The test program modules/audio_processing/test/audioproc_float.cc
defined the flag 'agc_compression_gain' and had checks if the
parameter was valid (audioproc_float). The flag was also copied to
webrtc::test::SimulationSettings of audio_processing_simulator.h. The
setting was however never applied to APM.
This change applies the setting on the GainControl submodule in the
same way as the agc_target_level is applied.
This is needed for e.g. testing the AGC fixed digital limiter with the
same configuration as it is (currently) used with in AudioMixerImpl.
Also added new flag '-experimental_agc'. This flag allows disabling the
experimental AGC, which is how the AGC is used in AudioMixerImpl.
ExperimentalAgc is enabled by default, exactly as it was prior to this change.
The change has been tested locally by listening tests and diff comparisons.
BUG=None
NOTRY=True # win_dbg bot not cooperating
Review-Url: https://codereview.webrtc.org/2684983004
Cr-Commit-Position: refs/heads/master@{#16603}
The files socketpool.h and diskcache.h also become unused, and are
deleted together with their sources.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2694753002
Cr-Commit-Position: refs/heads/master@{#16601}
downstream application depends on it.
Mark the old Port::AddAddress deprecated and will be removed after the
applications stop replying on it.
BUG=None.
R=deadbeef@webrtc.org
Review-Url: https://codereview.webrtc.org/2694103003 .
Cr-Commit-Position: refs/heads/master@{#16598}
If android_setsocknetwork() is available, and it fails, then bind()
should *not* be called, and an error should be returned.
If it succeeds, then bind should be called, but with an "any" address.
This is to prevent cases where sockets are sent with a source address
that doesn't match the network interface they're sent on. See bug below.
This CL also changes "NetworkBinderResults" to an enum class, and
renames it to "NetworkBinderResult".
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2646863005
Cr-Commit-Position: refs/heads/master@{#16597}
To ensure compliance with older version high profile should appear in local SDP
before baseline profile.
BUG=b/34816463
Review-Url: https://codereview.webrtc.org/2696733002
Cr-Commit-Position: refs/heads/master@{#16596}
Reason for revert:
Breaks downstream application's build
Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> The url of the ICE server is added to the IceCandiate class.
> This can be used to tell which server this candidate was gathered from.
>
> BUG=webrtc:7128
>
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2692993002
Cr-Commit-Position: refs/heads/master@{#16595}
Makes sure video decoder software fallback handles InitDecode()
failures, and properly releases the pointer after ::Release() so that
another decode failure will properly reinitialize the decoder.
Also makes sure to not call Decode() without a previous InitDecode()
succeeding.
BUG=webrtc:7154
R=noahric@chromium.org, sophiechang@chromium.org
Review-Url: https://codereview.webrtc.org/2690183004 .
Cr-Commit-Position: refs/heads/master@{#16594}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16593}
The URL of the ICE server will be reconstructed by the Port and the URL
attribute is added to the cricket::Candidate struct so that we can tell
which ICE server the candidate was gathered from.
This CL only changes the native C++ code. The Java and Objc wrapper will
be created in separate CLs.
BUG=webrtc::7128
Review-Url: https://codereview.webrtc.org/2685053004
Cr-Commit-Position: refs/heads/master@{#16591}
Lateness is determined by the length of the send-side history, currently
set to 60 seconds.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2684353004
Cr-Commit-Position: refs/heads/master@{#16588}
This avoids redoing RTP header parsing already done in Call, for video.
The next step is to convert other types of receive streams, i.e.,
audio and flexfec, to use a compatible OnRtpPacket method. We can then
introduce a shared base interface, and simplify media-independent
receive processing in Call.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2681673004
Cr-Commit-Position: refs/heads/master@{#16583}
The only implementation which used a nullptr was a mock used in tests,
so add a dummy instance there instead.
Remove tests for stats_proxy_ in vie_encoder and just dcheck in the
constructor instead.
BUG=None
Review-Url: https://codereview.webrtc.org/2695643002
Cr-Commit-Position: refs/heads/master@{#16577}
SslSocketFactory is unused since https://codereview.webrtc.org/2506983002, and it's the last
user of AutoDetectProxy.
Also move HttpListenServer and SocksProxyServer to the rtc_base_tests_utils gn target, since they're used by tests only.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2541453002
Cr-Commit-Position: refs/heads/master@{#16576}
Updated comment.
Don't call AdaptUp/AdaptDown in tests without first emitting a frame.
Handle frame received precondition in AdaptUp/AdaptDown with DCHECK
instead of return.
BUG=webrtc:4172, webrtc:6850
Review-Url: https://codereview.webrtc.org/2690023002
Cr-Commit-Position: refs/heads/master@{#16572}
Other minor changes:
- Define locks after stuff it is protecting
- Use explicit default dtors
- Replace unnecessary lock in DelayedEncoder with SequencedTaskChecker
BUG=webrtc:7130
Review-Url: https://codereview.webrtc.org/2686103002
Cr-Commit-Position: refs/heads/master@{#16554}
