Commit Graph

4245 Commits

Author SHA1 Message Date
7f52a6ef2b Split the implementation of VP8Encoder|Decoder::Create into a seperated file
(vp8_factory.cc).

R=fischman@webrtc.org, marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-24 23:56:39 +00:00
4f0801bd39 AviRecorder is missing a critical section.
BUG=2885
TEST=AUTOTEST
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-24 09:19:36 +00:00
33af96c5c2 Removed unused mock methods in audio_processing
TESTED=trybots,modules_unittests
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8999005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5597 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 23:56:05 +00:00
0f2809a5ac Add RTCP packet class.
Adds packet types: sr, rr, bye, fir.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5592 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 08:14:45 +00:00
c0907eff42 MIPS optimizations for AEC audio processing module
The resulting output streams obtained by testing with audioproc test application
are bit-exact with generic C code output streams.

Performance gain achieved:
- mips32 ~ 17%
- mips32r2 ~ 20%
- mipsdsp & mipsdspr2 ~ 21%

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7359004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5591 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 00:13:31 +00:00
d617a44a4f Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
- Transition scoped_ptr_mallocs to scoped_ptr.
- AlignedFreeDeleter matches Chromium's version.

TESTED=try bots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8969005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5587 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 21:08:36 +00:00
04a691adac Removing a variable that was never read
In NetEq4, the local variable discard_count in
PacketBuffer::DiscardOldPackets() was incremented but never read.
Removing it.

TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5584 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 15:27:00 +00:00
66061992fb ifdef the alsa code based on macro USE_X11
BUG=none
TEST=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 03:05:05 +00:00
78f0db4710 Fix the break caused by r5579.
TBR=tlegrand@google.com
BUG=

Review URL: https://webrtc-codereview.appspot.com/8939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5581 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 23:07:31 +00:00
c2d69d3229 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable.
BUG=2944
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5579 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 20:31:17 +00:00
97e7a640d8 Make WindowCapturerLinux handling window resize events.
We need to re-initialize the XServerPixelBuffer to the new size
when a window resize event is received.

BUG=https://code.google.com/p/chromium/issues/detail?id=339953
R=sergeyu@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 17:28:41 +00:00
056287eee0 This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both.
BUG=issue2874
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5576 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 13:45:54 +00:00
8098e07478 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
Add counter to RTCP sender and RTCP receiver.
Add video api GetRtcpPacketTypes().

BUG=2638
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 11:59:02 +00:00
e384104166 Fix DesktopAndCursorComposer not to crash
DesktopAndCursorComposer was crashing when screen/window
capturer returns a NULL frame due to an error.

BUG=crbug.com/344093
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 23:26:34 +00:00
27c6980239 Move the volume quantization workaround from VoE to AGC.
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.

Add a test to verify the behavior.

TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 20:24:56 +00:00
c320027d6a Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called
twice with the same settings.

Without this change, setting up a call with the new video API will
print a trace warning.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5566 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 14:51:00 +00:00
2086e0fbf3 Remove unnecessary warnings.
BUG=
TEST=try job
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8719005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 14:22:20 +00:00
346094cb01 Incorrect overhead calculation when using FEC + RTP extension headers.
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.

BUG=2899
R=phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8769005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 08:40:33 +00:00
340746aa13 Misc small nits in NetEq
Fixing a few small things found recently. This is mostly cosmetics.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8749005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5558 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 11:37:16 +00:00
f92aaff104 AudioProcessing is not a Module.
Remove Module as the base class of AudioProcessing. The inherited
methods were all no-ops.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/8779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5556 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-15 04:22:49 +00:00
e2fc13e42f Refactoring common_audio/signal_processing: Removed two macros used by isac only.
Removed a macro for malloc() and one for free().  They are only used by the audio codec isac, where I replaced the macro with its implementation.
Further, the includes were updated with full paths and put in alphabetical order.

BUG=N/A
TESTED=trybots,module_tests,module_unittests
R=turaj@webrtc.org, turajs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5554 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 23:12:34 +00:00
38bf249049 Initialize output_will_be_muted_.
TBR=aluebs

Review URL: https://webrtc-codereview.appspot.com/8659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5546 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 17:43:44 +00:00
fcfc6a990e Small refactoring of NetEq unittest for CNG with clock drift
Converting the test to a method within the test fixture, and setting
up two tests that call this method. One for positive and one for
negative clock drift. The one with positive clock drift is disabled
for now since it does not pass, but will be re-enabled shortly.

This change is only made for NetEq4.

R=tlegrand@google.com

Review URL: https://webrtc-codereview.appspot.com/8599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5541 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 11:42:28 +00:00
17342e5092 Add a method to inform AudioProcessing that its output will be muted.
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5538 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 22:28:31 +00:00
de782180b0 Change the type of propagation delta from int64 to int.
The delta value never exceeds the range of int. Changing it to int will save memory and copying cost.

