Commit Graph

3672 Commits

Author SHA1 Message Date
1ff218fac3 audio_processing/aec: Do not scale target delay at startup when on Android
When running AEC in extended_filter mode there is no startup phase to evaluate the reported system delay values.
Instead we simply use the first value and scale by two to avoid over compensating when synchronizing render and capture.
We don't need to be too accurate since we have extended the filter length.

On Android we use fixed (measured) reported delay values.
There is no need to be extra conservative here, because that is already built-in in the measured value.
In fact, the difference between devices is large and with such an extra conservative approach the true delay can not be caught by the filter length.
With this change we can improve performance on some devices.

BUG=4472
TESTED=offline on recordings from various devices
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49909004

Cr-Commit-Position: refs/heads/master@{#9144}
2015-05-06 10:08:50 +00:00
532531b656 audio_processing/delay_estimator: Always update robust validation statistics
The delay estimator has a robust_validation mode used to deliver more stable delay etimates. The cost is increased reaction time when we have a delay jump.
This mode can be turned on and off on the fly, but statistics are not updated while disabled. This makes the estimator unreliable if it is enabled on the fly.

This CL makes sure the update is always done.

BUG=4472
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50889004

Cr-Commit-Position: refs/heads/master@{#9143}
2015-05-06 09:58:09 +00:00
40a6d593d2 audio_processing/tests: Adds a flag to unpack input data to text file
For quick and easy aecdump verifiation storing data as text speeds up the issue tracking process, since anyone can simply view values like mic volume.

BUG=4609
TESTED=verified unpacking an aecdump with flag --txt stores that data in text files
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50849004

Cr-Commit-Position: refs/heads/master@{#9142}
2015-05-06 08:51:47 +00:00
9695d8523b Added VP9FrameBufferPool, a memory pool that is shared between libvpx and webrtc. Using the VP9 codec, the libvpx decoder will obtain its buffers from our memory pool. This lets us reuse the same buffers for our I420VideoFrames and not have to copy a frame for every decode (from libvpx buffers to webrtc/I420VideoFrame buffers).
(This is similar to chromium's MemoryPool in vpx_video_decoder.cc.)

BUG=1128
R=kjellander@webrtc.org, magjed@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48149004

Cr-Commit-Position: refs/heads/master@{#9141}
2015-05-06 08:42:22 +00:00
f242e665b4 Replace asm NEON function by intrinsics implementation on ARMv7
Passed building isac_neon and modules_unittests on Android ARMv7.
Passed modules_unittests with following filters:
  --gtest_filter=FiltersTest*
  --gtest_filter=LpcMaskingModelTest*
  --gtest_filter=TransformTest*
  --gtest_filter=FilterBanksTest*

WebRtcIsacfix_CalculateResidualEnergyNeon is removed, refer more in
Issue 4224.

The old review url is at: https://webrtc-codereview.appspot.com/37259004/

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48319005

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

Change-Id: I4c16e15930f1b3449d67b67bf023fac28121dff8
Cr-Commit-Position: refs/heads/master@{#9140}
2015-05-06 08:39:37 +00:00
589699eea2 Fix bug in transform_neon.c in iSAC codec.
The bug causes AcmReceiverBitExactness and AcmSenderBitExactness test
failed in modules_unittests.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Change-Id: I18b00056c53cf4158c186d449e5ab785065cca94

Review URL: https://webrtc-codereview.appspot.com/49889004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

Cr-Commit-Position: refs/heads/master@{#9138}
2015-05-06 02:25:20 +00:00
ab00404571 VCMEncodedFrame::VerifyAndAllocate: Use size_t instead of uint32_t for size argument
BUG=484432
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53379004

Cr-Commit-Position: refs/heads/master@{#9135}
2015-05-05 09:37:17 +00:00
01b488831b Use padding to achieve bitrate probing if the initial key frame has too few packets.
BUG=4350
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44879004

Cr-Commit-Position: refs/heads/master@{#9134}
2015-05-05 08:21:32 +00:00
c56ac1ec29 rtc::Buffer: Remove backwards compatibility band-aids
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:

  1. Change default return type of rtc::Buffer::data() from char* to
     uint8_t*. uint8_t is a more natural type for bytes, and won't
     accidentally convert to a string. (Chromium previously expected
     the default return type to be char, which is why
     rtc::Buffer::data() initially got char as default return type in
     9478437f, but that's been fixed now.)

