Commit Graph

11283 Commits

Author SHA1 Message Date
81d93f37a5 Remove the unused and untested functions from VoERTP_RTCP.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2667423004
Cr-Commit-Position: refs/heads/master@{#16606}
2017-02-14 11:44:57 +00:00
0706813280 Android AppRTCMobile: Add setting for selecting H264 Baseline or High profile
BUG=webrtc:6337
R=glaznev@webrtc.org

Review-Url: https://codereview.webrtc.org/2658243002 .
Cr-Commit-Position: refs/heads/master@{#16605}
2017-02-14 11:41:35 +00:00
1f16ee38c1 Compile ios helpers on mac as well.
BUG=webrtc:5582

Review-Url: https://codereview.webrtc.org/2586433002
Cr-Commit-Position: refs/heads/master@{#16604}
2017-02-14 11:07:57 +00:00
a3b2add27d Added handling of 'agc_compression_gain' flag in audioproc_f.
The test program modules/audio_processing/test/audioproc_float.cc
defined the flag 'agc_compression_gain' and had checks if the
parameter was valid (audioproc_float). The flag was also copied to
webrtc::test::SimulationSettings of audio_processing_simulator.h. The
setting was however never applied to APM.

This change applies the setting on the GainControl submodule in the
same way as the agc_target_level is applied.

This is needed for e.g. testing the AGC fixed digital limiter with the
same configuration as it is (currently) used with in AudioMixerImpl.

Also added new flag '-experimental_agc'. This flag allows disabling the
experimental AGC, which is how the AGC is used in AudioMixerImpl.
ExperimentalAgc is enabled by default, exactly as it was prior to this change.

The change has been tested locally by listening tests and diff comparisons.

BUG=None
NOTRY=True # win_dbg bot not cooperating

Review-Url: https://codereview.webrtc.org/2684983004
Cr-Commit-Position: refs/heads/master@{#16603}
2017-02-14 10:07:49 +00:00
ad9010c983 Make sure initial framedrop is off where quality scaling is off.
BUG=chromium:689972,chromium:689915

Review-Url: https://codereview.webrtc.org/2684683004
Cr-Commit-Position: refs/heads/master@{#16602}
2017-02-14 08:46:51 +00:00
1a95e61e37 Delete httpclient.c and related files.
The files socketpool.h and diskcache.h also become unused, and are
deleted together with their sources.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2694753002
Cr-Commit-Position: refs/heads/master@{#16601}
2017-02-14 08:23:10 +00:00
77282a87a1 Delete fileutils_mock.h.
It became unused in cl https://codereview.webrtc.org/2541453002/.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2690093002
Cr-Commit-Position: refs/heads/master@{#16599}
2017-02-14 07:21:52 +00:00
2758c664b4 Fix the build break by keeping the old Port::AddAddress method since the
downstream application depends on it.
Mark the old Port::AddAddress deprecated and will be removed after the
applications stop replying on it.

BUG=None.
R=deadbeef@webrtc.org

Review-Url: https://codereview.webrtc.org/2694103003 .
Cr-Commit-Position: refs/heads/master@{#16598}
2017-02-14 01:33:27 +00:00
c874d1296a Fixing logic for using android_setsocknetwork() with bind().
If android_setsocknetwork() is available, and it fails, then bind()
should *not* be called, and an error should be returned.

If it succeeds, then bind should be called, but with an "any" address.

This is to prevent cases where sockets are sent with a source address
that doesn't match the network interface they're sent on. See bug below.

This CL also changes "NetworkBinderResults" to an enum class, and
renames it to "NetworkBinderResult".

BUG=webrtc:7026

Review-Url: https://codereview.webrtc.org/2646863005
Cr-Commit-Position: refs/heads/master@{#16597}
2017-02-13 23:41:59 +00:00
06b7e5ce1f Add H.264 high profile to the list of supported codecs before baseline profile.
To ensure compliance with older version high profile should appear in local SDP
before baseline profile.

BUG=b/34816463

Review-Url: https://codereview.webrtc.org/2696733002
Cr-Commit-Position: refs/heads/master@{#16596}
2017-02-13 23:13:24 +00:00
b856794be7 Revert of Add the url attribute to the IceCandidate (Java Wrapper) (patchset #3 id:60001 of https://codereview.webrtc.org/2690593002/ )
Reason for revert:
Breaks downstream application's build

Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> The url of the ICE server is added to the IceCandiate class.
> This can be used to tell which server this candidate was gathered from.
>
> BUG=webrtc:7128
>
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88

TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128

Review-Url: https://codereview.webrtc.org/2692993002
Cr-Commit-Position: refs/heads/master@{#16595}
2017-02-13 22:31:38 +00:00
f812e45d8f Handle InitDecode and reset in fallback decoder.
Makes sure video decoder software fallback handles InitDecode()
failures, and properly releases the pointer after ::Release() so that
another decode failure will properly reinitialize the decoder.

