Commit Graph

3529 Commits

Author SHA1 Message Date
d68fa65d76 Improve cleaning for Android demo applications
There are a bunch of directories that are not cleaned between
builds since they're added to .gitignore.

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40999004

Cr-Commit-Position: refs/heads/master@{#8542}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8542 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-28 11:18:20 +00:00
f7bb6e723b Use new API from BoringSSL to get RFC name of cipher.
This CL uses the new API "SSL_CIPHER_get_rfc_name" from BoringSSL to
get the RFC-compliant cipher name instead of having a custom hardcoded
list of cipher names.

BUG=none
R=juberti@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40829004

Cr-Commit-Position: refs/heads/master@{#8541}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8541 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-28 01:41:49 +00:00
d31250518a Test to try to track down the alignment problem on Mac 10.9.
There's no code change here, I'm rearranging member variables of the
trace class and adding a sizeof check to the CriticalSection
class + alignment attribute for the mutex, on Mac only.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38339004

Cr-Commit-Position: refs/heads/master@{#8540}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8540 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-28 00:01:56 +00:00
07dcf60ee0 Revert 8532 "Ensure only temporary IPv6 address is selected as t..."
> Ensure only temporary IPv6 address is selected as the best IP.
> 
> The current logic of IPv6 selection could still have a small chance for non-temporary address to be selected for candidate. The scenario is that when there is no non-deprecated temporary IP, the global ones could be selected.
> 
> Global ones don't necessarily carry MAC. However, instead of comparing whether it has the MAC in it (sometimes 5 out of 6 elements from a MAC are the same, only one diffs), we should just err on the safe side.
> 
> BUG=4348
> R=juberti@webrtc.org, pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/38289004

TBR=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38319004

Cr-Commit-Position: refs/heads/master@{#8534}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8534 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 18:42:46 +00:00
385a7ceb1f Ensure only temporary IPv6 address is selected as the best IP.
The current logic of IPv6 selection could still have a small chance for non-temporary address to be selected for candidate. The scenario is that when there is no non-deprecated temporary IP, the global ones could be selected.

Global ones don't necessarily carry MAC. However, instead of comparing whether it has the MAC in it (sometimes 5 out of 6 elements from a MAC are the same, only one diffs), we should just err on the safe side.

BUG=4348
R=juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38289004

Cr-Commit-Position: refs/heads/master@{#8532}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8532 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 18:10:43 +00:00
fbef5c6293 Remove lock in ViEFrameProviderBase::IsFrameCallbackRegistered.
After looking at how this class is used I discovered that the threading model isn't completely crazy and that we can remove a lock from a method that's causing the most contention in getStats right now (and probably many other places).  IsFrameCallbackRegistered() looks innocent enough, but it contented with the frame encoder, which can make it wait for tens of milliseconds, unnecessarily.

In addition to removing the lock, I:

* Documented the threading model
* Added checks to guard against regressions
* Reduced lock scope in the Deregister function (not calling out to the notification while holding the lock).
*  made virtual functions non-virtual (since they're not really being overriden) and removed unnecessary method that, if called by other threads than the ctor thread, returned a value that wasn't necessarily correct.

BUG=2822
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40859004

Cr-Commit-Position: refs/heads/master@{#8531}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8531 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 15:43:23 +00:00
7400e0b876 Revert "I420VideoFrame: Remove functions set_width, set_height, and ResetSize"
This reverts commit r8434.

Reason for revert: Introduced a race condition. If ViECaptureProcess() -> SwapCapturedAndDeliverFrameIfAvailable() is called twice without a call to OnIncomingCapturedFrame() in between (with both captured_frame_ and deliver_frame_ populated), an old frame will be delivered again, since captured_frame_->IsZeroSize() will never be true.

BUG=4352
TBR=perkj@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40129004

Cr-Commit-Position: refs/heads/master@{#8530}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8530 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 15:19:18 +00:00
4b3618c7f3 Remove TraceImpl logging thread.
Simplifies TraceImpl significantly. Chromium uses trace callbacks and
logs directly through the trace callback interface. Added slowdowns when
logging to file are expected to only affect test targets.

BUG=
R=andresp@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34329004

Cr-Commit-Position: refs/heads/master@{#8529}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8529 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 15:05:50 +00:00
6c2e506cf4 Workaround Mac align bug for observer_ and crit_.
Unblocks rolling WebRTC into Chromium so we can debug what's actually
going on.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39329004

Cr-Commit-Position: refs/heads/master@{#8528}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8528 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 14:28:26 +00:00
3985f0151a ProcessThread improvements.
* Added a way to notify a Module that it's been attached to a ProcessThread.
  The benefit of this is to give the module a way to wake up the thread
  when it needs work to happen on the worker thread, immediately.
  Today, module instances are typically registered with a process thread
  outside the control of the modules themselves.  I.e. they typically
  don't know about the process thread they're attached to.

