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webrtc_m130/modules/audio_coding/neteq/tools
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Danil Chapovalov 6f5b0f920a Remove rtc::Optional alias and api:optional target 
Update left-overs where old target still was used.

Bug: webrtc:9078
Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
Reviewed-on: https://webrtc-review.googlesource.com/84740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23913}
2018-07-10 18:02:23 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_loop.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
Replace rtc::Optional with absl::optional in modules/audio_coding
2018-06-19 12:46:20 +00:00
encode_neteq_input.h
Replace rtc::Optional with absl::optional in modules/audio_coding
2018-06-19 12:46:20 +00:00
fake_decode_from_file.cc
Remove a legacy DCHEC in FakeDecodeFromFile.
2018-07-09 19:56:58 +00:00
fake_decode_from_file.h
Implement PacketDuration() for FakeDecoderFromFile.
2018-06-29 08:32:36 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
input_audio_file.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
neteq_delay_analyzer.cc
Aligning time in audio jitter buffer plot to other plots in rtc event log visualizer.
2018-06-21 14:23:53 +00:00
neteq_delay_analyzer.h
Aligning time in audio jitter buffer plot to other plots in rtc event log visualizer.
2018-06-21 14:23:53 +00:00
neteq_event_log_input.cc
Break out NetEqEventLogInput to separate source files
2018-07-02 14:15:29 +00:00
neteq_event_log_input.h
Break out NetEqEventLogInput to separate source files
2018-07-02 14:15:29 +00:00
neteq_external_decoder_test.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
neteq_external_decoder_test.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
neteq_input.cc
Remove rtc::Optional alias and api:optional target
2018-07-10 18:02:23 +00:00
neteq_input.h
Remove rtc::Optional alias and api:optional target
2018-07-10 18:02:23 +00:00
neteq_packet_source_input.cc
Break out NetEqEventLogInput to separate source files
2018-07-02 14:15:29 +00:00
neteq_packet_source_input.h
Break out NetEqEventLogInput to separate source files
2018-07-02 14:15:29 +00:00
neteq_performance_test.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
neteq_quality_test.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
neteq_replacement_input.cc
Replace rtc::Optional with absl::optional in modules/audio_coding
2018-06-19 12:46:20 +00:00
neteq_replacement_input.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
neteq_rtpplay.cc
Break out NetEqEventLogInput to separate source files
2018-07-02 14:15:29 +00:00
neteq_stats_getter.cc
Adding NetEq lifetime stats to event log visualizer.
2018-06-26 11:27:09 +00:00
neteq_stats_getter.h
Adding NetEq lifetime stats to event log visualizer.
2018-06-26 11:27:09 +00:00
neteq_test.cc
Reland "NetEq: Deprecate playout modes Fax, Off and Streaming"
2018-07-02 10:20:33 +00:00
neteq_test.h
Reland "NetEq: Deprecate playout modes Fax, Off and Streaming"
2018-07-02 10:20:33 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
packet.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
packet.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
Split LoggedBweProbeResult into -Success and -Failure.
2018-05-29 13:41:04 +00:00
rtc_event_log_source.h
Reland "Create new API for RtcEventLogParser."
2018-04-27 14:46:51 +00:00
rtp_analyze.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_encode.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtp_file_source.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtp_generator.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +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