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fix iOS build issue #164 #27

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fe4f694
Add missing overrides to QualityTestVideoEncoder
May 7, 2019
133f7e7
Rename "average_freeze_duration" metric to "freeze_duration_average"
May 8, 2019
8fc92e6
Add lifetime concealment stats to NetEqStatsPlotter.
May 8, 2019
6cdbf3f
Fix typo in SupportsEncoderFrameDropping's documentation
May 8, 2019
60f14ce
Do not use absl::flat_hash_map in DefaultVideoQualityAnalyzer.
MirkoBonadei May 8, 2019
26ab9d6
Roll chromium_revision db92e07547..8939017df7 (656805:657653)
May 8, 2019
d8b9ed7
Promote RtcEventLogOutputFile to api/
May 8, 2019
58e0657
Add decode/render frame rate metrics
May 8, 2019
8f119ca
Enable experiments with audio bitrate priority.
May 8, 2019
bd7046c
Remove redundant feedback_packet_seq_num_set_ in RtpVideoSender
May 7, 2019
daac582
Remove -Wno-undef and -Wno-extra-semi.
MirkoBonadei May 8, 2019
0f4f055
Don't remove or retransmit packets in the pacer queue.
May 8, 2019
86384fa
Roll chromium_revision 8939017df7..733884772b (657653:657800)
May 8, 2019
6853b56
Roll chromium_revision 733884772b..f5b58f6cdf (657800:657906)
May 8, 2019
ea5cbb5
Roll chromium_revision f5b58f6cdf..e2dc9e7e32 (657906:658007)
May 9, 2019
d7dd49f
RateControlSettings: add option to set max QP for libvpx vp8.
May 8, 2019
d61f2a7
Update SCTP status with transport whenever transport changes.
May 8, 2019
035ee11
Delete left-over tests NetEqExternalDecoderUnitTest
May 9, 2019
bf47f34
Add comments to clarify argument meanings in APM impl test
May 9, 2019
f3d828e
Make balanced degradation settings configurable through field trial.
May 6, 2019
780c136
Move OverUseDetectorOptions out of common_types.h
May 9, 2019
e42246a
Roll chromium_revision e2dc9e7e32..4b6421eedd (658007:658122)
May 9, 2019
de20b96
Revert "Reland "Copy video frames metadata between encoded and plain …
May 9, 2019
37f2b43
Reland "Version 2 "Refactoring DataContentDescription class""
May 9, 2019
0fa8605
Add DCHECK on the port allocator in P2PTransportChannel.
May 9, 2019
159d04e
Revert "Rename configurations_ to vpx_configs_ in LibvpxVp8Encoder"
May 9, 2019
14a9966
Roll chromium_revision 4b6421eedd..a91b44f5ab (658122:658232)
May 9, 2019
3d622d6
Revert "Refactor handling of configuration overrides from Vp8FrameBuf…
May 9, 2019
097c1f3
Roll chromium_revision a91b44f5ab..5f38b3b2db (658232:658382)
May 10, 2019
952b571
Delete unused class InsecureCryptStringImpl
May 9, 2019
e779847
Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN
May 9, 2019
fb08781
Reland "Refactor handling of configuration overrides from Vp8FrameBuf…
May 2, 2019
1992958
Qualify cmath function calls
oprypin Apr 23, 2019
418f580
Move kRtpCsrcSize from common_types.h to rtp_headers.h
May 8, 2019
aeedcc7
Roll chromium_revision 5f38b3b2db..def6b1e7f7 (658382:658489)
May 10, 2019
e1225d3
Reland "Rename configurations_ to vpx_configs_ in LibvpxVp8Encoder"
May 10, 2019
fb8c856
Revert "Change SimpleStringBuilder::Append to not use strcpyn and SIZ…
May 10, 2019
0da1156
Simplify WindowsCommandLineArguments, and move to example code.
May 10, 2019
8da35a6
Deprecate owned naked pointers to cricket::SessionDescription
May 10, 2019
eb9bf41
Fix problem in WebRTC-Bwe-AlrLimitedBackoff experiment
perkj May 10, 2019
ca36285
Add PlayoutVolumeChange RuntimeSetting.
May 10, 2019
2a1f020
Remove RtcEventLogImpl::owner_sequence_checker_
May 10, 2019
52490e3
Renaming inferred route change events.
jonex May 10, 2019
efe9314
Roll chromium_revision def6b1e7f7..b149f49431 (658489:658619)
May 10, 2019
46afbf9
Revert "Reland "Version 2 "Refactoring DataContentDescription class"""
steveanton May 10, 2019
b8aef84
Roll chromium_revision b149f49431..48b83dda01 (658619:658736)
May 10, 2019
a36591c
Roll chromium_revision 48b83dda01..f9876c4483 (658736:658883)
May 11, 2019
f00ca1a
Make the output_period_ms argument to StartRtcEventLog optional
May 10, 2019
fa61d80
Update visibility for java targets in sdk/android
