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webrtc/audio/test
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Philipp Hancke af512281b1 audio: make packets lost a signed integer 
as it is defined in RFC 3550. This avoids implicit casts
between signed and unsigned definitions.

BUG=webrtc:8626

Change-Id: I919b7c38ede1aa8d32f8e31b55660f540e5f5a6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279240
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38522}
2022-11-01 11:46:49 +00:00
..
unittests Migrate some scripts to python3 2022-06-01 10:09:36 +00:00
audio_bwe_integration_test.cc CallTest: migrate timeouts to TimeDelta. 2022-08-16 12:06:54 +00:00
audio_bwe_integration_test.h Propagate task queue to create test::DirectTransport by TaskQueueBase interface 2019-09-30 03:23:07 +00:00
audio_end_to_end_test.cc CallTest: migrate timeouts to TimeDelta. 2022-08-16 12:06:54 +00:00
audio_end_to_end_test.h Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
audio_stats_test.cc audio: make packets lost a signed integer 2022-11-01 11:46:49 +00:00
low_bandwidth_audio_test.cc Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
low_bandwidth_audio_test.py Allow low_bandwith_audio_test.py to pass unknown arg to the test. 2022-02-22 09:31:47 +00:00
low_bandwidth_audio_test_flags.cc Migrate WebRTC test infra to ABSL_FLAG. 2019-07-19 06:54:04 +00:00
nack_test.cc Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
non_sender_rtt_test.cc Rename AudioReceiveStream to AudioReceiveStreamInterface 2022-05-23 08:44:26 +00:00
OWNERS Reland "Use gtest_parallel with 1 worker for webrtc_perf_tests." 2021-12-07 10:08:16 +00:00
pc_low_bandwidth_audio_test.cc Add missing export to the perf output file 2022-09-27 10:53:51 +00:00