  2. Stop accepting void* inputs in constructors and methods. While
     this is convenient, it's also dangerous since any pointer type
     will implicitly convert to void*.

(This was previously committed (9e1a6d7c) but had to be reverted
(cbf09274) because Chromium on Android wasn't quite ready for it).

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47109004

Cr-Commit-Position: refs/heads/master@{#9132}
2015-05-04 12:54:56 +00:00
f75f0cf36a Enable GoogleWifiTrace3Mbps simulations.
BUG=3277
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50829004

Cr-Commit-Position: refs/heads/master@{#9131}
2015-05-04 12:26:26 +00:00
cbf0927473 Revert "rtc::Buffer: Remove backwards compatibility band-aids"
This reverts commit 9e1a6d7c236c9a8a322bef54d4ec2a087e5baa07, because
Chromium for Android still isn't happy with it.

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49869004

Cr-Commit-Position: refs/heads/master@{#9122}
2015-04-30 14:01:01 +00:00
9e1a6d7c23 rtc::Buffer: Remove backwards compatibility band-aids
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:

  1. Change default return type of rtc::Buffer::data() from char* to
     uint8_t*. uint8_t is a more natural type for bytes, and won't
     accidentally convert to a string. (Chromium previously expected
     the default return type to be char, which is why
     rtc::Buffer::data() initially got char as default return type in
     9478437f, but that's been fixed now.)

  2. Stop accepting void* inputs in constructors and methods. While
     this is convenient, it's also dangerous since any pointer type
     will implicitly convert to void*.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44269004

Cr-Commit-Position: refs/heads/master@{#9121}
2015-04-30 12:25:06 +00:00
ff019b0b55 Move rtc::AtomicOps to webrtc/base/atomicops.h.
Removes FixedSizeLockFreeQueue which isn't used anymore. This enabled
moving rtc::AtomicOps to webrtc/base/atomicops.h where they should be.

BUG=4330
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51789004

Cr-Commit-Position: refs/heads/master@{#9120}
2015-04-30 12:16:14 +00:00
3cfa756f37 audio_processing/aec: Fixes an incorrect sampling rate multiplier when processing in 48 kHz
In AEC a fixed fft size is used, but processing can in the lower band be in either 8 or 16 kHz.
Therefore we need a multiplier/rate factor to, for example, map frequency bands in Hz to frequency bins.

The multiplier/rate factor can only be either 1 or 2, but when 48 kHz support was added it was assigned 3.

BUG=crbug.com/482424
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43329004

Cr-Commit-Position: refs/heads/master@{#9117}
2015-04-29 18:22:50 +00:00
adf89b7e33 Added SetBitRate function to VoE API to allow changing the audio bitrate.
If the requested bitrate is not valid for the codec, the codec will decide on
an appropriate value.
Updated VoE command line tool to use new SetBitRate function.
Includes unittests for SetBitRate function.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kwiberg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50789004

Cr-Commit-Position: refs/heads/master@{#9115}
2015-04-29 14:03:45 +00:00
dea11f9c43 Add per flow throughput and delay metrics.
BUG=4548
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48299004

Cr-Commit-Position: refs/heads/master@{#9112}
2015-04-29 12:27:38 +00:00
97f13c5f7f Fixed incorrect RBSP parsing. The original version would eat 0x3 as an emulation byte in places where it shouldn't, whereas the real parsing is only supposed to eat 0x3 preceded by 0x0 0x0.
Also, now that BitBuffer is getting a writer (https://webrtc-codereview.appspot.com/45259005/), I wrote a function that creates a fake SPS of a given resolution. The created SPS has an emulation 0x3 and a real 0x3, so it ensures the parser has the correct behavior.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44349004

Cr-Commit-Position: refs/heads/master@{#9108}
2015-04-29 00:55:43 +00:00
31dc737d7a Platform dependent way of generating the seed for srand for simulations, so that they can be run in parallel.
The seed generated for Win won't be good enough to run the simulations in parallel.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49829004

Cr-Commit-Position: refs/heads/master@{#9101}
2015-04-28 13:43:44 +00:00
88de4792d0 AudioEncoderIsac: Print error code if CHECK for successful encoding fails
This will hopefully make the crash in bug 4577 easier to understand if
it happens again.

