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}
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}
The primary fix in this CL is to remove the dangling |thread_| pointer in AndroidVideoCapturerJni. That thread is not safe to use after Stop() has been called. Even after Stop() has been called, we must still be able to return late frames to Java in order to not leak them, so that path has been made thread safe instead. To make sure that we always return frames, the Java frame should be wrapped in a scoped_refptr as quickly as possible, so this CL moves the wrapping from AndroidVideoCapturer to AndroidVideoCapturerJni. This also removes the need for the interface function AndroidVideoCapturerDelegate::ReturnBuffer().
Some other minor changes are:
* Remove |valid_global_refs_| and all logic related to that. Now that rtc::Bind() captures method objects as scoped_refptr, the destructor of AndroidVideoCapturerJni will not be called before all frames are returned.
* Remove global ref |j_frame_observer_|. No need for this, we don’t call it and it is kept alive with standard Java memory management.
* Add helper function ShallowCenterCrop() for VideoFrameBuffers. This functionality already exists in the constructor of WrappedI420Buffer, but it’s more convenient to have it as a separate function.
BUG=webrtc:4742,webrtc:4909
R=glaznev@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1307973002 .
Cr-Commit-Position: refs/heads/master@{#9784}
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}
By converting three raw pointers to scoped_ptrs, we can eliminate the
need for a manually-defined destructor, and generally sleep better at
night.
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1310213003 .
Cr-Commit-Position: refs/heads/master@{#9776}
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}
AudioDeviceTemplate doesn't initialize `output_` and `input_` if the
initialization of `audio_manager_` succeeds. Similarly, it doesn't
terminate `input_` and `audio_manager_` if the termination of `output_`
succeeds. This CL fixes this.
BUG=
Review URL: https://codereview.webrtc.org/1296693003
Cr-Commit-Position: refs/heads/master@{#9760}
This makes us use the clever code that adapts the path depending on where we build, so it works in google3 too.
BUG=None
Review URL: https://codereview.webrtc.org/1291283003
Cr-Commit-Position: refs/heads/master@{#9752}
Negative values from TimeUntilNextProcess indicate that the module
wanted to run sooner than possible, not that an invalid error code was
returned. As such it's not a contract error.
BUG=webrtc:4879
NOTRY=true
Review URL: https://codereview.webrtc.org/1257833004
Cr-Commit-Position: refs/heads/master@{#9740}
This CL adds calculation and logging of average excess buffer delay
and number of delayed packet outage events per minute.
The first is the average of time spent in the packet buffer for all
packets. The average is calculated for intervals of one minute, and
the result is logged to the UMA stat
WebRTC.Audio.AverageExcessBufferDelayMs.
The second is a counter of delayed packet outage events that is
restarted every minute, and the result is logged to the UMA stat
WebRTC.Audio.DelayedPacketOutageEventsPerMinute. For a description of
delayed packet outages, see previous CL implementing a duration log
for these events.
BUG=webrtc:4915, chromium:488124
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1287333005 .
Cr-Commit-Position: refs/heads/master@{#9731}
Measures the duration of each packet loss concealment (a.k.a. expand)
event that is not followed by a merge operation.
Having decoded and played packet m−1, the next expected packet is
m. If packet m arrives after some time of packet loss concealment, we
have a delayed packet outage event. However, if instead packet n>m
arrives, we have a lost packet outage event. In NetEq, the two outage
types results in different operations. Both types start with expand
operations to generate audio to play while the buffer is empty. When a
lost packet outage happens, the expand operation(s) are followed by
one merge operation. For delayed packet outages, merge is not done,
and the expand operations are immediately followed by normal
operations.
This change also includes unit tests for the new statistics.
BUG=webrtc:4915, chromium:488124
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1290113002 .
Cr-Commit-Position: refs/heads/master@{#9725}
- Integrates intelligibility into audio_processing.
- Allows modification of reverse stream if intelligibility enabled.
- Makes intelligibility available in audioproc_float test.
- Adds reverse stream processing to audioproc_float.
- (removed) Makes intelligibility toggleable in real time in voe_cmd_test.
- Cleans up intelligibility construction, parameters, constants and dead code.
TBR=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1234463003
Cr-Commit-Position: refs/heads/master@{#9713}
Especially the VP9 codec currently may overshoot bitrate target at sudden picture changes, resulting in frames over 800 packets.
This limit should be reduced again once the codec behaves.
BUG=webrtc:4889
Review URL: https://codereview.webrtc.org/1266353003
Cr-Commit-Position: refs/heads/master@{#9675}
