8353 Commits

Author SHA1 Message Date
Karl Wiberg
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
Peter Boström
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
Peter Boström
f16fcbec73 Remove ViECapture usage in VideoSendStream.
Instead a ViECapturer object is allocated and directly operated on. This
additionally exposes ViESharedData to Call to access the module
ProcessThread, moving towards Call ownership of shared resources.

BUG=1695
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45339004

Cr-Commit-Position: refs/heads/master@{#9119}
2015-04-30 10:16:11 +00:00
Jelena Marusic
46bd31b994 VoE: VoENetwork unit test
Changes:
1. Added MockTransport.
2. Added VoiceEngineFixture class to be used in VoE unit tests. This class is in no way final as it is - it will grow as we add more unit tests and our understanding of the Voice Engine grows.
3. Modified VoEBase unit test to use VoiceEngineFixture.
4. Added VoENetwork unit test.

R=henrika@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49759004

Cr-Commit-Position: refs/heads/master@{#9118}
2015-04-30 08:57:17 +00:00
Bjorn Volcker
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
Erik Språng
efbde3775b Don't use CPU adaptation for screen content in the new API.
BUG=4605
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48309004

Cr-Commit-Position: refs/heads/master@{#9116}
2015-04-29 14:21:32 +00:00
Ivo Creusen
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
Fredrik Solenberg
23fba1ffa0 Add AudioReceiveStream to Call API.
BUG=4574
R=kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51749004

Cr-Commit-Position: refs/heads/master@{#9114}
2015-04-29 13:24:10 +00:00
Henrik Kjellander
10ba3eec5a Roll chromium_revision a12e1e1..0cb2549 (326495:327252)
https://codereview.chromium.org/1051343002 adds a dependency
on Chromium's third_party/junit into base/ which affects our
Android tests that uses that code.

The precompiled JUnit 4.11 JAR file that is only by the
libjingle_peerconnection_java_unittest target on Linux has been
moved to third_party/junit-jar, since it collided with the expected
path for the JUnit dependency mentioned above.
It had to be kept since the Chromium JUnit is only possible to build
when OS==android.

This CL also brings in Mockito and Robolectric, which should be
useful for our Android tests.

Other relevant changes:
* src/buildtools: 3b302fe..15308f4
* src/third_party/libjpeg_turbo: 034e9a9..9e9058b
* src/third_party/libyuv: 32ad6e0..01db3d1
Details: a12e1e1..0cb2549/DEPS

Clang version was not updated in this roll.

BUG=4499
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48239004

Cr-Commit-Position: refs/heads/master@{#9113}
2015-04-29 12:48:00 +00:00
Stefan Holmer
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
Peter Boström
94cc1fe4af Remove ViEImageProcess usage in VideoSendStream.
Replaces interface usage with direct calls on ViEEncoder removing a
layer of indirection. Also removing some methods from ViEImageProcess
that were only added for Video{Send,Receive}Stream usage.

BUG=1695
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45319004

Cr-Commit-Position: refs/heads/master@{#9111}
2015-04-29 12:08:49 +00:00
Henrik Kjellander
c444de6276 Make setup_links.py handle non-link directories during cleanup
The trybots ended up in a state that could not be cleaned up
when other tryjobs were altering the checkout with different
variations of dependencies.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46139004

Cr-Commit-Position: refs/heads/master@{#9110}
2015-04-29 09:27:29 +00:00
Bjorn Volcker
1ba344a070 Adds a MediaConstraint for the AudioOption aec_dump
Alson includes
- a test verifying that the option is set
- changed the test verifying delay_agnostic_aec option is set to use non-default value

BUG=4555
TESTED=locally through AppRTCDemo on N7 and Android One
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46059004

Cr-Commit-Position: refs/heads/master@{#9109}
2015-04-29 05:28:22 +00:00
Noah Richards
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
Noah Richards
86153c26a0 Added a BitBufferWriter subclass that contains methods for writing bit and byte-sized data, along with exponential golomb encoded data.
This pattern (read-only base, writable subclass) was picked to maintain a *Buffer option that doesn't copy the source bits when parsing. ByteBuffer and Buffer both copy. I'm open to discussion on what the type relationship would be, though :)

Tests have been added to ensure the symmetric nature of read/write operations.