BUG=2910
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 19:19:23 +00:00
07b5950c12 Initialize key_pressed_.
Was resulting in an error on Mac Asan:
[ RUN      ] ApmTest.DebugDump
[libprotobuf FATAL ../../third_party/protobuf/src/google/protobuf/message_lite.cc:224] CHECK failed: !coded_out.HadError():

TBR=aluebs

Review URL: https://webrtc-codereview.appspot.com/8539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5536 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 16:41:13 +00:00
ce8e077cf0 Add a keypress field to the audioproc debug proto.
Log the value in AudioProcessing, and unpack it to a new file in the
unpacking tool.

TESTED=
- The new tool can unpack old dumps.
- The old tool can unpack new dumps (without keypress.bool).
- Unpacking a new dump from voe_cmd_test produces a keypress.bool that
appears correct when examined.

R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5535 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 15:28:30 +00:00
75dd2885c5 Add an interface for accepting keypress signals to AudioProcessing.
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5529 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 20:52:30 +00:00
8685af7ea0 Remove "Too long processing time of Incoming frame" logspam.
This isn't indicative of anything actionable and spams android logcat with times
in the 10-30ms range several times per second.

BUG=2732
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 17:48:11 +00:00
a80be4b23c Add boundary checking to supress gcc 4.8.3 warning.
BUG=2888
Test=try, voe_cmd_test

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5526 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 16:38:45 +00:00
82ebb463fd Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.

It make the simliar feature's implementation consistent.

R=andrew@webrtc.org, fischman@webrtc.org, fischman@chromium.org
BUG=334447

Committed: https://code.google.com/p/webrtc/source/detail?r=5517

Review URL: https://webrtc-codereview.appspot.com/7769006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5524 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 04:48:27 +00:00
a65abf9d3a Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file."
This reverts commit 7686f0ddda717a9e776be0e219f039f68a10f9ed.

BUG=

TBR=andrew@webrtc.org, fischman@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/8369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5520 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:26:26 +00:00
1f64f06784 Add stats of incoming frame delays for debugging bandwidth estimation.
BUG=crbug/338380
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:12:14 +00:00
7686f0ddda Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.

It make the simliar feature's implementation consistent.

R=andrew@webrtc.org, fischman@webrtc.org, fischman@chromium.org
BUG=334447

Review URL: https://webrtc-codereview.appspot.com/7769006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5517 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 17:42:34 +00:00
ad3035fc9e Fix WindowCapturerWin to unselect bitmap before destroying DC.
BUG=https://code.google.com/p/webrtc/issues/detail?id=2901
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5504 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 21:24:04 +00:00
77c917a6ee Plot the capacity of a trace-based delivery filter.
Breaks out the instantaneous rate counters to its own class.

R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7999005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5494 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-06 16:34:47 +00:00
c88d3368d5 Add delay and send/receive throughput plots to BWE simulation.
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 15:57:14 +00:00
75642fcd9a Implementing replacement audio support in neteq_rtpplay
This CL makes it possible to replace the payload in an RTP stream
with audio from another (PCM) file. The new payload will be encoded as
PCM16b. The RTP headers will be updated to reflect this change of
payload type.

BUG=2834
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5490 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 08:49:13 +00:00
e6ab21b9ca Fixing a bug in DummyRTPpacket
This bug caused writing outside allocated memory when RTP header
extensions were used.

BUG=2834
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8009005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5489 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 08:46:46 +00:00
54744918ef Update AudioProcessing::Create docs.
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/8039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5488 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 06:30:29 +00:00
20a60ea39d Fix a cursor capturing issue on Windows.
The input position to WindowFromPoint should be relative to the desktop, not
relative to the window; if the result from WindowFromPoint is a child window
of the shared top window, it should be captured.

BUG=
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5487 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 17:49:12 +00:00
0e5a2b5de6 Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered.
This can happen when switching between multiple streams and a single while getting feedback from the receiver.

BUG=2881
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5486 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 14:38:25 +00:00
f6a638e001 Trivial rename of non-compile time consts.
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7669006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5482 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 01:31:28 +00:00
422fdbf502 Wire up feedback to VideoSender.
BUG=
R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5474 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 16:33:50 +00:00
c9ee412070 Re-enabling audio processing tests
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5473 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 14:41:57 +00:00
c1e28038ba Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-02 15:30:20 +00:00
1af5ea0538 Implement single monitor capture on Mac.
BUG=2787, 2824
TESTED=MacBook Pro Retina with an external monitor; verified changing display configuration while capturing; add/remove monitor while capturing; verified cursor position.
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5471 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-01 02:03:24 +00:00
83aee8f450 Fixing test name for NetEqPerformanceTest
The naming did not follow conventions.

BUG=2859
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5469 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 11:46:34 +00:00
1dd9b4d98e Add BWE tools for parsing RTP files.
bwe_rtp_play feeds packets from an RTP file into the BWE and prints the estimates.
bwe_rtp_to_text parses an RTP file and outputs the result to a text file.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7689006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5466 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 09:15:48 +00:00
bda5fa77af Fix the mouse cursor offset issue on Mac.
The problem is that MouseCursorMonitor returns coordinates in DIPs, while DisplayAndMouseComposer assumes that they are in physical pixels. The fix is to convert the position to physical pixels in MouseCursorMonitorMac.

R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7739006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5463 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-30 23:27:35 +00:00