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webrtc/modules/audio_coding/neteq/tools
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Yves Gerey 665174fdbb Reformat the WebRTC code base 
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +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 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 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 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 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 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 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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
neteq_stats_getter.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
neteq_stats_getter.h Reformat the WebRTC code base 2018-06-19 14:00:39 +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 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 Reformat the WebRTC code base 2018-06-19 14:00:39 +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