#define G729_FRAME_DURATION_MS (10)
static gboolean
-gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps);
+gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps);
static GstFlowReturn
-gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf);
+gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf);
-
-static const GstElementDetails gst_rtp_g729_pay_details =
-GST_ELEMENT_DETAILS ("RTP G.729 payloader",
- "Codec/Payloader/Network",
- "Packetize G.729 audio into RTP packets",
- "Olivier Crete <olivier.crete@collabora.co.uk>");
+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",
"clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"")
);
-GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, 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_rtp_g729_pay_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);
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_g729_pay_sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_g729_pay_src_template));
- gst_element_class_set_details (element_class, &gst_rtp_g729_pay_details);
+ g_object_unref (pay->adapter);
- GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
- "G.729 RTP Payloader");
+ G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
{
- GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
+ 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->finalize = gst_rtp_g729_pay_finalize;
+
+ gstelement_class->change_state = gst_rtp_g729_pay_change_state;
+
+ 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_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_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass)
+gst_rtp_g729_pay_init (GstRTPG729Pay * pay)
{
- GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
- GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
+ GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
payload->pt = GST_RTP_PAYLOAD_G729;
- gst_basertppayload_set_options (payload, "audio", FALSE, "G729", 8000);
+ gst_rtp_base_payload_set_options (payload, "audio", FALSE, "G729", 8000);
- gst_base_rtp_audio_payload_set_frame_based (audiopayload);
- gst_base_rtp_audio_payload_set_frame_options (audiopayload,
- G729_FRAME_DURATION_MS, G729_FRAME_SIZE);
+ pay->adapter = gst_adapter_new ();
+}
+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_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
+gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
GstStructure *structure;
payload->pt = pt;
payload->dynamic = pt != GST_RTP_PAYLOAD_G729;
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static GstFlowReturn
-gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
+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;
- GstBaseRTPAudioPayload *basertpaudiopayload =
- GST_BASE_RTP_AUDIO_PAYLOAD (payload);
+ GstRTPG729Pay *rtpg729pay = GST_RTP_G729_PAY (payload);
GstAdapter *adapter = NULL;
guint payload_len;
guint available;
guint minptime_octets = 0;
guint min_payload_len;
guint max_payload_len;
+ gsize size;
+ GstClockTime timestamp;
- available = GST_BUFFER_SIZE (buf);
+ size = gst_buffer_get_size (buf);
- if (available % G729_FRAME_SIZE != 0 &&
- available % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE)
+ 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
(int) (ptime_ms / G729_FRAME_DURATION_MS);
if (maxptime_octets < G729_FRAME_SIZE) {
- GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %" G_GINT64_FORMAT
+ 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_BASE_RTP_PAYLOAD_MTU
- (basertpaudiopayload), 0, 0) / G729_FRAME_SIZE) * G729_FRAME_SIZE,
+ (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);
}
/* If the ptime is specified in the caps, tried to adhere to it exactly */
- if (payload->abidata.ABI.ptime) {
- guint64 ptime = payload->abidata.ABI.ptime / GST_MSECOND;
+ if (payload->ptime) {
+ guint64 ptime = payload->ptime / GST_MSECOND;
guint ptime_in_bytes = G729_FRAME_SIZE *
(guint) (ptime / G729_FRAME_DURATION_MS);
min_payload_len = max_payload_len = ptime_in_bytes;
}
- GST_LOG_OBJECT (basertpaudiopayload,
+ GST_LOG_OBJECT (payload,
"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);
+ 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 (gst_adapter_available (adapter) == 0)
- basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
-
- if (gst_adapter_available (adapter) == 0 &&
- 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));
+ 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);
- g_object_unref (adapter);
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) {
- guint num;
-
/* We send as much as we can */
if (available <= max_payload_len) {
payload_len = available;
(available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
}
- ret = gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
- basertpaudiopayload->base_ts);
-
- num = payload_len / G729_FRAME_SIZE;
- basertpaudiopayload->base_ts += G729_FRAME_DURATION * num;
-
- available = gst_adapter_available (adapter);
+ ret = gst_rtp_g729_pay_push (rtpg729pay,
+ gst_adapter_take (adapter, payload_len), payload_len);
+ available -= payload_len;
}
- g_object_unref (adapter);
-
return ret;
/* ERRORS */
("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 %u", available));
+ " 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