BUG=4577
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52389004

Cr-Commit-Position: refs/heads/master@{#9100}
2015-04-28 13:43:43 +00:00
bcbcd84888 Improve TCP implementation by adding ssthresh and make it possible to start it with an offset.
Add a propagation delay to tests and make the run-time configurable for the fairness tests.

Handle losses in-between feedback messages.

BUG=4549
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49819004

Cr-Commit-Position: refs/heads/master@{#9099}
2015-04-28 12:38:31 +00:00
beb9798ab4 audio_processing: Fixed incorrect usage of SetExtraOptions() in offline tool
The way SetExtraOptions() is used today only applies for any one configuration change. The correct way is to set it after all flags have been scanned.

The prefered way to solve this is to use gflags and scan once, followed by applying the configuration when creating audio_processing. This is what is done in the new test tool audioproc_float.cc, but there are still some things left to do before we can replace this one.

BUG=N/A
TESTED=locally
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45279004

Cr-Commit-Position: refs/heads/master@{#9097}
2015-04-28 11:52:30 +00:00
143cec1cc6 Set correct encoder-specific settings for vpx in the new API.
Also, make VideoEncoderConfig::ContentType an enum class.

BUG=4569
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46069004

Cr-Commit-Position: refs/heads/master@{#9093}
2015-04-28 08:01:14 +00:00
e8a197bd07 Enable isac NEON building on Aarch64
Passed building isac_neon and modules_unittests on Android ARM64 and
ARMv7.
Passed modules_unittests with following filters:
--gtest_filter=FiltersTest*
--gtest_filter=LpcMaskingModelTest*
--gtest_filter=TransformTest*
--gtest_filter=FilterBanksTest*

WebRtcIsacfix_CalculateResidualEnergyNeon is not enabled due to Issue
4224.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44229004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

Cr-Commit-Position: refs/heads/master@{#9092}
2015-04-28 06:42:04 +00:00
5a92aa8440 Add 3-band filter-bank implementation
The implementation is a FIR filter bank with DCT modulation, similar to the proposed in "Multirate Signal Processing for Communication Systems" by Fredric J Harris.
The lowpass filter prototype has these characteristics:
* Passband ripple = 0.3dB
* Passband frequency = 0.147 (7kHz at 48kHz)
* Stopband attenuation = 40dB
* Stopband frequency = 0.192 (9.2kHz at 48kHz)
* Delay = 24 samples (500us at 48kHz)
* Linear phase

This filter bank does not satisfy perfect reconstruction. The SNR after analysis and synthesis (with no processing in between) is approximately 9.5dB depending on the input signal after compensating for the delay.

The performance on my workstation of AudioProcessing (with AGC and NS enabled) on a 413s recording compared to previous versions is as follows:
* Input signal has 32kHz sample rate: 3.01s
* Resampling 48kHz to 32kHz: 3.56s
* Today's temporary filter bank: 5.67s
* This filter-bank: 4.62s

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

Review URL: https://webrtc-codereview.appspot.com/48999005

Cr-Commit-Position: refs/heads/master@{#9090}
2015-04-27 18:34:16 +00:00
e6cefb60f8 GYP variables for building expat, icu, libsrtp, usrsctp
This makes the build more flexible when linking against
prebuilt external libraries.

Use existing build_* variables for libyuv and json in talk/
(already in use in webrtc/).

Also make it possible to avoid building the GTK parts of the Linux build.

BUG=4242
R=andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44179005

Cr-Commit-Position: refs/heads/master@{#9087}
2015-04-27 12:38:37 +00:00
61be2a4016 Clean up RTCPSender.
Reformat to current code style, remove non-const references, use
scoped_ptr, remove empty comments and dead code, etc..

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49019004

Cr-Commit-Position: refs/heads/master@{#9086}
2015-04-27 11:32:31 +00:00
12e0329007 Do not use Magnifier if there are multiple screens since it sometimes crashes.
BUG=crbug/478825
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/43289004

Cr-Commit-Position: refs/heads/master@{#9082}
2015-04-24 15:46:15 +00:00
f955b5d3f5 Add h.264 AVC SPS parsing for resolution (re-land)
Re-land of noharic@'s CL at  https://webrtc-codereview.appspot.com/48129004
which was reverted due to a Mac compile error which most
likely was a Goma flake (it passed on all trybots).

