/* GStreamer RTP SBC payloader * BlueZ - Bluetooth protocol stack for Linux * * Copyright (C) 2004-2010 Marcel Holtmann * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA * */ #ifdef HAVE_CONFIG_H #include #endif #include "gstrtpsbcpay.h" #include #include #define RTP_SBC_PAYLOAD_HEADER_SIZE 1 #define DEFAULT_MIN_FRAMES 0 #define RTP_SBC_HEADER_TOTAL (12 + RTP_SBC_PAYLOAD_HEADER_SIZE) #if G_BYTE_ORDER == G_LITTLE_ENDIAN /* FIXME: this seems all a bit over the top for a single byte.. */ struct rtp_payload { guint8 frame_count:4; guint8 rfa0:1; guint8 is_last_fragment:1; guint8 is_first_fragment:1; guint8 is_fragmented:1; } __attribute__ ((packed)); #elif G_BYTE_ORDER == G_BIG_ENDIAN struct rtp_payload { guint8 is_fragmented:1; guint8 is_first_fragment:1; guint8 is_last_fragment:1; guint8 rfa0:1; guint8 frame_count:4; } __attribute__ ((packed)); #else #error "Unknown byte order" #endif enum { PROP_0, PROP_MIN_FRAMES }; GST_DEBUG_CATEGORY_STATIC (gst_rtp_sbc_pay_debug); #define GST_CAT_DEFAULT gst_rtp_sbc_pay_debug #define parent_class gst_rtp_sbc_pay_parent_class G_DEFINE_TYPE (GstRtpSBCPay, gst_rtp_sbc_pay, GST_TYPE_RTP_BASE_PAYLOAD); static GstStaticPadTemplate gst_rtp_sbc_pay_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-sbc, " "rate = (int) { 16000, 32000, 44100, 48000 }, " "channels = (int) [ 1, 2 ], " "channel-mode = (string) { mono, dual, stereo, joint }, " "blocks = (int) { 4, 8, 12, 16 }, " "subbands = (int) { 4, 8 }, " "allocation-method = (string) { snr, loudness }, " "bitpool = (int) [ 2, 64 ]") ); static GstStaticPadTemplate gst_rtp_sbc_pay_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) audio," "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) { 16000, 32000, 44100, 48000 }," "encoding-name = (string) SBC") ); static void gst_rtp_sbc_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_sbc_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gint gst_rtp_sbc_pay_get_frame_len (gint subbands, gint channels, gint blocks, gint bitpool, const gchar * channel_mode) { gint len; gint join; len = 4 + (4 * subbands * channels) / 8; if (strcmp (channel_mode, "mono") == 0 || strcmp (channel_mode, "dual") == 0) len += ((blocks * channels * bitpool) + 7) / 8; else { join = strcmp (channel_mode, "joint") == 0 ? 1 : 0; len += ((join * subbands + blocks * bitpool) + 7) / 8; } return len; } static gboolean gst_rtp_sbc_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps) { GstRtpSBCPay *sbcpay; gint rate, subbands, channels, blocks, bitpool; gint frame_len; const gchar *channel_mode; GstStructure *structure; sbcpay = GST_RTP_SBC_PAY (payload); structure = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (structure, "rate", &rate)) return FALSE; if (!gst_structure_get_int (structure, "channels", &channels)) return FALSE; if (!gst_structure_get_int (structure, "blocks", &blocks)) return FALSE; if (!gst_structure_get_int (structure, "bitpool", &bitpool)) return FALSE; if (!gst_structure_get_int (structure, "subbands", &subbands)) return FALSE; channel_mode = gst_structure_get_string (structure, "channel-mode"); if (!channel_mode) return FALSE; frame_len = gst_rtp_sbc_pay_get_frame_len (subbands, channels, blocks, bitpool, channel_mode); sbcpay->frame_length = frame_len; gst_rtp_base_payload_set_options (payload, "audio", TRUE, "SBC", rate); GST_DEBUG_OBJECT (payload, "calculated frame length: %d ", frame_len); return gst_rtp_base_payload_set_outcaps (payload, NULL); } static GstFlowReturn gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay) { GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; guint available; guint max_payload; GstBuffer *outbuf; guint8 *payload_data; guint frame_count; guint payload_length; struct rtp_payload *payload; if (sbcpay->frame_length == 0) { GST_ERROR_OBJECT (sbcpay, "Frame length is 0"); return GST_FLOW_ERROR; } available = gst_adapter_available (sbcpay->adapter); max_payload = gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (sbcpay) - RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0); max_payload = MIN (max_payload, available); frame_count = max_payload / sbcpay->frame_length; payload_length = frame_count * sbcpay->frame_length; if (payload_length == 0) /* Nothing to send */ return GST_FLOW_OK; outbuf = gst_rtp_buffer_new_allocate (payload_length + RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0); /* get payload */ gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (sbcpay)); /* write header and copy data into payload */ payload_data = gst_rtp_buffer_get_payload (&rtp); payload = (struct rtp_payload *) payload_data; memset (payload, 0, sizeof (struct rtp_payload)); payload->frame_count = frame_count; gst_adapter_copy (sbcpay->adapter, payload_data + RTP_SBC_PAYLOAD_HEADER_SIZE, 0, payload_length); gst_rtp_buffer_unmap (&rtp); gst_adapter_flush (sbcpay->adapter, payload_length); /* FIXME: what about duration? */ GST_BUFFER_PTS (outbuf) = sbcpay->timestamp; GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", payload_length); return gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (sbcpay), outbuf); } static GstFlowReturn gst_rtp_sbc_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer) { GstRtpSBCPay *sbcpay; guint available; /* FIXME check for negotiation */ sbcpay = GST_RTP_SBC_PAY (payload); sbcpay->timestamp = GST_BUFFER_PTS (buffer); gst_adapter_push (sbcpay->adapter, buffer); available = gst_adapter_available (sbcpay->adapter); if (available + RTP_SBC_HEADER_TOTAL >= GST_RTP_BASE_PAYLOAD_MTU (sbcpay) || (available > (sbcpay->min_frames * sbcpay->frame_length))) return gst_rtp_sbc_pay_flush_buffers (sbcpay); return GST_FLOW_OK; } static gboolean gst_rtp_sbc_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event) { GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY (payload); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: gst_rtp_sbc_pay_flush_buffers (sbcpay); break; default: break; } return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event); } static void gst_rtp_sbc_pay_finalize (GObject * object) { GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY (object); g_object_unref (sbcpay->adapter); GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object)); } static void gst_rtp_sbc_pay_class_init (GstRtpSBCPayClass * klass) { GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GObjectClass *gobject_class = G_OBJECT_CLASS (klass); gobject_class->finalize = gst_rtp_sbc_pay_finalize; gobject_class->set_property = gst_rtp_sbc_pay_set_property; gobject_class->get_property = gst_rtp_sbc_pay_get_property; payload_class->set_caps = GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_set_caps); payload_class->handle_buffer = GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_handle_buffer); payload_class->sink_event = GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_sink_event); /* properties */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MIN_FRAMES, g_param_spec_int ("min-frames", "minimum frame number", "Minimum quantity of frames to send in one packet " "(-1 for maximum allowed by the mtu)", -1, G_MAXINT, DEFAULT_MIN_FRAMES, G_PARAM_READWRITE)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_sbc_pay_sink_factory)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_sbc_pay_src_factory)); gst_element_class_set_static_metadata (element_class, "RTP packet payloader", "Codec/Payloader/Network", "Payload SBC audio as RTP packets", "Thiago Sousa Santos "); GST_DEBUG_CATEGORY_INIT (gst_rtp_sbc_pay_debug, "rtpsbcpay", 0, "RTP SBC payloader"); } static void gst_rtp_sbc_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpSBCPay *sbcpay; sbcpay = GST_RTP_SBC_PAY (object); switch (prop_id) { case PROP_MIN_FRAMES: sbcpay->min_frames = g_value_get_int (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_sbc_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpSBCPay *sbcpay; sbcpay = GST_RTP_SBC_PAY (object); switch (prop_id) { case PROP_MIN_FRAMES: g_value_set_int (value, sbcpay->min_frames); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_sbc_pay_init (GstRtpSBCPay * self) { self->adapter = gst_adapter_new (); self->frame_length = 0; self->timestamp = 0; self->min_frames = DEFAULT_MIN_FRAMES; } gboolean gst_rtp_sbc_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpsbcpay", GST_RANK_NONE, GST_TYPE_RTP_SBC_PAY); }