Add audio_format_conversion to deps for audio_decoder_factory_interface.

webrtc/modules/audio_coding/BUILD.gn

 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../../build/webrtc.gni")
 import("audio_coding.gni")
@@ skipping 50 matching lines @@
   ]
 }
 
 rtc_source_set("audio_decoder_factory_interface") {
   sources = [
     "codecs/audio_decoder_factory.h",
   ]
   deps = [
     ":audio_decoder_interface",
     ":audio_format",
+    ":audio_format_conversion",

[kwiberg-webrtc, 2016/11/02 10:00:42]

Thanks, but this doesn't look right. audio_decoder_factory.h, the only source
file in this build target, doesn't #include audio_format_conversion.h. It's the
build targets that contain the following files that ought to depend on
audio_format_conversion:

$ git grep -nH -l -e '#include.*audio_format_conversion.h'
webrtc/modules/audio_coding/acm2/acm_receive_test.cc
webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
webrtc/modules/audio_coding/test/APITest.cc
webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
webrtc/modules/audio_coding/test/TestAllCodecs.cc
webrtc/modules/audio_coding/test/TestRedFec.cc
webrtc/modules/audio_coding/test/TestStereo.cc
webrtc/modules/audio_coding/test/TestVADDTX.cc
webrtc/modules/audio_coding/test/TwoWayCommunication.cc
webrtc/modules/audio_coding/test/delay_test.cc
webrtc/modules/audio_coding/test/iSACTest.cc
webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
webrtc/modules/audio_coding/test/opus_test.cc
webrtc/modules/utility/source/coder.cc
webrtc/voice_engine/channel.cc

Or am I misunderstanding something?

[kjellander_webrtc, 2016/11/02 10:15:08]

You're right. But as there are now downstream projects depending on
audio_decoder_factory_interface target and its audio_format_conversion.cc
implementation, this would be one way to solve the GYP->GN switch quickly. I
think this is fine if we add a TODO to clean this up when downstream projects
are updated to properly depend on the audio_format_conversion target.

A better, but slower, way to solve this would be to:
1. Create the audio_format_conversion target in GYP
2. Update downstream projects to depend on that target.

[kwiberg-webrtc, 2016/11/02 10:23:14]

Right. I'm not usually an "end justifies the means" kind of person, but for the
GYP->GN transition I can make an exception... It'll be less work in total to do
this quick fix to unblock the transition, and then clean up afterwards, is what
you're saying?

     "../../base:rtc_base_approved",
   ]
 }
 
 rtc_static_library("builtin_audio_decoder_factory") {
   sources = [
     "codecs/builtin_audio_decoder_factory.cc",
     "codecs/builtin_audio_decoder_factory.h",
   ]
   deps = [
@@ skipping 1591 matching lines @@
       "../../test:test_support_main",
       "//testing/gtest",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }