webrtc/audio
Elad Alon d8d3248d95 Reland "Delete test/constants.h"
This reverts commit 4f36b7a478.

Reason for revert: Failing tests fixed.

Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a3.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}

TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}
2019-02-19 08:51:20 +00:00
..
test Reland "Delete test/constants.h" 2019-02-19 08:51:20 +00:00
utility Receive-side ready for multiple channels. 2019-01-29 12:43:23 +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 Propagate base minimum delay to audio_receiver_stream 2019-02-06 11:07:42 +00:00
audio_receive_stream.h Propagate base minimum delay to audio_receiver_stream 2019-02-06 11:07:42 +00:00
audio_receive_stream_unittest.cc Add some missing includes and dependencies. 2019-01-18 15:30:26 +00:00
audio_send_stream.cc Reland "Always offer transport sequence number header extension for audio"" 2019-02-15 10:57:38 +00:00
audio_send_stream.h Make AudioSendStream to be OverheadObserver 2019-02-05 00:25:12 +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 Reland "Always offer transport sequence number header extension for audio"" 2019-02-15 10:57:38 +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 Reland "Always offer transport sequence number header extension for audio"" 2019-02-15 10:57:38 +00:00
channel_receive.cc Revert "Add Sender and Receiver interfaces for MediaTransport audio" 2019-02-18 09:52:40 +00:00
channel_receive.h Propagate base minimum delay to audio_receiver_stream 2019-02-06 11:07:42 +00:00
channel_send.cc Adds CallEncoder to ChannelSend. 2019-02-13 15:01:53 +00:00
channel_send.h Adds CallEncoder to ChannelSend. 2019-02-13 15:01:53 +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 Adds CallEncoder to ChannelSend. 2019-02-13 15:01:53 +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