Also makes sure to not call Decode() without a previous InitDecode()
succeeding.

BUG=webrtc:7154
R=noahric@chromium.org, sophiechang@chromium.org

Review-Url: https://codereview.webrtc.org/2690183004 .
Cr-Commit-Position: refs/heads/master@{#16594}
2017-02-13 22:11:08 +00:00
8586c8ee88 Add the url attribute to the IceCandidate (Java Wrapper)
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.

BUG=webrtc:7128

Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16593}
2017-02-13 22:04:50 +00:00
26d99c2e28 Add the URL attribute to cricket::Candiate.
The URL of the ICE server will be reconstructed by the Port and the URL
attribute is added to the cricket::Candidate struct so that we can tell
which ICE server the candidate was gathered from.

This CL only changes the native C++ code. The Java and Objc wrapper will
be created in separate CLs.

BUG=webrtc::7128

Review-Url: https://codereview.webrtc.org/2685053004
Cr-Commit-Position: refs/heads/master@{#16591}
2017-02-13 20:47:27 +00:00
39e14da919 Changing some PeerConnection-related comments.
As recommended by nisse@ in comments on this CL:
https://codereview.webrtc.org/2685093002/

BUG=None
NOTRY=True
TBR=nisse@webrtc.org

Review-Url: https://codereview.webrtc.org/2692923002
Cr-Commit-Position: refs/heads/master@{#16589}
2017-02-13 17:49:58 +00:00
e3a5567230 Reduce the BWE with 50% when feedback is received too late.
Lateness is determined by the length of the send-side history, currently
set to 60 seconds.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2684353004
Cr-Commit-Position: refs/heads/master@{#16588}
2017-02-13 17:08:22 +00:00
bcd88dbc01 WebRtcVoiceEngineTest: Changed a static_cast to a checked_cast.
Also two spelling fixes.
This is a follow-up to https://codereview.webrtc.org/2669583002/

TBR=kwiberg@webrtc.org
BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2697453004
Cr-Commit-Position: refs/heads/master@{#16586}
2017-02-13 15:04:05 +00:00
9de49e317e Add clearImage method to SurfaceViewRenderer.
BUG=None

Review-Url: https://codereview.webrtc.org/2691533002
Cr-Commit-Position: refs/heads/master@{#16584}
2017-02-13 14:15:02 +00:00
38cc1d6b31 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.
This avoids redoing RTP header parsing already done in Call, for video.

The next step is to convert other types of receive streams, i.e.,
audio and flexfec, to use a compatible OnRtpPacket method. We can then
introduce a shared base interface, and simplify media-independent
receive processing in Call.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2681673004
Cr-Commit-Position: refs/heads/master@{#16583}
2017-02-13 13:59:46 +00:00
07a050f995 Add support for swapping feeds in Android AppRTCMobile.
BUG=webrtc:6937

Review-Url: https://codereview.webrtc.org/2682943006
Cr-Commit-Position: refs/heads/master@{#16582}
2017-02-13 13:58:27 +00:00
806a1a0c28 Add ifdef protection for iOS-only headers
BUG=webrtc:6841

Review-Url: https://codereview.webrtc.org/2553683008
Cr-Commit-Position: refs/heads/master@{#16580}
2017-02-13 13:09:01 +00:00
06f240bc4f Clean out platform specific things from voice_engine_defines.h.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2689183002
Cr-Commit-Position: refs/heads/master@{#16578}
2017-02-13 12:42:52 +00:00
552c7c70b0 Make SendStatisticsProxy paramter mandatory in ViEEncoder ctor.
The only implementation which used a nullptr was a mock used in tests,
so add a dummy instance there instead.
Remove tests for stats_proxy_ in vie_encoder and just dcheck in the
constructor instead.

BUG=None

Review-Url: https://codereview.webrtc.org/2695643002
Cr-Commit-Position: refs/heads/master@{#16577}
2017-02-13 12:41:45 +00:00
1458462303 Delete unused classes AutoDetectProxy and SslSocketFactory.
SslSocketFactory is unused since https://codereview.webrtc.org/2506983002, and it's the last
user of AutoDetectProxy.

Also move HttpListenServer and SocksProxyServer to the rtc_base_tests_utils gn target, since they're used by tests only.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2541453002
Cr-Commit-Position: refs/heads/master@{#16576}
2017-02-13 12:33:28 +00:00
be03724ae1 Fix nits in vie_encoder
Updated comment.
Don't call AdaptUp/AdaptDown in tests without first emitting a frame.
Handle frame received precondition in AdaptUp/AdaptDown with DCHECK
instead of return.

