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webrtc_m130/audio
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Niels Möller fa2b2d62d7 Delete use of RtpPayloadRegistry. 
Use in voe::Channel replaced by a std::map storing payload type frequencies.
This is a followup to
https://webrtc-review.googlesource.com/c/src/+/93820.

Bug: webrtc:7135
Change-Id: I874b706aee19fdc2d841db42a540e4f7aa2725f1
Reviewed-on: https://webrtc-review.googlesource.com/94508
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24315}
2018-08-16 14:44:25 +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
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +00:00
audio_receive_stream.cc
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +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
Makes treatment of received reports of packets lost signed.
2018-08-15 14:27:23 +00:00
audio_send_stream.cc
Delete class voe::RtcEventLogProxy
2018-08-15 09:59:15 +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
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +00:00
channel_proxy.h
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +00:00
channel.cc
Delete use of RtpPayloadRegistry.
2018-08-16 14:44:25 +00:00
channel.h
Delete use of RtpPayloadRegistry.
2018-08-16 14:44:25 +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
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
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