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webrtc/modules/audio_coding/codecs
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Jonas Olsson 366a50c4ef Remove simple stringstream usages. 
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.

The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.

Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
2018-09-06 12:53:19 +00:00
..
cng Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
g711 Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
g722 Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
ilbc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
isac Remove simple stringstream usages. 2018-09-06 12:53:19 +00:00
opus Moving LappedTransform, Blocker and AudioRingBuffer. 2018-08-31 15:27:50 +00:00
pcm16b Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
red Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
tools Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_decoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_encoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_format_conversion.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_format_conversion.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
builtin_audio_decoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
builtin_audio_encoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
legacy_encoded_audio_frame.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00