In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
const int16_t* data() const;
int16_t* mutable_data();
- data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames.
- mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_.
These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation.
This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later.
BUG=webrtc:7343
TBR=henrika
Review-Url: https://codereview.webrtc.org/2750783004
Cr-Commit-Position: refs/heads/master@{#18543}
With this CL, all tests and tools under the neteq/ folder are
converted to use RTPHeader instead of WebRtcRTPHeader. WebRtcRTPHeader
has an RTPHeader as a member. None of the other member in
WebRtcRTPHeader where used.
TBR=kjellander@webrtc.org
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2809153002
Cr-Commit-Position: refs/heads/master@{#17845}
Reason for revert:
Downstream roadblock should be cleared by now. Relanding original patch.
Original issue's description:
> Revert of Change NetEq::InsertPacket to take an RTPHeader (patchset #2 id:20001 of https://codereview.webrtc.org/2807273004/ )
>
> Reason for revert:
> Broke downstream dependencies.
>
> Original issue's description:
> > Change NetEq::InsertPacket to take an RTPHeader
> >
> > It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
> > a member. None of the other member in WebRtcRTPHeader where used in
> > NetEq.
> >
> > This CL adapts the production code; tests and tools will be converted
> > in a follow-up CL.
> >
> > BUG=webrtc:7467
> >
> > Review-Url: https://codereview.webrtc.org/2807273004
> > Cr-Commit-Position: refs/heads/master@{#17652}
> > Committed: 4d027576a6
>
> TBR=ivoc@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7467
>
> Review-Url: https://codereview.webrtc.org/2812933002
> Cr-Commit-Position: refs/heads/master@{#17657}
> Committed: 10d095d4f7R=ivoc@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2835093002 .
Cr-Commit-Position: refs/heads/master@{#17843}
Reason for revert:
Broke downstream dependencies.
Original issue's description:
> Change NetEq::InsertPacket to take an RTPHeader
>
> It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
> a member. None of the other member in WebRtcRTPHeader where used in
> NetEq.
>
> This CL adapts the production code; tests and tools will be converted
> in a follow-up CL.
>
> BUG=webrtc:7467
>
> Review-Url: https://codereview.webrtc.org/2807273004
> Cr-Commit-Position: refs/heads/master@{#17652}
> Committed: 4d027576a6TBR=ivoc@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2812933002
Cr-Commit-Position: refs/heads/master@{#17657}
It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
a member. None of the other member in WebRtcRTPHeader where used in
NetEq.
This CL adapts the production code; tests and tools will be converted
in a follow-up CL.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2807273004
Cr-Commit-Position: refs/heads/master@{#17652}
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}
And implement SampleRateHz in a bunch of mocks.
BUG=webrtc:5801
NOTRY=true
Review-Url: https://codereview.webrtc.org/2029543002
Cr-Commit-Position: refs/heads/master@{#13161}
This allows us to get rid of the function that computes it, which gets
us one step closer to getting rid of the NetEqDecoder type.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2021063002
Cr-Commit-Position: refs/heads/master@{#12974}
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 CL implements the muted output functionality in NetEq. Tests are
added. The feature is currently off by default, and AcmReceiver makes
sure that the muted state is not engaged.
BUG=webrtc:5608
Review-Url: https://codereview.webrtc.org/1965733002
Cr-Commit-Position: refs/heads/master@{#12711}
The type is included in the AudioFrame output parameter.
Rename the type NetEqOutputType to just OutputType, since it is now
internal to NetEq.
BUG=webrtc:5607
Review URL: https://codereview.webrtc.org/1769883002
Cr-Commit-Position: refs/heads/master@{#11903}
With this change, NetEq now uses AudioFrame as output type, like the
surrounding functions in ACM and VoiceEngine already do.
The computational savings is probably slim, since one memcpy is
removed while another one is added (both in AcmReceiver::GetAudio).
