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rtppayload: set standard payload type as default

Browse source 
Initialize the PT to the default value of the codec and check if
it is still the default before declaring the pt to be dynamic or
not when setting the caps.

Also use the PT constants from the rtp lib when possible

https://bugzilla.gnome.org/show_bug.cgi?id=747965
This commit is contained in:
Hyunjun Ko 2015-08-06 11:33:37 +09:00 committed by Thiago Santos
parent e0878d6325
commit 5a17572119
13 changed files with 55 additions and 45 deletions
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6
gst/rtp/gstrtpL16pay.c
View file

@ -208,16 +208,16 @@ gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
if (gst_structure_get_int (structure, "channels", &channels)) {
gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
} else if (gst_structure_get_int (structure, "payload", &pt)) {
if (pt == 10)
if (pt == GST_RTP_PAYLOAD_L16_STEREO)
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
else if (pt == 11)
else if (pt == GST_RTP_PAYLOAD_L16_MONO)
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
}
if (gst_structure_get_int (structure, "clock-rate", &rate)) {
gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
} else if (gst_structure_get_int (structure, "payload", &pt)) {
if (pt == 10 || pt == 11)
if (pt == GST_RTP_PAYLOAD_L16_STEREO || pt == GST_RTP_PAYLOAD_L16_MONO)
gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL);
}

11
gst/rtp/gstrtpg722pay.c
View file

@ -47,6 +47,11 @@ GST_STATIC_PAD_TEMPLATE ("src",
"media = (string) \"audio\", "
"encoding-name = (string) \"G722\", "
"payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
"clock-rate = (int) 8000; "
"application/x-rtp, "
"media = (string) \"audio\", "
"encoding-name = (string) \"G722\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000")
);
@ -92,6 +97,8 @@ gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay)
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg722pay);
GST_RTP_BASE_PAYLOAD (rtpg722pay)->pt = GST_RTP_PAYLOAD_G722;
/* tell rtpbaseaudiopayload that this is a sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
}
@ -136,8 +143,8 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
* RFC 3551 although the sampling rate is 16000 Hz */
clock_rate = 8000;
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "G722",
clock_rate);
gst_rtp_base_payload_set_options (basepayload, "audio",
basepayload->pt != GST_RTP_PAYLOAD_G722, "G722", clock_rate);
params = g_strdup_printf ("%d", channels);
#if 0

15
gst/rtp/gstrtpg723pay.c
View file

@ -29,9 +29,6 @@
#include "gstrtpg723pay.h"
#define GST_RTP_PAYLOAD_G723 4
#define GST_RTP_PAYLOAD_G723_STRING "4"
#define G723_FRAME_DURATION (30 * GST_MSECOND)
static gboolean gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload,
@ -107,7 +104,6 @@ gst_rtp_g723_pay_init (GstRTPG723Pay * pay)
pay->adapter = gst_adapter_new ();
payload->pt = GST_RTP_PAYLOAD_G723;
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "G723", 8000);
}
static void
@ -128,16 +124,9 @@ static gboolean
gst_rtp_g723_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
GstStructure *structure;
gint pt;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "payload", &pt))
pt = GST_RTP_PAYLOAD_G723;
payload->pt = pt;
payload->dynamic = pt != GST_RTP_PAYLOAD_G723;
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_G723, "G723", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;

11
gst/rtp/gstrtpg729pay.c
View file

@ -121,7 +121,6 @@ gst_rtp_g729_pay_init (GstRTPG729Pay * pay)
GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
payload->pt = GST_RTP_PAYLOAD_G729;
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "G729", 8000);
pay->adapter = gst_adapter_new ();
}
@ -140,15 +139,9 @@ static gboolean
gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
GstStructure *structure;
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;
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_G729, "G729", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);

3
gst/rtp/gstrtpgsmpay.c
View file

@ -107,7 +107,8 @@ gst_rtp_gsm_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
if (strcmp ("audio/x-gsm", stname))
goto invalid_type;
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "GSM", 8000);
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_GSM, "GSM", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;

3
gst/rtp/gstrtph261pay.c
View file

@ -1018,7 +1018,8 @@ gst_rtp_h261_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gst_rtp_base_payload_set_options (payload, "video", TRUE, "H261", 90000);
gst_rtp_base_payload_set_options (payload, "video",
payload->pt != GST_RTP_PAYLOAD_H261, "H261", 90000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;

5
gst/rtp/gstrtph263pay.c
View file

@ -440,6 +440,7 @@ gst_rtp_h263_pay_class_init (GstRtpH263PayClass * klass)
static void
gst_rtp_h263_pay_init (GstRtpH263Pay * rtph263pay)
{
GST_RTP_BASE_PAYLOAD_PT (rtph263pay) = GST_RTP_PAYLOAD_H263;
rtph263pay->prop_payload_mode = DEFAULT_MODE_A;
}
@ -476,8 +477,8 @@ gst_rtp_h263_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
framesize = g_strdup_printf ("%d-%d", width, height);
}
payload->pt = GST_RTP_PAYLOAD_H263;
gst_rtp_base_payload_set_options (payload, "video", TRUE, "H263", 90000);
gst_rtp_base_payload_set_options (payload, "video",
payload->pt != GST_RTP_PAYLOAD_H263, "H263", 90000);
if (framesize != NULL) {
res = gst_rtp_base_payload_set_outcaps (payload,

