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| author | Andreas Huber <andih@google.com> | 2010-08-10 12:59:27 -0700 |
|---|---|---|
| committer | Android (Google) Code Review <android-gerrit@google.com> | 2010-08-10 12:59:27 -0700 |
| commit | 18f0174ff4eef46f121e77a0dc1d7a8271621f48 (patch) | |
| tree | 8b26b96602be5cf3aa59097317b9e63c1ab4d49a /media/libstagefright/rtsp/ARTPSource.cpp | |
| parent | 605f402796afa9c88e2f91284e98d1f59a0726aa (diff) | |
| parent | f88f84414ae7baead03497f1d650ad8ea2f87688 (diff) | |
| download | frameworks_base-18f0174ff4eef46f121e77a0dc1d7a8271621f48.zip frameworks_base-18f0174ff4eef46f121e77a0dc1d7a8271621f48.tar.gz frameworks_base-18f0174ff4eef46f121e77a0dc1d7a8271621f48.tar.bz2 | |
Merge "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." into gingerbread
Diffstat (limited to 'media/libstagefright/rtsp/ARTPSource.cpp')
| -rw-r--r-- | media/libstagefright/rtsp/ARTPSource.cpp | 9 |
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) { |