[WebRTC] updated to upstream branch 49 and related.

this patch is based on 38d7a7883...226f2f0bd
This commit is contained in:
roytam1 2021-10-01 10:48:39 +08:00
commit a3239eaa4d
2838 changed files with 151221 additions and 159688 deletions

View file

@ -8,18 +8,18 @@ include('/build/gyp.mozbuild')
webrtc_non_unified_sources = [
'trunk/webrtc/common_audio/vad/vad_core.c', # Because of name clash in the kInitCheck variable
'trunk/webrtc/common_audio/vad/webrtc_vad.c', # Because of name clash in the kInitCheck variable
'trunk/webrtc/modules/audio_coding/acm2/codec_manager.cc', # Because of duplicate IsCodecRED/etc
'trunk/webrtc/modules/audio_coding/codecs/g722/g722_decode.c', # Because of name clash in the saturate function
'trunk/webrtc/modules/audio_coding/codecs/g722/g722_encode.c', # Because of name clash in the saturate function
'trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c', # Because of name clash in the kDampFilter variable
'trunk/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c', # Because of name clash in the kDampFilter variable
'trunk/webrtc/modules/audio_coding/main/acm2/codec_manager.cc', # Because of duplicate IsCodecRED/etc
'trunk/webrtc/modules/audio_coding/neteq/audio_vector.cc', # Because of explicit template specializations
'trunk/webrtc/modules/audio_device/android/audio_manager.cc', # Because of TAG redefinition
'trunk/webrtc/modules/audio_device/android/audio_record_jni.cc', # Becuse of commonly named module static vars
'trunk/webrtc/modules/audio_device/android/audio_track_jni.cc', # Becuse of commonly named module static vars
'trunk/webrtc/modules/audio_device/android/audio_track_jni.cc', # Becuse of commonly named module static vars
'trunk/webrtc/modules/audio_device/android/opensles_player.cc', # Because of TAG redefinition
'trunk/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc', # Because of LATE()
'trunk/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc',# Because of LATE()
'trunk/webrtc/modules/audio_device/opensl/opensles_input.cc', # Because of name clash in the kOption variable
'trunk/webrtc/modules/audio_device/opensl/opensles_output.cc', # Because of name clash in the kOption variable
'trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.cc', # Because of name clash with #define FF
'trunk/webrtc/modules/audio_device/win/audio_device_core_win.cc', # Because of ordering assumptions in strsafe.h
'trunk/webrtc/modules/audio_processing/aec/aec_core.c', # Because of name clash in the ComfortNoise function
@ -34,12 +34,15 @@ webrtc_non_unified_sources = [
'trunk/webrtc/modules/audio_processing/gain_control_impl.cc', # Because of name clash in the Handle typedef
'trunk/webrtc/modules/audio_processing/high_pass_filter_impl.cc', # Because of name clash in the Handle typedef
'trunk/webrtc/modules/audio_processing/noise_suppression_impl.cc', # Because of name clash in the Handle typedef
'trunk/webrtc/modules/remote_bitrate_estimator/mimd_rate_control.cc', # Because of duplicate definitions of static consts against aimd_rate_control.cc
'trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc', # Because of duplicate definitions of static consts against remote_bitrate_estimator_abs_send_time.cc
'trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc', # Because of identically named functions and vars between tmmbr.cc and tmmbn.cc in an anonymous namespaces
'trunk/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc', # Because of identically named functions and vars between tmmbr.cc and tmmbn.cc in an anonymous namespaces
'trunk/webrtc/modules/video_capture/android/device_info_android.cc', # Because of duplicate module static variable names
'trunk/webrtc/modules/video_capture/android/video_capture_android.cc', # Because of duplicate module static variable names
'trunk/webrtc/modules/video_capture/windows/device_info_ds.cc', # Because of the MEDIASUBTYPE_HDYC variable
'trunk/webrtc/modules/video_capture/windows/help_functions_ds.cc', # Because of initguid.h
'trunk/webrtc/modules/video_capture/windows/sink_filter_ds.cc', # Because of the MEDIASUBTYPE_HDYC variable and initguid.h
'trunk/webrtc/video_engine/overuse_frame_detector.cc', # Because of name clash with call_stats.cc on kWeightFactor
'trunk/webrtc/video/overuse_frame_detector.cc', # Because of name clash with call_stats.cc on kWeightFactor
]
GYP_DIRS += ['trunk']
@ -65,7 +68,6 @@ if CONFIG['MOZ_WEBRTC_SIGNALING']:
'signaling/src/common/browser_logging/CSFLog.cpp',
'signaling/src/jsep/JsepSessionImpl.cpp',
'signaling/src/media-conduit/AudioConduit.cpp',
'signaling/src/media-conduit/CodecStatistics.cpp',
'signaling/src/media-conduit/MediaCodecVideoCodec.cpp',
'signaling/src/media-conduit/VideoConduit.cpp',
'signaling/src/media-conduit/WebrtcMediaCodecVP8VideoCodec.cpp',