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}
Later steps in the refactoring will have the factory injected from the
outside rather than owned by NetEq.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/1928293002
Cr-Commit-Position: refs/heads/master@{#12604}
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.
NOPRESUBMIT=True
BUG=webrtc:3970
Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
Remove the deprecated EncodeInternal interface from AudioEncoder
Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
BUG=webrtc:5591
Review URL: https://codereview.webrtc.org/1881003003
Cr-Commit-Position: refs/heads/master@{#12409}
Reason for revert:
Broke import. Implementations of the old interface still exists somewhere.
Original issue's description:
> Remove the deprecated EncodeInternal interface from AudioEncoder
>
> Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
>
> BUG=webrtc:5591
>
> Committed: https://crrev.com/5222d315dbea8f3563c100cc9f2451907f70b05f
> Cr-Commit-Position: refs/heads/master@{#12329}
TBR=kwiberg@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5591
Review URL: https://codereview.webrtc.org/1883543002
Cr-Commit-Position: refs/heads/master@{#12330}
Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
BUG=webrtc:5591
Review URL: https://codereview.webrtc.org/1864993002
Cr-Commit-Position: refs/heads/master@{#12329}
Renamed the new variant of EncodeInternal to EncodeImpl, so that
subclasses implementing one of the EncodeInternal don't need to
explicitly contain 'using AudioEncoder::EncodeInternal' to avoid their
implementation hiding the other variant of EncodeInternal. This causes
a warning (treated as an error) when building using GCC.
Review URL: https://codereview.webrtc.org/1764583003
Cr-Commit-Position: refs/heads/master@{#11868}
For backwards compatibility, I've added kept the old interface to
Encode() and EncodeInternal and created default implementations of both
variants of EncodeInternal(), each calling the other. At least one of
the variants must be implemented in a subclass or we'll run out of stack
and explode. Would be nice if we could catch that before runtime. :/
The new interface to EncodeInternal() is protected, since it should
never be called from the outside.
Was unable to mark the old EncodeInternal() as RTC_DEPRECATED, since the
default implementaion of the new variant needs to call it to work around
old implementations. The old Encode() variant is deprecated, at least.
Added a test for backwards compatibility in audio_encoder_unittest.cc.
For the added test I broke out MockEncodeHelper from
audio_encoder_copy_red_unittest.cc and renamed it MockAudioEncoderHelper.
Review URL: https://codereview.webrtc.org/1725143003
Cr-Commit-Position: refs/heads/master@{#11823}
This makes the sequence of expected calls easier to read. Also, we can
save one line and get rid of a gmock warning by expecting the
MockAudioEncoder object to be destroyed at the end of the test instead
of making a final marker call.
Review URL: https://codereview.webrtc.org/1331793003
Cr-Commit-Position: refs/heads/master@{#9916}
There's no point in returning a status code, since the max playback rate
is only a suggestion that the encoder is free to disregard.
Review URL: https://codereview.webrtc.org/1332573003
Cr-Commit-Position: refs/heads/master@{#9900}
It makes more sense to combine the two interfaces, since there wasn't
a clear line separating them. The result is a combined interface with
just over a dozen methods, half of which need to be implemented by
every subclass, while the other half have sensible (and trivial)
default implementations and are implemented only by the few subclasses
that need non-default behavior.
Review URL: https://codereview.webrtc.org/1322973004
Cr-Commit-Position: refs/heads/master@{#9894}
The GetTargetBitrate implementation will return the
target bitrate of the codec. This may differ from the
desired target bitrate, as set by SetTargetBitrate, depending on implementation.
Tests are updated to exercise the new functionality.
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1184313002.
Cr-Commit-Position: refs/heads/master@{#9461}
Added method AudioEncoder::MaxEncodedBytes() and provided implementations in derived encoders. This method returns the number of bytes that can be produced by the encoder at each Encode() call.
Unit tests were updated to use the new method.
Buffer allocation was not changed in AudioCodingModuleImpl::Encode(). It will be done after additional investigation.
Other refactoring work that was done, that may not be obvious why:
1. Moved some code into AudioEncoderCng::EncodePassive() to make it more consistent with EncodeActive().
2. Changed the order of NumChannels() and RtpTimestampRateHz() declarations in AudioEncoderG722 and AudioEncoderCopyRed classes. It just bothered me that the order was not the same as in AudioEncoder class and its other derived classes.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40259005
Cr-Commit-Position: refs/heads/master@{#8671}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8671 4adac7df-926f-26a2-2b94-8c16560cd09d
Although sample_rate_hz(), num_channels(), and rtp_timestamp_rate_hz()
are simple accessors for almost all implementations of AudioEncoder,
they are virtual and not guaranteed to be just simple accessors. Thus,
it makes more sense to use the normal CamelCase naming scheme.
BUG=4235
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34239004
Cr-Commit-Position: refs/heads/master@{#8407}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8407 4adac7df-926f-26a2-2b94-8c16560cd09d
The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()
Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.
BUG=3926
COAUTHOR:kwiberg@webrtc.orgR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34049004
Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
