* Boston, MA 02111-1307, USA.
*/
-/*
- * This payloader assumes that the data will ALWAYS come as zero or more
- * 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
- * Any other buffer format won't work
- */
-
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#define GST_RTP_PAYLOAD_G723 4
#define GST_RTP_PAYLOAD_G723_STRING "4"
-/* According to RFC 3551, works only with G723 encoded with 6.3 kb/s high-rate */
-#define G723_FRAME_SIZE 24
-#define G723B_SID_FRAME_SIZE 4
#define G723_FRAME_DURATION (30 * GST_MSECOND)
-#define G723_FRAME_DURATION_MS (30)
-
-static gboolean
-gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps);
-static GstFlowReturn
-gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf);
+static gboolean gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload,
+ GstCaps * caps);
+static GstFlowReturn gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload *
+ payload, GstBuffer * buf);
static const GstElementDetails gst_rtp_g723_pay_details =
GST_ELEMENT_DETAILS ("RTP G.723 payloader",
"Codec/Payloader/Network",
- "Packetize 6.3kb/s G.723 audio into RTP packets",
- "Tiago Katcipis <tiago.katcipis@digitro.com.br>");
+ "Packetize G.723 audio into RTP packets",
+ "Wim Taymans <wim.taymans@gmail.com>");
static GstStaticPadTemplate gst_rtp_g723_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
"clock-rate = (int) 8000, " "encoding-name = (string) \"G723\"")
);
-static void
-gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass);
+static void gst_rtp_g723_pay_init (GstRTPG723Pay * pay,
+ GstRTPG723PayClass * klass);
+static void gst_rtp_g723_pay_finalize (GObject * object);
-GST_BOILERPLATE (GstRTPG723Pay, gst_rtp_g723_pay, GstBaseRTPAudioPayload,
- GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+static GstStateChangeReturn gst_rtp_g723_pay_change_state (GstElement * element,
+ GstStateChange transition);
+
+GST_BOILERPLATE (GstRTPG723Pay, gst_rtp_g723_pay, GstBaseRTPPayload,
+ GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_g723_pay_base_init (gpointer klass)
static void
gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
{
- GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPPayloadClass *payload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ payload_class = (GstBaseRTPPayloadClass *) klass;
+
+ gobject_class->finalize = gst_rtp_g723_pay_finalize;
+
+ gstelement_class->change_state = gst_rtp_g723_pay_change_state;
payload_class->set_caps = gst_rtp_g723_pay_set_caps;
payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass)
{
GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
- GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
+
+ pay->adapter = gst_adapter_new ();
payload->pt = GST_RTP_PAYLOAD_G723;
gst_basertppayload_set_options (payload, "audio", FALSE, "G723", 8000);
+}
- gst_base_rtp_audio_payload_set_frame_based (audiopayload);
- gst_base_rtp_audio_payload_set_frame_options (audiopayload,
- G723_FRAME_DURATION_MS, G723_FRAME_SIZE);
+static void
+gst_rtp_g723_pay_finalize (GObject * object)
+{
+ GstRTPG723Pay *pay;
+
+ pay = GST_RTP_G723_PAY (object);
+ g_object_unref (pay->adapter);
+ pay->adapter = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
}
+
static gboolean
gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
{
}
static GstFlowReturn
-gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
+gst_rtp_g723_pay_flush (GstRTPG723Pay * pay)
{
- GstFlowReturn ret = GST_FLOW_OK;
- GstBaseRTPAudioPayload *basertpaudiopayload =
- GST_BASE_RTP_AUDIO_PAYLOAD (payload);
- GstAdapter *adapter = NULL;
- guint payload_len;
- const guint8 *data = NULL;
- guint available;
- guint maxptime_octets = G_MAXUINT;
- guint minptime_octets = 0;
- guint min_payload_len;
- guint max_payload_len;
- gboolean use_adapter = FALSE;
-