BUG=webrtc:4172, webrtc:6850

Review-Url: https://codereview.webrtc.org/2690023002
Cr-Commit-Position: refs/heads/master@{#16572}
2017-02-13 10:38:17 +00:00
7041eed59f Add possibility to plot statistics from integration tests per codec type/implementation.
BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2688863002
Cr-Commit-Position: refs/heads/master@{#16571}
2017-02-13 09:37:57 +00:00
804c1af48b Move trackmediainfomap files from api/ to pc/.
It looks like this was left out of the original api/pc move CL since it
had been added recently.

BUG=webrtc:5883
TBR=ossu@webrtc.org

Review-Url: https://codereview.webrtc.org/2690793003
Cr-Commit-Position: refs/heads/master@{#16560}
2017-02-12 03:07:31 +00:00
49ce67c992 Do not encode frames in MultithreadedFakeH264Encoder after Release().
Other minor changes:
- Define locks after stuff it is protecting
- Use explicit default dtors
- Replace unnecessary lock in DelayedEncoder with SequencedTaskChecker

BUG=webrtc:7130

Review-Url: https://codereview.webrtc.org/2686103002
Cr-Commit-Position: refs/heads/master@{#16554}
2017-02-11 08:25:18 +00:00
6607d84b44 Move one CircularBuffer to webrtc::test namespace.
There are currently two webrtc::CircularBuffers defined:
- modules/audio_coding/test/utility.{h,cc}
- modules/audio_processing/echo_detector/circular_buffer.{h,cc}

This CL moves the former definition to the webrtc::test namespace,
to avoid link errors in a future build target.

BUG=None

Review-Url: https://codereview.webrtc.org/2667383008
Cr-Commit-Position: refs/heads/master@{#16553}
2017-02-11 08:24:10 +00:00
1a2183d0c3 Removing unnecessary parameters from CreateXChannel methods.
"bundle_transport_name" is no longer relevant here, and
"rtcp_mux_required" is implied by whether or not an RTCP transport is
passed in.

BUG=None

Review-Url: https://codereview.webrtc.org/2689503002
Cr-Commit-Position: refs/heads/master@{#16551}
2017-02-11 07:44:49 +00:00
757146baf1 Remove PC factory options param from LocalAudioSource::Create.
It wasn't being used at all, and there's no need to tie LocalAudioSource to
PeerConnection.

BUG=None

Review-Url: https://codereview.webrtc.org/2682253002
Cr-Commit-Position: refs/heads/master@{#16550}
2017-02-11 05:26:48 +00:00
112b2e99d8 Switching some interfaces to use std::unique_ptr<>.
This helps show where ownership is transfered between objects.

Specifically, this CL wraps cricket::VideoCapturer, MediaEngineInterface
and DataEngineInterface in unique_ptr.

BUG=None
TBR=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2685093002
Cr-Commit-Position: refs/heads/master@{#16548}
2017-02-11 04:13:37 +00:00
81baed36bf Add ability to return moved value from FunctorMessageHandler, Optional.
This functionality is desired for this CL:
https://codereview.webrtc.org/2675173003/

BUG=None

Review-Url: https://codereview.webrtc.org/2681283002
Cr-Commit-Position: refs/heads/master@{#16546}
2017-02-11 02:11:11 +00:00
a4549d6588 Fix SDP parsing crash due to missing track ID in "a=msid".
BUG=chromium:686405

Review-Url: https://codereview.webrtc.org/2676293003
Cr-Commit-Position: refs/heads/master@{#16545}
2017-02-11 01:26:22 +00:00
abdc857967 Update list of supported Android codecs based on field trial dynamically.
Currently filed trial value which controls H.264 high profile support is
read once only when factory is created. If field trial value is changed for
the next WebRTC call supported codec list need to be updated as well.

BUG=b/34816463

Review-Url: https://codereview.webrtc.org/2685183004
Cr-Commit-Position: refs/heads/master@{#16543}
2017-02-10 22:40:57 +00:00
9238245d9b Fix nr of bytes sent to Opus decoder in DTX mode
BUG=webrtc:7144

Review-Url: https://codereview.webrtc.org/2693453003
Cr-Commit-Position: refs/heads/master@{#16542}
2017-02-10 21:50:38 +00:00
90f1e1e0d7 Fixing SDP parsing crash due to invalid port numbers.
BUG=chromium:677029

Review-Url: https://codereview.webrtc.org/2675273003
Cr-Commit-Position: refs/heads/master@{#16541}
2017-02-10 20:35:05 +00:00
5bd5ca344e Rename "PacketTransportInterface" to "PacketTransportInternal".
This is the naming scheme we've been using for internal interfaces.

