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authorAndreas Huber <andih@google.com>2010-08-10 11:18:36 -0700
committerAndreas Huber <andih@google.com>2010-08-10 11:38:18 -0700
commitf88f84414ae7baead03497f1d650ad8ea2f87688 (patch)
tree3f492f927b167b6d8185a447111208936b9e9cea /media/libstagefright/rtsp/ARTPSource.cpp
parent259b4c860212dd528b25d1cce6e74be01afed85c (diff)
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We're now going to ignore timestamps completely in gtalk video conferencing, playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup.
Change-Id: Idc3df74b42000f7a6aa3eae090718dc9d9c4186f
Diffstat (limited to 'media/libstagefright/rtsp/ARTPSource.cpp')
-rw-r--r--media/libstagefright/rtsp/ARTPSource.cpp9
1 files changed, 1 insertions, 8 deletions
diff --git a/media/libstagefright/rtsp/ARTPSource.cpp b/media/libstagefright/rtsp/ARTPSource.cpp
index e08183e..225f6e8 100644
--- a/media/libstagefright/rtsp/ARTPSource.cpp
+++ b/media/libstagefright/rtsp/ARTPSource.cpp
@@ -98,7 +98,7 @@ void ARTPSource::timeUpdate(uint32_t rtpTime, uint64_t ntpTime) {
mNTPTime[mNumTimes] = ntpTime;
mRTPTime[mNumTimes++] = rtpTime;
- if (mNumTimes == 2) {
+ if (timeEstablished()) {
for (List<sp<ABuffer> >::iterator it = mQueue.begin();
it != mQueue.end(); ++it) {
sp<AMessage> meta = (*it)->meta();
@@ -112,13 +112,6 @@ void ARTPSource::timeUpdate(uint32_t rtpTime, uint64_t ntpTime) {
}
bool ARTPSource::queuePacket(const sp<ABuffer> &buffer) {
-#if 1
- if (mNumTimes != 2) {
- // Drop incoming packets until we've established a time base.
- return false;
- }
-#endif
-
uint32_t seqNum = (uint32_t)buffer->int32Data();
if (mNumTimes == 2) {