Commit Graph

1309 Commits

Author SHA1 Message Date
74767401f2 Fix a bug preventing FilePlayer from playing encoded wav files
A bug in ACM2 prevented decoding and playout of wav files where the
audio data was encoded (i.e., not just linear PCM 16 bit data).

This CL fixes the issue, and adds a unit test for the FilePlayer.

BUG=3386
R=henrike@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6248 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-26 13:37:45 +00:00
546961a9d3 Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix.
Arithmetic encoder does not right the last 2 or 3 bytes of |streamval| when terminating the bit-stream. Perhaps the last bytes makes no difference in decoding the stream. However, the decoder reads full |streamval| (int16_t) going out of boundary and reading uninitialized values. This avoids this problem. by inserting zero-bytes whenever decoder intends to read outside boundary.

BUG=1353,chrome373312,b/13468260
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6234 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:14:29 +00:00
aa5ea1c0f9 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED
2. Add two new APIs to configure codec internal FEC

3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it.

New ACM gives good result.
Old ACM does not use NetEq 4, so FEC won't be decoded.

BUG=
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 15:16:51 +00:00
1bb5da04fe Adds missing include of assert header.
BUG=3380
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/14569008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6221 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-22 14:31:14 +00:00
21f7d6d2fe WebRTCDemo: move the deletion of CritSect to end of the dtor to fix a crash in Android video renderer.
BUG=3368
TEST=Manual Test

Review URL: https://webrtc-codereview.appspot.com/21519005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6220 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-22 02:57:55 +00:00
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
7ca277b574 Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect.
BUG=https://code.google.com/p/webrtc/issues/detail?id=3196
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6213 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 16:02:31 +00:00
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..."
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
c2213b6a0f Revert 6208 "Patch from henrike@webrtc.org"
Wasn't enough. I'll have to revert the whole rev 6202.

> Patch from henrike@webrtc.org
> https://code.google.com/p/webrtc/source/detail?r=6202
> didn't work for at least one file and broke most of 
> the compile steps in the FYI bots. The file is reverted
> here.
> 
> TBR= henrike@webrtc.org, sergeyu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/17609004

TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6209 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 10:03:09 +00:00
86df8acc92 Patch from henrike@webrtc.org
https://code.google.com/p/webrtc/source/detail?r=6202
didn't work for at least one file and broke most of 
the compile steps in the FYI bots. The file is reverted
here.

TBR= henrike@webrtc.org, sergeyu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6208 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 08:40:56 +00:00
48438c2c90 Enabling NetEq bit-exactness test for Win x64
A new reference file (neteq4_universal_ref_win_64.pcm) was generated and
uploaded.

Also removing the old hack to have different reference files
for different version of Visual Studio. The test is now only supporting
VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit
output identical to the generic reference file
(neteq4_universal_ref.pcm), so the specialized one
(neteq4_universal_ref_win_32.pcm) could have been removed. However,
since the resources sync mechanism does not include removing of old
files, a client could pick up the old reference and fail. Therefore,
this cl also updates neteq4_universal_ref_win_32.pcm to be identical to
neteq4_universal_ref.pcm.

BUG=1458
R=kjellander@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 16:07:43 +00:00
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
70bb2d5755 Revert r6198 "Expose the original packet length in in the RTP play tools."
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6200 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 13:25:48 +00:00
e208458643 Expose the original packet length in in the RTP play tools.
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6198 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 13:09:16 +00:00
be4ab99a53 Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8.
BUG=3370
R=bjornv@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6197 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 12:42:01 +00:00
a36db970bd Suppress GMOCK printouts from TestVideoSenderWithVp8
Adding a missing EXPECT_CALL.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6196 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 11:16:10 +00:00
a826006132 Add NACK and RPSI packet types to RTCP packet builder.
Fixes bug found when parsing received RPSI packet.

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

Review URL: https://webrtc-codereview.appspot.com/17419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 09:53:51 +00:00
cb711f77d2 Add interface to propagate audio capture timestamp to the renderer.
BUG=3111
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:39:11 +00:00
ebb467fdc8 Avoid NACK-list flush error on keyframe packets.
Receiver code used to indicate a flush error even if the incoming packet
is a keyframe, forcing a request of a keyframe. Now it takes this
keyframe into account and doesn't error as the stream is decodable from
this point.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6188 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 15:28:02 +00:00
64339a7069 Don't crash if a frame returned from the decoder is too old.
BUG=crbug/371805
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6187 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 13:31:35 +00:00
725e582461 Use the new gyp_var_prefix local variable set by gyp instead of the
global GYP_VAR_PREFIX set by the makefiles, since the latter is not
guaranteed to still be the same value at the time the command is
executed. Also, use abspath instead of realpath to convert paths to
absolute, since realpath expands to the empty string if the target file
doesn't exist, complicating build debugging.

BUG=
R=andrew@webrtc.org, torne@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6186 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 17:56:10 +00:00
14abcc7322 libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict.
libvpx macro (UNUSED) can be found here:
http://src.chromium.org/viewvc/chrome/trunk/deps/third_party/libvpx/source/libvpx/vpx/vpx_codec.h

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6185 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 16:54:44 +00:00
a3b5673879 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16
This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed.

BUG=3348
TESTED=trybots, manual unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 12:11:20 +00:00
1b21a57902 common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16
Macro was only mapping a function used in one place.

BUG=3348
TESTED=trybots, unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6180 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 06:40:31 +00:00
d5da25063c Revert "Revert "Audio processing: Feed each processing step its choice
of int or float data"

This reverts commit 6142.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6172 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 11:17:21 +00:00
b4e80e095f Re-enable almost all NetEqDecodingTests for Android
All but three tests in NetEqDecodingTest could be re-enabled without
any changes. Also making sure that the TestNetworkStatistics test exits
on first diff. (Otherwise, the log output gets flooded with error
messages.)

