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guint cached_mtu;
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guint cached_min_ptime;
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guint cached_max_ptime;
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guint cached_ptime;
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guint cached_min_length;
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guint cached_max_length;
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guint cached_ptime_multiple;
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guint cached_align;
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guint cached_csrc_count;
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gboolean buffer_list;
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};
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static void gst_rtp_base_audio_payload_finalize (GObject * object);
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static void gst_rtp_base_audio_payload_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_rtp_base_audio_payload_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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/* bytes to time functions */
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static GstClockTime
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gst_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload *
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payload, guint64 bytes);
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static GstClockTime
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gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload *
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payload, guint64 bytes);
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/* bytes to RTP time functions */
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static guint32
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gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload *
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payload, guint64 bytes);
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static guint32
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gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload *
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payload, guint64 bytes);
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/* time to bytes functions */
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static guint64
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gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload *
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payload, GstClockTime time);
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static guint64
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gst_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload *
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payload, GstClockTime time);
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static GstFlowReturn gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload
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* payload, GstBuffer * buffer);
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static GstStateChangeReturn gst_rtp_base_payload_audio_change_state (GstElement
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* element, GstStateChange transition);
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static gboolean gst_rtp_base_payload_audio_sink_event (GstRTPBasePayload
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* payload, GstEvent * event);
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/* cached quark to avoid contention on the global quark table lock */
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#define META_TAG_AUDIO meta_tag_audio_quark
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static GQuark meta_tag_audio_quark;
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#define gst_rtp_base_audio_payload_parent_class parent_class
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G_DEFINE_TYPE_WITH_PRIVATE (GstRTPBaseAudioPayload, gst_rtp_base_audio_payload,
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GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_base_audio_payload_class_init (GstRTPBaseAudioPayloadClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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meta_tag_audio_quark = g_quark_from_static_string (GST_META_TAG_AUDIO_STR);
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gobject_class->finalize = gst_rtp_base_audio_payload_finalize;
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gobject_class->set_property = gst_rtp_base_audio_payload_set_property;
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gobject_class->get_property = gst_rtp_base_audio_payload_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST,
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g_param_spec_boolean ("buffer-list", "Buffer List",
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"Use Buffer Lists",
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DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_change_state);
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gstrtpbasepayload_class->handle_buffer =
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GST_DEBUG_FUNCPTR (gst_rtp_base_audio_payload_handle_buffer);
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gstrtpbasepayload_class->sink_event =
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GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_sink_event);
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GST_DEBUG_CATEGORY_INIT (rtpbaseaudiopayload_debug, "rtpbaseaudiopayload", 0,
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"base audio RTP payloader");
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}
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static void
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gst_rtp_base_audio_payload_init (GstRTPBaseAudioPayload * payload)
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{
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payload->priv = gst_rtp_base_audio_payload_get_instance_private (payload);
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/* these need to be set by child object if frame based */
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payload->frame_size = 0;
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payload->frame_duration = 0;
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/* these need to be set by child object if sample based */
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payload->sample_size = 0;
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payload->priv->adapter = gst_adapter_new ();
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payload->priv->buffer_list = DEFAULT_BUFFER_LIST;
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}
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static void
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gst_rtp_base_audio_payload_finalize (GObject * object)
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{
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GstRTPBaseAudioPayload *payload;
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payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);
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g_object_unref (payload->priv->adapter);
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GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
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}
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static void
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gst_rtp_base_audio_payload_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec)
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{
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GstRTPBaseAudioPayload *payload;
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payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);
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switch (prop_id) {
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case PROP_BUFFER_LIST:
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#if 0
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payload->priv->buffer_list = g_value_get_boolean (value);
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#endif
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payload->priv->buffer_list = FALSE;
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_base_audio_payload_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstRTPBaseAudioPayload *payload;
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payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);
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switch (prop_id) {
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case PROP_BUFFER_LIST:
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g_value_set_boolean (value, payload->priv->buffer_list);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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/**
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* gst_rtp_base_audio_payload_set_frame_based:
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* @rtpbaseaudiopayload: a pointer to the element.
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*
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* Tells #GstRTPBaseAudioPayload that the child element is for a frame based
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* audio codec
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*/
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void
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gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *
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rtpbaseaudiopayload)
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{
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g_return_if_fail (rtpbaseaudiopayload != NULL);
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g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL);
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g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL);
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g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL);
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rtpbaseaudiopayload->priv->bytes_to_time =
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gst_rtp_base_audio_payload_frame_bytes_to_time;
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rtpbaseaudiopayload->priv->bytes_to_rtptime =
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gst_rtp_base_audio_payload_frame_bytes_to_rtptime;
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rtpbaseaudiopayload->priv->time_to_bytes =
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gst_rtp_base_audio_payload_frame_time_to_bytes;
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}
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/**
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* gst_rtp_base_audio_payload_set_sample_based:
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* @rtpbaseaudiopayload: a pointer to the element.
