gstreamer/gst/rtp/gstrtpsbcpay.c

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/* GStreamer RTP SBC payloader
* BlueZ - Bluetooth protocol stack for Linux
*
* Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
*
* 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 <config.h>
#endif
#include <gst/audio/audio.h>
#include "gstrtpsbcpay.h"
#include <math.h>
#include <string.h>
#include "gstrtputils.h"
#define RTP_SBC_PAYLOAD_HEADER_SIZE 1
#define DEFAULT_MIN_FRAMES 0
#define RTP_SBC_HEADER_TOTAL (12 + RTP_SBC_PAYLOAD_HEADER_SIZE)
/* BEGIN: Packing for rtp_payload */
#ifdef _MSC_VER
#pragma pack(push, 1)
#endif
#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;
}
#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;
}
#else
#error "Unknown byte order"
#endif
#ifdef _MSC_VER
;
#pragma pack(pop)
#else
__attribute__ ((packed));
#endif
/* END: Packing for rtp_payload */
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;
sbcpay->frame_duration = ((blocks * subbands) * GST_SECOND) / rate;
sbcpay->last_timestamp = GST_CLOCK_TIME_NONE;
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, *paybuf;
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 (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_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (sbcpay->adapter, payload_length);
gst_rtp_copy_meta (GST_ELEMENT_CAST (sbcpay), outbuf, paybuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
outbuf = gst_buffer_append (outbuf, paybuf);
GST_BUFFER_PTS (outbuf) = sbcpay->last_timestamp;
GST_BUFFER_DURATION (outbuf) = frame_count * sbcpay->frame_duration;
GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes: %" GST_TIME_FORMAT,
payload_length, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
sbcpay->last_timestamp += frame_count * sbcpay->frame_duration;
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);
if (GST_BUFFER_IS_DISCONT (buffer)) {
/* Try to flush whatever's left */
gst_rtp_sbc_pay_flush_buffers (sbcpay);
/* Drop the rest */
gst_adapter_flush (sbcpay->adapter,
gst_adapter_available (sbcpay->adapter));
/* Reset timestamps */
sbcpay->last_timestamp = GST_CLOCK_TIME_NONE;
}
if (sbcpay->last_timestamp == GST_CLOCK_TIME_NONE)
sbcpay->last_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_static_pad_template (element_class,
&gst_rtp_sbc_pay_sink_factory);
gst_element_class_add_static_pad_template (element_class,
&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 <thiagoss@lcc.ufcg.edu.br>");
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->last_timestamp = GST_CLOCK_TIME_NONE;
self->min_frames = DEFAULT_MIN_FRAMES;
}
gboolean
gst_rtp_sbc_pay_plugin_init (GstPlugin * plugin)
{
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return gst_element_register (plugin, "rtpsbcpay", GST_RANK_NONE,
GST_TYPE_RTP_SBC_PAY);
}