May 13, 2019
d7c7d0e
Roll chromium_revision f9876c4483..7da20dc3cb (658883:658985)
May 13, 2019
d287962
Distinguish between missing packet and send failure.
May 10, 2019
166b45d
Adds route changes in event logs.
jonex May 13, 2019
18a6625
Fix typo in rtp_sender.h
May 13, 2019
157b781
Remove deprecated SetRates/SetRateAllocation from VideoEncoder.
May 13, 2019
0379d8c
Change a way, how receive stream is determined in DefaultAudioQuality…
May 13, 2019
46ac470
AEC3: Remove unused code
May 13, 2019
dab21c6
Fix for crash in frame matcher on short runs.
jonex May 13, 2019
2ebf523
Reland "Copy video frames metadata between encoded and plain frames i…
May 13, 2019
3dfb680
Use robust variance computation in RollingAccumulator.
May 13, 2019
863b2a2
Roll chromium_revision 7da20dc3cb..b06ee6f3e1 (658985:659110)
May 13, 2019
5fc28b1
Reland "Reland "Version 2 "Refactoring DataContentDescription class"""
May 13, 2019
e62a08a
Request key frame on all layers.
May 13, 2019
66fbb13
Roll chromium_revision b06ee6f3e1..a1c6ffe854 (659110:659234)
May 13, 2019
90d9feb
Roll chromium_revision a1c6ffe854..9500673d82 (659234:659367)
May 14, 2019
f11c8d1
Check for uninitialized audio unit in HandleInterruptionEnd.
kthelgason May 14, 2019
3b9aa66
Roll chromium_revision 9500673d82..1c80f902b4 (659367:659511)
May 14, 2019
83afeeb
Remove redundant capture time adjustment in RtpSender
May 14, 2019
c3f4820
Change default secure SCTP protocol to UDP/DTLS/SCTP
May 14, 2019
4aa1192
Change default SDP syntax for SCTP to spec-compliant.
May 14, 2019
39068db
Roll chromium_revision 1c80f902b4..279d455093 (659511:659647)
May 14, 2019
c136b06
Add datagram_transport and congestion_control interface
May 14, 2019
e88eefc
Roll chromium_revision 279d455093..3820725ad2 (659647:659764)
May 15, 2019
fbb45bd
Send and parse SCTP max-message-size in SDP
May 15, 2019
8567c39
Roll chromium_revision 3820725ad2..eacdc75a1b (659764:659866)
May 15, 2019
4fd4297
Fix metadata setting in H264 decoder
May 14, 2019
0ac1d99
Remove streaming_mode as it is always false.
May 14, 2019
a768345
Reduce flakiness of NetworkEmulationManagerTest.ThroughputStats
May 15, 2019
7581ff7
Add screen share support to PC level test framework
May 15, 2019
666290a
Roll chromium_revision eacdc75a1b..7991984f8d (659866:659976)
May 15, 2019
dafb4f8
Roll chromium_revision 7991984f8d..ee800955b4 (659976:660080)
May 15, 2019
9a7970d
Roll chromium_revision ee800955b4..34a275789d (660080:660187)
May 15, 2019
0f1a7ba
Thread safe crc32 table initialization
steveanton May 15, 2019
4e4fc62
Roll chromium_revision 34a275789d..609f581dc6 (660187:660301)
May 16, 2019
baf9c62
Add <cstdio> include to string_builder.cc to support Android NDK r17
mpsrig May 16, 2019
198cf00
Reland "Change SimpleStringBuilder::Append to not use strcpyn and SIZ…
May 10, 2019
fdd6d3e
Delete deprecated rtc::Thread default constructor
Apr 30, 2019
6e70f18
Revert "Delete deprecated rtc::Thread default constructor"
May 16, 2019
8d3d6cf
SCTP: Treat message size zero as "responder selects"
May 16, 2019
f204787
ReportBlockData and observer added, for stats collection in future CLs.
henbos May 16, 2019
9a4c93b
Add DCHECK in LibvpxVp8Encoder
May 16, 2019
67c76b2
AEC3: Minor code corrections
May 14, 2019
a24d934
Add the option to use raw RTP packetization without the generic header.
May 16, 2019
f0792ce
Roll chromium_revision 609f581dc6..0d85f6ad4e (660301:660414)
May 16, 2019
fe57f62
Roll chromium_revision 0d85f6ad4e..64564f7a42 (660414:660541)
May 16, 2019
45a8273
Roll chromium_revision 64564f7a42..5ab21b0a3a (660541:660661)
May 17, 2019
02ed529
Revert "Roll chromium_revision 64564f7a42..5ab21b0a3a (660541:660661)"
MirkoBonadei May 17, 2019
8e1a008
Roll chromium_revision 64564f7a42..4c9872694a (660541:660753)
May 17, 2019
1e193fa
Add DecelerationTargetLevelOffset Field Trial.
May 15, 2019
f13a096
Fix memory leak in Thread::PostTask.
May 17, 2019
fd26ef7
Delete unused RTPFragmentationHeader members
May 17, 2019
519d74a
