2015GitWebRTC编译实录3

2015.05.17 librtprtcp 编译通过
[702/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.bitrate.o
[703/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.fec_receiver_impl.o
[704/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.receive_statistics_impl.o
[705/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.remote_ntp_time_estimator.o
[706/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_header_parser.o
[707/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_rtcp_impl.o
[708/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtcp_packet.o
[710/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtcp_receiver.o
[711/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtcp_receiver_help.o
[712/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtcp_sender.o
[713/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtcp_utility.o
[714/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_header_extension.o
[715/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_receiver_impl.o
[716/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_sender.o
[717/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_utility.o
[718/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.ssrc_database.o
[719/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.tmmbr_help.o
[720/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.dtmf_queue.o
[721/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.forward_error_correction_internal.o
[722/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_receiver_audio.o
[723/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_sender_audio.o
[724/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.forward_error_correction.o
[725/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.producer_fec.o
[726/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.h264_sps_parser.o
[727/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_packet_history.o
[728/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_payload_registry.o
[729/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_receiver_strategy.o
[730/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_receiver_video.o
[731/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_sender_video.o
[732/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_format.o
[733/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_format_h264.o
[734/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_format_vp8.o
[735/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.rtp_format_video_generic.o
[736/1600 ] CXX obj /webrtc/modules/rtp_rtcp/source/rtp_rtcp.vp8_partition_aggregator.o
[745/1600 ] librtp_rtcp.a

2015.05.17 libsrtp 编译通过
[852/1600 ] CC obj/third_party/libsrtp/srtp/srtp/libsrtp.srtp.o
[855/1600 ] CC obj/third_party/libsrtp/srtp/srtp/libsrtp.ekt.o
[856/1600 ] CC obj/third_party/libsrtp/srtp/crypto/cipher/libsrtp.aes.o
[857/1600 ] CC obj/third_party/libsrtp/srtp/crypto/cipher/libsrtp.cipher.o
[858/1600 ] CC obj/third_party/libsrtp/srtp/crypto/cipher/libsrtp.null_cipher.o
[859/1600 ] CC obj/third_party/libsrtp/srtp/crypto/hash/libsrtp.auth.o
[860/1600 ] CC obj/third_party/libsrtp/srtp/crypto/hash/libsrtp.null_auth.o
[861/1600 ] CC obj/third_party/libsrtp/srtp/crypto/kernel/libsrtp.alloc.o
[862/1600 ] CC obj/third_party/libsrtp/srtp/crypto/kernel/libsrtp.crypto_kernel.o
[863/1600 ] CC obj/third_party/libsrtp/srtp/crypto/kernel/libsrtp.err.o
[864/1600 ] CC obj/third_party/libsrtp/srtp/crypto/kernel/libsrtp.key.o
[865/1600 ] CC obj/third_party/libsrtp/srtp/crypto/math/libsrtp.datatypes.o
[866/1600 ] CC obj/third_party/libsrtp/srtp/crypto/math/libsrtp.gf2_8.o
[867/1600 ] CC obj/third_party/libsrtp/srtp/crypto/math/libsrtp.stat.o
[868/1600 ] CC obj/third_party/libsrtp/srtp/crypto/replay/libsrtp.rdb.o
[869/1600 ] CC obj/third_party/libsrtp/srtp/crypto/replay/libsrtp.rdbx.o
[870/1600 ] CC obj/third_party/libsrtp/srtp/crypto/replay/libsrtp.ut_sim.o
[871/1600 ] CC obj/third_party/libsrtp/srtp/crypto/rng/libsrtp.rand_source.o
[872/1600 ] CC obj/third_party/libsrtp/srtp/crypto/cipher/libsrtp.aes_gcm_ossl.o
[873/1600 ] CC obj/third_party/libsrtp/srtp/crypto/cipher/libsrtp.aes_icm_ossl.o
[874/1600 ] CC obj/third_party/libsrtp/srtp/crypto/hash/libsrtp.hmac_ossl.o
[1264/1600 ] libsrtp.a

-DHAVE_CONFIG_H
-DHAVE_STDLIB_H
-DHAVE_STRING_H
-DOPENSSL
-DHAVE_INT16_T
-DHAVE_INT32_T
-DHAVE_INT8_T
-DHAVE_UINT16_T
-DHAVE_UINT32_T
-DHAVE_UINT64_T
-DHAVE_UINT8_T
-DHAVE_STDINT_H
-DHAVE_INTTYPES_H
-DHAVE_NETINET_IN_H
-DHAVE_ARPA_INET_H
-DHAVE_UNISTD_H

