static GstCaps *
gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter)
{
GstCaps *caps, *peercaps, *tcaps;
GstStructure *s;
const gchar *stereo;
if (pad == GST_RTP_BASE_PAYLOAD_SRCPAD (payload))
return
GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
(payload, pad, filter);
tcaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
peercaps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload),
tcaps);
gst_caps_unref (tcaps);
if (!peercaps)
return
GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
(payload, pad, filter);
if (gst_caps_is_empty (peercaps))
return peercaps;
caps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SINKPAD (payload));
s = gst_caps_get_structure (peercaps, 0);
stereo = gst_structure_get_string (s, "stereo");
if (stereo != NULL) {
caps = gst_caps_make_writable (caps);
if (!strcmp (stereo, "1")) {
GstCaps *caps2 = gst_caps_copy (caps);
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 1, NULL);
caps = gst_caps_merge (caps, caps2);
} else if (!strcmp (stereo, "0")) {
GstCaps *caps2 = gst_caps_copy (caps);
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 2, NULL);
caps = gst_caps_merge (caps, caps2);
}
}
gst_caps_unref (peercaps);
if (filter) {
GstCaps *tmp = gst_caps_intersect_full (caps, filter,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
GST_DEBUG_OBJECT (payload, "Returning caps: %" GST_PTR_FORMAT, caps);
return caps;
}
static gboolean
gst_rtp_base_payload_audio_sink_event (GstRTPBasePayload * basep,
GstEvent * event)
{
GstRTPBaseAudioPayload *payload;
gboolean res = FALSE;
payload = GST_RTP_BASE_AUDIO_PAYLOAD (basep);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* flush remaining bytes in the adapter */
gst_rtp_base_audio_payload_flush (payload, -1, -1);
break;
case GST_EVENT_FLUSH_STOP:
gst_adapter_clear (payload->priv->adapter);
break;
default:
break;
}
/* let parent handle the remainder of the event */
res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (basep, event);
return res;
}
static gboolean
gst_rtp_mp4g_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
case GST_EVENT_EOS:
/* This flush call makes sure that the last buffer is always pushed
* to the base payloader */
gst_rtp_mp4g_pay_flush (rtpmp4gpay);
break;
case GST_EVENT_FLUSH_STOP:
gst_rtp_mp4g_pay_reset (rtpmp4gpay);
break;
default:
break;
}
/* let parent handle event too */
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
}
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 GstFlowReturn
gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpL16Pay *rtpL16pay;
rtpL16pay = GST_RTP_L16_PAY (basepayload);
buffer = gst_buffer_make_writable (buffer);
if (rtpL16pay->order &&
!gst_audio_buffer_reorder_channels (buffer, rtpL16pay->info.finfo->format,
rtpL16pay->info.channels, rtpL16pay->info.position,
rtpL16pay->order->pos)) {
return GST_FLOW_ERROR;
}
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
buffer);
}
static gboolean
gst_rtp_mpa_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
gboolean ret;
GstRtpMPAPay *rtpmpapay;
rtpmpapay = GST_RTP_MPA_PAY (payload);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* make sure we push the last packets in the adapter on EOS */
gst_rtp_mpa_pay_flush (rtpmpapay);
break;
case GST_EVENT_FLUSH_STOP:
gst_rtp_mpa_pay_reset (rtpmpapay);
break;
default:
break;
}
ret = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
return ret;
}
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 <*****@*****.**>");
GST_DEBUG_CATEGORY_INIT (gst_rtp_sbc_pay_debug, "rtpsbcpay", 0,
"RTP SBC payloader");
}