Also, this CL will introduce a PacketTransportInterface in the webrtc namespace,
which would get too easily confused with the rtc:: one:
https://codereview.webrtc.org/2675173003/

BUG=None

Review-Url: https://codereview.webrtc.org/2679103006
Cr-Commit-Position: refs/heads/master@{#16539}
2017-02-10 19:31:50 +00:00
88df0bc591 Make functions in fileutils.h use "const std::string&".
This way, the strings are not copied everytime the function is called.

BUG=webrtc:7142
NOTRY=True

Review-Url: https://codereview.webrtc.org/2685583009
Cr-Commit-Position: refs/heads/master@{#16537}
2017-02-10 17:27:14 +00:00
46a0021e4e Retransmitted packets are now counted in receive time
BUG=chromium:690358

Review-Url: https://codereview.webrtc.org/2683423002
Cr-Commit-Position: refs/heads/master@{#16536}
2017-02-10 17:16:05 +00:00
adb374b4ed Remove henrik.lundin from webrtc/common_video/OWNERS
BUG=none
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2687883004
Cr-Commit-Position: refs/heads/master@{#16535}
2017-02-10 16:44:31 +00:00
087bd34d23 Move AudioDecoder and related stuff to the api/ directory
BUG=webrtc:5805, webrtc:6725

Review-Url: https://codereview.webrtc.org/2668523004
Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 16:15:44 +00:00
84a3759825 Change rtc::VideoSinkWants to have target and a max pixel count
The current method with max_pixel_count and max_pixel_count_step_up,
where only one should be used at a time and this first signaling an
inclusive upper bound and other other an exclusive lower bound, makes
for a lot of confusion.

I've updated this to have a desired target and a maximum instead. The
source should select a resolution as close to the target as possible,
but no higher than the maximum.

I intend to also add similar frame rate settings in an upcoming cl.

BUG=webrtc:4172,webrtc:6850

Review-Url: https://codereview.webrtc.org/2672793002
Cr-Commit-Position: refs/heads/master@{#16533}
2017-02-10 15:04:27 +00:00
e9ad271db4 Increase the send-time history to 60 seconds.
This helps us avoid time-outs on really bad networks with long queues.
Also adding periodic logging of the fake network pipe's queue in milliseconds.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2687013005
Cr-Commit-Position: refs/heads/master@{#16532}
2017-02-10 14:09:28 +00:00
4ca18696c5 Allow residual echo detector to be enabled/disabled using AudioOptions, and no longer disable it on mobile platforms.
BUG=webrtc:7136

Review-Url: https://codereview.webrtc.org/2679103007
Cr-Commit-Position: refs/heads/master@{#16531}
2017-02-10 13:11:09 +00:00
0d729b3039 Check for use_x11 before runnig desktop_capture_modules_tests on linux.
The tests need "x11/shared_x_display.h" which is not included when use_x11 is false and we're on linux.

The problem is:

screen_capturer_integration_test.cc
 - requires ->
screen_drawer.h
 - requires ->
screen_drawer_linux.cc
 - requires ->
x11/shared_x_display.h
 which is not included when use_x11 is false.

BUG=None
NOTRY=True

Review-Url: https://codereview.webrtc.org/2684683003
Cr-Commit-Position: refs/heads/master@{#16529}
2017-02-10 09:38:23 +00:00
38e9324e4e Add script for plotting statistics from webrtc integration test logs.
Add tests (plot_videoprocessor_integrationtest.cc) to be used to plot stats from (not yet used).

Move VideoProcessorIntegrationTest fixture to separate file. To be used by plot_videoprocessor_integrationtest.cc.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2643853002
Cr-Commit-Position: refs/heads/master@{#16528}
2017-02-10 09:37:17 +00:00
55c5be0a03 Remove unused methods in WebRtcVoiceEngine and VoiceMediaChannel.
These methods relate to typing detection but are not used anymore. Typing detection is enabled through the VoiceDetection module on the APM.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2684933008
Cr-Commit-Position: refs/heads/master@{#16527}
2017-02-10 09:20:25 +00:00
654d54c073 Use std::unique_ptr in VideoProcessor.
Add RTC_CHECKs for failures in VideoProcessor::Init.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2684223002
Cr-Commit-Position: refs/heads/master@{#16526}
2017-02-10 08:16:07 +00:00
faedf7f942 Getting rid of "benign blocking error" log spam.
For UDP sockets, instead of calling "recv" with "MSG_PEEK" to see if the socket
is alive, just check whether or not it's been closed. It seems that, at least
on some versions of Android, with some specific conditions involving loopback
sockets, calling "recv" with a buffer that's too small for a UDP packet causes
an EFAULT error.

BUG=webrtc:6715

Review-Url: https://codereview.webrtc.org/2678353006
Cr-Commit-Position: refs/heads/master@{#16522}
2017-02-09 23:09:22 +00:00