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*
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* Tells #GstRTPBaseAudioPayload that the child element is for a sample based
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* audio codec
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*/
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void
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gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *
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rtpbaseaudiopayload)
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{
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g_return_if_fail (rtpbaseaudiopayload != NULL);
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g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL);
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g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL);
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g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL);
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rtpbaseaudiopayload->priv->bytes_to_time =
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gst_rtp_base_audio_payload_sample_bytes_to_time;
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rtpbaseaudiopayload->priv->bytes_to_rtptime =
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gst_rtp_base_audio_payload_sample_bytes_to_rtptime;
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rtpbaseaudiopayload->priv->time_to_bytes =
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gst_rtp_base_audio_payload_sample_time_to_bytes;
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}
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/**
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* gst_rtp_base_audio_payload_set_frame_options:
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* @rtpbaseaudiopayload: a pointer to the element.
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* @frame_duration: The duraction of an audio frame in milliseconds.
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* @frame_size: The size of an audio frame in bytes.
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*
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* Sets the options for frame based audio codecs.
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*
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*/
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void
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gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload
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* rtpbaseaudiopayload, gint frame_duration, gint frame_size)
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{
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GstRTPBaseAudioPayloadPrivate *priv;
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g_return_if_fail (rtpbaseaudiopayload != NULL);
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priv = rtpbaseaudiopayload->priv;
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rtpbaseaudiopayload->frame_duration = frame_duration;
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priv->frame_duration_ns = frame_duration * GST_MSECOND;
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rtpbaseaudiopayload->frame_size = frame_size;
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priv->align = frame_size;
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gst_adapter_clear (priv->adapter);
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GST_DEBUG_OBJECT (rtpbaseaudiopayload, "frame set to %d ms and size %d",
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frame_duration, frame_size);
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}
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/**
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* gst_rtp_base_audio_payload_set_sample_options:
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* @rtpbaseaudiopayload: a pointer to the element.
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* @sample_size: Size per sample in bytes.
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*
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* Sets the options for sample based audio codecs.
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*/
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void
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gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload
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* rtpbaseaudiopayload, gint sample_size)
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{
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g_return_if_fail (rtpbaseaudiopayload != NULL);
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/* sample_size is in bits internally */
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gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
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sample_size * 8);
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}
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/**
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* gst_rtp_base_audio_payload_set_samplebits_options:
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* @rtpbaseaudiopayload: a pointer to the element.
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* @sample_size: Size per sample in bits.
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*
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* Sets the options for sample based audio codecs.
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*/
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void
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gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload
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* rtpbaseaudiopayload, gint sample_size)
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{
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guint fragment_size;
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GstRTPBaseAudioPayloadPrivate *priv;
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g_return_if_fail (rtpbaseaudiopayload != NULL);
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priv = rtpbaseaudiopayload->priv;
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rtpbaseaudiopayload->sample_size = sample_size;
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/* sample_size is in bits and is converted into multiple bytes */
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fragment_size = sample_size;
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while ((fragment_size % 8) != 0)
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fragment_size += fragment_size;
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priv->fragment_size = fragment_size / 8;
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priv->align = priv->fragment_size;
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gst_adapter_clear (priv->adapter);
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GST_DEBUG_OBJECT (rtpbaseaudiopayload,
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"Samplebits set to sample size %d bits", sample_size);
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}
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static void
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gst_rtp_base_audio_payload_set_meta (GstRTPBaseAudioPayload * payload,
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GstBuffer * buffer, guint payload_len, GstClockTime timestamp)
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{
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GstRTPBasePayload *basepayload;
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GstRTPBaseAudioPayloadPrivate *priv;
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GstRTPBuffer rtp = { NULL };
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basepayload = GST_RTP_BASE_PAYLOAD_CAST (payload);
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priv = payload->priv;
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/* set payload type */
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gst_rtp_buffer_map (buffer, GST_MAP_WRITE, &rtp);
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gst_rtp_buffer_set_payload_type (&rtp, basepayload->pt);
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/* set marker bit for disconts */
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if (priv->discont) {
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GST_DEBUG_OBJECT (payload, "Setting marker and DISCONT");
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gst_rtp_buffer_set_marker (&rtp, TRUE);
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GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
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priv->discont = FALSE;
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}
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gst_rtp_buffer_unmap (&rtp);
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GST_BUFFER_PTS (buffer) = timestamp;
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/* get the offset in RTP time */
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GST_BUFFER_OFFSET (buffer) = priv->bytes_to_rtptime (payload, priv->offset);
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priv->offset += payload_len;
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/* Set the duration from the size */
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GST_BUFFER_DURATION (buffer) = priv->bytes_to_time (payload, payload_len);
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/* remember the last rtptime/timestamp pair. We will use this to realign our
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* RTP timestamp after a buffer discont */
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priv->last_rtptime = GST_BUFFER_OFFSET (buffer);
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priv->last_timestamp = timestamp;
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}
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/**
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* gst_rtp_base_audio_payload_push:
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* @baseaudiopayload: a #GstRTPBasePayload
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* @data: (array length=payload_len): data to set as payload
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* @payload_len: length of payload
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* @timestamp: a #GstClockTime
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*
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* Create an RTP buffer and store @payload_len bytes of @data as the
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* payload. Set the timestamp on the new buffer to @timestamp before pushing
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* the buffer downstream.
