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webrtc/modules/audio_coding/neteq/tools
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Ilya Nikolaevskiy b661c658da Revert "Remove rtc::Optional alias and api:optional target" 
This reverts commit 6f5b0f920a.

Reason for revert: Breaks internal project.

Original change's description:
> 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}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org

Change-Id: I95f5ec33520b823c3d0c9cb83d945d6a15355367
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9078
Reviewed-on: https://webrtc-review.googlesource.com/88140
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23921}
2018-07-11 07:41:41 +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.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
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
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 Revert "Remove rtc::Optional alias and api:optional target" 2018-07-11 07:41:41 +00:00
neteq_input.h Revert "Remove rtc::Optional alias and api:optional target" 2018-07-11 07:41:41 +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.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
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
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