Original description:
Add proper lifetime of encoder-specific settings.
Permits passing VideoEncoderConfig between threads and not worry about
the lifetime of an underlying void pointer. Also adds type safety to
unpacking of codec-specific settings.
These settings are not yet propagating to VideoEncoder interfaces, but
the aim is to get rid of webrtc::VideoCodec for VideoEncoder.
BUG=webrtc:3424
R=perkj@webrtc.org, pbos@webrtc.orgTBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2347843002
Cr-Commit-Position: refs/heads/master@{#14396}
NetEqDecoder is still used in the external interfaces, but this change
opens up the ability to use SdpAudioFormats directly, once appropriate
interfaces have been added.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2355503002
Cr-Commit-Position: refs/heads/master@{#14368}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
The biggest change to NetEq is the move from a primary flag, to a
Priority with two separate levels: one set by RED splitting and one
set by the codec itself. This allows us to unambigously prioritize
"fallback" packets from these two sources. I've chosen what I believe
is the sensible ordering: packets that the codec prioritizes are
chosen first, regardless of if they are secondary RED packets or
not. So if we were to use Opus w/ FEC in RED, we'd only do Opus FEC
decoding if there was no RED packet that could cover the time slot.
With this change, PayloadSplitter now only deals with RED
packets. Maybe it should be renamed RedPayloadSplitter?
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2342443005
Cr-Commit-Position: refs/heads/master@{#14347}
audio_decoder.cc depends on LegacyEncodedAudioFrame and
LegacyEncodedAudioFrame depends on AudioDecoder::EncodedAudioFrame, so
there's no clear way to separate them as of now. This error is also
hodling up builds downstream. I expect we'll revisit these
dependencies as part of the upcoming larger restructuring effort.
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2359763002
Cr-Commit-Position: refs/heads/master@{#14329}
Reason for revert:
ODR violation
Original issue's description:
> Adding BitrateController to audio network adaptor.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/26b039a137be0a8703766f45b546b29323de714f
> Cr-Commit-Position: refs/heads/master@{#14293}
TBR=michaelt@webrtc.org,henrik.lundin@webrtc.org,krasin@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2352223002
Cr-Commit-Position: refs/heads/master@{#14327}
There's still some code run specifically for Opus w/ FEC. It will be
addressed in a separate CL.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2326003002
Cr-Commit-Position: refs/heads/master@{#14319}
Verifies that NetEq doesn't enter muted state when CNG mode is active
and the packet stream is suspended for a long time.
BUG=webrtc:5608
Review-Url: https://codereview.webrtc.org/2335343011
Cr-Commit-Position: refs/heads/master@{#14308}
It requires a new NetEq method, but it can no longer fail. And we no
longer need to use AcmReceiver::decoders_, which we're trying to
eliminate.
(This is a re-land of https://codereview.webrtc.org/2342313002.)
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2348233002
Cr-Commit-Position: refs/heads/master@{#14304}
This change adds a new statistic for logging how many calls to
NetEq::GetAudio resulted in a "muted output". A muted output happens
if the packet stream has been dead for some time (and the last decoded
packet was not comfort noise).
BUG=webrtc:5606
BUG=b/31256483
Review-Url: https://codereview.webrtc.org/2341293002
Cr-Commit-Position: refs/heads/master@{#14302}
It allows the decoder to split the input up into usable chunks before
they are put into NetEq's PacketBuffer. Eventually, all packet splitting
will move into ParsePayload.
There's currently a base implementation of ParsePayload. It will
generate a single Frame that calls the underlying AudioDecoder for
getting Duration() and to Decode.
BUG=webrtc:5805
BUG=chromium:428099
Review-Url: https://codereview.webrtc.org/2326953003
Cr-Commit-Position: refs/heads/master@{#14300}
Reason for revert:
Seems to have broken Chromium tests.
Original issue's description:
> AcmReceiver: Look up last decoder in NetEq's table of decoders
>
> AcmReceiver::decoders_ is now one step closer to being unused.
>
> BUG=webrtc:5801
>
> Committed: https://crrev.com/1e4d8b574cde64d93b98d89c7b817fb93185a307
> Cr-Commit-Position: refs/heads/master@{#14274}
TBR=ossu@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2348123002
Cr-Commit-Position: refs/heads/master@{#14279}
Reason for revert:
Seems to have broken Chromium tests.
Original issue's description:
> AcmReceiver: Ask NetEq to delete all decoders at once instead of one by one
>
> It requires a new NetEq method, but it can no longer fail. And we no
> longer need to use AcmReceiver::decoders_, which we're trying to
> eliminate.
