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Move RtpExtension to api/ directory and config.h/.cc to call/.

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BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Commit-Position: refs/heads/master@{#19639}
This commit is contained in:
Stefan Holmer
2017-09-01 15:29:28 +02:00
parent 801f722377
commit 1acbd68718
41 changed files with 602 additions and 450 deletions
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5
webrtc/modules/audio_coding/BUILD.gn
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@ -1569,8 +1569,11 @@ if (rtc_include_tests) {
":isac_fix",
":webrtc_opus",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_default",
"../../system_wrappers:system_wrappers_default",
"../../test:field_trial",
"../../test:test_main",
"../audio_processing",
"//testing/gtest",
@ -1702,6 +1705,8 @@ if (rtc_include_tests) {
"../..:webrtc_common",
"../../common_audio",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_default",
"../../test:field_trial",
]
configs += [ ":RTPencode_config" ]