13
gst/rtp/gstrtpjpegpay.c
View file

@ -54,7 +54,13 @@ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
" media = (string) \"video\", "
" payload = (int) 26 , "
" payload = (int) " GST_RTP_PAYLOAD_JPEG_STRING ", "
" clock-rate = (int) 90000, "
" encoding-name = (string) \"JPEG\", "
" width = (int) [ 1, 65536 ], " " height = (int) [ 1, 65536 ]; "
" application/x-rtp, "
" media = (string) \"video\", "
" payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
" clock-rate = (int) 90000, "
" encoding-name = (string) \"JPEG\", "
" width = (int) [ 1, 65536 ], " " height = (int) [ 1, 65536 ]")
@ -283,6 +289,8 @@ gst_rtp_jpeg_pay_init (GstRtpJPEGPay * pay)
pay->type = DEFAULT_JPEG_TYPE;
pay->width = -1;
pay->height = -1;
GST_RTP_BASE_PAYLOAD_PT (pay) = GST_RTP_PAYLOAD_JPEG;
}
static gboolean
@ -321,7 +329,8 @@ gst_rtp_jpeg_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
pay->width = GST_ROUND_UP_8 (width) / 8;
}
gst_rtp_base_payload_set_options (basepayload, "video", TRUE, "JPEG", 90000);
gst_rtp_base_payload_set_options (basepayload, "video",
basepayload->pt != GST_RTP_PAYLOAD_JPEG, "JPEG", 90000);
if (num > 0) {
gdouble framerate;

3
gst/rtp/gstrtpmp2tpay.c
View file

@ -112,7 +112,8 @@ gst_rtp_mp2t_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gst_rtp_base_payload_set_options (payload, "video", TRUE, "MP2T", 90000);
gst_rtp_base_payload_set_options (payload, "video",
payload->pt != GST_RTP_PAYLOAD_MP2T, "MP2T", 90000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;

5
gst/rtp/gstrtpmpapay.c
View file

@ -104,6 +104,8 @@ static void
gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
{
rtpmpapay->adapter = gst_adapter_new ();
GST_RTP_BASE_PAYLOAD (rtpmpapay)->pt = GST_RTP_PAYLOAD_MPA;
}
static void
@ -133,7 +135,8 @@ gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gst_rtp_base_payload_set_options (payload, "audio", TRUE, "MPA", 90000);
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_MPA, "MPA", 90000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;

7
gst/rtp/gstrtpmpvpay.c
View file

@ -45,6 +45,10 @@ GST_STATIC_PAD_TEMPLATE ("src",
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_MPV_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MPV\"; "
"application/x-rtp, "
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MPV\"")
);
@ -131,7 +135,8 @@ gst_rtp_mpv_pay_reset (GstRTPMPVPay * pay)
static gboolean
gst_rtp_mpv_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gst_rtp_base_payload_set_options (payload, "video", FALSE, "MPV", 90000);
gst_rtp_base_payload_set_options (payload, "video",
payload->pt != GST_RTP_PAYLOAD_MPV, "MPV", 90000);
return gst_rtp_base_payload_set_outcaps (payload, NULL);
}

6
gst/rtp/gstrtppcmapay.c
View file

@ -86,6 +86,7 @@ gst_rtp_pcma_pay_init (GstRtpPcmaPay * rtppcmapay)
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmapay);
GST_RTP_BASE_PAYLOAD (rtppcmapay)->pt = GST_RTP_PAYLOAD_PCMA;
GST_RTP_BASE_PAYLOAD (rtppcmapay)->clock_rate = 8000;
/* tell rtpbaseaudiopayload that this is a sample based codec */
@ -100,9 +101,8 @@ gst_rtp_pcma_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
payload->pt = GST_RTP_PAYLOAD_PCMA;
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "PCMA", 8000);
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_PCMA, "PCMA", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;

6
gst/rtp/gstrtppcmupay.c
View file

@ -86,6 +86,7 @@ gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay)
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmupay);
GST_RTP_BASE_PAYLOAD (rtppcmupay)->pt = GST_RTP_PAYLOAD_PCMU;
GST_RTP_BASE_PAYLOAD (rtppcmupay)->clock_rate = 8000;
/* tell rtpbaseaudiopayload that this is a sample based codec */
@ -100,9 +101,8 @@ gst_rtp_pcmu_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
payload->pt = GST_RTP_PAYLOAD_PCMU;
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "PCMU", 8000);
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_PCMU, "PCMU", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;