AcmReceiver::decoders_ is now one step closer to being unused.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2339953002
Cr-Commit-Position: refs/heads/master@{#14274}
Remove a large number of targets that are no longer built, to reduce maintenance.
Only targets that have a GN version were removed.
BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2340773003
Cr-Commit-Position: refs/heads/master@{#14231}
Declare resources for GN targets so that they can be isolated
NOTRY=True
BUG=chromium:497757
Review-Url: https://codereview.webrtc.org/2340753002
Cr-Commit-Position: refs/heads/master@{#14210}
The basic implementation of AudioNetworkAdaptor include the introduction of
1. Controller
2. ControllerManager
ControllerManager is to hold all needed controllers. It also orders them according to their significance in dealing with current network condition.
Controller provides an interface MakeDecision, which has to be implemented by specific controllers. AudioNetworkAdaptorImpl calls MakeDecision of the controllers in the order decided by ControllerManager to collect EncoderRuntimeConfig.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2306083002
Cr-Commit-Position: refs/heads/master@{#14201}
A left shift by 10 was assumed to never overflow, since "[s]imulation
of the 25 files shows that maximum value in the vector gain_lo_hiQ17[]
is 441344, which means that it is log2((2^31)/441344) = 12.2 shifting
bits from saturation." However, a fuzzer test succeeded in provoking
an overflow, which we ignore in this CL on the theory that only
"abnormal" inputs cause overflow.
Also had to replace a "foo << 1" with "foo * (1 << 1)" in
WEBRTC_SPL_MUL_16_32_RSFT15 because foo could be negative; this
problem showed up as soon as I'd asked UBSan to ignore the overflow
discussed above.
BUG=chromium:615819
Review-Url: https://codereview.webrtc.org/2314413002
Cr-Commit-Position: refs/heads/master@{#14162}
Reason for revert:
Broke all Chromium libFuzzer builds
https://bugs.chromium.org/p/chromium/issues/detail?id=645069
Original issue's description:
> Setting up an RTP input fuzzer for NetEq
>
> This CL introduces a new fuzzer target neteq_rtp_fuzzer that
> manipulates the RTP header fields before inserting the packets into
> NetEq. A few helper classes are also introduced.
>
> BUG=webrtc:5447
> NOTRY=True
>
> Committed: https://crrev.com/2d273f1e97cd5030ed1686f27ce1118291b66395
> Cr-Commit-Position: refs/heads/master@{#14103}
TBR=ivoc@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5447
Review-Url: https://codereview.webrtc.org/2328483002
Cr-Commit-Position: refs/heads/master@{#14131}
I could not find a single place it was used, outside of the unittests
for the sync packet support itself.
Review-Url: https://codereview.webrtc.org/2309303002
Cr-Commit-Position: refs/heads/master@{#14130}
With this CL, the NetEqReplacementInput class handles reordered and
missing packets in a better way than before, by storing the last
confirmed packet size and using that when the next packet size cannot
be calculated.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2319553003
Cr-Commit-Position: refs/heads/master@{#14122}
AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions.
This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor.
This CL illustrates the interfaces of the AudioNetworkAdaptor.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2308573002
Cr-Commit-Position: refs/heads/master@{#14115}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
This CL introduces a new fuzzer target neteq_rtp_fuzzer that
manipulates the RTP header fields before inserting the packets into
NetEq. A few helper classes are also introduced.
BUG=webrtc:5447
NOTRY=True
Review-Url: https://codereview.webrtc.org/2315633002
Cr-Commit-Position: refs/heads/master@{#14103}
We hit a fuzzer bug that caused numDecodedBytesLB + numDecodedBytesUB
> lenEncodedBytes, which is obviously bogus. Check for that, and for
the case whhere the UB decoder itself realized that something was
wrong. (The code already makes the corresponding check for the LB
decoder.)
BUG=chromium:637899
Review-Url: https://codereview.webrtc.org/2315693002
Cr-Commit-Position: refs/heads/master@{#14091}
If neteq_rtpplay is invoked with the --ssrc option to select packets
matching a specific SSRC, but no matching packets are found, this CL
provides a meaningful error message.
BUG=webrtc:2692
NOTRY=True
TBR=ivoc@webrtc.org
Review-Url: https://codereview.webrtc.org/2318503002
Cr-Commit-Position: refs/heads/master@{#14083}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Add "//build/config/compiler:optimize_max" to rtc_add_configs and
"//build/config/compiler:default_optimization" to rtc_remove_configs.
