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webrtc/modules/audio_coding/neteq/tools
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Danil Chapovalov b602123a5a Replace rtc::Optional with absl::optional in modules/audio_coding 
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'modules/audio_coding'

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 12:46:20 +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 Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +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 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 Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
input_audio_file.cc Fix for crash when reading from audio file in NetEq. 2018-03-23 18:29:05 +00:00
input_audio_file.h Replacing the legacy tool RTPencode with a new rtp_encode 2017-11-24 09:05:48 +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 Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
neteq_delay_analyzer.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
neteq_external_decoder_test.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq_external_decoder_test.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +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 Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
neteq_packet_source_input.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
neteq_packet_source_input.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
neteq_performance_test.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +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 Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +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 Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
neteq_rtpplay.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
neteq_stats_getter.cc Let NetEq stats getter provide time for each stats query. 2018-04-23 12:53:26 +00:00
neteq_stats_getter.h Let NetEq stats getter provide time for each stats query. 2018-04-23 12:53:26 +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.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
packet.h Fix circular dependencies in webrtc_common. 2017-12-15 14:33:26 +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
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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_encode.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
rtp_file_source.cc Remove a couple of unnecessary winsock2.h includes 2018-04-03 08:49:58 +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