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authorJoshua Colp <jcolp@digium.com>2016-09-14 08:59:51 -0400
committerJoshua Colp <jcolp@digium.com>2016-09-14 12:58:04 -0500
commit1cac856e175efb57585448f0266b5eedd21c7d40 (patch)
treede89613bc518e753ba2ca9dec16054a9f771e376
parenta5f7b07579093c22e5b2397451b46c5073d09810 (diff)
rtp: Preserve timestamps on video frames.
Currently when receiving video over RTP we store only a calculated samples on the frame. When starting the video it can take some time for this calculation to actually yield a value as it requires constant changing timestamps. As well if a video frame passes over multiple RTP packets this calculation will fail as the timestamp is the same as the previous RTP packet and the number of samples calculated will be 0. This change preserves the timestamp on the frame and allows it to pass through the core. When sending the video this timestamp is used instead of a new one being calculated. ASTERISK-26367 #close Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd
-rw-r--r--main/codec_builtin.c6
-rw-r--r--res/res_rtp_asterisk.c2
2 files changed, 8 insertions, 0 deletions
diff --git a/main/codec_builtin.c b/main/codec_builtin.c
index 1d329bc3b..973423d55 100644
--- a/main/codec_builtin.c
+++ b/main/codec_builtin.c
@@ -728,36 +728,42 @@ static struct ast_codec h261 = {
.name = "h261",
.description = "H.261 video",
.type = AST_MEDIA_TYPE_VIDEO,
+ .sample_rate = 1000,
};
static struct ast_codec h263 = {
.name = "h263",
.description = "H.263 video",
.type = AST_MEDIA_TYPE_VIDEO,
+ .sample_rate = 1000,
};
static struct ast_codec h263p = {
.name = "h263p",
.description = "H.263+ video",
.type = AST_MEDIA_TYPE_VIDEO,
+ .sample_rate = 1000,
};
static struct ast_codec h264 = {
.name = "h264",
.description = "H.264 video",
.type = AST_MEDIA_TYPE_VIDEO,
+ .sample_rate = 1000,
};
static struct ast_codec mpeg4 = {
.name = "mpeg4",
.description = "MPEG4 video",
.type = AST_MEDIA_TYPE_VIDEO,
+ .sample_rate = 1000,
};
static struct ast_codec vp8 = {
.name = "vp8",
.description = "VP8 video",
.type = AST_MEDIA_TYPE_VIDEO,
+ .sample_rate = 1000,
};
static struct ast_codec t140red = {
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index d05774fde..627f800a4 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -4764,6 +4764,8 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
/* Video -- samples is # of samples vs. 90000 */
if (!rtp->lastividtimestamp)
rtp->lastividtimestamp = timestamp;
+ ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
+ rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass.format) / 1000);
rtp->f.samples = timestamp - rtp->lastividtimestamp;
rtp->lastividtimestamp = timestamp;
rtp->f.delivery.tv_sec = 0;