commit 385b10bbaade624470e9ee6cf4844cc5740febb7
author Ivo Creusen <ivoc@webrtc.org> Fri Oct 13 12:37:27 2017 +0200
committer Commit Bot <commit-bot@chromium.org> Fri Oct 13 13:26:57 2017 +0000
tree c64eda709f386171708f807a63a0729d9ad6031e
parent e75922341816a79d66b729af41375c7273cd0876

Added experiment to improve handling of frame length changes in NetEq.

The field trial effects two things: after a frame length change the IAT
histogram is scaled to prevent an immediate change in target buffer
level. Also, the peak history in the delay peak detector is cleared, 
because the size of the peaks is stored in number of packets (which
will be incorrect after a frame length change).

Bug: webrtc:8381
Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8
Reviewed-on: https://webrtc-review.googlesource.com/8101
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20284}

modules/audio_coding/BUILD.gn

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index da055bb..b222ad7 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1143,14 +1143,10 @@
   sources = [
     "neteq/tools/fake_decode_from_file.cc",
     "neteq/tools/fake_decode_from_file.h",
-    "neteq/tools/input_audio_file.cc",
-    "neteq/tools/input_audio_file.h",
     "neteq/tools/neteq_delay_analyzer.cc",
     "neteq/tools/neteq_delay_analyzer.h",
     "neteq/tools/neteq_replacement_input.cc",
     "neteq/tools/neteq_replacement_input.h",
-    "neteq/tools/resample_input_audio_file.cc",
-    "neteq/tools/resample_input_audio_file.h",
   ]
 
   public_configs = [ ":neteq_tools_config" ]
@@ -1165,16 +1161,31 @@
     "../../api:array_view",
     "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
-    "../../common_audio",
     "../../rtc_base:rtc_base_approved",
     "../rtp_rtcp",
   ]
 
   public_deps = [
+    ":neteq_input_audio_tools",
     ":neteq_tools_minimal",
   ]
 }
 
+rtc_source_set("neteq_input_audio_tools") {
+  sources = [
+    "neteq/tools/input_audio_file.cc",
+    "neteq/tools/input_audio_file.h",
+    "neteq/tools/resample_input_audio_file.cc",
+    "neteq/tools/resample_input_audio_file.h",
+  ]
+
+  deps = [
+    "../..:webrtc_common",
+    "../../common_audio",
+    "../../rtc_base:rtc_base_approved",
+  ]
+}
+
 if (rtc_enable_protobuf) {
   rtc_static_library("rtc_event_log_source") {
     testonly = true