Drop data for disabled endpoints.
titov-artem May 17, 2019
df5731e
Improve spec compliance of SetStreamIDs in RtpSenderInterface
May 17, 2019
3525f86
Adds feedback generator.
jonex May 17, 2019
12f8866
Roll chromium_revision 4c9872694a..243a2094e7 (660753:660868)
May 17, 2019
b50d995
Add juberti@ to webrtc root owners
May 16, 2019
45b2e27
Remove non-source sources from binary targets
tanderson-google May 17, 2019
9fe1834
Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video.
henbos May 16, 2019
6b319e6
Add CreateDatagram to MediaTransportFactory
May 17, 2019
137f6c8
Introduce peer connection level webrtc video quality tests.
May 17, 2019
5a96a0e
Reland "Delete deprecated rtc::Thread default constructor"
Apr 30, 2019
ed4d158
Fix test names in pc_full_stack_tests.cc
May 20, 2019
60f4e29
Delete configuration of unused transport_sequence_number_allocator
May 20, 2019
03b4f9d
Update android tests to use single argument PeerConnectionFactory fac…
DanilChapovalov May 16, 2019
0ee0d1e
Roll chromium_revision 243a2094e7..f5d370078e (660868:660984)
MirkoBonadei May 20, 2019
39f4681
Remove unused dependency.
MirkoBonadei May 20, 2019
cc18917
Revert "Improve spec compliance of SetStreamIDs in RtpSenderInterface"
henrikand May 20, 2019
b9979a5
AGC2 RNN VAD: remove unused dep (KissFFT)
alebzk May 20, 2019
94079f8
Android: Add support for OpenGL ES 3
Hnoo112233 May 18, 2019
eb16697
AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate
May 16, 2019
1ff16c8
Add RtpSenderInterface.SetStreams
May 20, 2019
f4e085a
Using absl traits for checks and logging.
jonex May 20, 2019
871ac42
Refactor of GoogCC debug printer.
jonex May 17, 2019
9f864be
Roll chromium_revision f5d370078e..e7b2a8fc98 (660984:661399)
May 20, 2019
762076b
Add flag to use datagram transport (without implementation)
May 20, 2019
d9f02f6
Roll chromium_revision e7b2a8fc98..cc9f0ad182 (661399:661517)
May 20, 2019
053c371
Audio coding: Don't choke when RTP timestamp rate > sample rate
May 16, 2019
e860206
Roll chromium_revision cc9f0ad182..7a39eea5d8 (661517:661628)
May 21, 2019
0f78c6b
Delete NO_MAIN_THREAD_WRAPPING preprocessor define.
May 20, 2019
9d1840c
Revert "Delete NO_MAIN_THREAD_WRAPPING preprocessor define."
May 21, 2019
aaa1143
Use single argument PeerConnectionFactory factory in objc code
DanilChapovalov May 17, 2019
8abcf83
Adds IsEmpty to SampleStats.
jonex May 21, 2019
97716c0
Implement max-channels for SCTP datachannels.
May 21, 2019
4d29ef0
Add periodic alive message logging to prevent test infra think, that …
May 20, 2019
b0ac943
Avoid encrypting empty audio packet.
minyuel May 21, 2019
3be9da3
Make unpack_aecdump unpack RuntimeSettings
May 21, 2019
19da5ce
Formatting of WebRTC-Vp9InterLayerPred field trial.
May 20, 2019
d703cd0
Revert "Avoid encrypting empty audio packet."
minyuel May 21, 2019
04a3cc1
Delete rtc_base/unittest_main.cc
May 21, 2019
c701dec
Add GetTransportParametersOffer method for DatagramTransportInterface
May 21, 2019
acab559
Adds overuse predictor to GoogCC.
jonex May 21, 2019
3b112e2
Delete multi-parameter CreateModularPeerConnectionFactory
DanilChapovalov May 20, 2019
9c91887
Splits SendTimeHistory::AddAndRemoveOld into Add/Remove.
jonex May 20, 2019
4880e15
Roll chromium_revision 7a39eea5d8..0c18b1a229 (661628:661811)
May 21, 2019
4f08faa
Introduce MediaTransportConfig
May 21, 2019
5f19f8f
Roll chromium_revision 0c18b1a229..1216f271d5 (661811:661928)
May 21, 2019
bb90ccc
Roll chromium_revision 1216f271d5..15b783dc7c (661928:662034)
May 22, 2019
afb8d5c
Log average decoded and rendered framerate for a VideoReceiveStream.
May 21, 2019
b5d9183
Add RTP timestamp to contributing sources
May 21, 2019
04f3924
Delete no longer used windows helpers
May 21, 2019
23aff9b
Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget.
henbos May 20, 2019
b32f2c7
Publish rtc event log api and default factory for it in api/
DanilChapovalov May 22, 2019
040dc43
Fix shadowing of override_field_trials_ in WebRtcVideoEngineTest
May 22, 2019