More simplifications and clean-up will be done in
AcmReceiver::GetAudio in future CLs.
BUG=webrtc:5607
Review URL: https://codereview.webrtc.org/1750353002
Cr-Commit-Position: refs/heads/master@{#11874}
The new fields are default-populated for built-in decoders, but for
external decoders, the name can now be given when registering the
decoder.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1484343003
Cr-Commit-Position: refs/heads/master@{#10952}
Also removes virtual from VideoDecoder::Decode and updated mocks and
tests accordingly to use VideoDecoder::DecodeInternal instead.
BUG=webrtc:5167
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1512483003 .
Cr-Commit-Position: refs/heads/master@{#10935}
This operation was relatively simple, since no one was doing anything
fishy with this enum. A large number of lines had to be changed
because the enum values now live in their own namespace, but this is
arguably worth it since it is now much clearer what sort of constant
they are.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1424083002
Cr-Commit-Position: refs/heads/master@{#10449}
The Init() method was previously used to initialize and reset
decoders, and returned an error code. The new Reset() method is used
for reset only; the constructor is now responsible for fully
initializing the AudioDecoder.
Reset() doesn't return an error code; it turned out that none of the
functions it ended up calling could actually fail, so this CL removes
their error return codes as well.
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1319683002 .
Cr-Commit-Position: refs/heads/master@{#9798}
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
NetEQ can crash when decoder gives too many output samples than it can handle. A practical case this happens is when multiple opus packets are combined.
The best solution is to pass the max size to the ACM decode function and let it return a failure if the max size if too small.
BUG=4361
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45619004
Cr-Commit-Position: refs/heads/master@{#8730}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8730 4adac7df-926f-26a2-2b94-8c16560cd09d
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
This should be safe to land now that issue 4143 was resolved (in r8492).
This change effectively reverts 8488.
TBR=kwiberg@webrtc.org
Original commit message:
This CL changes the way the decoder sample rate is set and updated. In
practice, it only concerns the iSAC (float) codec.
One single iSAC decoder instance is used for both wideband and
super-wideband decoding, and the instance must be told to switch
output frequency if the payload type changes. This used to be done
through a call to UpdateDecoderSampleRate, but is now instead done in
the Decode call as an extra parameter.
Review URL: https://webrtc-codereview.appspot.com/39289004
Cr-Commit-Position: refs/heads/master@{#8496}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8496 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL changes the way the decoder sample rate is set and updated. In
practice, it only concerns the iSAC (float) codec.
One single iSAC decoder instance is used for both wideband and
super-wideband decoding, and the instance must be told to switch
output frequency if the payload type changes. This used to be done
through a call to UpdateDecoderSampleRate, but is now instead done in
the Decode call as an extra parameter.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34349004
Cr-Commit-Position: refs/heads/master@{#8476}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8476 4adac7df-926f-26a2-2b94-8c16560cd09d
Since r7255, it could happen that an old packet would block the decoding
process until enough packet was received for the buffer to flush. This
CL fixes that by:
- Partially reverting r7255;
- Remove recent old packets before taking a decision for GetAudio;
- Remove all old packets after a packet has been extracted for decoding;
- Adding tests for reordered packets.
BUG=chrome:423985
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
Before this change it could happen that a large jump in timestamp (a
jump not correlated to wall-clock change) caused the audio to go silent
without recovering. The reason was that all incoming packets after the
jump were considered too old compared to the last decoded packet, and
were deleted. With CL changes two things:
1. If the only available packet in the buffer is an old packet, NetEq
will do Expand instead of immediate reset. This is to avoid that one
late packet triggers a reset.
2. Old packets are discarded only when the decision to decode a packet
has been taken. This is to allow the buffer to grow and eventually
flush if no decodable packet has been found for some time.
This CL also includes a new unit test for this situation.
BUG=3785
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7255 4adac7df-926f-26a2-2b94-8c16560cd09d
