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webrtc/audio
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Niels Möller 87e2d785a0 Prepare for splitting FrameType into AudioFrameType and VideoFrameType 
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.

After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.

Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
2019-03-07 10:12:57 +00:00
..
test Don't inject worker queue into send streams. 2019-03-07 09:42:26 +00:00
utility Implicitly suppress //build/config/clang:find_bad_constructs. 2019-03-01 10:18:17 +00:00
audio_level.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_level.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_receive_stream.cc Expose relative packet arrival delay metric in stats API. 2019-03-06 16:35:16 +00:00
audio_receive_stream.h Delete a few return values from audio streams and video send streams. 2019-03-06 10:56:08 +00:00
audio_receive_stream_unittest.cc Delete a few return values from audio streams and video send streams. 2019-03-06 10:56:08 +00:00
audio_send_stream.cc Don't inject worker queue into send streams. 2019-03-07 09:42:26 +00:00
audio_send_stream.h Don't inject worker queue into send streams. 2019-03-07 09:42:26 +00:00
audio_send_stream_tests.cc Reland "Delete test/constants.h" 2019-02-19 08:51:20 +00:00
audio_send_stream_unittest.cc Don't inject worker queue into send streams. 2019-03-07 09:42:26 +00:00
audio_state.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_state.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_state_unittest.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_transport_impl.cc Receive-side ready for multiple channels. 2019-01-29 12:43:23 +00:00
audio_transport_impl.h Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
BUILD.gn Don't inject worker queue into send streams. 2019-03-07 09:42:26 +00:00
channel_receive.cc Modernize RtpRtcp factory function: use unique_ptr as return type 2019-03-06 14:38:39 +00:00
channel_receive.h Delete a few return values from audio streams and video send streams. 2019-03-06 10:56:08 +00:00
channel_send.cc Prepare for splitting FrameType into AudioFrameType and VideoFrameType 2019-03-07 10:12:57 +00:00
channel_send.h Move ownership of RTPSenderAudio to ChannelSend. 2019-03-06 17:15:00 +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 Move ownership of RTPSenderAudio to ChannelSend. 2019-03-06 17:15:00 +00:00
null_audio_poller.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
null_audio_poller.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
remix_resample.cc Receive-side ready for multiple channels. 2019-01-29 12:43:23 +00:00
remix_resample.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
remix_resample_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
transport_feedback_packet_loss_tracker.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
transport_feedback_packet_loss_tracker.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
transport_feedback_packet_loss_tracker_unittest.cc Stop using Googletest legacy APIs. 2019-01-31 13:23:33 +00:00