webrtc/modules/modules.gyp - Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

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Unified Diff: webrtc/modules/modules.gyp

Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Reverted unnecessary change to buffer_unittest.cc Created 4 years, 10 months ago

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Index: webrtc/modules/modules.gyp

diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp

index 68e4d11fe800dd7038b66c719c9588e47f6d2b7e..743b420d58dee267fd4af04588e3d408d4e225dc 100644

--- a/webrtc/modules/modules.gyp

+++ b/webrtc/modules/modules.gyp

@@ -167,6 +167,7 @@

'<(webrtc_root)/tools/tools.gyp:agc_test_utils',

'sources': [

+ 'audio_coding/codecs/audio_encoder_unittest.cc',

'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',

'audio_coding/acm2/acm_receiver_unittest_oldapi.cc',

'audio_coding/acm2/audio_coding_module_unittest_oldapi.cc',

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