* Improve ProcessThread's WakeUp algorithm to not call TimeUntilNextProcess
  when a WakeUp call is requested.  This is an optimization for the above
  case which avoids the module having to acquire a lock or do an interlocked
  operation before calling WakeUp(), which would ensure the module's
  TimeUntilNextProcess() implementation would return 0.

BUG=2822
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39239004

Cr-Commit-Position: refs/heads/master@{#8527}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8527 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 13:37:25 +00:00
abbdd520b0 AudioEncoder: documentation fix
Follow-up to https://webrtc-codereview.appspot.com/38279004/

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38309004

Cr-Commit-Position: refs/heads/master@{#8524}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8524 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 09:20:25 +00:00
3aca0b0b31 Add 48kHz support to Beamformer
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.

BUG=webrtc:3146
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35159004

Cr-Commit-Position: refs/heads/master@{#8522}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8522 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 21:53:00 +00:00
b1f0de30be AudioEncoder: change Encode and EncodeInternal return type to void
After code cleanup done on issues:
https://webrtc-codereview.appspot.com/34259004/
https://webrtc-codereview.appspot.com/43409004/
https://webrtc-codereview.appspot.com/34309004/
https://webrtc-codereview.appspot.com/34309004/
https://webrtc-codereview.appspot.com/36209004/
https://webrtc-codereview.appspot.com/40899004/
https://webrtc-codereview.appspot.com/39279004/
https://webrtc-codereview.appspot.com/42099005/
and the similar work done for AudioEncoderDecoderIsacT,  methods AudioEncoder::Encode and AudioEncoder::EncodeInternal will always succeed. Therefore, there is no need for them to return bool value that represents success or failure.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38279004

Cr-Commit-Position: refs/heads/master@{#8518}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8518 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 15:38:46 +00:00
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 14:43:50 +00:00
ac2d27d9ae Fix style violations in common_types.h and config.h
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.

The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.

BUG=163
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26089004

Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 14:01:28 +00:00
891d48393e Wire up target_media_bitrate in VideoSendStream.
Also wires up target_enc_bitrate in WebRtcVideoEngine2.

BUG=1667,1788
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42479004

Cr-Commit-Position: refs/heads/master@{#8515}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8515 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:16:17 +00:00
9dd0ebc379 Remove the default RTP module.
This CL removes the default module owned by ViEEncoder, functionality in
the module to register default modules and the final changes in
rtp_rtcp_impl using default/child modules.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42509004

Cr-Commit-Position: refs/heads/master@{#8514}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8514 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 12:58:24 +00:00
3e6e271ec3 Implement CpuOveruseMetrics as callbacks.
Adds avg_encode_ms and encode_usage_percent in WebRtcVideoEngine2 and
corresponding stats to VideoSendStream::Stats.

BUG=1667, 1788
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42429004

Cr-Commit-Position: refs/heads/master@{#8513}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8513 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 12:20:24 +00:00
38d9cc51d5 Add back return statement after FATAL()
Some compilers do not accept that non-void functions end with FATAL()
instead of a return statement. This change adds back a few return
statements that were removed in r8463.

BUG=4344
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42519004

Cr-Commit-Position: refs/heads/master@{#8509}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8509 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 09:43:19 +00:00
f09e7b8a4f WebRtcVideoFrame: DCHECK exclusive ownership for non-const pixel access
Add some const safety by DCHECK(HasOneRef()) in non-const GetYPlane. This CL also replaces all incorrect non-const calls with const calls for pixel data access in cricket::VideoFrame. It's easy to call the non-const version of e.g. GetYPlane by mistake, even if only const-access is needed. For example:
const scoped_ptr<cricket::VideoFrame> foo;
const uint8_t* y = foo->GetYPlane();
will actually call the non-const version of GetYPlane.

R=mflodman@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39079004

Cr-Commit-Position: refs/heads/master@{#8507}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8507 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 14:50:19 +00:00
6c66163567 Fix TestScaler PSNR tests
TestScaler::ComputeAvgSequencePSNR is currently a complex NOP, that always returns kPerfectPSNR. Two frames are read from files into arrays, and then converted into I420VideoFrames. However, the incorrect function ConvertFromI420 is used instead of ConvertToI420, resulting in two empty I420VideoFrames. I420PSNR on empty frames returns kPerfectPSNR.

