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webrtc/modules/audio_coding/acm2
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philipel 0a5fe77d23 Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. 
By removing it we can in turn (next CL) get rid of RTPTypeHeader, which is a
union that cause some problems.

Bug: none
Change-Id: I9246ecbfe2c8b7eda27497cccbc5f438958b64bf
Reviewed-on: https://webrtc-review.googlesource.com/83985
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23666}
2018-06-19 16:44:19 +00:00
..
acm_codec_database.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_codec_database.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receive_test.cc Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. 2018-06-19 16:44:19 +00:00
acm_receive_test.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receiver.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receiver.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
acm_receiver_unittest.cc Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. 2018-06-19 16:44:19 +00:00
acm_resampler.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_resampler.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
acm_send_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_send_test.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
audio_coding_module.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_coding_module_unittest.cc Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. 2018-06-19 16:44:19 +00:00
call_statistics.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
call_statistics.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
call_statistics_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
codec_manager.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
codec_manager.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
codec_manager_unittest.cc Removed Die mock from MockAudioEncoder 2018-02-22 12:53:38 +00:00
rent_a_codec.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
rent_a_codec.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
rent_a_codec_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00