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webrtc/modules/audio_coding/neteq/mock
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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
..
mock_buffer_level_filter.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_decoder_database.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_delay_manager.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_delay_peak_detector.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mock_dtmf_buffer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_dtmf_tone_generator.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_expand.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_external_decoder_pcm16b.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mock_packet_buffer.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
mock_red_payload_splitter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mock_statistics_calculator.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00