Alex Narest 2d06e366e8 Adds fixed PL loss mode to neteq_quality_test. ...

It will be available in all inheriting tests.
The mode allows setting start time and duration for every loss event.

Bug: webrtc:8877
Change-Id: Ife36db6d431387083ac22406a0814e02117100bc
Reviewed-on: https://webrtc-review.googlesource.com/51822
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22005}

2018-02-13 15:34:04 +00:00

..

audio_checksum.h

Use generic MessageDigest class instead of MD5 / SHA-1 specific classes.

2017-12-21 12:39:50 +00:00

audio_loop.cc

Fixing WebRTC after moving from src/webrtc to src/

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

audio_loop.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

audio_sink.cc

Fixing WebRTC after moving from src/webrtc to src/

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

audio_sink.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

constant_pcm_packet_source.cc

Fixing WebRTC after moving from src/webrtc to src/

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

constant_pcm_packet_source.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

DEPS

Fixing WebRTC after moving from src/webrtc to src/

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

encode_neteq_input.cc

Move some numeric utility code from rtc_base/ to rtc_base/numerics/

2017-11-22 11:21:47 +00:00

encode_neteq_input.h

Fixing WebRTC after moving from src/webrtc to src/

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

fake_decode_from_file.cc

Move some numeric utility code from rtc_base/ to rtc_base/numerics/

2017-11-22 11:21:47 +00:00

fake_decode_from_file.h

Fixing WebRTC after moving from src/webrtc to src/

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

input_audio_file_unittest.cc

Move some numeric utility code from rtc_base/ to rtc_base/numerics/

2017-11-22 11:21:47 +00:00

input_audio_file.cc

Replacing the legacy tool RTPencode with a new rtp_encode

2017-11-24 09:05:48 +00:00

input_audio_file.h

Replacing the legacy tool RTPencode with a new rtp_encode

2017-11-24 09:05:48 +00:00

neteq_delay_analyzer.cc

Add code to generate python visualization to neteq_rtpplay

2017-12-06 10:52:42 +00:00

neteq_delay_analyzer.h

Add code to generate python visualization to neteq_rtpplay

2017-12-06 10:52:42 +00:00

neteq_external_decoder_test.cc

Fixing WebRTC after moving from src/webrtc to src/

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

neteq_external_decoder_test.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

neteq_input.cc

Fixing WebRTC after moving from src/webrtc to src/

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

neteq_input.h

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

neteq_packet_source_input.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

neteq_packet_source_input.h

Fixing WebRTC after moving from src/webrtc to src/

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

neteq_performance_test.cc

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

neteq_performance_test.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

neteq_quality_test.cc

Adds fixed PL loss mode to neteq_quality_test.

2018-02-13 15:34:04 +00:00

neteq_quality_test.h

Adds fixed PL loss mode to neteq_quality_test.

2018-02-13 15:34:04 +00:00

neteq_replacement_input.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

neteq_replacement_input.h

Fixing WebRTC after moving from src/webrtc to src/

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

neteq_rtpplay.cc

Adds voice concealment periods reporting to neteq_rtpplay.

2018-02-07 18:41:42 +00:00

neteq_test.cc

Adds voice concealment periods reporting to neteq_rtpplay.

2018-02-07 18:41:42 +00:00

neteq_test.h

Adds voice concealment periods reporting to neteq_rtpplay.

2018-02-07 18:41:42 +00:00

output_audio_file.h

Fixing WebRTC after moving from src/webrtc to src/

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

output_wav_file.h

Fixing WebRTC after moving from src/webrtc to src/

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

packet_source.cc

Fixing WebRTC after moving from src/webrtc to src/

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

packet_source.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

packet_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

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

packet.cc

Fixing WebRTC after moving from src/webrtc to src/

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

packet.h

Fix circular dependencies in webrtc_common.

2017-12-15 14:33:26 +00:00

resample_input_audio_file.cc

Fixing WebRTC after moving from src/webrtc to src/

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

resample_input_audio_file.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

rtc_event_log_source.cc

Fix circular dependency in rtc_event_log.

2017-11-29 10:46:19 +00:00

rtc_event_log_source.h

Fixing WebRTC after moving from src/webrtc to src/

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

rtp_analyze.cc

Fixing WebRTC after moving from src/webrtc to src/

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

rtp_encode.cc

rtp_encode: Unify the encoder configs somewhat

2017-12-07 09:43:17 +00:00

rtp_file_source.cc

Fixing WebRTC after moving from src/webrtc to src/

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

rtp_file_source.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

rtp_generator.cc

Fixing WebRTC after moving from src/webrtc to src/

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

rtp_generator.h

Fix circular dependencies in webrtc_common.

2017-12-15 14:33:26 +00:00

rtp_jitter.cc

Replacing the legacy tool RTPjitter with a new rtp_jitter

2017-11-24 13:38:59 +00:00

rtpcat.cc

Fixing WebRTC after moving from src/webrtc to src/

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