We already had a special case for android, but it only worked for arm32.
BUG=webrtc:4198, webrtc:4199
Review URL: https://codereview.webrtc.org/1512833003
Cr-Commit-Position: refs/heads/master@{#10989}
As a step toward fixing webrtc:3987, here we update the RTPencode to allow Opus RTP payloads.
BUG=webrtc:3987, webrtc:2692
Review URL: https://codereview.webrtc.org/1516653003
Cr-Commit-Position: refs/heads/master@{#10987}
They were meant to be run if we have either iSAC float or fix, but the
typo made them run for just float.
BUG=webrtc:4198, webrtc:4199
Review URL: https://codereview.webrtc.org/1513483005
Cr-Commit-Position: refs/heads/master@{#10969}
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}
The bug hasn't caused us any problems, since we don't run CNG together with Opus (our only real 48 kHz codec), but would cause problems if used with PCB16b @ 48 kHz.
BUG=webrtc:5303
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1496243002 .
Cr-Commit-Position: refs/heads/master@{#10929}
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}
This change moves the logics for keeping track of the last ouput
sample rate from AcmReceiver to NetEq, where it fits better. The
getter function AcmReceiver::current_sample_rate_hz() is renamed to
last_output_sample_rate_hz().
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1467163002
Cr-Commit-Position: refs/heads/master@{#10754}
Also clean up some include_dir entries and update the few
references to them with absolute include paths instead.
Finally fixed a few lint errors and invalid header guards.
None of these are used downstream.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_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
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1438663003 .
Cr-Commit-Position: refs/heads/master@{#10700}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=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 -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
Reading of PCAP (Wireshark) files was not possible due to a bug in the
parsing of files. This change fixes that by adding new validator methods
to RtpFileSource that can be used to determine the input file type.
R=ivoc@webrtc.org
Review URL: https://codereview.webrtc.org/1427923003
Cr-Commit-Position: refs/heads/master@{#10490}
This change avoids calling neteq_->EnableVad() and DisableVad from the
AcmReceiver constructor. Instead, the new member
enable_post_decode_vad is added to NetEq's config struct. It is
disabled by defualt, but ACM sets it to enabled. This preserves the
behavior both of NetEq stand-alone (i.e., in tests) and of ACM.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1425133002
Cr-Commit-Position: refs/heads/master@{#10476}
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}
Negative acknowledgement (NACK) has up to now been implemented in
ACM. But, since NetEq is in charge of the actual packet buffer, it
makes more sense to have the NACK functionlaity in there.
This CL does the following:
- Move nack.{h,cc} and the unit tests from main/acm2 to neteq.
- Move the NACK related code in ACM into NetEq.
- NACK related functions in AcmReceiver are changed to simple
forwarding APIs.
- Remove unused members in AcmReceiver.
- Remove unused API functions in NetEq.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1410073006
Cr-Commit-Position: refs/heads/master@{#10448}
This is no longer used. Related code in the iSAC codec itself will be
deleted a follow-up CL.
BUG=4210
Review URL: https://codereview.webrtc.org/1404463003
Cr-Commit-Position: refs/heads/master@{#10272}
This CL restructures the RtcEventLog protobuf format, by removing the DebugEvent message. This is done by moving the LOG_START and LOG_END events to the EventType enum and making a seperate message for audio playout events. In addition to these changes, some fields were added to the AudioReceiveConfig and AudioSendConfig messages, but these are for future use and are not currently logged yet.
This is a follow-up to CL 1340283002 which adds a SSRC to AudioPlayout events in the RtcEventLog.
BUG=webrtc:4741
R=henrik.lundin@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/1348113003 .
Cr-Commit-Position: refs/heads/master@{#10221}
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means,
likely just the linker omitting compilation units with no incoming
references.
(This was previously landed as revisions 10046 and 10060, and got
reverted because it broke several of the Chromium FYI bots.)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1368843003
Cr-Commit-Position: refs/heads/master@{#10127}
This updates the isolate.gypi copies we have to maintain in our
code repo to Chromium's revision 310ea93.
The changes about generating .isolated.gen.json files are needed
to support running with Swarming (https://www.chromium.org/developers/testing/isolated-testing)
Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that's added to our links
script.
In order to use isolate_driver.py, the .isolate files must be in the
same directory as the test_name_run target is defined, which meant
I had to move around some of the isolate files and targets below
webrtc/modules.
BUG=497757
R=maruel@chromium.orgTBR=henrik.lundin@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
TESTED=Clobbered trybots:
git cl try -c --bot=linux_compile_rel --bot=mac_compile_rel --bot=win_compile_rel --bot=android_compile_rel --bot=ios_rel -m tryserver.webrtc
Review URL: https://codereview.webrtc.org/1373513002 .
Cr-Commit-Position: refs/heads/master@{#10081}
Reason for revert:
Breaking Chromium FYI bots.
Original issue's description:
> Don't link with audio codecs that we don't use
>
> We used to link with all audio codecs unconditionally (except Opus);
> this patch makes gyp and gn only link to the ones that are used.
