Separate the null implementation from rtp_rtcp_defines.h, and follow chromium style guide for virtual functions.
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2400993002
Cr-Commit-Position: refs/heads/master@{#14738}
The renaming is to reflect this class is only used for RTCP interaction
and not for other transports.
This Cl will be followed by multiple CLs moving all send-side RTP
functionality to a separate class, rtp module ownership away from
VideoSendStream and use TaskQueue instead of ProcessThread for RTP.
BUG=webrtc:6456
Review-Url: https://codereview.webrtc.org/2390463002
Cr-Commit-Position: refs/heads/master@{#14556}
I'll be doing some changes to code it tests (rtp_receiver_audio,
specifically) and want to make sure there are tests in place before I
touch anything.
Fixed test_api_audio not properly checking payload data. Required a
fix to LoopBackTransport in test_api to as to act like the regular
audio and video parts of WebRTC and separate payload from header data.
Also added a test for CNG and cleaned up constants.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2378403004
Cr-Commit-Position: refs/heads/master@{#14529}
- Split out reading/writing of FEC headers to classes separate
from ForwardErrorCorrection. This makes ForwardErrorCorrection
oblivious to what FEC header scheme is used, and lets it focus on
encoding/decoding the FEC payloads.
- Add unit tests for FEC header readers/writers.
- Split ForwardErrorCorrection::XorPackets into XorHeaders and
XorPayloads and reuse these functions for both encoding and
decoding.
- Rename AttemptRecover -> AttemptRecovery in ForwardErrorCorrection.
BUG=webrtc:5654
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2260803002 .
Cr-Commit-Position: refs/heads/master@{#14316}
- Rename GenerateFec -> EncodeFec in ForwardErrorCorrection. This naming
is more consistent with DecodeFec.
- Add appropriate using directives, to reduce clutter in tests.
- Move ConstructMediaPackets to fec_test_helper.{h,cc}. This will help
future tests of ULPFEC/FlexFEC header formatters.
- Generalize tests in rtp_fec_unittest.cc to typed tests. This will help
testing ForwardErrorCorrection with both ULPFEC and FlexFEC.
This CL should not impact functionality or performance.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2267393002
Cr-Commit-Position: refs/heads/master@{#14314}
- Rename GetNumberOfFecPackets -> NumFecPackets and
PacketOverhead -> MaxPacketOverhead in ForwardErrorCorrection.
- Rename FECPacketOverhead -> FecPacketOverhead in ProducerFec.
- Move ownership of ForwardErrorCorrection from RTPSenderVideo
to ProducerFec.
- Make MaxPacketOverhead a member function of ForwardErrorCorrection.
This will allow for changing it, based on FEC header types, later on.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2275443002
Cr-Commit-Position: refs/heads/master@{#14194}
The last in-tree call site recently disappeared, so they were unused.
BUG=webrtc:5922
Review-Url: https://codereview.webrtc.org/2066473002
Cr-Commit-Position: refs/heads/master@{#13751}
This CL converts the ForwardErrorCorrection class to use std::unique_ptr
for memory management, rather than manually delete'ing allocated memory.
It further renames some data structures and types to distinguish between
generated FEC packets (i.e. coming from the encode operation) and
received FEC packets (i.e. coming in over the wire, intended for the
decode operation).
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2099243003
Cr-Commit-Position: refs/heads/master@{#13687}
This CL mainly updates the FEC code to use more C++11 features and
to be more in line with the style guide. These changes should
have no impact on the functionality provided by the FEC.
Summary of style fixes:
- Use range-based for loops, where applicable.
- Use auto type deduction for iterator type names.
- Use RTC_DCHECK instead of assert.
- Rename FEC to Fec, where applicable.
- Update test_fec.cc to use variable_names rather than variableNames.
- Avoid redefining the PacketList types outside ForwardErrorCorrection.
Another minor change is that storage for the packet masks, as these
are generated, now is provided by a member variable, rather than
being dynamically allocated on every call to GenerateFec.
BUG=webrtc:5654
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2080553003 .
Cr-Commit-Position: refs/heads/master@{#13403}
Reason for revert:
Breaks user code. Said code needs to stop using scoped_ptr!
Original issue's description:
> Remove webrtc/base/scoped_ptr.h
>
> BUG=webrtc:5520
>
> NOTRY=True
>
> Committed: https://crrev.com/65fc62e9dd8a8716db625aaef76ab92f542ecc5a
> Cr-Commit-Position: refs/heads/master@{#12684}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520
Review-Url: https://codereview.webrtc.org/1965063003
Cr-Commit-Position: refs/heads/master@{#12686}
Reason for revert:
Revert because it breaks downstream code.
Original issue's description:
> Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
> Cr-Commit-Position: refs/heads/master@{#12015}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1812453002
Cr-Commit-Position: refs/heads/master@{#12016}
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1782053002
Cr-Commit-Position: refs/heads/master@{#11953}
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1722253002
Cr-Commit-Position: refs/heads/master@{#11927}
With this change the following tests have been successfully
passing in the iOS Simulator for iPhone 5 and iOS 9:
* audio_decoder_unittests
* common_video_unittests
* modules_tests
* rtc_api_objc_tests
* rtc_pc_unittests
* system_wrappers_unittests
* voice_engine_unittests
The modules_unittests and common_audio_unittests are
handled in https://codereview.webrtc.org/1698033002/
BUG=webrtc:4755
NOTRY=True
Review URL: https://codereview.webrtc.org/1694353003
Cr-Commit-Position: refs/heads/master@{#11646}
rand() usage replaced with new Random class, which also makes it clearer what interval random number is in.
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1519503002
Cr-Commit-Position: refs/heads/master@{#11019}
except rand() function that is subject of CL#1519503002
and namespace that is fixed in CL#1506823002
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1511413005
Cr-Commit-Position: refs/heads/master@{#11012}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