- available = GST_BUFFER_SIZE (buf);
-
- if (available % G723_FRAME_SIZE != 0 &&
- available % G723_FRAME_SIZE != G723B_SID_FRAME_SIZE)
- goto invalid_size;
+ GstBuffer *outbuf;
+ GstFlowReturn ret;
+ guint8 *payload;
+ guint avail;
+
+ avail = gst_adapter_available (pay->adapter);
+
+ outbuf = gst_rtp_buffer_new_allocate (avail, 0, 0);
+ payload = gst_rtp_buffer_get_payload (outbuf);
+
+ GST_BUFFER_TIMESTAMP (outbuf) = pay->timestamp;
+ GST_BUFFER_DURATION (outbuf) = pay->duration;
- /* max number of bytes based on given ptime, has to be multiple of
- * frame_duration */
- if (payload->max_ptime != -1) {
- guint ptime_ms = payload->max_ptime / 1000000;
+ /* copy G723 data as payload */
+ gst_adapter_copy (pay->adapter, payload, 0, avail);
- maxptime_octets = G723_FRAME_SIZE *
- (int) (ptime_ms / G723_FRAME_DURATION_MS);
+ /* flush bytes from adapter */
+ gst_adapter_flush (pay->adapter, avail);
+ pay->timestamp = GST_CLOCK_TIME_NONE;
+ pay->duration = 0;
- if (maxptime_octets < G723_FRAME_SIZE) {
- GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %" G_GINT64_FORMAT
- " is smaller than minimum %d ns, overwriting to minimum",
- payload->max_ptime, G723_FRAME_DURATION_MS);
- maxptime_octets = G723_FRAME_SIZE;
- }
+ /* set discont and marker */
+ if (pay->discont) {
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ gst_rtp_buffer_set_marker (outbuf, TRUE);
+ pay->discont = FALSE;
}
- max_payload_len = MIN (
- /* MTU max */
- (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
- (basertpaudiopayload), 0, 0) / G723_FRAME_SIZE) * G723_FRAME_SIZE,
- /* ptime max */
- maxptime_octets);
+ ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (pay), outbuf);
- /* min number of bytes based on a given ptime, has to be a multiple
- of frame duration */
- {
- guint64 min_ptime;
+ return ret;
+}
- g_object_get (G_OBJECT (payload), "min-ptime", &min_ptime, NULL);
+/* 00 high-rate speech (6.3 kb/s) 24
+ * 01 low-rate speech (5.3 kb/s) 20
+ * 10 SID frame 4
+ * 11 reserved 0 */
+static const guint size_tab[4] = {
+ 24, 20, 4, 0
+};
- min_ptime = min_ptime / 1000000;
- minptime_octets = G723_FRAME_SIZE *
- (int) (min_ptime / G723_FRAME_DURATION_MS);
+static GstFlowReturn
+gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ guint8 *data;
+ guint size;
+ guint8 HDR;
+ GstRTPG723Pay *pay;
+ GstClockTime packet_dur, timestamp;
+ guint payload_len, packet_len;
+
+ pay = GST_RTP_G723_PAY (payload);
+
+ size = GST_BUFFER_SIZE (buf);
+ data = GST_BUFFER_DATA (buf);
+ timestamp = GST_BUFFER_TIMESTAMP (buf);
+
+ if (GST_BUFFER_IS_DISCONT (buf)) {
+ /* flush everything on discont */
+ gst_adapter_clear (pay->adapter);
+ pay->timestamp = GST_CLOCK_TIME_NONE;
+ pay->duration = 0;
+ pay->discont = TRUE;
}
- min_payload_len = MAX (minptime_octets, G723_FRAME_SIZE);
+ /* should be one of these sizes */
+ if (size != 4 && size != 20 && size != 24)
+ goto invalid_size;
- if (min_payload_len > max_payload_len) {
- min_payload_len = max_payload_len;
- }
+ /* check size by looking at the header bits */
+ HDR = data[0] & 0x3;
+ if (size_tab[HDR] != size)
+ goto wrong_size;
- GST_DEBUG_OBJECT (basertpaudiopayload,
- "Calculated min_payload_len %u and max_payload_len %u",
- min_payload_len, max_payload_len);
-
- adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload);
-
- if (adapter && gst_adapter_available (adapter)) {
- /* If there is always data in the adapter, we have to use it */
- gst_adapter_push (adapter, buf);
- available = gst_adapter_available (adapter);
- use_adapter = TRUE;
- } else {
- /* let's set the base timestamp */
- basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
-
- /* If buffer fits on an RTP packet, let's just push it through */
- /* this will check against max_ptime and max_mtu */
