Niels Möller 7008287219 Delete struct webrtc::PacketTime. ...

Replaced by a int64_t representing time in us.

Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}

2018-08-07 10:07:15 +00:00

..

test

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

utility

Delete root header file typedef.h.

2018-07-25 14:59:26 +00:00

audio_level.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_level.h

Delete AudioMonitor and related code.

2018-01-30 09:48:29 +00:00

audio_receive_stream_unittest.cc

Delete struct webrtc::PacketTime.

2018-08-07 10:07:15 +00:00

audio_receive_stream.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_receive_stream.h

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00

audio_send_stream_tests.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_send_stream_unittest.cc

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

audio_send_stream.cc

Better handle target audio bitrate allocation.

2018-07-13 09:06:06 +00:00

audio_send_stream.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_state_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_state.cc

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

audio_state.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_transport_impl.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

audio_transport_impl.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

BUILD.gn

Remove memcheck.

2018-08-07 07:40:08 +00:00

channel_proxy.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

channel_proxy.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

channel.cc

Delete telephone-event handling from RTPReceiverAudio.

2018-08-02 12:55:40 +00:00

channel.h

Delete telephone-event handling from RTPReceiverAudio.

2018-08-02 12:55:40 +00:00

conversion.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

DEPS

Move remaining traces of VoiceEngine

2018-01-17 13:27:47 +00:00

mock_voe_channel_proxy.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

null_audio_poller.cc

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +00:00

null_audio_poller.h

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)

2017-11-01 11:04:26 +00:00

OWNERS

…

remix_resample_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

remix_resample.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

remix_resample.h

Remove dependencies on modules:module_api from AudioProcessing.

2018-04-12 22:05:27 +00:00

time_interval_unittest.cc

Replacing rtc::TimeDelta with webrtc::TimeDelta.

2018-05-08 13:22:53 +00:00

time_interval.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

time_interval.h

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00

transport_feedback_packet_loss_tracker_unittest.cc

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

transport_feedback_packet_loss_tracker.cc

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00

transport_feedback_packet_loss_tracker.h

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00