/* GStreamer
+ * Copyright (C) <2007> Nokia Corporation
+ * Copyright (C) <2007> Collabora Ltd
+ * @author: Olivier Crete <olivier.crete@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* 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"
+#include <config.h>
#endif
-#include "gstrtpg729pay.h"
+#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/base/gstadapter.h>
-/* elementfactory information */
-static GstElementDetails gst_rtpg729pay_details = {
- "RTP Payloader for G729 Audio",
- "Codec/Payloader/Network",
- "Packetize G729 audio streams into RTP packets",
- "Laurent Glayal <spglegle@yahoo.fr>"
-};
+#include "gstrtpg729pay.h"
GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug);
#define GST_CAT_DEFAULT (rtpg729pay_debug)
-static GstStaticPadTemplate gst_rtpg729pay_sink_template =
+#define G729_FRAME_SIZE 10
+#define G729B_CN_FRAME_SIZE 2
+#define G729_FRAME_DURATION (10 * GST_MSECOND)
+#define G729_FRAME_DURATION_MS (10)
+
+static gboolean
+gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps);
+static GstFlowReturn
+gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf);
+
+static GstStateChangeReturn
+gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition);
+
+static GstStaticPadTemplate gst_rtp_g729_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/G729, channels=(int)1, rate=(int)8000")
+ GST_STATIC_CAPS ("audio/G729, " /* according to RFC 3555 */
+ "channels = (int) 1, " "rate = (int) 8000")
);
-static GstStaticPadTemplate gst_rtpg729pay_src_template =
+static GstStaticPadTemplate gst_rtp_g729_pay_src_template =
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) 8000, " "encoding-name = (string) \"G729\";"
+ "payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
+ "clock-rate = (int) 8000, "
+ "encoding-name = (string) \"G729\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) " GST_RTP_PAYLOAD_G729_STRING ", "
- "clock-rate = (int) 8000")
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"")
);
-static gboolean gst_rtpg729pay_setcaps (GstBaseRTPPayload * payload,
- GstCaps * caps);
-
-GST_BOILERPLATE (GstRtpG729Pay, gst_rtpg729pay, GstBaseRTPAudioPayload,
- GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+#define gst_rtp_g729_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRTPG729Pay, gst_rtp_g729_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
-gst_rtpg729pay_base_init (gpointer klass)
+gst_rtp_g729_pay_finalize (GObject * object)
{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstRTPG729Pay *pay = GST_RTP_G729_PAY (object);
+
+ g_object_unref (pay->adapter);
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtpg729pay_sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtpg729pay_src_template));
- gst_element_class_set_details (element_class, &gst_rtpg729pay_details);
+ G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
-gst_rtpg729pay_class_init (GstRtpG729PayClass * klass)
+gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
{
- GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
- GstBaseRTPPayloadClass *gstbasertppayload_class;
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstElementClass *gstelement_class = (GstElementClass *) klass;
+ GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass);
+
+ GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
+ "G.729 RTP Payloader");
- gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+ gobject_class->finalize = gst_rtp_g729_pay_finalize;
- parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
+ gstelement_class->change_state = gst_rtp_g729_pay_change_state;
- gstbasertppayload_class->set_caps = gst_rtpg729pay_setcaps;
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_g729_pay_sink_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_g729_pay_src_template));
- GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
- "G729 audio RTP payloader");
+ gst_element_class_set_details_simple (gstelement_class, "RTP G.729 payloader",
+ "Codec/Payloader/Network/RTP",
+ "Packetize G.729 audio into RTP packets",
+ "Olivier Crete <olivier.crete@collabora.co.uk>");
+
+ payload_class->set_caps = gst_rtp_g729_pay_set_caps;
+ payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer;
}
static void
-gst_rtpg729pay_init (GstRtpG729Pay * rtpg729pay, GstRtpG729PayClass * klass)
+gst_rtp_g729_pay_init (GstRTPG729Pay * pay)
{
- GstBaseRTPPayload *basertppayload;
- GstBaseRTPAudioPayload *basertpaudiopayload;
+ GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
- basertppayload = GST_BASE_RTP_PAYLOAD (rtpg729pay);
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg729pay);
+ payload->pt = GST_RTP_PAYLOAD_G729;
+ gst_rtp_base_payload_set_options (payload, "audio", FALSE, "G729", 8000);
- /* we don't set the payload type, it should be set by the application using
- * the pt property or the default 96 will be used */
- basertppayload->clock_rate = 8000;
+ pay->adapter = gst_adapter_new ();
+}
- /* tell basertpaudiopayload that this is a frame based codec */
- gst_base_rtp_audio_payload_set_frame_based (basertpaudiopayload);
- gst_basertppayload_set_options (basertppayload, "audio", FALSE, "G729", 8000);
- gst_base_rtp_audio_payload_set_frame_options (basertpaudiopayload, 10, 10);
+static void
+gst_rtp_g729_pay_reset (GstRTPG729Pay * pay)
+{
+ gst_adapter_clear (pay->adapter);
+ pay->discont = FALSE;
+ pay->next_rtp_time = 0;
+ pay->first_ts = GST_CLOCK_TIME_NONE;
+ pay->first_rtp_time = 0;
}
static gboolean
-gst_rtpg729pay_setcaps (GstBaseRTPPayload * basertppayload, GstCaps * caps)
+gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
- GstRtpG729Pay *rtpg729pay;
- GstBaseRTPAudioPayload *basertpaudiopayload;
- gboolean ret;
+ gboolean res;
GstStructure *structure;
- const char *payload_name;
-
- rtpg729pay = GST_RTP_G729_PAY (basertppayload);
- basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basertppayload);
+ gint pt;
structure = gst_caps_get_structure (caps, 0);
+ if (!gst_structure_get_int (structure, "payload", &pt))
+ pt = GST_RTP_PAYLOAD_G729;
+
+ payload->pt = pt;
+ payload->dynamic = pt != GST_RTP_PAYLOAD_G729;
+
+ res = gst_rtp_base_payload_set_outcaps (payload, NULL);
+
+ return res;
+}
+
+static GstFlowReturn
+gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay,
+ const guint8 * data, guint payload_len)
+{
+ GstRTPBasePayload *basepayload;
+ GstClockTime duration;
+ guint frames;
+ GstBuffer *outbuf;
+ guint8 *payload;
+ GstFlowReturn ret;
+ GstRTPBuffer rtp = { NULL };
+
+ basepayload = GST_RTP_BASE_PAYLOAD (rtpg729pay);
+
+ GST_DEBUG_OBJECT (rtpg729pay, "Pushing %d bytes ts %" GST_TIME_FORMAT,
+ payload_len, GST_TIME_ARGS (rtpg729pay->next_ts));
+
+ /* create buffer to hold the payload */
+ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+
+ gst_rtp_buffer_map (outbuf, GST_MAP_READWRITE, &rtp);
+
+ /* copy payload */
+ payload = gst_rtp_buffer_get_payload (&rtp);
+ memcpy (payload, data, payload_len);
+
+ /* set metadata */
+ frames =
+ (payload_len / G729_FRAME_SIZE) + ((payload_len % G729_FRAME_SIZE) >> 1);
+ duration = frames * G729_FRAME_DURATION;
+ GST_BUFFER_TIMESTAMP (outbuf) = rtpg729pay->next_ts;
+ GST_BUFFER_DURATION (outbuf) = duration;
+ GST_BUFFER_OFFSET (outbuf) = rtpg729pay->next_rtp_time;
+ rtpg729pay->next_ts += duration;
+ rtpg729pay->next_rtp_time += frames * 80;
+
+ if (G_UNLIKELY (rtpg729pay->discont)) {
+ GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ gst_rtp_buffer_set_marker (&rtp, TRUE);
+ rtpg729pay->discont = FALSE;
+ }
+ gst_rtp_buffer_unmap (&rtp);
+
+ ret = gst_rtp_base_payload_push (basepayload, outbuf);
+
+ return ret;
+}
+
+static void
+gst_rtp_g729_pay_recalc_rtp_time (GstRTPG729Pay * rtpg729pay, GstClockTime time)
+{
+ if (GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts)
+ && GST_CLOCK_TIME_IS_VALID (time) && time >= rtpg729pay->first_ts) {
+ GstClockTime diff;
+ guint32 rtpdiff;
+
+ diff = time - rtpg729pay->first_ts;
+ rtpdiff = (diff / GST_MSECOND) * 8;
+ rtpg729pay->next_rtp_time = rtpg729pay->first_rtp_time + rtpdiff;
+ GST_DEBUG_OBJECT (rtpg729pay,
+ "elapsed time %" GST_TIME_FORMAT ", rtp %" G_GUINT32_FORMAT ", "
+ "new offset %" G_GUINT32_FORMAT, GST_TIME_ARGS (diff), rtpdiff,
+ rtpg729pay->next_rtp_time);
+ }
+}
+
+static GstFlowReturn
+gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstRTPG729Pay *rtpg729pay = GST_RTP_G729_PAY (payload);
+ GstAdapter *adapter = NULL;
+ guint payload_len;
+ guint available;
+ guint maxptime_octets = G_MAXUINT;
+ guint minptime_octets = 0;
+ guint min_payload_len;
+ guint max_payload_len;
+ gsize size;
+ GstClockTime timestamp;
+
+ size = gst_buffer_get_size (buf);
+
+ if (size % G729_FRAME_SIZE != 0 &&
+ size % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE)
+ goto invalid_size;
+
+ /* 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 / GST_MSECOND;
- payload_name = gst_structure_get_name (structure);
