Unwrap uint16_t to int64_t, based on delta and last sequence number.
This can make application logic, putting packets in maps etc, much
simpler.
BUG=
Review URL: https://codereview.webrtc.org/1209623002
Cr-Commit-Position: refs/heads/master@{#9887}
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}
Handling the case when encoder drops only the higher layer.
Added options to screenshare loopback test to discard high temporal or spatial layers (to view the lower layers).
Review URL: https://codereview.webrtc.org/1287643002
Cr-Commit-Position: refs/heads/master@{#9883}
Part of work removing dependency on Chromium's base.
BUG=468375 (in particular comment #37)
Review URL: https://codereview.webrtc.org/1327023002
Cr-Commit-Position: refs/heads/master@{#9880}
The easy way also happens to be more efficient if we have to
reallocate, but that's a minor concern here.
Review URL: https://codereview.webrtc.org/1327053002
Cr-Commit-Position: refs/heads/master@{#9876}
This CL contains major modifications of the audio output parts for WebRTC on iOS:
- general code cleanup
- improves thread handling (added thread checks, remove critical section, atomic ops etc.)
- reduces loopback latency of iPhone 6 from ~90ms to ~60ms ;-)
- improves selection of audio parameters on iOS
- reduces complexity by removing complex and redundant delay estimates
- now instead uses fixed delay estimates if for some reason the SW EAC must be used
- adds AudioFineBuffer to compensate for differences in native output buffer size and
the 10ms size used by WebRTC. Same class as is used today on Android and we have unit tests for
this class (the old code was buggy and we have several issue reports of crashes related to it)
Similar improvements will be done for the recording sid as well in a separate CL.
I will also add support for 48kHz in an upcoming CL since that will improve Opus performance.
BUG=webrtc:4796,webrtc:4817,webrtc:4954, webrtc:4212
TEST=AppRTC demo and iOS modules_unittests using --gtest_filter=AudioDevice*
R=pbos@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1254883002 .
Cr-Commit-Position: refs/heads/master@{#9875}
Updating full stack test to optionally save metadata for each frame and save it
to a file with given filename (controlled from the new full_stack_samples
executable).
Adding a Python script that reads the output generated by full stack test
and plots the graph(s).
Review URL: https://codereview.webrtc.org/1289933003
Cr-Commit-Position: refs/heads/master@{#9874}
The disabling of the sin,cos,sinf,cosf functions had the wrong
condition for GN. This fixes that and also makes the condition
in common.gypi a bit more readable.
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1307633008 .
Cr-Commit-Position: refs/heads/master@{#9871}
base/logging.h dumped to stderr by default in debug mode, but webrtc
"trace" (via system_wrappers/../logging.h) has that feature disabled by
default. This makes the two consistent.
Bonus: log the filename:line in base/logging.h, which exists in the
system_wrappers variant.
TEST=neteq_impl.cc logs (which use base/logging.h) no longer appear in
debug mode, unless --logs=true is passed. Filenames appear correctly.
Review URL: https://codereview.webrtc.org/1331503002
Cr-Commit-Position: refs/heads/master@{#9868}
DesktopFrameWin::Create() may return nullptr when it fails to allocate
windows bitmap. ScreenCapturerWinGdi wasn't handling that case properly.
BUG=527660
Review URL: https://codereview.webrtc.org/1309143007
Cr-Commit-Position: refs/heads/master@{#9865}
Telling the encoder to adjust the parameters for the screen content.
Also, telling the encoder to skip the encoding of very flat/low content blocks. For now only for screensharing. (number 8 in VP8E_SET_STATIC_THRESHOLD is correct)
Review URL: https://codereview.webrtc.org/1308753006
Cr-Commit-Position: refs/heads/master@{#9860}
This will be used for the send side bitrate estimation. Storing various
meta-data about packets that can be retreived when arrival time feeback
arrives.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1288033008
Cr-Commit-Position: refs/heads/master@{#9859}
Value was incorrectly truncated to 16 bits when serializing the message.
Fixed, with added regression tests.
BUG=
Review URL: https://codereview.webrtc.org/1294393002
Cr-Commit-Position: refs/heads/master@{#9858}
Pitfalls:
* Left shift of signed integer has undefined behavior
* Right-shift of signed integer has platform-specific behavior is value is negative
* Cast from unsigned to signed has undefined behavior if value is negative
BUG=webrtc:4824
Review URL: https://codereview.webrtc.org/1226993003
Cr-Commit-Position: refs/heads/master@{#9854}
Hopefully, this will make it easier to figure out what's wrong the
next time this happens.
BUG=526478
Review URL: https://codereview.webrtc.org/1313073008
Cr-Commit-Position: refs/heads/master@{#9844}
This test only currently works because stun.l.google.com has an IPv4
address and the TURN port is created with an IPv6 address. But the test
would start failing if/when it starts providing an IPv6 address. Which
may already be happening, as indicated by a recent test failure.
Review URL: https://codereview.webrtc.org/1290233003
Cr-Commit-Position: refs/heads/master@{#9841}
of decoder factory class.
- Add new Peer connection factory method to initialize shared
EGL context.
This provides an option to use single peer connection factory
in the application and create peer connections from the same
factory and reinitialize shared EGL context for video
decoding HW acceleration.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1304063011 .
Cr-Commit-Position: refs/heads/master@{#9838}
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}
At least Android try bots seem to have a timeout that will forcibly shut
down the executable if no output has been observed for 60s. Since full
stack test typically run for 60s we need to output give some to avoid
racy shutdown.
BUG=chromium:513170
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1288453003 .
Cr-Commit-Position: refs/heads/master@{#9822}
Cleaning AudioConferenceMixer APIs to match Chromium style guide.
Main changes:
1. change all mutable references to pointers
2. add const to all non-mutable references
3. add const to as many methods as possible
BUG=
R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1311733003 .
Cr-Commit-Position: refs/heads/master@{#9821}
