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gerrit-public.fairphone.software
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platform
/
external
/
webrtc
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119e2197b75efa28a901231d4812258940a98a16
/
audio
86d053c
Use source_sets in component builds and static_library in release builds.
by Mirko Bonadei
· 5 years ago
dabdde6
Avoid running NullAudioPoller without receiving streams
by Gustaf Ullberg
· 5 years ago
9429888
Delete deprecated bytes_sent/bytes_rcvd stat values
by Niels Möller
· 5 years ago
f39c815
Cleanup: Replacing set extension status bool with CHECK.
by Sebastian Jansson
· 5 years ago
ac0a4cb
Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
by Niels Möller
· 5 years ago
eb90e6f
Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest
by Danil Chapovalov
· 5 years ago
ef0627f
Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
by Mirko Bonadei
· 5 years ago
fbde32e
Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
by Niels Möller
· 5 years ago
4b64411
NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate
by Karl Wiberg
· 5 years ago
cd0eedb
Don't allocate audio if we have no transport sequence number.
by Sebastian Jansson
· 5 years ago
0a6510d
Removes rtp_transport checks in AudioSendStream
by Sebastian Jansson
· 5 years ago
35cf9e7
Replaces static modifier functions in AudioSendStream.
by Sebastian Jansson
· 5 years ago
ea55b08
Adds support for passing a vector of packets to the paced sender.
by Erik Språng
· 5 years ago
0429f78
Base overhead calculation for audio priority rate on available data.
by Sebastian Jansson
· 5 years ago
f23131f
Removing AudioAllocationSettings moving functionality to AudioSendStream.
by Sebastian Jansson
· 5 years ago
62aee93
Adds trial to calculate audio overhead based on available data.
by Sebastian Jansson
· 5 years ago
44db436
Propagate task queue to create test::DirectTransport by TaskQueueBase interface
by Danil Chapovalov
· 5 years ago
01dd885
Moves contents of bitrate_controller to goog_cc
by Sebastian Jansson
· 5 years ago
40de3cc
Propagating TargetRate struct to BitrateAllocator.
by Sebastian Jansson
· 5 years ago
93b1ea2
Using struct for bitrate allocation limits.
by Sebastian Jansson
· 5 years ago
ee5ec9a
Replacing local closure classes with C++14 moving capture lambdas.
by Sebastian Jansson
· 5 years ago
317a1f0
Use std::make_unique instead of absl::make_unique.
by Mirko Bonadei
· 5 years ago
eaaaf41
Introduce api/crypto/BUILD.gn.
by Mirko Bonadei
· 5 years ago
65f17ca
Move MediaTransportInterface out of the libjingle_peerconnection_api target
by Niels Möller
· 5 years ago
6516f76
Deprecate SingleThreadedTaskQueueForTesting class.
by Yves Gerey
· 5 years ago
a837030
Split out RtpSource from libjingle_peerconnection_api
by Niels Möller
· 5 years ago
65024d9
Remove clock drift metric from NetEq.
by Jakob Ivarsson
· 5 years ago
b6220d9
Delete unused logic for audio RtcpMode::kOff
by Niels Möller
· 5 years ago
f13df86
Delete audio methods SignalNetworkState
by Niels Möller
· 5 years ago
b4a6128
Delete unneeded dependencies on libjingle_peerconnection_api
by Niels Möller
· 5 years ago
6dcd4dc
New target for api/rtp_parameters.h and api/media_types.h.
by Niels Möller
· 5 years ago
fac7e31
Removes TransportSequenceNumberAllocator
by Erik Språng
· 5 years ago
4208a13
Removes deprecated InsertPacket/TimeToSendPacket/TimeToSendPadding
by Erik Språng
· 5 years ago
d77cc24
New const method StreamStatistician::GetStats
by Niels Möller
· 5 years ago
224c69d
Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo
by Niels Möller
· 5 years ago
70efdde
Set local ssrc at construction of Rtp module
by Erik Språng
· 5 years ago
54d5d2c
Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc
by Erik Språng
· 5 years ago
71c6b56
Allow sending abs-send-time for audio streams.
by Sebastian Jansson
· 5 years ago
58b496b
Let StreamStatistician::GetReceiveStreamDataCounters return counters by value
by Niels Möller
· 5 years ago
5b5d97c
Reland of "Reporting of decoding_codec_plc events""
by Alex Narest
· 5 years ago
b168678
Add RTC_ prefix to non-standard format specifier macro "PRIdNS"
by Oleh Prypin
· 5 years ago
83bbe91
Delete deprecated rtc_event_log header
by Danil Chapovalov
· 5 years ago
ed44f54
In ChannelReceive, use AcmReceiver directly, not AudioCodingModule
by Niels Möller
· 5 years ago
fedd625
Change 2g network pc audio test to more realistic network
by Artem Titov
· 5 years ago
054e3bb
Reland "Replace the implementation of `GetContributingSources()` on the audio side."
by Chen Xing
· 5 years ago
da4f093
Reland "Only include payload in bytes sent/received."
by Bjorn A Mellem
· 5 years ago
bedb7a8
Revert "Reporting of decoding_codec_plc events"
by Mirko Bonadei
· 5 years ago
bcd068d
Revert "Only include payload in bytes sent/received."