This CL replaces ConvertFromI420 with ConvertToI420 and actually measures the PSNR. Unfortunately, some tests do not pass when we use the real psnr. The tests that fail are the ones that scale back and forth to a different aspect ratio. webrtc::Scaler has been changed to preserve aspect ratio, and this means that we will end up with a cropped frame if scale and rescale to a different target aspect ratio. I simply removed those tests to make it pass. Having some working tests instead of a lot of dummy tests seems like a win.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35339004

Cr-Commit-Position: refs/heads/master@{#8506}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8506 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 14:10:47 +00:00
96abda0316 Removing FEC functionality from the default RTP module.
This CL removes the last default module methods used from ViEEncoder and
the default module itself will be removed in a separate CL.

BUG=769
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35309004

Cr-Commit-Position: refs/heads/master@{#8505}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8505 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 13:50:51 +00:00
9b969e167d AudioEncoderCopyRed: CHECK that encode call doesn't fail
Call to AudioEncoder::Encode fails only if fed bad input, so instead of handling failure, we can just CHECK.
There is also no need to handle case where size of encoded data is larger than allowed maximum, so we just CHECK.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42099005

Cr-Commit-Position: refs/heads/master@{#8504}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8504 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 11:53:45 +00:00
749c60217d Moved gypi to avoid presubmit warning about '..' when touching the files.
R=kjellander@webrtc.org,mflodman@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39299004

Cr-Commit-Position: refs/heads/master@{#8503}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8503 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 11:50:44 +00:00
5c928ebd1d Let first packet through to avoid getting key frame requests (and no nacks) for EndToEndTest.ReceivedFecPacketsNotNacked.
BUG=4328
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38259004

Cr-Commit-Position: refs/heads/master@{#8502}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8502 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 11:47:43 +00:00
09c77b95bb Add decoder-timing stats to VideoReceiveStream.
Also breaks out SsrcStats from VideoReceiveStream::Stats as they don't
have that much overlap.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667, 1788

Review URL: https://webrtc-codereview.appspot.com/40819004

Cr-Commit-Position: refs/heads/master@{#8501}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8501 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 10:42:45 +00:00
c5558b7021 Remove AudioCodingModule's dependency on the Module interface
BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42069004

Cr-Commit-Position: refs/heads/master@{#8500}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8500 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 10:37:46 +00:00
af82f75690 Let Add10MsData method do the encoding work as well
This change essentially makes the Process method a no-op. All it does
now is to return a stored value from the last encoding.

The purpose of this change is to forge the Add... and Process methods
into one and the same.

BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38229004

Cr-Commit-Position: refs/heads/master@{#8499}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8499 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 10:33:42 +00:00
8d350d4bc4 Add new AcmGenericCodecTest and verify output from Encode function
The test specifically verifies that the output is as expected when
DTX/CNG is used.

COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38219004

Cr-Commit-Position: refs/heads/master@{#8497}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8497 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 10:06:20 +00:00
1eda4e3db6 Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"
This should be safe to land now that issue 4143 was resolved (in r8492).
This change effectively reverts 8488.

TBR=kwiberg@webrtc.org

Original commit message:
This CL changes the way the decoder sample rate is set and updated. In
practice, it only concerns the iSAC (float) codec.

One single iSAC decoder instance is used for both wideband and
super-wideband decoding, and the instance must be told to switch
output frequency if the payload type changes. This used to be done
through a call to UpdateDecoderSampleRate, but is now instead done in
the Decode call as an extra parameter.

Review URL: https://webrtc-codereview.appspot.com/39289004

Cr-Commit-Position: refs/heads/master@{#8496}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8496 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 10:03:19 +00:00
0a3ff7976b New AudioTrack implementation now works on pre-Lollipop devices.
The previous version used an AudioTrack.write() implementation that required API Level 21. This is now fixed.

BUG=4339
R=magjed@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42459004

Cr-Commit-Position: refs/heads/master@{#8494}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8494 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 09:28:20 +00:00
d4dfba8ea1 iSAC Decode: Prevent Memcheck from complaining about uninitialized value
Without this patch, Valgrind's Memcheck was complaining that the test
for whether we should return -1 following the call to
WebRtcIsac_DecodeLb made a conditional branch or move based on the
value of numSamplesLB, which was uninitialized if WebRtcIsac_DecodeLb
failed.

