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}
This CL makes the Java render interface asynchronous by requiring every call to renderFrame() to be followed by an explicit renderFrameDone() call. In JNI, this is implemented with cricket::VideoFrame::Copy() before calling renderFrame(), and a corresponding call to delete in renderFrameDone(). This CL is primarily done to prepare for a new renderer implementation.
BUG=webrtc:4742, webrtc:4909
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1313563002 .
Cr-Commit-Position: refs/heads/master@{#9814}
Replaced the wav file dumping functionality in aec_core.c with the newly added corresponding macros
Added macros for logging of AEC internal data
BUG=
Review URL: https://codereview.webrtc.org/1272403003
Cr-Commit-Position: refs/heads/master@{#9808}
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}
Why the replacements? Mainly two reasons:
1) RTCCertificate owns the identity and as long as things are referencing the identity there should be a scoped_refptr reference to the RTCCertificate. Handing out raw pointers is less memory safe.
2) With the latest RFC, an RTCCertificate should be sufficient for specifying a crypto cert and the code should be updated to use RTCCertificate instead of SSLIdentity directly.
This replace work is split up into multiple CLs. In this CL...
- WebRtcSessionDescriptionFactory is updated to use RTCCertificate over SSLIdentity.
- WebRtcSessionDescriptionFactory::SignalCertificateReady is connected to WebRtcSession::OnCertificateReady and WebRtcSession is updated to use RTCCertificate.
- The cricket::Transport and related classes are updated to use RTCCertificate. These are called from WebRtcSession::OnCertificateReady.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1312643004 .
Cr-Commit-Position: refs/heads/master@{#9794}
There are 2 ways to design this.
1. TCP Only mode: this means that we disable all UDP protocols across board.
2. disallow TURN over UDP. Along with DISABLE_UDP, DISABLE_STUN, we should achieve the same result.
I'm going with #2.
BUG=webrtc:4784
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1311153003 .
Cr-Commit-Position: refs/heads/master@{#9791}
The Buffer is saved between calls, so after the initial allocation
it'll already be allocated and of the right size. The stack-allocated
array had the advantage of requiring no heap allocation at all, but
for most popular encoders it ended up allocating about 15 kB too much,
and now that we allow user-defined encoders there was also the
(remote) possibility that the buffer would actually be too small.
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1303413003 .
Cr-Commit-Position: refs/heads/master@{#9789}
VideoFrameBuffer currently has two overloaded data() functions for pixel access, one for const and one for non-const. Unfortunately, it will default to the non-const version, even when 'const scoped_refptr<VideoFrameBuffer>&' is used. This is a problem, because many subclasses use RTC_NOTREACHED() in the non-const version.
This CL makes the non-const version of data() explicit with a different, longer function name MutableData().
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1304143003 .
Cr-Commit-Position: refs/heads/master@{#9787}