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*
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* Returns: a #GstFlowReturn
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*/
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GstFlowReturn
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gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
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const guint8 * data, guint payload_len, GstClockTime timestamp)
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{
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GstRTPBasePayload *basepayload;
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GstBuffer *outbuf;
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guint8 *payload;
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GstFlowReturn ret;
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GstRTPBuffer rtp = { NULL };
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basepayload = GST_RTP_BASE_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_base_payload_allocate_output_buffer (basepayload,
|
|
payload_len, 0, 0);
|
|
|
|
/* copy payload */
|
|
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
|
|
payload = gst_rtp_buffer_get_payload (&rtp);
|
|
memcpy (payload, data, payload_len);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
/* set metadata */
|
|
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
|
|
timestamp);
|
|
|
|
ret = gst_rtp_base_payload_push (basepayload, outbuf);
|
|
|
|
return ret;
|
|
}
|
|
|
|
typedef struct
|
|
{
|
|
GstRTPBaseAudioPayload *pay;
|
|
GstBuffer *outbuf;
|
|
} CopyMetaData;
|
|
|
|
static gboolean
|
|
foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)
|
|
{
|
|
CopyMetaData *data = user_data;
|
|
GstRTPBaseAudioPayload *pay = data->pay;
|
|
GstBuffer *outbuf = data->outbuf;
|
|
const GstMetaInfo *info = (*meta)->info;
|
|
const gchar *const *tags = gst_meta_api_type_get_tags (info->api);
|
|
|
|
if (info->transform_func && (!tags || (g_strv_length ((gchar **) tags) == 1
|
|
&& gst_meta_api_type_has_tag (info->api, META_TAG_AUDIO)))) {
|
|
GstMetaTransformCopy copy_data = { FALSE, 0, -1 };
|
|
GST_DEBUG_OBJECT (pay, "copy metadata %s", g_type_name (info->api));
|
|
/* simply copy then */
|
|
info->transform_func (outbuf, *meta, inbuf,
|
|
_gst_meta_transform_copy, ©_data);
|
|
} else {
|
|
GST_DEBUG_OBJECT (pay, "not copying metadata %s", g_type_name (info->api));
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
|
|
baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
|
|
{
|
|
GstRTPBasePayload *basepayload;
|
|
GstRTPBaseAudioPayloadPrivate *priv;
|
|
GstBuffer *outbuf;
|
|
guint payload_len;
|
|
GstFlowReturn ret;
|
|
|
|
priv = baseaudiopayload->priv;
|
|
basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
|
|
|
|
payload_len = gst_buffer_get_size (buffer);
|
|
|
|
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
|
|
payload_len, GST_TIME_ARGS (timestamp));
|
|
|
|
/* create just the RTP header buffer */
|
|
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
|
|
|
|
/* set metadata */
|
|
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
|
|
timestamp);
|
|
|
|
if (priv->buffer_list) {
|
|
GstBufferList *list;
|
|
guint i, len;
|
|
|
|
list = gst_buffer_list_new ();
|
|
len = gst_buffer_list_length (list);
|
|
|
|
for (i = 0; i < len; i++) {
|
|
/* FIXME */
|
|
g_warning ("bufferlist not implemented");
|
|
gst_buffer_list_add (list, outbuf);
|
|
gst_buffer_list_add (list, buffer);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
|
|
ret = gst_rtp_base_payload_push_list (basepayload, list);
|
|
} else {
|
|
CopyMetaData data;
|
|
|
|
/* copy payload */
|
|
data.pay = baseaudiopayload;
|
|
data.outbuf = outbuf;
|
|
gst_buffer_foreach_meta (buffer, foreach_metadata, &data);
|
|
outbuf = gst_buffer_append (outbuf, buffer);
|
|
|
|
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
|
|
ret = gst_rtp_base_payload_push (basepayload, outbuf);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_audio_payload_flush:
|
|
* @baseaudiopayload: a #GstRTPBasePayload