- if (g_strcasecmp ("audio/G729", payload_name) != 0)
- goto wrong_name;
+ maxptime_octets = G729_FRAME_SIZE *
+ (int) (ptime_ms / G729_FRAME_DURATION_MS);
- ret = gst_basertppayload_set_outcaps (basertppayload, NULL);
+ if (maxptime_octets < G729_FRAME_SIZE) {
+ GST_WARNING_OBJECT (payload, "Given ptime %" G_GINT64_FORMAT
+ " is smaller than minimum %d ns, overwriting to minimum",
+ payload->max_ptime, G729_FRAME_DURATION_MS);
+ maxptime_octets = G729_FRAME_SIZE;
+ }
+ }
+
+ max_payload_len = MIN (
+ /* MTU max */
+ (int) (gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU
+ (payload), 0, 0) / G729_FRAME_SIZE)
+ * G729_FRAME_SIZE,
+ /* ptime max */
+ maxptime_octets);
+
+ /* min number of bytes based on a given ptime, has to be a multiple
+ of frame duration */
+ {
+ guint64 min_ptime = payload->min_ptime;
+
+ min_ptime = min_ptime / GST_MSECOND;
+ minptime_octets = G729_FRAME_SIZE *
+ (int) (min_ptime / G729_FRAME_DURATION_MS);
+ }
+
+ min_payload_len = MAX (minptime_octets, G729_FRAME_SIZE);
+
+ 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 (payload->ptime) {
+ guint64 ptime = payload->ptime / GST_MSECOND;
+ guint ptime_in_bytes = G729_FRAME_SIZE *
+ (guint) (ptime / G729_FRAME_DURATION_MS);
+
+ /* 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;
+ }
+
+ GST_LOG_OBJECT (payload,
+ "Calculated min_payload_len %u and max_payload_len %u",
+ min_payload_len, max_payload_len);
+
+ adapter = rtpg729pay->adapter;
+ available = gst_adapter_available (adapter);
+
+ timestamp = GST_BUFFER_TIMESTAMP (buf);
+
+ /* resync rtp time on discont or a discontinuous cn packet */
+ if (GST_BUFFER_IS_DISCONT (buf)) {
+ /* flush remainder */
+ if (available > 0) {
+ gst_rtp_g729_pay_push (rtpg729pay,
+ gst_adapter_take (adapter, available), available);
+ available = 0;
+ }
+ rtpg729pay->discont = TRUE;
+ gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
+ }
+
+ if (size < G729_FRAME_SIZE)
+ gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
+
+ if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts))) {
+ rtpg729pay->first_ts = timestamp;
+ rtpg729pay->first_rtp_time = rtpg729pay->next_rtp_time;
+ }
+
+ /* let's reset the base timestamp when the adapter is empty */
+ if (available == 0)
+ rtpg729pay->next_ts = timestamp;
+
+ if (available == 0 && size >= min_payload_len && size <= max_payload_len) {
+ GstMapInfo map;
+
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ ret = gst_rtp_g729_pay_push (rtpg729pay, map.data, map.size);
+ gst_buffer_unmap (buf, &map);
+ gst_buffer_unref (buf);
+ return ret;
+ }
+
+ gst_adapter_push (adapter, buf);
+ available = gst_adapter_available (adapter);
+
+ /* as long as we have full frames */
+ /* this loop will push all available buffers till the last frame */
+ while (available >= min_payload_len ||
+ available % G729_FRAME_SIZE == G729B_CN_FRAME_SIZE) {
+ /* We send as much as we can */
+ if (available <= max_payload_len) {
+ payload_len = available;
+ } else {
+ payload_len = MIN (max_payload_len,
+ (available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
+ }
+
+ ret = gst_rtp_g729_pay_push (rtpg729pay,
+ gst_adapter_take (adapter, payload_len), payload_len);
+ available -= payload_len;
+ }
return ret;
/* ERRORS */
-wrong_name:
+invalid_size:
{
- GST_ERROR_OBJECT (rtpg729pay, "wrong name, expected 'audio/G729', got '%s'",
- payload_name);
- return FALSE;
+ GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
+ ("Invalid input buffer size"),
+ ("Invalid buffer size, should be a multiple of"
+ " G729_FRAME_SIZE(10) with an optional G729B_CN_FRAME_SIZE(2)"
+ " added to it, but it is %" G_GSIZE_FORMAT, size));
+ gst_buffer_unref (buf);
+ return GST_FLOW_ERROR;
+ }
+}
+
+static GstStateChangeReturn
+gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition)
+{
+ GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+
+ /* handle upwards state changes here */
+ switch (transition) {
+ default:
+ break;
}
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ /* handle downwards state changes */
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_rtp_g729_pay_reset (GST_RTP_G729_PAY (element));
+ break;
+ default:
+ break;
+ }
+
+ return ret;
}
gboolean
gst_rtp_g729_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg729pay",
- GST_RANK_NONE, GST_TYPE_RTP_G729_PAY);
+ GST_RANK_SECONDARY, GST_TYPE_RTP_G729_PAY);
}
projects / platform / upstream / gst-plugins-good.git / blobdiff