- if (GST_BUFFER_SIZE (buf) >= min_payload_len &&
- GST_BUFFER_SIZE (buf) <= max_payload_len) {
- ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
- GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
- GST_BUFFER_TIMESTAMP (buf));
- gst_buffer_unref (buf);
- return ret;
- }
-
- available = GST_BUFFER_SIZE (buf);
- data = (guint8 *) GST_BUFFER_DATA (buf);
- }
+ /* calculate packet size and duration */
+ payload_len = gst_adapter_available (pay->adapter) + size;
+ packet_dur = pay->duration + G723_FRAME_DURATION;
+ packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
- /* as long as we have full frames */
- /* this loop will push all available buffers till the last frame */
- while (available >= min_payload_len ||
- available % G723_FRAME_SIZE == G723B_SID_FRAME_SIZE) {
- guint num;
-
- /* We send as much as we can */
- if (available <= max_payload_len) {
- payload_len = available;
- } else {
- payload_len = MIN (max_payload_len,
- (available / G723_FRAME_SIZE) * G723_FRAME_SIZE);
- }
-
- if (use_adapter) {
- data = gst_adapter_peek (adapter, payload_len);
- }
-
- ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data,
- payload_len, basertpaudiopayload->base_ts);
-
- num = payload_len / G723_FRAME_SIZE;
- basertpaudiopayload->base_ts += G723_FRAME_DURATION * num;
-
- if (use_adapter) {
- gst_adapter_flush (adapter, payload_len);
- available = gst_adapter_available (adapter);
- } else {
- available -= payload_len;
- data += payload_len;
- }
+ if (gst_basertppayload_is_filled (payload, packet_len, packet_dur)) {
+ /* size or duration would overflow the packet, flush the queued data */
+ ret = gst_rtp_g723_pay_flush (pay);
}
- if (!use_adapter) {
- if (available != 0 && adapter) {
- GstBuffer *buf2;
- buf2 = gst_buffer_create_sub (buf,
- GST_BUFFER_SIZE (buf) - available, available);
- gst_adapter_push (adapter, buf2);
- } else {
- gst_buffer_unref (buf);
- }
+ /* update timestamp, we keep the timestamp for the first packet in the adapter
+ * but are able to calculate it from next packets. */
+ if (timestamp != GST_CLOCK_TIME_NONE && pay->timestamp == GST_CLOCK_TIME_NONE) {
+ if (timestamp > pay->duration)
+ pay->timestamp = timestamp - pay->duration;
+ else
+ pay->timestamp = 0;
}
- if (adapter) {
- g_object_unref (adapter);
+ /* add packet to the queue */
+ gst_adapter_push (pay->adapter, buf);
+ pay->duration = packet_dur;
+
+ /* check if we can flush now */
+ if (pay->duration >= payload->min_ptime) {
+ ret = gst_rtp_g723_pay_flush (pay);
}
return ret;
- /* ERRORS */
+ /* WARNINGS */
invalid_size:
{
- GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
+ GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
("Invalid input buffer size"),
- ("Invalid buffer size, should be a multiple of"
- " G723_FRAME_SIZE(24) with an optional G723B_SID_FRAME_SIZE(4)"
- " added to it, but it is %u", available));
+ ("Input size should be 4, 20 or 24, got %u", size));
gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
+ return GST_FLOW_OK;
}
+wrong_size:
+ {
+ GST_ELEMENT_WARNING (pay, STREAM, WRONG_TYPE,
+ ("Wrong input buffer size"),
+ ("Expected input buffer size %u but got %u", size_tab[HDR], size));
+ gst_buffer_unref (buf);
+ return GST_FLOW_OK;
+ }
+}
+
+static GstStateChangeReturn
+gst_rtp_g723_pay_change_state (GstElement * element, GstStateChange transition)
+{
+ GstStateChangeReturn ret;
+ GstRTPG723Pay *pay;
+
+ pay = GST_RTP_G723_PAY (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ gst_adapter_clear (pay->adapter);
+ pay->timestamp = GST_CLOCK_TIME_NONE;
+ pay->duration = 0;
+ pay->discont = TRUE;
+ 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 (pay->adapter);
+ break;
+ default:
+ break;
+ }
+
+ return ret;
}
/*Plugin init functions*/
projects / platform / upstream / gstreamer.git / commitdiff