|
|
* @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
|
|
*/
|
|
GstFlowReturn
|
|
gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
|
|
guint payload_len, GstClockTime timestamp)
|
|
{
|
|
GstRTPBasePayload *basepayload;
|
|
GstRTPBaseAudioPayloadPrivate *priv;
|
|
GstBuffer *outbuf;
|
|
GstFlowReturn ret;
|
|
GstAdapter *adapter;
|
|
guint64 distance;
|
|
|
|
priv = baseaudiopayload->priv;
|
|
adapter = priv->adapter;
|
|
|
|
basepayload = GST_RTP_BASE_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_pts (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_rtp_base_audio_payload_push_buffer (baseaudiopayload, buffer,
|
|
timestamp);
|
|
} else {
|
|
GstBuffer *paybuf;
|
|
CopyMetaData data;
|
|
|
|
|
|
/* create buffer to hold the payload */
|
|
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
|
|
|
|
paybuf = gst_adapter_take_buffer_fast (adapter, payload_len);
|
|
|
|
data.pay = baseaudiopayload;
|
|
data.outbuf = outbuf;
|
|
gst_buffer_foreach_meta (paybuf, foreach_metadata, &data);
|
|
outbuf = gst_buffer_append (outbuf, paybuf);
|
|
|
|
/* set metadata */
|
|
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
|
|
timestamp);
|
|
|
|
ret = gst_rtp_base_payload_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_rtp_base_audio_payload_get_lengths (GstRTPBasePayload * basepayload,
|
|
guint csrc_count, guint * min_payload_len, guint * max_payload_len,
|
|
guint * align)
|
|
{
|
|
GstRTPBaseAudioPayload *payload;
|
|
GstRTPBaseAudioPayloadPrivate *priv;
|
|
guint max_mtu, mtu;
|
|
guint maxptime_octets;
|
|
guint minptime_octets;
|
|
guint ptime_mult_octets;
|
|
|
|
payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);
|
|
priv = payload->priv;
|
|
|
|
if (priv->align == 0)
|
|
return FALSE;
|
|
|
|
mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);
|
|
|
|
/* check cached values. Since csrc_count may vary for each packet, we only
|
|
* check whether the new value exceeds the cached value and thus result in
|
|
* smaller payload. */
|
|
if (G_LIKELY (priv->cached_mtu == mtu
|
|
&& priv->cached_ptime_multiple ==
|
|
basepayload->ptime_multiple
|
|
&& priv->cached_ptime == basepayload->ptime
|
|
&& priv->cached_max_ptime == basepayload->max_ptime
|
|
&& priv->cached_min_ptime == basepayload->min_ptime
|
|
&& priv->cached_csrc_count >= csrc_count)) {
|
|
/* if nothing changed, return cached values */
|
|
*min_payload_len = priv->cached_min_length;
|
|
*max_payload_len = priv->cached_max_length;
|
|
*align = priv->cached_align;
|
|
return TRUE;
|
|
}
|
|
|
|
ptime_mult_octets = priv->time_to_bytes (payload,
|
|
basepayload->ptime_multiple);
|
|
*align = ALIGN_DOWN (MAX (priv->align, ptime_mult_octets), priv->align);
|
|
|
|
/* 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, csrc_count);
|
|
/* 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->ptime) {
|
|
guint ptime_in_bytes = priv->time_to_bytes (payload,
|
|
basepayload->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->ptime;
|
|
priv->cached_min_ptime = basepayload->min_ptime;
|
|
priv->cached_max_ptime = basepayload->max_ptime;
|
|
priv->cached_ptime_multiple = basepayload->ptime_multiple;
|
|
priv->cached_min_length = *min_payload_len;
|
|
priv->cached_max_length = *max_payload_len;
|
|
priv->cached_align = *align;
|
|
priv->cached_csrc_count = csrc_count;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* frame conversions functions */
|
|
static GstClockTime
|
|
gst_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload *
|
|
payload, guint64 bytes)
|
|
{
|
|
guint64 framecount;
|
|
|
|
framecount = bytes / payload->frame_size;
|
|
if (G_UNLIKELY (bytes % payload->frame_size))
|
|
framecount++;
|
|
|
|
return framecount * payload->priv->frame_duration_ns;
|
|
}
|
|
|
|
static guint32
|
|
gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload *
|
|
payload, guint64 bytes)
|
|
{
|
|
guint64 framecount;
|
|
guint64 time;
|
|
|
|
framecount = bytes / payload->frame_size;
|
|
if (G_UNLIKELY (bytes % payload->frame_size))
|
|
framecount++;
|
|
|
|
time = framecount * payload->priv->frame_duration_ns;
|
|
|
|
