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Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798

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R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrike@webrtc.org
2014-08-11 21:06:30 +00:00
parent 820f8e9ca7
commit 6ac22e6b47
27 changed files with 92 additions and 104 deletions
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5
webrtc/modules/video_coding/main/test/rtp_player.cc
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@ -348,8 +348,9 @@ class RtpPlayerImpl : public RtpPlayerInterface {
virtual int NextPacket(int64_t time_now) {
// Send any packets ready to be resent.
RawRtpPacket* packet;
while ((packet = lost_packets_.NextPacketToResend(time_now))) {
for (RawRtpPacket* packet = lost_packets_.NextPacketToResend(time_now);
packet != NULL;
packet = lost_packets_.NextPacketToResend(time_now)) {
int ret = SendPacket(packet->data(), packet->length());
if (ret > 0) {
printf("Resend: %08x:%u\n", packet->ssrc(), packet->seq_num());