mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 13:48:55 +00:00
This is required to add RTP header extensions from the meta automatically. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/674>
385 lines
11 KiB
C
385 lines
11 KiB
C
/* GStreamer RTP SBC payloader
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* BlueZ - Bluetooth protocol stack for Linux
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*
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* Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
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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 Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 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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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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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 <gst/audio/audio.h>
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#include "gstrtpsbcpay.h"
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#include <math.h>
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#include <string.h>
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#include "gstrtputils.h"
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#define RTP_SBC_PAYLOAD_HEADER_SIZE 1
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#define DEFAULT_MIN_FRAMES 0
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#define RTP_SBC_HEADER_TOTAL (12 + RTP_SBC_PAYLOAD_HEADER_SIZE)
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/* BEGIN: Packing for rtp_payload */
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#ifdef _MSC_VER
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#pragma pack(push, 1)
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#endif
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#if G_BYTE_ORDER == G_LITTLE_ENDIAN
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/* FIXME: this seems all a bit over the top for a single byte.. */
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struct rtp_payload
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{
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guint8 frame_count:4;
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guint8 rfa0:1;
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guint8 is_last_fragment:1;
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guint8 is_first_fragment:1;
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guint8 is_fragmented:1;
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}
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#elif G_BYTE_ORDER == G_BIG_ENDIAN
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struct rtp_payload
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{
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guint8 is_fragmented:1;
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guint8 is_first_fragment:1;
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guint8 is_last_fragment:1;
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guint8 rfa0:1;
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guint8 frame_count:4;
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}
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#else
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#error "Unknown byte order"
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#endif
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#ifdef _MSC_VER
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;
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#pragma pack(pop)
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#else
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__attribute__ ((packed));
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#endif
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/* END: Packing for rtp_payload */
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enum
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{
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PROP_0,
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PROP_MIN_FRAMES
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};
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_sbc_pay_debug);
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#define GST_CAT_DEFAULT gst_rtp_sbc_pay_debug
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#define parent_class gst_rtp_sbc_pay_parent_class
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G_DEFINE_TYPE (GstRtpSBCPay, gst_rtp_sbc_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static GstStaticPadTemplate gst_rtp_sbc_pay_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-sbc, "
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"rate = (int) { 16000, 32000, 44100, 48000 }, "
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"channels = (int) [ 1, 2 ], "
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"channel-mode = (string) { mono, dual, stereo, joint }, "
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"blocks = (int) { 4, 8, 12, 16 }, "
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"subbands = (int) { 4, 8 }, "
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"allocation-method = (string) { snr, loudness }, "
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"bitpool = (int) [ 2, 64 ]")
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);
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static GstStaticPadTemplate gst_rtp_sbc_pay_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) audio,"
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) { 16000, 32000, 44100, 48000 },"
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"encoding-name = (string) SBC")
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);
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static void gst_rtp_sbc_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_sbc_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gint
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gst_rtp_sbc_pay_get_frame_len (gint subbands, gint channels,
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gint blocks, gint bitpool, const gchar * channel_mode)
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{
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gint len;
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gint join;
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len = 4 + (4 * subbands * channels) / 8;
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if (strcmp (channel_mode, "mono") == 0 || strcmp (channel_mode, "dual") == 0)
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len += ((blocks * channels * bitpool) + 7) / 8;
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else {
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join = strcmp (channel_mode, "joint") == 0 ? 1 : 0;
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len += ((join * subbands + blocks * bitpool) + 7) / 8;
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}
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return len;
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}
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static gboolean
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gst_rtp_sbc_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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GstRtpSBCPay *sbcpay;
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gint rate, subbands, channels, blocks, bitpool;
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gint frame_len;
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const gchar *channel_mode;
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GstStructure *structure;
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sbcpay = GST_RTP_SBC_PAY (payload);
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "rate", &rate))
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return FALSE;
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if (!gst_structure_get_int (structure, "channels", &channels))
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return FALSE;
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if (!gst_structure_get_int (structure, "blocks", &blocks))
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return FALSE;
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if (!gst_structure_get_int (structure, "bitpool", &bitpool))
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return FALSE;
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if (!gst_structure_get_int (structure, "subbands", &subbands))
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return FALSE;
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channel_mode = gst_structure_get_string (structure, "channel-mode");
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if (!channel_mode)
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return FALSE;
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frame_len = gst_rtp_sbc_pay_get_frame_len (subbands, channels, blocks,
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bitpool, channel_mode);
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sbcpay->frame_length = frame_len;
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sbcpay->frame_duration = ((blocks * subbands) * GST_SECOND) / rate;
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sbcpay->last_timestamp = GST_CLOCK_TIME_NONE;
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gst_rtp_base_payload_set_options (payload, "audio", TRUE, "SBC", rate);
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GST_DEBUG_OBJECT (payload, "calculated frame length: %d ", frame_len);
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return gst_rtp_base_payload_set_outcaps (payload, NULL);
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}
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static GstFlowReturn
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gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
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{
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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guint available;
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guint max_payload;
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GstBuffer *outbuf, *paybuf;
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guint8 *payload_data;
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guint frame_count;
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guint payload_length;
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struct rtp_payload *payload;
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GstFlowReturn res;
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if (sbcpay->frame_length == 0) {
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GST_ERROR_OBJECT (sbcpay, "Frame length is 0");
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return GST_FLOW_ERROR;
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}
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do {
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available = gst_adapter_available (sbcpay->adapter);