4ed7e51
Revert "Add ability to cap the video jitter estimate to a max value."
May 22, 2019
9ce451a
End NetEq simulation if there are no more packets to decode.
May 22, 2019
58c71db
Fix for crash in event log when using scenario tests.
jonex May 22, 2019
646fda0
Implement RTCMediaSourceStats and friends in standard getStats().
henbos May 22, 2019
7e7c5c3
WebRTC Opus C interface: Add support for non-48 kHz encode sample rate
May 21, 2019
ecd3054
Replace a broken assumption in candidate gathering for mDNS candidates.
May 22, 2019
4024440
Lowercase windows includes in desktop_capture/.
May 23, 2019
2799e63
Add sizes of spatial layer frames to EncodedImage
May 17, 2019
39ece6d
Delete unused method ModuleRtpRtcpImpl::SendCompoundRTCP
May 23, 2019
e7e3601
Remove hex_encode functions with raw buffer output from the header file
May 23, 2019
a352248
Add a config flag to disable the audio ALR probing request.
rodbro May 23, 2019
fe68daa
Add option to configure raw RTP packetization per payload type.
May 23, 2019
eb180f8
Fix incorrect libvpx vp9 dynamic rate control settings
May 23, 2019
74b373f
Delete STACK_ARRAY macro, and use of alloca
May 23, 2019
e9a2ee2
Improve NetEq network adaptation in the beginning of the call.
May 22, 2019
890bc30
Cleanup of video packet overhead calculation.
jonex May 23, 2019
62ce035
RtpVideoSender nits
May 23, 2019
2988aca
Fix chromium autoroller to parse new clang revision format
May 23, 2019
c1b3666
Revert "Delete STACK_ARRAY macro, and use of alloca"
henbos May 23, 2019
51f5790
Roll chromium_revision 15b783dc7c..b82a501520 (662034:662691)
May 23, 2019
4163317
[PeerConnection::AddIceCandidate()] Use mid to look up contents in re…
May 23, 2019
c1c0d6d
Roll chromium_revision b82a501520..8b25075ed7 (662691:662811)
May 23, 2019
316f3ac
Datagram Transport Integration
May 23, 2019
4c55c89
Roll chromium_revision 8b25075ed7..8ae1a64b43 (662811:662926)
May 24, 2019
f3db34d
Revert "Cleanup of video packet overhead calculation."
jonex May 24, 2019
eb1754c
VP9 screenshare: Don't base layers frame-rate on input frame-rate
May 24, 2019
ce72323
Revert "VP9 screenshare: Don't base layers frame-rate on input frame-…
May 24, 2019
f25df35
Reland "Delete STACK_ARRAY macro, and use of alloca"
May 24, 2019
479c055
Let RtpVideoStreamReceiver implement KeyFrameRequestSender
May 23, 2019
5c18a5f
Reland "VP9 screenshare: Don't base layers frame-rate on input frame-…
May 24, 2019
815b1a6
Use preprocessor to strip H264 implementation.
MirkoBonadei May 24, 2019
fadb181
Negotiate use of RTCP loss notification feedback (LNTF)
May 24, 2019
a8cf3b7
Ensure CpuInfo::DetectNumberOfCores is > 0 and thread safe.
MirkoBonadei May 24, 2019
0730872
Roll chromium_revision 8ae1a64b43..e1ec78e27e (662926:663034)
May 24, 2019
a0e9943
Negotiation of LNTF controls instantiation of RTPSenderVideo::rtp_seq…
May 24, 2019
4b27648
Avoid the render lock in AudioProcessingImpl::ProcessStream
May 23, 2019
4c29546
Add test to cover bug in vp9 wrapper, triggered by field trial
May 24, 2019
d9b4f33
Cleanup of AudioAllocationSettings flags.
jonex May 23, 2019
039a714
VP9 screenshare: drop base layer separately
May 24, 2019
34cd485
Delete the remaining ORTC interfaces.
May 24, 2019
8b096a0
LogToSderr by default in WebRTC tests
May 23, 2019
3a1b927
Remove rtp_ and rtcp_packet_transport() from the RtpTransport interface.
May 24, 2019
15baf5e
Remove last mention of ortc from the codebase.
MirkoBonadei May 25, 2019
ad44b75
Roll chromium_revision e1ec78e27e..60cc82f9b7 (663034:663509)
May 27, 2019
87da109
Make ReceiveStatisticsImpl::SetMaxReorderingThreshold apply per ssrc
May 24, 2019
e0eb325
AudioEncoderOpusImpl: Remove unused static methods
May 23, 2019
87e3f9d
[video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo
henbos May 27, 2019
6e436d1
[audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo
henbos May 27, 2019
883eefc
Implement RTCRemoteInboundRtpStreamStats for both audio and video.
henbos May 27, 2019
126f2b3