>
> BUG=webrtc:5801
>
> Committed: https://crrev.com/f6232b43a176e1717354b671a0a52b887d70de59
> Cr-Commit-Position: refs/heads/master@{#14275}
TBR=ossu@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2349973002
Cr-Commit-Position: refs/heads/master@{#14278}
Reason for revert:
Seems to have broken Chromium tests.
Original issue's description:
> AcmReceiver::DecoderByPayloadType: Ask NetEq for decoder
>
> Instead of looking in AcmReceiver::decoders_, which we're trying to
> get rid of.
>
> BUG=webrtc:5801
>
> Committed: https://crrev.com/07772e4738ef8007280f97a0245eef34b9ca9391
> Cr-Commit-Position: refs/heads/master@{#14276}
TBR=ossu@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2346173002
Cr-Commit-Position: refs/heads/master@{#14277}
Instead of looking in AcmReceiver::decoders_, which we're trying to
get rid of.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2341283002
Cr-Commit-Position: refs/heads/master@{#14276}
It requires a new NetEq method, but it can no longer fail. And we no
longer need to use AcmReceiver::decoders_, which we're trying to
eliminate.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2342313002
Cr-Commit-Position: refs/heads/master@{#14275}
AcmReceiver::decoders_ is now one step closer to being unused.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2339953002
Cr-Commit-Position: refs/heads/master@{#14274}
Remove a large number of targets that are no longer built, to reduce maintenance.
Only targets that have a GN version were removed.
BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2340773003
Cr-Commit-Position: refs/heads/master@{#14231}
Declare resources for GN targets so that they can be isolated
NOTRY=True
BUG=chromium:497757
Review-Url: https://codereview.webrtc.org/2340753002
Cr-Commit-Position: refs/heads/master@{#14210}
The basic implementation of AudioNetworkAdaptor include the introduction of
1. Controller
2. ControllerManager
ControllerManager is to hold all needed controllers. It also orders them according to their significance in dealing with current network condition.
Controller provides an interface MakeDecision, which has to be implemented by specific controllers. AudioNetworkAdaptorImpl calls MakeDecision of the controllers in the order decided by ControllerManager to collect EncoderRuntimeConfig.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2306083002
Cr-Commit-Position: refs/heads/master@{#14201}
A left shift by 10 was assumed to never overflow, since "[s]imulation
of the 25 files shows that maximum value in the vector gain_lo_hiQ17[]
is 441344, which means that it is log2((2^31)/441344) = 12.2 shifting
bits from saturation." However, a fuzzer test succeeded in provoking
an overflow, which we ignore in this CL on the theory that only
"abnormal" inputs cause overflow.
Also had to replace a "foo << 1" with "foo * (1 << 1)" in
WEBRTC_SPL_MUL_16_32_RSFT15 because foo could be negative; this
problem showed up as soon as I'd asked UBSan to ignore the overflow
discussed above.
BUG=chromium:615819
Review-Url: https://codereview.webrtc.org/2314413002
Cr-Commit-Position: refs/heads/master@{#14162}
Reason for revert:
Broke all Chromium libFuzzer builds
https://bugs.chromium.org/p/chromium/issues/detail?id=645069
Original issue's description:
> Setting up an RTP input fuzzer for NetEq
>
> This CL introduces a new fuzzer target neteq_rtp_fuzzer that
> manipulates the RTP header fields before inserting the packets into
> NetEq. A few helper classes are also introduced.
>
> BUG=webrtc:5447
> NOTRY=True
>
> Committed: https://crrev.com/2d273f1e97cd5030ed1686f27ce1118291b66395
> Cr-Commit-Position: refs/heads/master@{#14103}
TBR=ivoc@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5447
Review-Url: https://codereview.webrtc.org/2328483002
Cr-Commit-Position: refs/heads/master@{#14131}
I could not find a single place it was used, outside of the unittests
for the sync packet support itself.
Review-Url: https://codereview.webrtc.org/2309303002
Cr-Commit-Position: refs/heads/master@{#14130}
With this CL, the NetEqReplacementInput class handles reordered and
missing packets in a better way than before, by storing the last
confirmed packet size and using that when the next packet size cannot
be calculated.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2319553003
Cr-Commit-Position: refs/heads/master@{#14122}
AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions.
This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor.
This CL illustrates the interfaces of the AudioNetworkAdaptor.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2308573002
Cr-Commit-Position: refs/heads/master@{#14115}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
This CL introduces a new fuzzer target neteq_rtp_fuzzer that
manipulates the RTP header fields before inserting the packets into
NetEq. A few helper classes are also introduced.
BUG=webrtc:5447
NOTRY=True
Review-Url: https://codereview.webrtc.org/2315633002
Cr-Commit-Position: refs/heads/master@{#14103}