return gst_util_uint64_scale_int (time,
|
|
GST_RTP_BASE_PAYLOAD (payload)->clock_rate, GST_SECOND);
|
|
}
|
|
|
|
static guint64
|
|
gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload *
|
|
payload, GstClockTime time)
|
|
{
|
|
return gst_util_uint64_scale (time, payload->frame_size,
|
|
payload->priv->frame_duration_ns);
|
|
}
|
|
|
|
/* sample conversion functions */
|
|
static GstClockTime
|
|
gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload *
|
|
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_RTP_BASE_PAYLOAD (payload)->clock_rate);
|
|
}
|
|
|
|
static guint32
|
|
gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload *
|
|
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_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload *
|
|
payload, guint64 time)
|
|
{
|
|
guint64 samples;
|
|
|
|
samples = gst_util_uint64_scale_int (time,
|
|
GST_RTP_BASE_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_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
|
|
basepayload, GstBuffer * buffer)
|
|
{
|
|
GstRTPBaseAudioPayload *payload;
|
|
GstRTPBaseAudioPayloadPrivate *priv;
|
|
guint payload_len;
|
|
GstFlowReturn ret;
|
|
guint available;
|
|
guint min_payload_len;
|
|
guint max_payload_len;
|
|
guint align;
|
|
guint size;
|
|
gboolean discont;
|
|
GstClockTime timestamp;
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);
|
|
priv = payload->priv;
|
|
|
|
timestamp = GST_BUFFER_PTS (buffer);
|
|
discont = GST_BUFFER_IS_DISCONT (buffer);
|
|
if (discont) {
|
|
|
|
GST_DEBUG_OBJECT (payload, "Got DISCONT");
|
|
/* flush everything out of the adapter, mark DISCONT */
|
|
ret = gst_rtp_base_audio_payload_flush (payload, -1, -1);
|
|
priv->discont = TRUE;
|
|
|
|
/* 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 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_rtp_base_audio_payload_get_lengths (basepayload,
|
|
gst_rtp_base_payload_get_source_count (basepayload, buffer),
|
|
&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_get_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) &&
|
|
(size % align == 0)) {
|
|
/* 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_rtp_base_audio_payload_push_buffer (payload, buffer, timestamp);
|
|
} 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 */
|
|
/* TODO: Use buffer lists here */
|
|
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_rtp_base_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_rtp_base_payload_audio_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstRTPBaseAudioPayload *rtpbasepayload;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtpbasepayload = GST_RTP_BASE_AUDIO_PAYLOAD (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
rtpbasepayload->priv->cached_mtu = -1;
|
|
rtpbasepayload->priv->last_rtptime = -1;
|
|
rtpbasepayload->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 (rtpbasepayload->priv->adapter);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_base_payload_audio_sink_event (GstRTPBasePayload * basep,
|
|
GstEvent * event)
|
|
{
|
|
GstRTPBaseAudioPayload *payload;
|
|
gboolean res = FALSE;
|
|
|
|
payload = GST_RTP_BASE_AUDIO_PAYLOAD (basep);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
/* flush remaining bytes in the adapter */
|
|
gst_rtp_base_audio_payload_flush (payload, -1, -1);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_adapter_clear (payload->priv->adapter);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* let parent handle the remainder of the event */
|
|
res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (basep, event);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtp_base_audio_payload_get_adapter:
|
|
* @rtpbaseaudiopayload: a #GstRTPBaseAudioPayload
|
|
*
|
|
* Gets the internal adapter used by the depayloader.
|
|
*
|
|
* Returns: (transfer full): a #GstAdapter.
|
|
*/
|
|
GstAdapter *
|
|
gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload
|
|
* rtpbaseaudiopayload)
|
|
{
|
|
GstAdapter *adapter;
|
|
|
|
if ((adapter = rtpbaseaudiopayload->priv->adapter))
|
|
g_object_ref (adapter);
|
|
|
|
return adapter;
|
|
}
|