There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.
It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
Reason for revert:
Internal bots failed.
Original issue's description:
> Reland "Move smoothing filter to common audio".
>
> The original CL was this https://codereview.webrtc.org/2484153002/
>
> Due to failure with internal trial servers, it was reverted. This CL tries to reland it.
>
> BUG=webrtc:6443
>
> Committed: https://crrev.com/223641f1b903e41e77d88f03199b4cdb65255ec8
> Cr-Commit-Position: refs/heads/master@{#15227}
TBR=tommi@webrtc.org,solenberg@webrtc.org,terelius@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:6443
Review-Url: https://codereview.webrtc.org/2529943002
Cr-Commit-Position: refs/heads/master@{#15228}
This change adds code that lets Opus increase the complexity setting
at low bitrates (only relevant for mobile where the default complexity
is not already maximum). The feature is default off.
Also adding a performance test to make sure the complexity adaptation
has desired effect.
BUG=webrtc:6708
Review-Url: https://codereview.webrtc.org/2503443002
Cr-Commit-Position: refs/heads/master@{#15182}
It turns out that that audio network adaptor can always be created with knowledge of receiver frame length range. There is no need to keep some infrastructure that is used for runtime setting of receiver frame length ranges.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2429503002
Cr-Commit-Position: refs/heads/master@{#14748}
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}
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}
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}
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}
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}
We hit a fuzzer bug that caused numDecodedBytesLB + numDecodedBytesUB
> lenEncodedBytes, which is obviously bogus. Check for that, and for
the case whhere the UB decoder itself realized that something was
wrong. (The code already makes the corresponding check for the LB
decoder.)
BUG=chromium:637899
Review-Url: https://codereview.webrtc.org/2315693002
Cr-Commit-Position: refs/heads/master@{#14091}
The sample uses are from when I debugged bug 617124. The change in neteq_network_stats_unittest.cc is a fix for a minor unrelated bug found by the try bots when I tried to land this CL (a test was passing uninitialized packet data to NetEq).
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2293893002
Cr-Commit-Position: refs/heads/master@{#14034}
We have RTC_CHECK and RTC_DCHECK for C now, so we should use it. It's
one fewer difference between our C and C++ code.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2274083002
Cr-Commit-Position: refs/heads/master@{#13930}
We detect an unreasonable state (caused by a bad encoded stream)
before it can lead to problems, and handle it by resetting the
decoder.
NOPRESUBMIT=true
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2255203002
Cr-Commit-Position: refs/heads/master@{#13888}
So that we don't have to use assert(). Includes one sample call site.
NOTRY=true
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2262173002
Cr-Commit-Position: refs/heads/master@{#13862}
scale1 == 31 if and only if w10 == 0. So even though 1 << scale1
overflows, we know that the result of the multiplication should be 0.
Handle that case.
BUG=chromium:615818
Review-Url: https://codereview.webrtc.org/2258543002
Cr-Commit-Position: refs/heads/master@{#13847}
Reland of https://codereview.webrtc.org/2072753002/ which broke
chromium due to how their build was setup. This issue should now be
resolved.
Changed WebRtcVoiceEngine to present receive codecs from the formats
provided by its decoder factory. Added supported formats to
BuiltinAudioDecoderFactory. Added helper functions for creating some
simple decoder factories for mocking.
Created a PayloadTypeMapper for assigning payload types to formats. I
think we'll eventually want to use this further up, or possibly in
both the audio and video sides. It would be best if the engines didn't
have to talk payload types at all, but it might be more difficult to
get around when payload types depend on each-other, like the RTX
codecs for video.
BUG=webrtc:5805
TBR=ivoc@webrtc.org
Review-Url: https://codereview.webrtc.org/2250683002
Cr-Commit-Position: refs/heads/master@{#13793}
Reason for revert:
For some reason, payload_type_mapper.cc is not being picked up in Chrome builds, leading to undefined references. Reverting while investigating.
Original issue's description:
> WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.
>
> Changed WebRtcVoiceEngine to present receive codecs from the formats
> provided by its decoder factory. Added supported formats to
> BuiltinAudioDecoderFactory. Added helper functions for creating some
> simple decoder factories for mocking.
>
> Created a PayloadTypeMapper for assigning payload types to formats. I
> think we'll eventually want to use this further up, or possibly in
> both the audio and video sides. It would be best if the engines didn't
> have to talk payload types at all, but it might be more difficult to
> get around when payload types depend on each-other, like the RTX
> codecs for video.
>
> This CL also includes some changes to rtc::Optional. I've put them in
> a separate CL that should (or should not) land first, making these
> changes void.
> See: https://codereview.webrtc.org/2072713002/
>
> BUG=webrtc:5805
>
> Committed: https://crrev.com/95eb1ba0db79d8fd134ae61b0a24648598684e8a
> Cr-Commit-Position: refs/heads/master@{#13459}
TBR=ivoc@webrtc.org,tina.legrand@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2151453002
Cr-Commit-Position: refs/heads/master@{#13460}
Changed WebRtcVoiceEngine to present receive codecs from the formats
provided by its decoder factory. Added supported formats to
BuiltinAudioDecoderFactory. Added helper functions for creating some
simple decoder factories for mocking.
Created a PayloadTypeMapper for assigning payload types to formats. I
think we'll eventually want to use this further up, or possibly in
both the audio and video sides. It would be best if the engines didn't
have to talk payload types at all, but it might be more difficult to
get around when payload types depend on each-other, like the RTX
codecs for video.
This CL also includes some changes to rtc::Optional. I've put them in
a separate CL that should (or should not) land first, making these
changes void.
See: https://codereview.webrtc.org/2072713002/
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2072753002
Cr-Commit-Position: refs/heads/master@{#13459}
If the specification for the speech encoder hasn't changed, we should
reuse it instead of recreating it. Otherwise, we lose its state. (This
problem was originally discovered because AudioEncoderOpus instances
would forget that they were supposed to be using DTX.)
BUG=webrtc:6020, chromium:622647
Review-Url: https://codereview.webrtc.org/2089183002
Cr-Commit-Position: refs/heads/master@{#13273}
We didn't really want it; it was only necessary because we wanted to
use rtc::Optional<SdpAudioFormat>, and Optional used to require the
contained type to be default constructable. But as of May 9th
(https://codereview.webrtc.org/1896833004), it no longer does.
Review-Url: https://codereview.webrtc.org/2066233002
Cr-Commit-Position: refs/heads/master@{#13211}