The tests that are still disabled are:
NetEqDecodingTest.TestBitExactness
NetEqDecodingTest.TestNetworkStatistics
NetEqDecodingTest.DecoderError

BUG=3343
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6168 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 07:14:00 +00:00
7cb4752184 WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process.
This cl is to teach videocapture android how to deinitialize and allow it to be re-initializable.

BUG=3284
TEST=ManualTest
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6167 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 03:18:15 +00:00
21299d4e00 Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.

Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc

Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.

BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 19:00:59 +00:00
88abf11cad Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe.
BUG=3111
TEST=try bots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6152 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 16:53:51 +00:00
a36ad6929d Add webrtc field trials API.
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.

Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.

Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.

BUG=crbug/367114
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:24:04 +00:00
2fa17015d1 Re-enable NetEqExternalDecoderTest for Android
The test runs without problems now.

BUG=3343
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16519005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6144 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 11:45:22 +00:00
bf93fb3176 Re-enable NetEQ DecoderDatabase test for Android
The test was failing because iLBC is not enabled on Android. Now, the
test is using PCM16B instead.

BUG=3343
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6143 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 10:42:03 +00:00
b1a66d166c Revert "Audio processing: Feed each processing step its choice of int or float data"
This reverts r6138.

tbr=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6142 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:39:56 +00:00
db60434b31 Re-enable the BitrateEstimatorTest cases for the Call API.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6141 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:15:19 +00:00
5c49c64de5 Remove all use of AudioFrame::energy_ from AudioCodingModule
Since r6117, the energy is always calculated in the mixer module,
regardless of the value that ACM sets for energy_.

This part of the the aftermath of issue 3255.

BUG=3255
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6140 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:06:52 +00:00
934a265a47 Audio processing: Feed each processing step its choice of int or float data
Each audio processing step is given a pointer to an AudioBuffer, where
it can read and write int data. This patch adds corresponding
AudioBuffer methods to read and write float data; the buffer will
automatically convert the stored data between int and float as
necessary.

This patch also modifies the echo cancellation step to make use of the
new methods (it was already using floats internally; now it doesn't
have to convert from and to ints anymore).

(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the echo canceller no longer unnecessarily
converts float data to int and then immediately back to float for each
iteration in the loop in EchoCancellationImpl::ProcessCaptureAudio.)

BUG=
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6138 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:01:35 +00:00
3d5cb33da4 Remove WEBRTC_TRACE use in video_capture/
Does not touch platform-specific code.

BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6137 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 08:42:07 +00:00
c3e8abda7c Deleting all NetEq3 files
NetEq3 is deprecated and replaced by NetEq4
(webrtc/modules/audio_coding/neteq4/).

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14469007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6118 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 10:40:52 +00:00
4d363ae305 The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy.
This part of the the aftermath of issue 3255.

BUG=3255
R=andrew@webrtc.org, henrike@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6117 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:50:02 +00:00
3a5825909d Deleting all ACM1 files
ACM1 is deprecated and replaced by ACM2
(webrtc/modules/audio_coding/acm2/).

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6115 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:08:56 +00:00
924e81f797 Echo cancellation functions docs: Follow style guide w.r.t. placement of *
The style guide says to use "void* x", not void *x", and the code in
these files already do so, but the comments do not. Fix that.

Also, in the interest of reducing eye strain, I fixed the vertical
alignment in a small number of cases.

BUG=
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6101 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 09:55:19 +00:00
b9863ce6ba One of the NetEq methods needs to be virtual.
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6099 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-10 00:58:49 +00:00
17bf9a2c5e Modifying neteq.gyp
|codecs| list is introduced, which is used in both |neteq_dependencies| and AudiDecoder unittests dependencies. This way, AudioDecoder unittests depend only on |codecs| and not on the entire |neteq_dependencies|, which is unnecessary.

TEST=trybots
BUG=
R=andrew@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6094 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-09 18:04:50 +00:00
4cc763621e AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_
data_was_mixed_ was always false, so it can be removed. That makes the
role of data_ simpler, but not so simple that it doesn't merit an
explanation.

BUG=
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6076 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-08 07:10:11 +00:00
66773a032a Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
BUG=3111
TEST=try bots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 17:09:44 +00:00
94f1d4cd55 Fix odd codes in video_capture on Mac.
BUG=3272
TEST=vie_auto_test
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6070 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 02:57:13 +00:00
b1eb43142e video_render.gypi: clean up some libraries directives to be more specific.
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6068 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 00:09:30 +00:00
ed4cb56575 Remove timestamp_extrapolator's dependency to Clock and vcm defines.
TEST=existing tests
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6058 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-06 04:50:49 +00:00
382c0c209d Allow the RTP level indicator computation to work at any sample rate.
Break out the computation to a separate class, and call directly into
this from channel.cc rather than going through AudioProcessing. This
circumvents AudioProcessing's sample rate limitations.

We now compute the RMS over all samples rather than downmixing to a
single channel. This makes the call point in channel.cc easier, is
more "correct" and should have similar (negligible) complexity.

This caused slight changes in the RMS output, so the ApmTest.Process
reference has been updated. Snippet of the failing output:

[ RUN      ] ApmTest.Process
Running test 4 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 5 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 6 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 10 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 11 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 12 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27

BUG=3290
TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc
produce reasonable printed out results from RMS().

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-05 18:22:21 +00:00
4220434d37 Implement the Windows screen capturer using the Magnification API.
The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.

BUG=2789
TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
R=sergeyu@chromium.org, wez@chromium.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6048

Review URL: https://webrtc-codereview.appspot.com/12149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6053 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-05 16:08:47 +00:00