BUG=
R=bcornell@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45259005

Cr-Commit-Position: refs/heads/master@{#9107}
2015-04-28 22:13:50 +00:00
Wei Zhong
80154f6b28 Set correct .type directive for asm functions.
This is required per binutil manual.
https://sourceware.org/binutils/docs/as/Type.html#Type

Otherwise, runtime SEGV_MAPERR if linked by GNU bfd linker.

BUG=b/20218337
R=wzhong@chromium.org

Review URL: https://webrtc-codereview.appspot.com/47059004

Patch from Wei Zhong <wzhong@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#9106}
2015-04-28 20:51:57 +00:00
Alex Glaznev
faa6d076b7 Remove a few verbose log messages from webrtcvideoengine2.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49189004

Cr-Commit-Position: refs/heads/master@{#9105}
2015-04-28 16:40:51 +00:00
Peter Thatcher
019087f5bb Add safeguards against signalling peer-reflexive candidates.
BUG=4208
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/50799004

Cr-Commit-Position: refs/heads/master@{#9104}
2015-04-28 16:06:34 +00:00
Henrik Kjellander
ae331349c6 Always specify current OS when syncing Chromium.
Recently, we've hit a rare case of a dependency being
present only below 'unix' and not in the 'android' section
of the DEPS file in Chromium (src/third_party/junit/src).

This exposed a bug in our sync_chromium.py script, which is that
when the --deps flag is passed to gclient sync, _only_ that platform
is processed. That means our Android builders synced with --deps=android
and thus only getting the Android deps, not the 'unix' ones (Linux).
That behavior is different from a regular gclient sync, where both
the current platform's DEPS and the one specified by appending a target_os
variable into .gclient is used.

Ensuring that we pass both the current platform and the optionally specified
DEPS platform into our sync_chromium.py sync, gives us the same behavior
as a regular sync for the special Chromium sync we use.

TESTED=
First:
rm chromium/.last_sync_chromium
Then I ran without this patch and with target_os = ["android"] in .gclient. Verified
that --deps=android was passed.
With this patch, I repeated the above and verified --deps=unix,android was passed.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52379004

Cr-Commit-Position: refs/heads/master@{#9103}
2015-04-28 14:08:29 +00:00
Peter Boström
8786f637b2 Roll gtest-parallel.
Rolls to 48e584a52bb9db1d1c915ea33463e9e4e1b36d1b which includes fix for
running TEST_P functions.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50819004

Cr-Commit-Position: refs/heads/master@{#9102}
2015-04-28 13:43:45 +00:00
Stefan Holmer
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
Karl Wiberg
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
Stefan Holmer
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
Erik Språng
9d657cfd66 Fix dangling pointer in screenshare_loopback
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52369004

Cr-Commit-Position: refs/heads/master@{#9098}
2015-04-28 12:00:29 +00:00
Bjorn Volcker
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
Peter Boström
ddbddbdee6 Remove ViENetwork usage in VideoSendStream.
Replaces interface usage with direct calls on ViEEncoder and ViEChannel
removing a layer of indirection.

BUG=1695
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50809004

Cr-Commit-Position: refs/heads/master@{#9096}
2015-04-28 10:35:15 +00:00
Peter Boström
038df3c5d7 Remove ViEExternalCodec usage in VideoSendStream.
Replaces interface usage with direct calls on ViEEncoder removing a
layer of indirection.

BUG=1695
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49179004

Cr-Commit-Position: refs/heads/master@{#9095}
2015-04-28 09:58:18 +00:00
Peter Boström
4a9cb6b67d Prevent zero-timestamps in captured_frame_.
ViECapturer::OnIncomingCapturedFrame can abort if two frames without a
timestamp are delivered at the same time (the same NTP timestamp is
generated). This currently causes captured_frame_ to be in a state
where timestamps (NTP time etc.) are set, but RTP timestamp is not. This
is an issue if the previous frame in captured_frame_ has yet to be
picked up by the capturer thread, as it will pick up captured_frame_ in
a state where the RTP timestamp has not been set.

