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}
This gets rid of a bit of codec-specific code in VoE.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2355483003
Cr-Commit-Position: refs/heads/master@{#14614}
NetEq already uses SdpAudioFormat internally; this CL adds an
AudioCodingModule::RegisterReceiveCodec overload that accepts
SdpAudioFormat, and propagates it through AcmReceiver into NetEq.
The intention is to get rid of the other ways to specify decoders and
always use SdpAudioFormat. (And eventually to do the same for encoders
too.)
NOTRY=true
BUG=5801
Review-Url: https://codereview.webrtc.org/2365653004
Cr-Commit-Position: refs/heads/master@{#14506}
It's a very general type, and we're about to start needing it in other
places besides AudioCodingModule.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2380463003
Cr-Commit-Position: refs/heads/master@{#14423}
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}
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}
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}
The new method returns the current total delay (packet buffer and sync
buffer) in ms, with smoothing applied to even out short-time
fluctuations due to jitter. The packet buffer part of the delay is not
updated during DTX/CNG periods.
This CL also pipes the new metric through ACM and uses it in
VoiceEngine. It replaces the previous method of estimating the buffer
delay (where an inserted packet's RTP timestamp was compared with the
last played timestamp from NetEq). The new method works better under
periods of DTX/CNG.
Review-Url: https://codereview.webrtc.org/2262203002
Cr-Commit-Position: refs/heads/master@{#13855}
There was a fast path in PreprocessToAddData that would just use the
input timestamps if the input format was equal to the required format of
the encoder. This works well as long as the codec never changes. If we
are first doing resampling (specifically upsampling) and then change to
a codec that does not require resampling, we'll need to stick to
whatever input timestamp we left off at, rather than silently accepting
whatever we're sent.
BUG=622435
Review-Url: https://codereview.webrtc.org/2119393002
Cr-Commit-Position: refs/heads/master@{#13398}
AudioCodingModuleImpl is the only implementation of the
AudioCodingModule interface (except for test mocks). So it's a good
fit to put it in an anonymous namespace in the interface's .cc file,
to ensure that no one except AudioCodingModule::Create ever references
it.
Except for moving code, this CL introduces two other small changes:
* It cleans up the set of #includes in audio_coding_module.cc.
Specifically, I removed #includes that were already present in
audio_coding_module.h, and did not bring along any #includes from
audio_coding_module_impl.h and .cc except those that were
necessary to get it to compile.
* It moves AudioCodingModuleImpl from the webrtc::acm2 to the
webrtc::<anonymous> namespace. This means I had to qualify a few
things it references with acm2::.
Review-Url: https://codereview.webrtc.org/2069723003
Cr-Commit-Position: refs/heads/master@{#13191}
Channel's API remains unchanged, but the creation of a BuiltinAudioDecoderFactory is now in Channel. The next step would be to amend Channel's API (through CreateChannel, I believe) to allow an AudioDecoderFactory to be sent along.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1992763002
Cr-Commit-Position: refs/heads/master@{#12893}
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