This is the default optimization in GYP, and might help explain a 82.5%
regression in webrtc_perf_tests at 13946:13946
BUG=chromium:641966
NOTRY=True
Review-Url: https://codereview.webrtc.org/2307283002
Cr-Commit-Position: refs/heads/master@{#14067}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Remove //build/config/sanitizers:deps as a dependency for
all rtc_executable targets and add it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2308553002
Cr-Commit-Position: refs/heads/master@{#14048}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
The sample uses are from when I debugged bug 617124. The change in neteq_network_stats_unittest.cc is a fix for a minor unrelated bug found by the try bots when I tried to land this CL (a test was passing uninitialized packet data to NetEq).
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2293893002
Cr-Commit-Position: refs/heads/master@{#14034}
If a CNG packet is received first, followed by a speech packet with
another sample rate, NetEq should treat this as a change of codec, flush
out the CNG packet and reset the sample rate to that of the speech
packet.
BUG=webrtc:5447
NOTRY=True
Review-Url: https://codereview.webrtc.org/2307493002
Cr-Commit-Position: refs/heads/master@{#14032}
That is, rather than keeping a separate pointer and size.
This helps automate memory management in NetEq and will be useful in the
work to minimize the AudioDecoder interface as part of the injectable
audio codec work.
I'm planning a follow-up that will change the current management of Packet* to wrapping them in unique_ptr instead.
Review-Url: https://codereview.webrtc.org/2289093003
Cr-Commit-Position: refs/heads/master@{#14002}
The compiler optimization for Windows is O1 by default in GN, but O2 in GYP.
This might help explain the regression observed on neteq_performance_unittest.
NOTRY=True
BUG=641966
Review-Url: https://codereview.webrtc.org/2291253003
Cr-Commit-Position: refs/heads/master@{#13999}
With this change, the value 0xFF is no longer used to flag that the RTP
type is unknown. Instead, an empty value for the rtc::Optional is used.
Review-Url: https://codereview.webrtc.org/2290153002
Cr-Commit-Position: refs/heads/master@{#13989}
In order to get resource files to be properly packaged into
the .app for a unit test on iOS, the resource files needs
to be listed as sources in a bundle_data target.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2292853002
Cr-Commit-Position: refs/heads/master@{#13968}
This adds a new file, webrtc/modules/audio_coding/neteq/tools/packet_source.cc, so that I'll have somewhere to put the new non-inlined methods.
NOTRY=true
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2290593002
Cr-Commit-Position: refs/heads/master@{#13956}
This is still a tiny lie, since it checks for kCodecArbitrary to avoid
scaling a codec with an external decoder, because of missing information
in that case. The main point is still true, though. Once the next CL is
in, removing NetEqDecoder usage completely in DecoderDatabase, another
workaround will be in place for external decoders until we can fully
deprecate them.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2270063006
Cr-Commit-Position: refs/heads/master@{#13952}
the number of points that need to be mocked for testing.
For the now non-virtual methods, DecoderDatabase now does a lookup
through GetDecoderInfo and then delegates to the appropriate method in
the DecoderInfo object, if one is found.
A few other methods were also changed to look up through GetDecoderInfo.
Also moved the audio decoder factory into DecoderInfo, so that
DecoderInfo::GetDecoder can be used directly.
Review-Url: https://codereview.webrtc.org/2276913002
Cr-Commit-Position: refs/heads/master@{#13933}
We have RTC_CHECK and RTC_DCHECK for C now, so we should use it. It's
one fewer difference between our C and C++ code.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2274083002
Cr-Commit-Position: refs/heads/master@{#13930}
Checksums were updated for NetEq and ACM bitexactness tests (after
verifying the audio quality).
BUG=webrtc:5447
Review-Url: https://codereview.webrtc.org/2266293005
Cr-Commit-Position: refs/heads/master@{#13928}
This removes the warning printouts about unknown header extensions.
BUG=webrtc:2692
Review-Url: https://codereview.webrtc.org/2266403005
Cr-Commit-Position: refs/heads/master@{#13912}
The current_rtp_payload_type_ should only be updated when the packet is
actually inserted into the packet buffer, since then the payload type
has been validated. This CL removes an unvalidated setting of this value
that happened after SSRC change or upon first packet.
BUG=webrtc:5447
Review-Url: https://codereview.webrtc.org/2270793003
Cr-Commit-Position: refs/heads/master@{#13910}
If an error happens in the GetAudio call, for instance when corrupt
payloads are inserted, GetAudio wil return an error. In this case, the
audio frame is not always correctly populated, which is to be expected.
BUG=webrtc:5447
Review-Url: https://codereview.webrtc.org/2272963002
Cr-Commit-Position: refs/heads/master@{#13902}
This implementation interprets payloads of size 1 as codec-internal SID
frames, marking the start of a CNG period. Changes were made to other
parts of the test payload chain, since it had to make use of the virtual
payload size in the case of header-only RTP files.
BUG=webrtc:2692
Review-Url: https://codereview.webrtc.org/2275903002
Cr-Commit-Position: refs/heads/master@{#13901}
