Commit f1fe0e71 authored by Mark Nauwelaerts's avatar Mark Nauwelaerts

rtpg729pay: avoid basertppayload perfect-rtptime mode

G729 packets may only occur intermittently (e.g. cn packets), and as such
do not allow for perfect-rtptime calculating rtp times based on frame or byte
count.  In particular, do not use rtp audio base payloader as base class, but
rather base payloader directly.
parent 6405df0c
......@@ -48,6 +48,9 @@ gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps);
static GstFlowReturn
gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * 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,
......@@ -71,8 +74,8 @@ static GstStaticPadTemplate gst_rtp_g729_pay_src_template =
"clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"")
);
GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_g729_pay_base_init (gpointer klass)
......@@ -92,11 +95,27 @@ gst_rtp_g729_pay_base_init (gpointer klass)
"G.729 RTP Payloader");
}
static void
gst_rtp_g729_pay_finalize (GObject * object)
{
GstRTPG729Pay *pay = GST_RTP_G729_PAY (object);
g_object_unref (pay->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
gobject_class->finalize = gst_rtp_g729_pay_finalize;
gstelement_class->change_state = gst_rtp_g729_pay_change_state;
payload_class->set_caps = gst_rtp_g729_pay_set_caps;
payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer;
}
......@@ -105,15 +124,18 @@ static void
gst_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass)
{
GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
payload->pt = GST_RTP_PAYLOAD_G729;
gst_basertppayload_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;
}
static gboolean
......@@ -135,12 +157,49 @@ gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
return res;
}
static GstFlowReturn
gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay,
const guint8 * data, guint payload_len, GstClockTime timestamp,
GstClockTime duration)
{
GstBaseRTPPayload *basepayload;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
basepayload = GST_BASE_RTP_PAYLOAD (rtpg729pay);
GST_DEBUG_OBJECT (rtpg729pay, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, data, payload_len);
/* set metadata */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = duration;
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 (outbuf, TRUE);
rtpg729pay->discont = FALSE;
}
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * 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;
......@@ -164,7 +223,7 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
(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;
......@@ -174,7 +233,8 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
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,
(payload), 0, 0) / G729_FRAME_SIZE)
* G729_FRAME_SIZE,
/* ptime max */
maxptime_octets);
......@@ -207,25 +267,26 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
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);
if (GST_BUFFER_IS_DISCONT (buf))
rtpg729pay->discont = TRUE;
adapter = rtpg729pay->adapter;
/* let's reset the base timestamp when the adapter is empty */
if (gst_adapter_available (adapter) == 0)
basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
rtpg729pay->next_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,
ret = gst_rtp_g729_pay_push (rtpg729pay,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
GST_BUFFER_TIMESTAMP (buf));
GST_BUFFER_TIMESTAMP (buf), GST_BUFFER_DURATION (buf));
gst_buffer_unref (buf);
g_object_unref (adapter);
return ret;
}
......@@ -236,7 +297,7 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
/* 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;
GstClockTime duration;
/* We send as much as we can */
if (available <= max_payload_len) {
......@@ -246,17 +307,16 @@ gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
(available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
}
ret = gst_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
basertpaudiopayload->base_ts);
duration = (payload_len / G729_FRAME_SIZE) * G729_FRAME_DURATION;
rtpg729pay->next_ts += duration;
num = payload_len / G729_FRAME_SIZE;
basertpaudiopayload->base_ts += G729_FRAME_DURATION * num;
ret = gst_rtp_g729_pay_push (rtpg729pay,
gst_adapter_take (adapter, payload_len), payload_len,
rtpg729pay->next_ts, duration);
available = gst_adapter_available (adapter);
}
g_object_unref (adapter);
return ret;
/* ERRORS */
......@@ -272,6 +332,31 @@ invalid_size:
}
}
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)
{
......
......@@ -43,6 +43,10 @@ typedef struct _GstRTPG729PayClass GstRTPG729PayClass;
struct _GstRTPG729Pay
{
GstBaseRTPAudioPayload audiopayload;
GstAdapter *adapter;
GstClockTime next_ts;
gboolean discont;
};
struct _GstRTPG729PayClass
......
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