This test is to verify that the debug dump can perfectly reproduce APM states if the recording is made from the first input sample.
BUG=
Review URL: https://codereview.webrtc.org/1393353003
Cr-Commit-Position: refs/heads/master@{#10506}
Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.
Size of WebRtcRTPHeader: 4352 -> 1784 bytes.
BUG=webrtc:5144, chromium:500602
Review URL: https://codereview.webrtc.org/1427253002
Cr-Commit-Position: refs/heads/master@{#10504}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
Reading of PCAP (Wireshark) files was not possible due to a bug in the
parsing of files. This change fixes that by adding new validator methods
to RtpFileSource that can be used to determine the input file type.
R=ivoc@webrtc.org
Review URL: https://codereview.webrtc.org/1427923003
Cr-Commit-Position: refs/heads/master@{#10490}
This method is no longer called. With that gone, a number of other
methods and member variables are obsoleted, and removed.
Methods deleted:
AcmReceiver::InsertStreamOfSyncPackets
AcmReceiver::GetNumSyncPacketToInsert()
AcmReceiver::GetSilence, never called
Member variables deleted:
missing_packets_sync_stream_
late_packets_sync_stream_
av_sync_
initial_delay_manager_
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1419573013
Cr-Commit-Position: refs/heads/master@{#10484}
This is a re-land of https://codereview.webrtc.org/1353263005/
which was reverted because of perf-regressions. Changes since that CL:
* Change LayerFilteringTransport to send a padding packet instead of
dropping it for data that should be filtered out. This prevents
confusion due to changed sequence numbers.
* Changed timing of stats poller thread in VideoAnalyzer. Startup was
racy wrt initializion of send_stream_.
* Minor formatting issues.
PERF NOTE: This change will affect some performance numbers slightly.
In particular, {encode_frame_rate, encode_time_ms,
encode_usage_percent, media_bitrate_bps} will change due to timing
of the measurements.
BUG=
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1412233003
Cr-Commit-Position: refs/heads/master@{#10483}
This change avoids calling neteq_->EnableVad() and DisableVad from the
AcmReceiver constructor. Instead, the new member
enable_post_decode_vad is added to NetEq's config struct. It is
disabled by defualt, but ACM sets it to enabled. This preserves the
behavior both of NetEq stand-alone (i.e., in tests) and of ACM.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1425133002
Cr-Commit-Position: refs/heads/master@{#10476}
The end goal is to remove AcmReceiver::SetInitialDelay. This change is
in preparation for that goal. It turns out that
AcmReceiver::SetInitialDelay was only invoked through the following
call chain, where each method in the chain is never referenced from
anywhere else (except from tests in some cases):
ViEChannel::SetReceiverBufferingMode
-> ViESyncModule::SetTargetBufferingDelay
-> VoEVideoSync::SetInitialPlayoutDelay
-> Channel::SetInitialPlayoutDelay
-> AudioCodingModule::SetInitialPlayoutDelay
-> AcmReceiver::SetInitialDelay
The start of the chain, ViEChannel::SetReceiverBufferingMode was never
referenced.
This change deletes all the methods above except
AcmReceiver::SetInitialDelay itself, which will be handled in a
follow-up change.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1421013006
Cr-Commit-Position: refs/heads/master@{#10471}
The former is very similar to the latter, but less general (mostly in
naming).
This CL, which is the first to use Maybe at scale, also removes the implicit conversion from T to Maybe<T>, since it was agreed that the increased verbosity increased legibility.
Review URL: https://codereview.webrtc.org/1430433004
Cr-Commit-Position: refs/heads/master@{#10461}
Add a new interface to abstract away file operations. This CL temporarily
removes support for dumping the output of reverse streams. It will be easy to
restore in the new framework, although we may decide to only allow it with
the aecdump format.
We also now require the user to specify the output format, rather than
defaulting to the input format.
TEST=Bit-exact output to the previous audioproc_f version using an input wav
file, and to the legacy audioproc using an aecdump file.