AudioEncoderOpus: Add support for 16 kHz input sample rate
May 24, 2019
6019d43
Removes using imports from flexfec_receiver.
jonex May 27, 2019
2e8d78c
Allow overriding subsets of probing field trials
May 27, 2019
36bc4f8
Add thread guards to cricket::P2PTransportChannel.
May 27, 2019
2370242
Enable flex fec support in PC quality test framework
May 27, 2019
8b27910
Include downlink delay into congestion window size.
yingwang May 27, 2019
3a072de
Roll chromium_revision 60cc82f9b7..13f6824c51 (663509:663612)
May 27, 2019
ca2c430
Allow both LNTF to coexist with NACKs and key frame requests
May 27, 2019
f2e9cab
Fix BWE simulation graph in event log visualization
May 27, 2019
a33a860
Deprecate functions returning cricket::DataContentDescription.
May 28, 2019
4ffed7c
Add field trial for selecting potentially useful packets as padding.
May 28, 2019
9a57350
Use ';' to escape '/' characters in path to dumped received video stream
May 28, 2019
9ab520e
Reland "Avoid encrypting empty audio packet."
minyuel May 28, 2019
0b97e17
Cleanup of CongestionWindowDownlinkDelay trial.
jonex May 28, 2019
787f4b2
Fix text logging of ALR detector experiment settings.
May 28, 2019
07fc398
Roll chromium_revision 13f6824c51..9809faf8ca (663612:663719)
May 28, 2019
ce33b6a
Implement QualityLimitationReasonTracker and expose "reason".
henbos May 28, 2019
f94e3d9
Roll chromium_revision 9809faf8ca..09fae7ef1b (663719:663849)
May 28, 2019
64e97cf
Roll chromium_revision 09fae7ef1b..9b60f86c15 (663849:663961)
May 28, 2019
44bd71c
Create a composite implementation of RtpTransportInternal.
May 28, 2019
686be20
Fix ICE connection in datagram_transport.
May 28, 2019
e4470cd
Roll chromium_revision 9b60f86c15..99181c0bec (663961:664078)
May 29, 2019
6737841
Add jitterBufferDelay and jitterBufferEmittedCount stats for video
May 28, 2019
98266a4
Roll chromium_revision 99181c0bec..d4906ebd49 (664078:664184)
May 29, 2019
232b6a1
Propagate screenshare info into video track and it's source.
May 29, 2019
a1d1a1e
WebRTC Opus C interface: Add support for non-48 kHz decode sample rate
May 28, 2019
28f0eb2
Move H.264 SPS VUI rewriting to FrameEncodeMetadataWriter.
May 28, 2019
e8e7d7b
Move Connection into it's own .h/.cc file.
May 29, 2019
2f5554d
Make KeyFrameRequestSender injectable in RtpVideoStreamReceiver
May 29, 2019
ed69d41
Remove deprecated RtcEventLog Create functions
DanilChapovalov May 29, 2019
7eb0a5e
AudioDecoderOpus: Add support for 16 kHz output sample rate
May 29, 2019
b3b3e3f
Add acked bandwidth estimator config for sample uncertainty in ALR.
rodbro May 29, 2019
845c6aa
Add support for early loss detection using transport feedback.
May 29, 2019
85a9d91
Add ability to set min/start/max bitrate on peer's PC in PC quality t…
May 29, 2019
6806550
Fix build with recent linux kernel.
emilio May 29, 2019
740cc35
Roll chromium_revision d4906ebd49..8891f34d24 (664184:664289)
May 29, 2019
72055b1
Roll chromium_revision 8891f34d24..2d1120f0c1 (664289:664417)
May 29, 2019
74bebc5
Add OnDatagramAcked interface
May 29, 2019
a913c12
Roll chromium_revision 2d1120f0c1..81e506385d (664417:664522)
May 29, 2019
0c1c1b4
Move ownership of ICE from DtlsTransport to JsepTransport.
May 30, 2019
57dc02a
Add receive_timestamp to DatagramAcks.
May 30, 2019
d91cdbd
[getStats] Make remote-inbound-rtp.ssrc match outbound-rtp.ssrc.
henbos Jun 24, 2019
1720171
Merge to M76: Partially revert of ColorSpace information copying arou…
Jul 12, 2019
9863f3d
Merge to M76: Use the dummy address 0.0.0.0:9 in the c= and the m= li…
Jul 15, 2019
9ca8f80
Some changes for building with OWT. (#13)
jianjunz Jul 29, 2019
ae00694
Update webrtc visiblity to SDK (#23)
taste1981 Aug 18, 2019
00de26d
H265 enabling step1: SPS parser for hevc
taste1981 May 20, 2016
6c5fb02
Rebase all changes for HEVC to M70
taste1981 Oct 22, 2018
a7f6e53
Fix ios build issue #164
taste1981 Aug 21, 2019
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Update android tests to use single argument PeerConnectionFactory fac…
…tory