However, as can be seen in the source, the control flow only depends
on the value of numSamplesLB if numDecodedBytesLB >= 0; i.e., if
WebRtcIsac_DecodeLb returned successfully, in which case numSamplesLB
is always initialized. The discrepancy is due to the fact that
Valgrind works on the generated machine code, which contains spurious
such dependencies. The generated code for this test:

  if ((numDecodedBytesLB < 0) || (numDecodedBytesLB > lenEncodedLBBytes) ||
      (numSamplesLB > MAX_FRAMESAMPLES)) {

looks like this:

  95:   0f bf 45 d6             movswl -0x2a(%rbp),%eax
  99:   3d c0 03 00 00          cmp    $0x3c0,%eax
  9e:   0f 8f 45 01 00 00       jg     1e9 <Decode+0x1e9>
  a4:   44 89 f0                mov    %r14d,%eax
  a7:   c1 e0 10                shl    $0x10,%eax
  aa:   0f 88 39 01 00 00       js     1e9 <Decode+0x1e9>
  b0:   41 0f bf ce             movswl %r14w,%ecx
  b4:   89 8d 98 e1 ff ff       mov    %ecx,-0x1e68(%rbp)
  ba:   41 0f bf c7             movswl %r15w,%eax
  be:   39 c1                   cmp    %eax,%ecx
  c0:   0f 8f 23 01 00 00       jg     1e9 <Decode+0x1e9>

Note how the compiler has seemingly ignored the C language's guarantee
that the arguments to || must be evaluated in left-to-right order, and
compares numSamplesLB (%eax) with MAX_FRAMESAMPLES (0x3c0, a.k.a. 960)
before the other two conditions! If the uninitialized value in
numSamplesLB happens to be greater than 960, we'll jump to
Decode+0x1e9 (where we'll return -1) without even looking at the other
two conditions. Has the compiler generated broken code?

Well, no. If numDecodedBytesLB is < 0 so that numSamplesLB is
uninitialized, we'll end up jumping to 1e9 whether that value is
greater than 960 or not; we'll just do it with different jump
instructions. This is entirely invisible as far as the C language is
concerned, but the dependency on the uninitialized value is visible at
the machine code level, which is why Memcheck complains.

This patch solves the problem by pragmatically initializing
numSamplesLB before the call even though it isn't necessary other than
for placating Memcheck.

BUG=4143
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36309004

Cr-Commit-Position: refs/heads/master@{#8492}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8492 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 08:09:28 +00:00
87a592dc50 Fix dependencies of media_file module and move gypi into the right dir to
avoid submit warnings referencing files with '..'.

TBR=kjellander@webrtc.org
R=kjellander@webrtc.org
BUG=4185

Review URL: https://webrtc-codereview.appspot.com/40919004

Cr-Commit-Position: refs/heads/master@{#8491}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8491 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 03:18:44 +00:00
49096de442 DCHECK send DataCountersUpdated for valid SSRCs.
Also updates RTPSender to not update RTX stats when RTX is disabled.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42399004

Cr-Commit-Position: refs/heads/master@{#8489}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8489 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 22:38:22 +00:00
903182bd8e Revert r8476 "Set decoder output frequency in AudioDecoder::Decode call"
This change uncovered issue 4143, evading the Memcheck suppression
since the signature is changed in the Decode function.

A fix for this is in the making; see
https://review.webrtc.org/36309004. This CL will be re-landed once the
fix is in place.

BUG=4143
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42089004

Cr-Commit-Position: refs/heads/master@{#8488}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8488 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 21:18:44 +00:00
b9c18d5643 Set decoder output frequency in AudioDecoder::Decode call
This CL changes the way the decoder sample rate is set and updated. In
practice, it only concerns the iSAC (float) codec.

One single iSAC decoder instance is used for both wideband and
super-wideband decoding, and the instance must be told to switch
output frequency if the payload type changes. This used to be done
through a call to UpdateDecoderSampleRate, but is now instead done in
the Decode call as an extra parameter.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34349004

Cr-Commit-Position: refs/heads/master@{#8476}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8476 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 15:59:20 +00:00
f88791d783 AudioEncoderCng: CHECK that encode calls don't fail
Calls to WebRtcCng_Encode, AudioEncoder::Encode and Vad::VoiceActivity fail only if fed bad input, so instead of handling failure, we can just CHECK. This also makes it unnecessary for methods AudioEncoderCng::EncodePassive and AudioEncoderCng::EncodeActive to return a value, so we can make them void.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39279004