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max_payload =
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gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (sbcpay) -
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RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
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max_payload = MIN (max_payload, available);
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frame_count = max_payload / sbcpay->frame_length;
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payload_length = frame_count * sbcpay->frame_length;
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if (payload_length == 0) /* Nothing to send */
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return GST_FLOW_OK;
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outbuf =
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gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
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(sbcpay), RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
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/* get payload */
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gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
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gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (sbcpay));
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/* write header and copy data into payload */
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payload_data = gst_rtp_buffer_get_payload (&rtp);
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payload = (struct rtp_payload *) payload_data;
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memset (payload, 0, sizeof (struct rtp_payload));
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payload->frame_count = frame_count;
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gst_rtp_buffer_unmap (&rtp);
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paybuf = gst_adapter_take_buffer_fast (sbcpay->adapter, payload_length);
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gst_rtp_copy_audio_meta (sbcpay, outbuf, paybuf);
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outbuf = gst_buffer_append (outbuf, paybuf);
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GST_BUFFER_PTS (outbuf) = sbcpay->last_timestamp;
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GST_BUFFER_DURATION (outbuf) = frame_count * sbcpay->frame_duration;
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GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes: %" GST_TIME_FORMAT,
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payload_length, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
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sbcpay->last_timestamp += frame_count * sbcpay->frame_duration;
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res = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (sbcpay), outbuf);
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/* try to send another RTP buffer if available data exceeds MTU size */
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} while (res == GST_FLOW_OK);
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return res;
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}
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static GstFlowReturn
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gst_rtp_sbc_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
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{
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GstRtpSBCPay *sbcpay;
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guint available;
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/* FIXME check for negotiation */
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sbcpay = GST_RTP_SBC_PAY (payload);
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if (GST_BUFFER_IS_DISCONT (buffer)) {
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/* Try to flush whatever's left */
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gst_rtp_sbc_pay_flush_buffers (sbcpay);
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/* Drop the rest */
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gst_adapter_flush (sbcpay->adapter,
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gst_adapter_available (sbcpay->adapter));
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/* Reset timestamps */
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sbcpay->last_timestamp = GST_CLOCK_TIME_NONE;
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}
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if (sbcpay->last_timestamp == GST_CLOCK_TIME_NONE)
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sbcpay->last_timestamp = GST_BUFFER_PTS (buffer);
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gst_adapter_push (sbcpay->adapter, buffer);
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available = gst_adapter_available (sbcpay->adapter);
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if (available + RTP_SBC_HEADER_TOTAL >=
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GST_RTP_BASE_PAYLOAD_MTU (sbcpay) ||
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(available > (sbcpay->min_frames * sbcpay->frame_length)))
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return gst_rtp_sbc_pay_flush_buffers (sbcpay);
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return GST_FLOW_OK;
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}
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static gboolean
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gst_rtp_sbc_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
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{
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GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY (payload);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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gst_rtp_sbc_pay_flush_buffers (sbcpay);
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break;
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default:
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break;
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}
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return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
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}
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static void
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gst_rtp_sbc_pay_finalize (GObject * object)
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{
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GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY (object);
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g_object_unref (sbcpay->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_sbc_pay_class_init (GstRtpSBCPayClass * klass)
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{
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GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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gobject_class->finalize = gst_rtp_sbc_pay_finalize;
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gobject_class->set_property = gst_rtp_sbc_pay_set_property;
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gobject_class->get_property = gst_rtp_sbc_pay_get_property;
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payload_class->set_caps = GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_set_caps);
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payload_class->handle_buffer =
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GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_handle_buffer);
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payload_class->sink_event = GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_sink_event);
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/* properties */
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_MIN_FRAMES,
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g_param_spec_int ("min-frames", "minimum frame number",
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"Minimum quantity of frames to send in one packet "
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"(-1 for maximum allowed by the mtu)",
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-1, G_MAXINT, DEFAULT_MIN_FRAMES, G_PARAM_READWRITE));
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_sbc_pay_sink_factory);
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_sbc_pay_src_factory);
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gst_element_class_set_static_metadata (element_class, "RTP packet payloader",
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"Codec/Payloader/Network", "Payload SBC audio as RTP packets",
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"Thiago Sousa Santos <thiagoss@lcc.ufcg.edu.br>");
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GST_DEBUG_CATEGORY_INIT (gst_rtp_sbc_pay_debug, "rtpsbcpay", 0,
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"RTP SBC payloader");
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}
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static void
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gst_rtp_sbc_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpSBCPay *sbcpay;
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sbcpay = GST_RTP_SBC_PAY (object);
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switch (prop_id) {
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case PROP_MIN_FRAMES:
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sbcpay->min_frames = g_value_get_int (value);
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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_sbc_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpSBCPay *sbcpay;
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sbcpay = GST_RTP_SBC_PAY (object);
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switch (prop_id) {
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case PROP_MIN_FRAMES:
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g_value_set_int (value, sbcpay->min_frames);
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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_sbc_pay_init (GstRtpSBCPay * self)
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{
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self->adapter = gst_adapter_new ();
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self->frame_length = 0;
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self->last_timestamp = GST_CLOCK_TIME_NONE;
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self->min_frames = DEFAULT_MIN_FRAMES;
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}
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gboolean
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gst_rtp_sbc_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpsbcpay", GST_RANK_NONE,
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GST_TYPE_RTP_SBC_PAY);
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}
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