Bug: webrtc:10284
Change-Id: Ifd3e2322f6fe01ed7ad9254c7d4e8cddca59b491
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137051
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27985}
  • Loading branch information
DanilChapovalov authored and Commit Bot committed May 20, 2019
commit 03b4f9d1f8a8f990c143fbc36f3b667635503143
7 changes: 2 additions & 5 deletions examples/androidnativeapi/BUILD.gn
Original file line number Diff line number Diff line change
@@ -49,14 +49,11 @@ if (is_android) {
":generated_jni",
"../../api:scoped_refptr",
"//api:libjingle_peerconnection_api",
"//api/audio_codecs:builtin_audio_decoder_factory",
"//api/audio_codecs:builtin_audio_encoder_factory",
"//api/video:builtin_video_bitrate_allocator_factory",
"//api/task_queue:default_task_queue_factory",
"//logging:rtc_event_log_impl_base",
"//media:rtc_audio_video",
"//media:rtc_internal_video_codecs",
"//modules/audio_processing",
"//modules/audio_processing:api",
"//media:rtc_media_engine_defaults",
"//modules/utility",
"//pc:libjingle_peerconnection",
"//rtc_base",
39 changes: 22 additions & 17 deletions examples/androidnativeapi/jni/android_call_client.cc
Original file line number Diff line number Diff line change
@@ -13,16 +13,14 @@
#include <utility>