by Bjorn Mellem
· 5 years ago
0a88ea0
Reporting of decoding_codec_plc events
by Alex Narest
· 5 years ago
1704801
Prevent concurrent access to AudioSendStream's configuration.
by Yves Gerey
· 5 years ago
8f319a3
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
by Alessio Bazzica
· 5 years ago
fab3460
Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
by Alessio Bazzica
· 5 years ago
9973933
Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
by Chen Xing
· 5 years ago
aa59eca
Move RtpPacketSender and merge it with RtpPacketPacer.
by Erik Språng
· 5 years ago
74a1b4b
Only include payload in bytes sent/received.
by Bjorn A Mellem
· 5 years ago
cbc91ef
Improve low bandwidth audio test instrumentatin, fix PC test
by Artem Titov
· 5 years ago
2ab97f6
Migrate WebRTC test infra to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
0182a03
Reland "Remove the injectable bitrate allocation strategy API."
by Jonas Olsson
· 5 years ago
4c2c412
Set local ssrc at construction (audio)
by Erik Språng
· 5 years ago
24192c2
Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
by Ivo Creusen
· 5 years ago
e95b57c
Revert "Remove the injectable bitrate allocation strategy API."
by Mirko Bonadei
· 5 years ago
52e240e
Use 16000Hz audio in PC test when specified
by Artem Titov
· 5 years ago
b1f2d60
Reland "Fix collection of audio metrics from PC test framework for audio test"
by Artem Titov
· 5 years ago
80cb3f6
Remove the injectable bitrate allocation strategy API.
by Jonas Olsson
· 5 years ago
4876cb2
Revert "Fix collection of audio metrics from PC test framework for audio test"
by Mirko Bonadei
· 5 years ago
d0679bd
Enables usage of ChannelMixer in WebRTC's output mixer.
by henrika
· 5 years ago
2d0880b
Fix collection of audio metrics from PC test framework for audio test
by Artem Titov
· 5 years ago
4a126e4
Rename tests to prevent clashing with old audio test
by Artem Titov
· 5 years ago
a4d8737
Format almost everything.
by Jonas Olsson
· 5 years ago
c8263e0
Introduce PC level audio quality test.
by Artem Titov
· 5 years ago
2250b05
Adding support for channel mixing between different channel layouts.
by henrika
· 5 years ago
d2c336f
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
by Henrik Boström
· 5 years ago
67008df
Revert "Replace the implementation of `GetContributingSources()` on the audio side."
by Artem Titov
· 5 years ago
8fa7151
Replace the implementation of `GetContributingSources()` on the audio side.
by Chen Xing
· 5 years ago
3e8ef94
Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
by Chen Xing
· 5 years ago
225842c
Initialize signal processing function pointers statically
by Karl Wiberg
· 5 years ago
3472b9a
Delete RTCInboundRTPStreamStats::fraction_lost
by Niels Möller
· 5 years ago
f48bca7
Avoid triggering a false error logging when using encryptor and sending DTX.
by Minyue Li
· 5 years ago
59b8654
Switch from RtpPacketSender to RtpPacketPacer interface usage.
by Erik Språng
· 5 years ago
08fa953
Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
by Danil Chapovalov
· 5 years ago
fd5166c
Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
by Philip Eliasson
· 5 years ago
fc96135
Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
by Danil Chapovalov
· 5 years ago
9ab520e
Reland "Avoid encrypting empty audio packet."
by Minyue Li
· 5 years ago
6e436d1
[audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo
by Henrik Boström
· 5 years ago
87da109
Make ReceiveStatisticsImpl::SetMaxReorderingThreshold apply per ssrc
by Niels Möller
· 5 years ago
a352248
Add a config flag to disable the audio ALR probing request.
by Christoffer Rodbro
· 5 years ago
b32f2c7
Publish rtc event log api and default factory for it in api/
by Danil Chapovalov
· 5 years ago
b5d9183
Add RTP timestamp to contributing sources
by Johannes Kron
· 5 years ago
4f08faa
Introduce MediaTransportConfig
by Anton Sukhanov
· 5 years ago
d703cd0
Revert "Avoid encrypting empty audio packet."
by Minyue Li
· 5 years ago
b0ac943
Avoid encrypting empty audio packet.
by Minyue Li
· 5 years ago
8f119ca
Enable experiments with audio bitrate priority.
by Jonas Olsson
· 5 years ago
9356252
Ensure that we always set values for min and max audio bitrate.
by Daniel Lee
· 5 years ago
8d8ffdb
Expose new audio stats on the API
by Ivo Creusen
· 5 years ago
44125fa
Reland "Piping audio interruption metrics to API layer"
by Henrik Lundin
· 5 years ago
fc02a79
Revert "Piping audio interruption metrics to API layer"
by Henrik Andreassson
· 5 years ago
413ccc4
Stop DCHECK which occurs in ANA BitrateController when overhead is zero.
by Bjorn A Mellem
· 5 years ago
299c4e6
Piping audio interruption metrics to API layer
by Henrik Lundin
· 5 years ago
c35b6e6
Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData
by Niels Möller
· 5 years ago
30a276b
Add RTP sequence number to TransportFeedbackObserver::AddPacket()
by Erik Språng
· 5 years ago
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