>
> (This unfortunately fails to have a measurable impact on Chromium
> binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
> fix were already being excluded from Chromium by some other means
> (likely just the linker omitting compilation units with no incoming
> references).)
>
> BUG=webrtc:4557
>
> Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809
> Cr-Commit-Position: refs/heads/master@{#10046}
TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1368933002
Cr-Commit-Position: refs/heads/master@{#10069}
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
(This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means
(likely just the linker omitting compilation units with no incoming
references).)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1349393003
Cr-Commit-Position: refs/heads/master@{#10046}
Bug 4985 revealed two flaws
1. Opus duration estimate did not return correct length for DTX packets,
2. NetEq DoCodecInternalCng did not assign enough buffer.
P.S.
Generalizing problem 1, current NetEq decode function checks memory size by calling the duration estimate function. This is not ideal. A better way is to let codec's decode function to receive buffer size and return failure if it is not enough. This can be made in a separate CL.
BUG=webrtc:4985
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1334303005 .
Cr-Commit-Position: refs/heads/master@{#10031}
* Make sure they're all final and don't allow copying or assignment.
* Get rid of the single-channel PCM decoder classes.
* Move some includes from .h to .cc files where possible.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1353803002
Cr-Commit-Position: refs/heads/master@{#10021}
Currently, it's sitting in AudioEncoderIsac*'s files, which is less
than obvious. This CL puts the encoder and decoder in separate files
together with the C implementation; CLs are afoot to make it so for
the other built-in codecs as well.
BUG=webrtc:4557
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1339253003 .
Cr-Commit-Position: refs/heads/master@{#10018}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1348613003
Cr-Commit-Position: refs/heads/master@{#10015}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1346993002
Cr-Commit-Position: refs/heads/master@{#9966}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1348053002
Cr-Commit-Position: refs/heads/master@{#9961}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
This CL adds support for simulating neteq using stored RTP packets as well as calls to GetAudio from an RtcEventLog, using the stored timestamps.
The type of the input file is detected automatically.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1316903002
Cr-Commit-Position: refs/heads/master@{#9886}
In some cases, the number of samples (per channel) in NetEq's sync
buffer could fall below the allowed minimum (5 samples for narrowband,
scaling for other rates). If the number of samples extracted from the
buffer was smaller than the desired number, an error is
returned. However, if the decoder returns fewer samples than expected,
it could happen that the sync buffer level falls under the minimum,
but enough samples are extracted. This triggered an assert. With this
change, the minimum level of the sync buffer is always enforced.
A test is implemented to trigger the problem. It made the assert fire
without this fix, but it now passes.
BUG=webrtc:4840
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1324453002 .
Cr-Commit-Position: refs/heads/master@{#9828}
These asserts cover error cases that are also handled by the code
after the assert. Should not have both assert and error handling.
BUG=webrtc:4840
Review URL: https://codereview.webrtc.org/1321023002
Cr-Commit-Position: refs/heads/master@{#9804}
This is a bug that was introduced in
https://codereview.webrtc.org/1230503003, where the variable "int
temp_bufsize" was changed to a size_t. If the packet buffer was
flushed while inserting a packet, temp_bufsize became negative, which
was tested later in an if-statement. Now, with size_t instead, it
would just become very large, and the if-statement would never see a
negative value. The effect was that the packet size in samples could
be updated with a very large positive number, causing an overflow
which triggered rtc::checked_cast in
StatisticsCalculator::GetNetworkStatistics, line 220.
Also adding a test to reproduce the crash. Without the fix, the test
results in the above mentioned checked_cast to trigger. With the fix,
everything works fine.
BUG=chromium:525260
Review URL: https://codereview.webrtc.org/1307893004
Cr-Commit-Position: refs/heads/master@{#9802}
This was not implemented before. It returns the current total delay
(packet buffer and sync buffer) of NetEq. This is the same information
that was already available in
NetEqNetworkStatistics::current_buffer_size_ms, that can be obtained
through NetEq::NetworkStatistics(). But, since the current delay is a
key metric of NetEq, it is convenient to have it available in a
simpler way.
This is a re-landing of r9359,
https://webrtc-codereview.appspot.com/51149004, which was reverted in
r9360. The refactoring made in r9669 facilitated the relanding.
TBR=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1313873003
Cr-Commit-Position: refs/heads/master@{#9801}
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}
With this change, the aggregates for packet waiting times are
calculated in NetEq's StatisticsCalculator insead of in
AcmReceiver. This simplifies things somewhat, and avoids having to
copy the raw data on polling.
R=ivoc@webrtc.org, minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1296633002 .
Cr-Commit-Position: refs/heads/master@{#9778}
The only shared state is now the bandwidth estimation info.
This reduces the amount and complexity of the locking
substantially.
Review URL: https://codereview.webrtc.org/1208993010
Cr-Commit-Position: refs/heads/master@{#9762}