#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/peer_connection_interface.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "examples/androidnativeapi/generated_jni/jni/CallClient_jni.h"
#include "logging/rtc_event_log/rtc_event_log_factory.h"
#include "media/engine/internal_decoder_factory.h"
#include "media/engine/internal_encoder_factory.h"
#include "media/engine/webrtc_media_engine.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "media/engine/webrtc_media_engine_defaults.h"
#include "sdk/android/native_api/jni/java_types.h"
#include "sdk/android/native_api/video/wrapper.h"

@@ -156,19 +154,26 @@ void AndroidCallClient::CreatePeerConnectionFactory() {
signaling_thread_->SetName("signaling_thread", nullptr);
RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread";

std::unique_ptr<cricket::MediaEngineInterface> media_engine =
cricket::WebRtcMediaEngineFactory::Create(
nullptr /* adm */, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
absl::make_unique<webrtc::InternalEncoderFactory>(),
absl::make_unique<webrtc::InternalDecoderFactory>(),
nullptr /* audio_mixer */, webrtc::AudioProcessingBuilder().Create());
RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get();

pcf_ = CreateModularPeerConnectionFactory(
network_thread_.get(), worker_thread_.get(), signaling_thread_.get(),
std::move(media_engine), webrtc::CreateCallFactory(),
webrtc::CreateRtcEventLogFactory());
webrtc::PeerConnectionFactoryDependencies pcf_deps;
pcf_deps.network_thread = network_thread_.get();
pcf_deps.worker_thread = worker_thread_.get();
pcf_deps.signaling_thread = signaling_thread_.get();
pcf_deps.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
pcf_deps.call_factory = webrtc::CreateCallFactory();
pcf_deps.event_log_factory = absl::make_unique<webrtc::RtcEventLogFactory>(
pcf_deps.task_queue_factory.get());

cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory = pcf_deps.task_queue_factory.get();
media_deps.video_encoder_factory =
absl::make_unique<webrtc::InternalEncoderFactory>();
media_deps.video_decoder_factory =
absl::make_unique<webrtc::InternalDecoderFactory>();
webrtc::SetMediaEngineDefaults(&media_deps);
pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
RTC_LOG(LS_INFO) << "Media engine created: " << pcf_deps.media_engine.get();

pcf_ = CreateModularPeerConnectionFactory(std::move(pcf_deps));
RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_;
}