TBR=stefan@webrtc.org, noharic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44329005

Cr-Commit-Position: refs/heads/master@{#9079}
2015-04-24 11:56:44 +00:00
c043afc605 Cleanup inside IncomingVideoStream.
Removes logging, thread annotates members (finding 2 bugs where
variables were not protected by the correct critsect) and adds consts
and scoped_ptrs.

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

Review URL: https://webrtc-codereview.appspot.com/43219004

Cr-Commit-Position: refs/heads/master@{#9078}
2015-04-24 11:54:05 +00:00
a96f02b6f3 Make sure histograms in jitter buffer are only updated if running.
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46089004

Cr-Commit-Position: refs/heads/master@{#9076}
2015-04-24 06:51:52 +00:00
e3827f27c3 Revert "Add h.264 AVC SPS parsing for resolution."
The Mac64 Debug builder is broken for an unknown failure (trybot is
green, no failure obvious in the commit break). Reverting this CL to see
if it goes green again, and then relanding to see if it is just some
weird flaky build issue.

This reverts commit 5ea8eff55ec21a1d81aaf7d29c0106fe13256150.

BUG=
TBR=rollback

Review URL: https://webrtc-codereview.appspot.com/47019004

Cr-Commit-Position: refs/heads/master@{#9074}
2015-04-24 01:14:56 +00:00
5ea8eff55e Add h.264 AVC SPS parsing for resolution.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48129004

Cr-Commit-Position: refs/heads/master@{#9073}
2015-04-23 23:48:42 +00:00
9728241e6a Record H264 NALU type in the h264 header.
BUG=
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48999004

Cr-Commit-Position: refs/heads/master@{#9072}
2015-04-23 18:14:46 +00:00
fe7a80c38c Prevent sender RTCP signals for receive-only channels.
Since RTCP packets are delivered to both senders and receivers that
correspond the receivers currently log that NACKed packets are missing,
since they have no direct connection to the sending side or the RTP
packet history. Also preventing triggering on SR requests and PLI/FIR.

BUG=
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45249004

Cr-Commit-Position: refs/heads/master@{#9071}
2015-04-23 15:52:58 +00:00
d3e8eda839 (Re-land) AudioEncoderDecoderIsac: Merge the two config structs
This reverts commit 599beb86, which in turn reverted 7c324cac. What
makes it work this time is that we don't remove the option of setting
bit_rate to 0 in order to ask for the default value.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4228, chromium:478161
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48199004

Cr-Commit-Position: refs/heads/master@{#9068}
2015-04-23 12:06:46 +00:00
92f9eacd13 g722 and red encoders: Use rtc::Buffer instead of scoped_ptr<uint8_t[]>
It's a win for red, and a toss-up for g722 since it never resizes its
buffer.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45219005

Cr-Commit-Position: refs/heads/master@{#9067}
2015-04-23 11:53:02 +00:00
352595459d Use short include paths for icu headers.
This makes it possible to build with icu located
in another absolute path.

BUG=4242
R=andresp@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46079004

Cr-Commit-Position: refs/heads/master@{#9063}
2015-04-23 06:58:02 +00:00
5a3178042b Reformatting RTPtimeshift.cc file.
BUG=2692
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45239004

Cr-Commit-Position: refs/heads/master@{#9052}
2015-04-22 11:11:39 +00:00
ac69016b0f Improve TCP by adding a real timeout to in flight packets.
Note that the timeout should depend on the smoothed RTT, but for now is hard coded to 1000 ms.

This solves issues where a full cwnd gets lost.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51739004

Cr-Commit-Position: refs/heads/master@{#9051}
2015-04-22 11:11:28 +00:00
fb49451014 Disables mic bump-up level if not built with chromium
In http://chromegw.corp.google.com/viewvc/chrome-internal?view=rev&revision=61016 a feature to bump up low input audio levels to a fixed value of 33%. In https://webrtc-codereview.appspot.com/43109004/ a configuration to choose an arbitrary level was added, but still using 33% as default.
The original bump-up feature was added to fix audio issues in chrome, but affected also non-chrome users. This CL disables the feature for non-chrome applications.

