| /external/webrtc/webrtc/modules/utility/source/ |
| helpers_ios.mm | 175 LOG(LS_WARNING) << "Failed to find device name (" << raw_name << ")";
|
| /external/webrtc/webrtc/base/ |
| proxydetect.cc | 311 LOG(LS_WARNING) << "Proxy address too long [" << start << "]";
321 LOG(LS_WARNING) << "Proxy address without port [" << buffer << "]";
329 LOG(LS_WARNING) << "Proxy address with invalid port [" << buffer << "]";
341 LOG(LS_WARNING) << "Proxy address with unknown protocol ["
537 LOG_F(LS_WARNING) << "Unparsed pref [" << buffer << "]";
[all...] |
| httpclient.cc | 240 LOG_F(LS_WARNING) << "Malformed cache header";
377 LOG(LS_WARNING) << "Couldn't obtain absolute uri";
386 LOG(LS_WARNING) << "Couldn't obtain relative uri";
461 LOG_F(LS_WARNING) << "Couldn't lock cache";
556 LOG_F(LS_WARNING) << "Cache failure, continuing with normal request";
|
| opensslstreamadapter.cc | 344 LOG(LS_WARNING) << "Unknown digest algorithm: " << digest_alg;
850 LOG(LS_WARNING) << "OpenSSLStreamAdapter::Error("
870 LOG(LS_WARNING) << "SSL_shutdown failed, error = "
1058 LOG(LS_WARNING) << "Failed to compute peer cert digest.";
1064 LOG(LS_WARNING) << "Rejected peer certificate due to mismatched digest.";
[all...] |
| win32socketserver.cc | 284 LOG(LS_WARNING) << "GetLocalAddress: unable to get local addr, socket="
299 LOG(LS_WARNING) << "GetRemoteAddress: unable to get remote addr, socket="
610 LOG(LS_WARNING) << "Socket::OPT_DSCP not supported.";
|
| logging.cc | 294 current_level = LS_WARNING;
397 case LS_WARNING:
|
| autodetectproxy.cc | 56 LOG(LS_WARNING) << "AutoDetectProxy removing http prefix on proxy host";
|
| dbus.cc | 42 LOG(LS_WARNING) << "Failed to load dbus-glib symbol table.";
|
| httpserver.cc | 33 LOG(LS_WARNING) << "HttpServer::CloseAll has not completed";
|
| macsocketserver.cc | 288 LOG_E(LS_WARNING, OS, result) << "ReceiveNextEvent";
|
| socketpool.cc | 90 LOG_F(LS_WARNING) << "(" << events << ", " << err
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| forward_error_correction.cc | 122 LOG(LS_WARNING) << "Can't protect " << num_media_packets
135 LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes "
143 LOG(LS_WARNING) << "Media packet " << media_packet->length << " bytes "
568 LOG(LS_WARNING) << "FEC packet has an all-zero packet mask.";
653 LOG(LS_WARNING)
667 LOG(LS_WARNING) << "Incorrect FEC protection length, dropping.";
[all...] |
| rtcp_utility.cc | 475 LOG(LS_WARNING) << "Too little data (" << size_bytes << " byte"
484 LOG(LS_WARNING) << "Invalid RTCP header: Version must be "
497 LOG(LS_WARNING) << "Buffer too small (" << size_bytes
507 LOG(LS_WARNING) << "Invalid RTCP header: Padding bit set but 0 payload "
514 LOG(LS_WARNING) << "Invalid RTCP header: Too many padding bytes ("
[all...] |
| /external/webrtc/webrtc/p2p/base/ |
| turnserver.cc | 221 LOG(LS_WARNING) << "Received invalid STUN message";
608 LOG_J(LS_WARNING, this) << "Invalid TURN message type received: "
677 LOG_J(LS_WARNING, this) << "Received invalid send indication";
686 LOG_J(LS_WARNING, this) << "Received send indication without permission"
777 LOG_J(LS_WARNING, this) << "Received channel data for invalid channel, id="
808 LOG_J(LS_WARNING, this) << "Received external packet without permission, "
|
| /external/webrtc/webrtc/sound/ |
| pulseaudiosoundsystem.cc | 497 LOG(LS_WARNING) << "Buffer overflow on capture stream " << stream;
527 LOG(LS_WARNING) << "Ignoring extra GetVolumeCallback";
558 LOG(LS_WARNING) << "Ignoring extra GetSourceChannelCountCallback";
870 LOG(LS_WARNING) << "Buffer underflow on playback stream "
941 LOG(LS_WARNING) << "Ignoring extra GetVolumeCallback";
978 LOG(LS_WARNING) << "Failed to load symbol table";
[all...] |
| alsasoundsystem.cc | 121 LOG(LS_WARNING) << "Timeout while waiting on stream";
133 LOG(LS_WARNING) << "Spurious wake-up";
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| neteq_impl.cc | 863 LOG(LS_WARNING) << "Output array is too short. " << max_length << " < " <<
[all...] |
| decoder_database.cc | 253 LOG(LS_WARNING) << "CheckPayloadTypes: unknown RTP payload type "
|
| /external/webrtc/talk/media/base/ |
| videocapturer.cc | 158 LOG(LS_WARNING) << "UpdateAspectRatio ignored invalid ratio: "
193 LOG(LS_WARNING) << "Cannot unpause a camera that hasn't been paused.";
201 LOG(LS_WARNING) << "Camera cannot be unpaused while muted.";
|
| /external/webrtc/talk/media/sctp/ |
| sctpdataengine_unittest.cc | 97 LOG(LS_WARNING) << "Unsupported: SctpFakeNetworkInterface::SendRtcp.";
102 LOG(LS_WARNING) << "Unsupported: SctpFakeNetworkInterface::SetOption.";
106 LOG(LS_WARNING) << "Unsupported: SctpFakeNetworkInterface::SetOption.";
|
| /external/webrtc/talk/app/webrtc/ |
| datachannel.cc | 355 LOG(LS_WARNING) << "DataChannel received unexpected CONTROL message, "
365 LOG(LS_WARNING) << "DataChannel failed to parse OPEN_ACK message, sid = "
|
| /external/webrtc/talk/session/media/ |
| channel.cc | 669 LOG(LS_WARNING) << "Got EWOULDBLOCK from socket.";
746 LOG(LS_WARNING) << "Can't process incoming " << PacketType(rtcp)
900 LOG(LS_WARNING) << "DTLS-SRTP key export failed";
926 LOG(LS_WARNING) << "GetSslRole failed";
[all...] |
| /external/webrtc/webrtc/modules/audio_device/ios/ |
| audio_device_ios.mm | 350 LOG(LS_WARNING) << "Object is destructed with an active audio session";
688 LOG(LS_WARNING) << "Unable to set the preferred sample rate";
[all...] |
| /external/webrtc/talk/app/webrtc/java/jni/ |
| androidvideocapturer_jni.cc | 129 LOG(LS_WARNING) << method_name << "() called for closed capturer.";
|
| /external/webrtc/talk/media/webrtc/ |
| simulcast.cc | 263 LOG(LS_WARNING) << "Unable to parse WebRTC-ScreenshareLayerRates"
|