5 changes: 2 additions & 3 deletions sdk/android/BUILD.gn
Original file line number Diff line number Diff line change
@@ -1550,16 +1550,15 @@ if (is_android) {
":opensles_audio_device_module",
":video_jni",
"../../api:scoped_refptr",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",
"../../api/task_queue:default_task_queue_factory",
"../../api/video:video_frame",
"../../logging:rtc_event_log_impl_base",
"../../media:rtc_audio_video",
"../../media:rtc_internal_video_codecs",
"../../media:rtc_media_base",
"../../media:rtc_media_engine_defaults",
"../../modules/audio_device",
"../../modules/audio_device:mock_audio_device",
"../../modules/audio_processing",
"../../modules/audio_processing:api",
"../../modules/utility",
"../../pc:libjingle_peerconnection",
Original file line number Diff line number Diff line change
@@ -10,14 +10,13 @@
#include "sdk/android/native_api/peerconnection/peer_connection_factory.h"

#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "logging/rtc_event_log/rtc_event_log_factory.h"
#include "media/base/media_engine.h"
#include "media/engine/internal_decoder_factory.h"
#include "media/engine/internal_encoder_factory.h"
#include "media/engine/webrtc_media_engine.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "media/engine/webrtc_media_engine_defaults.h"
#include "rtc_base/logging.h"
#include "sdk/android/generated_native_unittests_jni/jni/PeerConnectionFactoryInitializationHelper_jni.h"
#include "sdk/android/native_api/audio_device_module/audio_device_android.h"
@@ -42,20 +41,29 @@ rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> CreateTestPCF(
// webrtc/rtc_base/ are convoluted, we simply wrap here to avoid having to
// think about ramifications of auto-wrapping there.
rtc::ThreadManager::Instance()->WrapCurrentThread();
auto adm = CreateJavaAudioDeviceModule(jni, GetAppContextForTest(jni).obj());

std::unique_ptr<cricket::MediaEngineInterface> media_engine =
cricket::WebRtcMediaEngineFactory::Create(
adm, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
absl::make_unique<webrtc::InternalEncoderFactory>(),
absl::make_unique<webrtc::InternalDecoderFactory>(),
nullptr /* audio_mixer */, webrtc::AudioProcessingBuilder().Create());
RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get();

auto factory = CreateModularPeerConnectionFactory(
network_thread, worker_thread, signaling_thread, std::move(media_engine),
webrtc::CreateCallFactory(), webrtc::CreateRtcEventLogFactory());

PeerConnectionFactoryDependencies pcf_deps;
pcf_deps.network_thread = network_thread;
pcf_deps.worker_thread = worker_thread;
pcf_deps.signaling_thread = signaling_thread;
pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory();
pcf_deps.call_factory = CreateCallFactory();
pcf_deps.event_log_factory =
absl::make_unique<RtcEventLogFactory>(pcf_deps.task_queue_factory.get());

cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory = pcf_deps.task_queue_factory.get();
media_deps.adm =
CreateJavaAudioDeviceModule(jni, GetAppContextForTest(jni).obj());
media_deps.video_encoder_factory =
absl::make_unique<webrtc::InternalEncoderFactory>();
media_deps.video_decoder_factory =
absl::make_unique<webrtc::InternalDecoderFactory>();
SetMediaEngineDefaults(&media_deps);
pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
RTC_LOG(LS_INFO) << "Media engine created: " << pcf_deps.media_engine.get();

auto factory = CreateModularPeerConnectionFactory(std::move(pcf_deps));
RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << factory;
RTC_CHECK(factory) << "Failed to create the peer connection factory; "
<< "WebRTC/libjingle init likely failed on this device";