Note that the default value is set to 0, but any value up to 12 will do. Zero was selected because it is more clear that the feature is turned off.

BUG=4529
R=andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43259004

Cr-Commit-Position: refs/heads/master@{#9048}
2015-04-22 04:39:47 +00:00
8f85dbcce4 Reduce the number of registers used in MIPS optimizations.
This change is needed by ChromeOS as it introduces -fno-omit-frame-pointer
flag (see code.google.com/p/chromium/issues/detail?id=477749). This causes
compile error for MIPS, as some MIPS optimization blocks use maximum possible
number of available registers.
Also, this change contains minor GN build fix for MIPS platform regarding the
pitch_filter_mips.c / pitch_filter_c.c file inclusion.

BUG=477749
R=andrew@webrtc.org, djordje.pesut@imgtec.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48139004

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

Cr-Commit-Position: refs/heads/master@{#9047}
2015-04-21 23:52:26 +00:00
61b4d518af Dynamic resolution change for VP8 HW encode.
Off by default for now.

BUG=
R=glaznev@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45849004

Cr-Commit-Position: refs/heads/master@{#9045}
2015-04-21 22:29:53 +00:00
5464a6e548 Remove VideoCodingModule::InitializeReceiver.
This code is no longer used to reset, so we can just initialize the
object in the constructor.

BUG=4391
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43249004

Cr-Commit-Position: refs/heads/master@{#9044}
2015-04-21 14:35:34 +00:00
9dbbcfbcb5 Remove VideoCodingModule::InitializeSender.
This code is no longer used to reset, so we can just initialize the
object in the constructor.

BUG=4391
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51619005

Cr-Commit-Position: refs/heads/master@{#9043}
2015-04-21 13:54:56 +00:00
95702246d7 Fix broken perf prints.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51709004

Cr-Commit-Position: refs/heads/master@{#9042}
2015-04-21 13:44:01 +00:00
5f92051f06 Fix bug in TCP implementation (simulations).
The problem was that only ACKed packets were subtracted from in_flight_, but lost packets were never removed, which caused TCP to stop sending eventually.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43239004

Cr-Commit-Position: refs/heads/master@{#9041}
2015-04-21 12:48:07 +00:00
e62202fedf Support handling multiple RTX but only generate SDP with RTX associated with VP8.
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.

Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.

BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36889004

Cr-Commit-Position: refs/heads/master@{#9040}
2015-04-21 12:25:42 +00:00
9478437fde rtc::Buffer improvements
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
     int8_t*, and char*. Previously, they accepted void*, meaning that
     any kind of pointer was accepted. I think requiring an explicit
     cast in cases where the input array isn't already of a byte-sized
     type is a better compromise between convenience and safety.

  2. data() can now return a uint8_t* instead of a char*, which seems
     more appropriate for a byte array, and is harder to mix up with
     zero-terminated C strings. data<int8_t>() is also available so
     that callers that want that type instead won't have to cast, as
     is data<char>() (which remains the default until all existing
     callers have been fixed).

  3. Constructors, SetData(), and AppendData() now accept arrays
     natively, not just decayed to pointers. The advantage of this is
     that callers don't have to pass the size separately.

  4. There are new constructors that allow setting size and capacity
     without initializing the array. Previously, this had to be done
     separately after construction.

  5. Instead of TransferTo(), Buffer now supports swap(), and move
     construction and assignment, and has a Pass() method that works
     just like std::move(). (The Pass method is modeled after
     scoped_ptr::Pass().)

R=jmarusic@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42989004

Cr-Commit-Position: refs/heads/master@{#9033}
2015-04-20 12:03:00 +00:00
599beb8687 Revert "AudioEncoderDecoderIsac: Merge the two config structs"
Reason for revert - breaks Hangouts

This reverts commit 7c324cac50ac38122b3f3b26455bc55ad834bfc0.

BUG=chromium:478161

Review URL: https://webrtc-codereview.appspot.com/43209004

Cr-Commit-Position: refs/heads/master@{#9030}
2015-04-17 21:13:59 +00:00
a51e8f490c Fix some simulation issues.
Don't default to an infinite queue.
Make sure the computation of missing packets is correct.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49059004

Cr-Commit-Position: refs/heads/master@{#9028}
2015-04-17 13:48:58 +00:00