eladalon 760a076a52 Create unit tests for RtpDemuxer ...

1. Create unit tests for RtpDemuxer.
2. Add an RTC_DCHECK in RtpDemuxer that makes sure that the sink<->ssrc multimap does not allow multiple instances of the same association.

BUG=None

Review-Url: https://codereview.webrtc.org/2902823004
Cr-Commit-Position: refs/heads/master@{#18357}

2017-05-31 16:12:25 +00:00

..

audio_receive_stream.h

Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )

2017-04-10 14:39:05 +00:00

audio_send_stream.cc

Injectable audio encoders: WebRtcVoiceEngine and company

2017-04-27 09:08:52 +00:00

audio_send_stream.h

Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.

2017-05-02 13:46:30 +00:00

audio_state.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

bitrate_allocator_unittest.cc

Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ )

2017-02-21 15:28:31 +00:00

bitrate_allocator.cc

Renaming probing_interval to bwe_period globally.

2017-05-18 21:32:41 +00:00

bitrate_allocator.h

Renaming probing_interval to bwe_period globally.

2017-05-18 21:32:41 +00:00

bitrate_estimator_tests.cc

Making FakeNetworkPipe demux audio and video packets.

2017-04-10 23:57:57 +00:00

BUILD.gn

Create unit tests for RtpDemuxer

2017-05-31 16:12:25 +00:00

call_perf_tests.cc

Delete helper class MediaTypePacketReceiver.

2017-05-09 12:12:00 +00:00

call_unittest.cc

Store/restore RTP state for audio streams with same SSRC within a call

2017-05-23 13:07:11 +00:00

call.cc

This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.

2017-05-30 10:52:10 +00:00

call.h

This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008).

2017-05-08 18:52:38 +00:00

DEPS

Passed AudioMixer to AudioState::Config.

2016-11-17 14:48:56 +00:00

fake_rtp_transport_controller_send.h

Simple tests for Call::SetBitrateConfig.

2017-05-17 18:49:12 +00:00

flexfec_receive_stream_impl.cc

Don't allocate any RTPSender object for a receive only RtpRtcp module.

2017-03-20 10:52:39 +00:00

flexfec_receive_stream_impl.h

New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

2017-05-16 11:47:04 +00:00

flexfec_receive_stream_unittest.cc

Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.

2017-02-16 14:52:32 +00:00

flexfec_receive_stream.h

Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.

2017-01-13 15:41:19 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00

rampup_tests.cc

Making FakeNetworkPipe demux audio and video packets.

2017-04-10 23:57:57 +00:00

rampup_tests.h

Move SelectMediaType from RampUpTester to BaseTest.

2017-04-05 07:33:40 +00:00

rtp_demuxer_unittest.cc

Create unit tests for RtpDemuxer

2017-05-31 16:12:25 +00:00

rtp_demuxer.cc

Create unit tests for RtpDemuxer

2017-05-31 16:12:25 +00:00

rtp_demuxer.h

New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

2017-05-16 11:47:04 +00:00

rtp_transport_controller_send_interface.h

Move RtpTransportControllerSend to a new file.

2017-04-24 12:53:20 +00:00

rtp_transport_controller_send.cc

Address some violations of chromium-style.

2017-05-31 09:24:52 +00:00

rtp_transport_controller_send.h

Address some violations of chromium-style.

2017-05-31 09:24:52 +00:00

rtx_receive_stream_unittest.cc

New class RtxReceiveStream.

2017-05-19 13:15:19 +00:00

rtx_receive_stream.cc

Address some violations of chromium-style.

2017-05-31 09:24:52 +00:00

rtx_receive_stream.h

Address some violations of chromium-style.

2017-05-31 09:24:52 +00:00

syncable.cc

Stop using VoEVideoSync in Call/VideoReceiveStream.

2017-01-31 11:58:40 +00:00

syncable.h

Stop using VoEVideoSync in Call/VideoReceiveStream.

2017-01-31 11:58:40 +00:00