Review URL: https://codereview.webrtc.org/1409943002
Cr-Commit-Position: refs/heads/master@{#10460}
When estimating that we can send more than the codec max bitrate (under
the assumption that codec max bitrate is good enough for the quality),
we should use additional bitrate so that we can maintain good quality.
Global bitrate caps should still be enforced through bitrate caps (b=AS)
and not codec max bitrates.
BUG=webrtc:5102
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1428473002
Cr-Commit-Position: refs/heads/master@{#10457}
This operation was relatively simple, since no one was doing anything
fishy with this enum. A large number of lines had to be changed
because the enum values now live in their own namespace, but this is
arguably worth it since it is now much clearer what sort of constant
they are.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1424083002
Cr-Commit-Position: refs/heads/master@{#10449}
Negative acknowledgement (NACK) has up to now been implemented in
ACM. But, since NetEq is in charge of the actual packet buffer, it
makes more sense to have the NACK functionlaity in there.
This CL does the following:
- Move nack.{h,cc} and the unit tests from main/acm2 to neteq.
- Move the NACK related code in ACM into NetEq.
- NACK related functions in AcmReceiver are changed to simple
forwarding APIs.
- Remove unused members in AcmReceiver.
- Remove unused API functions in NetEq.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1410073006
Cr-Commit-Position: refs/heads/master@{#10448}
Following CLs will finish the takeover completely. After that,
RentACodec will also start creating and owning codecs, at which point
its name will start making sense.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1412683006
Cr-Commit-Position: refs/heads/master@{#10432}
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419193002
Cr-Commit-Position: refs/heads/master@{#10430}
We have decided not to do a switch from old (AudioCodingModule) to new
(AudioCoding) API. Instead, we will gradually evolve the old API to
meet the new design goals.
As a consequence of this decision, the AudioCoding and AudioCodingImpl
classes are deleted. Also removing associated unit test sources. No
test coverage is lost with this operation, since the tests for the
"old" API are testing more than the deleted tests did.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1415163002
Cr-Commit-Position: refs/heads/master@{#10406}
Matches the include order in webrtc/base/criticalsection.h and makes use
of winsock2.h instead of winsock.h for consistency.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1407053008
Cr-Commit-Position: refs/heads/master@{#10389}
Before this change, UpdateEstimate would repeatedly decrease bitrate
even though there's no fresh corresponding RTCP loss report, triggering
multiple reactions to a single indication of high packet loss.
BUG=webrtc:5101
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1417723005
Cr-Commit-Position: refs/heads/master@{#10374}
Xvfb is needed for the screen capture tests in modules_unittests,
which also brings in xdisplaycheck used by testing/xvfb.py.
libjingle_media_unittest was missing a resource video in the .isolate
file.
BUG=chromium:497757
R=stip@chromium.org
Review URL: https://codereview.webrtc.org/1415603005 .
Cr-Commit-Position: refs/heads/master@{#10365}
We don't allow more than one retransmission within one RTT, but the RTT
estimate might be off. Reasonably, the remote end will not send a NACK
until the packet after has been received - so always resend on first
request.
Review URL: https://codereview.webrtc.org/1414563003
Cr-Commit-Position: refs/heads/master@{#10362}
Reports show that we see full echo from the OnePlus 2 device.
Disabling hardware effects and revert to WebRTC-based
components instead as a test to see if it helps.
R=tommi@webrtc.org
TBR=tommi
BUG=b/25096456
Review URL: https://codereview.webrtc.org/1417093002 .
Cr-Commit-Position: refs/heads/master@{#10357}
This patch also also ensures that audio is restored after an incoming
GSM call.
BUG=webrtc:5058, webrtc:5012
TEST=Manual tests using modified AppRTCDemo and three different BT headsets
Review URL: https://codereview.webrtc.org/1401963002
Cr-Commit-Position: refs/heads/master@{#10354}