Cr-Commit-Position: refs/heads/master@{#8475}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8475 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 14:59:19 +00:00
5e3fea1049 Fixing WebRTC engine demo JNI symbol export.
R=henrika@webrtc.org
BUG=None

Review URL: https://webrtc-codereview.appspot.com/39269004

Cr-Commit-Position: refs/heads/master@{#8474}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8474 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 14:51:20 +00:00
db8e605c16 Break out BWE test models to separate files
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36299004

Cr-Commit-Position: refs/heads/master@{#8471}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8471 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 13:24:56 +00:00
ccd7c7c45d Remove more unused code in ACM
This CL removes a lot of unused code in AudioCodingModuleImpl and
ACMGenericCodec.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40889004

Cr-Commit-Position: refs/heads/master@{#8470}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8470 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 12:02:18 +00:00
13ca5f6db2 AudioEncoderOpus: CHECK that encode call doesn't fail
WebRtcOpus_Encode will only ever fail if fed bad input, and since we don't do that, we can CHECK that it doesn't fail instead of having code that tries to handle failure.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40899004

Cr-Commit-Position: refs/heads/master@{#8469}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8469 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 09:57:18 +00:00
d324546ced Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36179004

Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 21:29:45 +00:00
722739108a Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
Includes GN changes from
https://webrtc-codereview.appspot.com/39249004/

Android changes for JNI were required due to
https://codereview.chromium.org/843103003

Other relevant changes:
* src/buildtools: 5c5e924..93b3d0a
* src/third_party/boringssl/src: d306f16..b180ee9
* src/third_party/icu: 4e3266f..2081ee6
* src/third_party/libvpx: 5cdd302..33bbffe
* src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7
* src/tools/gyp: 4d7c139..3464008
* src/tools/swarming_client: bdad118..1b7bfec
Details: b0c3ed3..2c3ffb2/DEPS

Clang version was not updated in this roll.

R=dpranke@chromium.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40079004

Cr-Commit-Position: refs/heads/master@{#8466}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 19:09:22 +00:00
829a6f4ac2 Merge ACMGenericCodec and ACMGenericCodecWrapper
ACMGenericCodecWrapper was the only remaining subclass of
ACMGenericCodec, and was the only class that was ever instantiated.
This CL merges the two, essentially keeping the function implementations
from ACMGenericCodecWrapper except where the base class's code was
invoked.

As it turns out, a lot of functions were never used, but in some cases
they were refernced in AudioCodingModuleImpl. In these cases, the
referencing code is commented out and marked FATAL(). This will be
further cleaned up in follow-up CLs.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38209004

Cr-Commit-Position: refs/heads/master@{#8463}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8463 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 16:33:49 +00:00
f3a306b5bc g722: Enhanced documentation. Added CHECK.
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43409004

Cr-Commit-Position: refs/heads/master@{#8462}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8462 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 15:41:49 +00:00
2acec4cc32 Enhanced documentation. Replaced DCHECK with CHECK.
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34309004

Cr-Commit-Position: refs/heads/master@{#8461}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8461 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 15:28:14 +00:00
962c62475e Refactoring WebRTC Java/JNI audio track in C++ and Java.
This CL is part II in a major refactoring effort. See https://webrtc-codereview.appspot.com/33969004 for part I.

- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioTrack (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Simplified the delay estimate
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup

Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).

BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39169004

Cr-Commit-Position: refs/heads/master@{#8460}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8460 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 11:54:41 +00:00
8278c072b6 Enable NACK under SendsAndReceivesH264.
Decoding with errors has a bug that triggers an assert during packet
loss. Switching to NACK since that is what we expected to be running.

BUG=4337
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43399004

Cr-Commit-Position: refs/heads/master@{#8458}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8458 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 11:12:18 +00:00
fa58745445 Delete all codec-specific subclasses of ACMGenericCodec
They have all been replaced by AudioEncoder subclasses, accessed throgh
ACMGenericCodecWrapper objects. After this change, the only subclass of
ACMGenericCodec is ACMGenericCodecWrapper. (The two will be consolidated
in a future cl.)

This CL also deletes acm_opus_unittest.cc. This test file was already
replaced audio_encoder_opus_unittest.cc	in r8244.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40729004

Cr-Commit-Position: refs/heads/master@{#8457}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8457 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 09:26:51 +00:00
2a5cfc2167 Replaced unnecessary check with an explicit CHECK.
WebRtcIlbcfix_Encode method that is called returns an error code only if a packet with more than 3 frames is passed, which is illegal.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36209004

Cr-Commit-Position: refs/heads/master@{#8456}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8456 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 08:53:15 +00:00