移走不需要的代码
../../../mysrc/third_party/libsrtp/srtp/crypto/rng/rand_linux_kernel.c:57:3: warning: implicit declaration
of function ‘get_random_bytes‘ is invalid in C99 [-Wimplicit-function-declaration]

时间: 2024-07-31 15:28:26

2015GitWebRTC编译实录3的相关文章

2015GitWebRTC编译实录13

2015.07.21 libboringssl.a 编译通过主要是生成路径,去除test文件比较啰嗦,后继测试需要重点跟进下 CC obj/third_party/boringssl/boringssl.err_data.oCC obj/third_party/boringsslsrc/crypto/aes/boringssl.aes.oCC obj/third_party/boringsslsrc/crypto/aes/boringssl.mode_wrappers.oCC obj/third

2015GitWebRTC编译实录14

libvpx 尝试用脚本编译了下,发现有问题,就偃旗息鼓,改用他自己的configure了,在网上找了下,Git上有个现成的,直接用,更好些. https://github.com/brion/VPX-iOS 注意他这里有个子模块,需要更新下,其他的还好说. [230/1600 ] CC obj/third_party/libvpx/source/libvpx/vp8/common/libvpx.blockd.o[231/1600 ] CC obj/third_party/libvpx/sour

2015GitWebRTC编译实录5

2015.07.20 libaudio_encoder_interface/libaudio_decoder_interface 编译通过将encoder,decoder两个lib合并了,后面需要看看是否合理.[1/1600 ] CXX obj /webrtc/modules/audio_coding/codecs/audio_encoder_interface.audio_encoder.o[380/1600 ] CXX obj /webrtc/modules/audio_coding/cod

2015GitWebRTC编译实录10

2015.07.20 rtc_p2p编译通过[879/1600 ] CXX obj /webrtc/p2p/client/rtc_p2p.httpportallocator.o[880/1600 ] CXX obj /webrtc/p2p/client/rtc_p2p.basicportallocator.o[881/1600 ] libisac_neon.a[882/1600 ] CXX obj /webrtc/p2p/client/rtc_p2p.connectivitychecker.o[

2015GitWebRTC编译实录11

2015.07.21 ilbc 编译通过注意有几个win32打头的文件,其实都是要编进去的[429/1600 ] CC obj ilbc.abs_quant.o[430/1600 ] CXX obj ilbc.audio_encoder_ilbc.o[432/1600 ] CC obj ilbc.abs_quant_loop.o[433/1600 ] CC obj ilbc.cb_mem_energy_augmentation.o[434/1600 ] CC obj ilbc.augmented

2015GitWebRTC编译实录7

2015.07.20 libvoiceengine 编译通过去除了mock测试代码,mock是用来进行测试的,意义不大.另外会报一个常量错误,需要定义WEBRTC_MAC宏,只定义WEBRTC_IOS宏是有问题的[834/1600 ] CXX obj /webrtc/voice_engine/voice_engine.transmit_mixer.o[835/1600 ] CXX obj /webrtc/voice_engine/voice_engine.channel.o[836/1600 ]

2015GitWebRTC编译实录4

2015.07.17 libg711 编译通过[422/1600 ] CC obj /webrtc/modules/audio_coding/codecs/g711/g711.g711.o[423/1600 ] CC obj /webrtc/modules/audio_coding/codecs/g711/g711.g711_interface.o[424/1600 ] CXX obj /webrtc/modules/audio_coding/codecs/g711/g711.audio_enc

2015GitWebRTC编译实录2

2015.07.17libyuvneon编译通过,可能需要验证才行.先继续下一个lib commonaudio[170/1600 ] CXX obj /webrtc/common_audio/common_audio.audio_util.o[171/1600 ] CXX obj /webrtc/common_audio/common_audio.audio_converter.o[172/1600 ] CXX obj /webrtc/common_audio/common_audio.audi

2015GitWebRTC编译实录12

2015.07.20 libjingle_peerconnection 编译通过[1382/1600 ] CXX obj/talk/app /webrtc/libjingle_peerconnection.mediaconstraintsinterface.o[1451/1600 ] CXX obj/talk/app /webrtc/libjingle_peerconnection.jsepicecandidate.o[1452/1600 ] CXX obj/talk/app /webrtc/l

2015GitWebRTC编译实录6

2015.07.20 libbitrate_controller 编译通过依赖system_wrappers lib,编写测试代码时需要注意.[425/1600 ] CXX obj /webrtc/modules/bitrate_controller/bitrate_controller.bitrate_controller_impl.o[426/1600 ] CXX obj /webrtc/modules/bitrate_controller/bitrate_controller.bitrat