The playout modes other than Normal have not been reachable for a long
time, other than through tests. It is time to deprecate them.
The only meaningful use was that Fax mode was sometimes set from
tests, in order to avoid time-stretching operations (accelerate and
pre-emptive expand) from messing with the test results. With this CL,
a new config is added instead, which lets the user specify exactly
this: don't do time-stretching.
As a result of Fax and Off modes being removed, the following code
clean-up was done:
- Fold DecisionLogicNormal into DecisionLogic.
- Remove AudioRepetition and AlternativePlc operations, since they can
no longer be reached.
Bug: webrtc:9421
Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
Reviewed-on: https://webrtc-review.googlesource.com/84123
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23704}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
NetEq tapers down the audio produced through loss concealment when the
expansion has been going on for some time. When the audio packets starts
coming in again, there is a ramp-up that happens. This ramp-up could
before this change extend over more than one 10 ms block, which made
keeping track of the scaling factor necessary. With this change, we make
this ramp-up quicker in the rare cases when it lasted more than 10 ms,
so that it always ramps up to 100% within one block. This way, we can
remove the mute_factor_array.
This change breaks bit-exactness, but careful listening could not reveal
an audible difference.
This change is a part of a larger refactoring of NetEq's PLC code.
Bug: webrtc:9180
Change-Id: I4c513ce3ed8d66f9beec2abfb1f0c7ffaac7a21e
Reviewed-on: https://webrtc-review.googlesource.com/77180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23342}
The overflow currently does not cause any problems, but it has been
found that it can cause crashes after a refactoring that is coming in
the near future.
Bug: webrtc:9180
Change-Id: Ia2c4e545c062c4f8ad13cbc47b8796c6e8a4e906
Reviewed-on: https://webrtc-review.googlesource.com/77667
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23327}
NetEq was (up until this CL) capable of fading over to generating a
constant background noise when voice expansion had lasted too long.
However, the code has for a really long time only ever used the "off"
mode, which meant that long expansions are faded down to complete
silence (only zeros), i.e., background noise fill was not used.
Removing the other two modes ("on" and "fade") simplifies the code.
Bug: webrtc:9180
Change-Id: Ia2d46960208f3d75c9659ad3f027c52e5ecfb6b0
Reviewed-on: https://webrtc-review.googlesource.com/71485
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22969}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
This change makes the fix for too long delays during Opus DTX periods
permanent. The fix has up until now been under an experiment, named
WebRTC-NetEqOpusDtxDelayFix.
Bug: webrtc:8488,chromium:780849
Change-Id: I006abb67f96d9d7880bf2215d7d6b52db6cbbfbc
Reviewed-on: https://webrtc-review.googlesource.com/44420
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21786}
This allows removing the specific classes in a later CL.
Bug: webrtc:8677
Change-Id: I3b9c1f3191c38e6d31a3de990e2d882505e79adc
Reviewed-on: https://webrtc-review.googlesource.com/35040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21412}
This tests NetEq with a stream encoded with Opus using it's internal
DTX/CNG.
Also adding a new resource file which is encoded using Opus with DTX.
Bug: webrtc:8488
Change-Id: Icfba5bc5dc7f9c9d0e637a90f4df674e8ba40358
Reviewed-on: https://webrtc-review.googlesource.com/26028
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20905}
Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
Internally in NetEq, an FEC packet looks very similar to a split packet, which caused NetEq to miscalculate the frame length of FEC packets. This incorrect framelength calculation was incorrectly handled as a framelength change by NetEq.
Bug: webrtc:8410
Change-Id: Icaea961d055e49d7726b87811881db0b9149805b
Reviewed-on: https://webrtc-review.googlesource.com/12420
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20373}
We added a new bot to client.webrtc.fyi (https://build.chromium.org/p/client.webrtc.fyi/builders/Win%20%28more%20configs%29).
It seems it is spotting some unsafe conversions and this CL is a test to see if we can use rtc::dchecked_cast to fix them:
../../modules/audio_coding/neteq/neteq_unittest.cc(547): error C2220: warning treated as error - no 'object' file generated
../../modules/audio_coding/neteq/neteq_unittest.cc(547): warning C4267: '=': conversion from 'size_t' to 'uint16_t', possible loss of data
../../modules/audio_coding/neteq/neteq_unittest.cc(548): warning C4267: '=': conversion from 'size_t' to 'uint32_t', possible loss of data
../../modules/audio_coding/neteq/neteq_unittest.cc(977): warning C4267: '+=': conversion from 'size_t' to 'uint32_t', possible loss of data
../../modules/audio_coding/neteq/neteq_unittest.cc(979): warning C4267: '+=': conversion from 'size_t' to 'uint32_t', possible loss
Bug: chromium:759980
Change-Id: Icd0f32ccf620c7c6642fadff797dc2482918648d
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/12921
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20335}
When the test was created, it was disabled for mobile platforms from
the beginning. This is likely a copy-paste from the related
NetEqDecodingTest.TestBitExactness which includes testing codecs not
supported on mobile platforms (e.g., iLBC). This restriction is not
needed for the Opus-only test.
The test remains disabled for iOS, since none of the bots actually run
the relevant test binary on actual iOS devices.
Bug: none
Change-Id: I9071e0e32c83b62c8c7af59ac1cb3e46227f8e8e
Reviewed-on: https://webrtc-review.googlesource.com/8561
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20264}
Description of this stat can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay
Bug: webrtc:8281
Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023
Reviewed-on: https://webrtc-review.googlesource.com/3381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20069}
When NetEq performs a merge operation, it will usually have to correct
the stats for number of concealment samples produced, sometimes with
decreasing it.
This does not make sense in the context of the stats spec, and
stats-consuming applications may not be prepared for it. With this
change, only positive corrections are allowed for the
concealed_samples value. This will sometimes lead to a small positive
bias, but it will be negligible over time.
Bug: webrtc:8253
Change-Id: Ie9de311ab16401f1a4b435f6269725901b8cf561
Reviewed-on: https://webrtc-review.googlesource.com/1583
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19941}
The number of concealment events. This counter increases every time a concealed sample is
synthesized after a non-concealed sample. That is, multiple consecutive concealed samples
will increase the concealedSamples count multiple times but is a single concealment event.
Bug: webrtc:8246
Change-Id: I7ef404edab765218b1f11e3128072c5391e83049
Reviewed-on: https://webrtc-review.googlesource.com/1221
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19881}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
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}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/audio_coding/neteq/neteq_unittest.cc (Browse further)