BUG=chromium:480953
R=magjed@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45299004

Cr-Commit-Position: refs/heads/master@{#9094}
2015-04-28 08:48:13 +00:00
Erik Språng
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
Zhongwei Yao
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
Jiayang Liu
d7e5c44e94 STUN allocation should not be disabled when using shared port and TURN servers are provided.
BUG=
R=juberti@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48229004

Cr-Commit-Position: refs/heads/master@{#9091}
2015-04-27 18:46:58 +00:00
Alejandro Luebs
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
Tommi
494f20977e Move CriticalSection into rtc_base_approved.
This class is being used from both libjingle and webrtc but we recently had a regression when we added dependency on libjingle's Thread class. This cleans that up and moves the implementation of CriticalSection and helper classes into the source file.

I'm also improving debugging facilities and constness.

BUG=
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51769004

Cr-Commit-Position: refs/heads/master@{#9089}
2015-04-27 15:39:00 +00:00
Peter Boström
59d91dc951 Remove ViERTP_RTCP usage in VideoSendStream.
Replaces interface usage with direct calls on ChannelGroup, ViEEncoder
and ViEChannel removing a layer of indirection.

BUG=1695
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43319004

Cr-Commit-Position: refs/heads/master@{#9088}
2015-04-27 15:24:12 +00:00
Henrik Kjellander
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
Erik Språng
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
Åsa Persson
3c391cbabb Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api.
Add tests for verifying that video histograms are updated.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44309004

Cr-Commit-Position: refs/heads/master@{#9085}
2015-04-27 08:09:27 +00:00
Peter Boström
52ef9d7738 Stop IncomingVideoStream on delete.
Fixes race between VideoReceiveStream destruction and pending
IncomingVideoStream frames.

BUG=
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45259004

Cr-Commit-Position: refs/heads/master@{#9084}
2015-04-24 16:06:33 +00:00
Andrew MacDonald
23dc68e515 Add the rtc_build_openmax_dl variable to the GN build.
For symmetry with the gyp build.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49109005

Cr-Commit-Position: refs/heads/master@{#9083}
2015-04-24 15:46:31 +00:00
Jiayang Liu
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
Tommi
77d444a433 Handle the case when hoststring is empty.
BUG=chromium:480536
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46109004

Cr-Commit-Position: refs/heads/master@{#9081}
2015-04-24 13:38:17 +00:00
Peter Boström
c4188fd3c7 Use IncomingVideoStream in VideoReceiveStream.
Decouples VideoReceiveStream further from webrtc/video_engine/ as well
as most of webrtc/modules/video_render/ resulting in a simpler setup.

BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50749004

Cr-Commit-Position: refs/heads/master@{#9080}
2015-04-24 13:15:40 +00:00
Henrik Kjellander
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
Peter Boström
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
Henrik Kjellander
a9ae0dfe12 Roll chromium_revision d5098d0..a12e1e1 (326014:326495)
Relevant changes:
* src/third_party/libvpx: 0816cf2..c600ca7
* src/tools/swarming_client: 13e7c88..f222001
Details: d5098d0..a12e1e1/DEPS

Clang version was not updated in this roll.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49149004

Cr-Commit-Position: refs/heads/master@{#9077}
2015-04-24 07:27:38 +00:00
Åsa Persson
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
Bjorn Volcker
affcfb2f16 Refactor common_audio/signal_processing: Removed usage of trivial macro WEBRTC_SPL_MUL_16_16
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)

Also includes
- style changes
- replaced pointer operations with direct element access

BUG=3348,3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48949004

Cr-Commit-Position: refs/heads/master@{#9075}
2015-04-24 06:11:50 +00:00
Noah Richards
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
Noah Richards
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
Noah Richards
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