Revert of Don't link with audio codecs that we don't use (patchset #4 id:60001 of https://codereview.webrtc.org/1349393003/ )

Reason for revert:
Breaking Chromium FYI bots.

Original issue's description:
> Don't link with audio codecs that we don't use
>
> We used to link with all audio codecs unconditionally (except Opus);
> this patch makes gyp and gn only link to the ones that are used.
>
> (This unfortunately fails to have a measurable impact on Chromium
> binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
> fix were already being excluded from Chromium by some other means
> (likely just the linker omitting compilation units with no incoming
> references).)
>
> BUG=webrtc:4557
>
> Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809
> Cr-Commit-Position: refs/heads/master@{#10046}

TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1368933002

Cr-Commit-Position: refs/heads/master@{#10069}

webrtc/build/common.gypi

@@ -42,13 +42,9 @@
       'webrtc_vp9_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp9',
       'include_opus%': 1,
       'opus_dir%': '<(DEPTH)/third_party/opus',
-
-      # Enable to use the Mozilla internal settings.
-      'build_with_mozilla%': 0,
     },
     'build_with_chromium%': '<(build_with_chromium)',
     'build_with_libjingle%': '<(build_with_libjingle)',
-    'build_with_mozilla%': '<(build_with_mozilla)',
     'webrtc_root%': '<(webrtc_root)',
     'apk_tests_path%': '<(apk_tests_path)',
     'modules_java_gyp_path%': '<(modules_java_gyp_path)',
@@ -102,6 +98,9 @@
     # Disable by default
     'have_dbus_glib%': 0,
 
+    # Enable to use the Mozilla internal settings.
+    'build_with_mozilla%': 0,
+
     # Make it possible to provide custom locations for some libraries.
     'libvpx_dir%': '<(DEPTH)/third_party/libvpx',
     'libyuv_dir%': '<(DEPTH)/third_party/libyuv',

webrtc/engine_configurations.h

@@ -17,6 +17,27 @@
 //                              Voice and Video
 // ============================================================================
 
+// ----------------------------------------------------------------------------
+//  [Voice] Codec settings
+// ----------------------------------------------------------------------------
+
+// iSAC and G722 are not included in the Mozilla build, but in all other builds.
+#ifndef WEBRTC_MOZILLA_BUILD
+#ifdef WEBRTC_ARCH_ARM
+#define WEBRTC_CODEC_ISACFX  // Fix-point iSAC implementation.
+#else
+#define WEBRTC_CODEC_ISAC  // Floating-point iSAC implementation (default).
+#endif  // WEBRTC_ARCH_ARM
+#define WEBRTC_CODEC_G722
+#endif  // !WEBRTC_MOZILLA_BUILD
+
+// iLBC and Redundancy coding are excluded from Chromium and Mozilla
+// builds to reduce binary size.
+#if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_MOZILLA_BUILD)
+#define WEBRTC_CODEC_ILBC
+#define WEBRTC_CODEC_RED
+#endif  // !WEBRTC_CHROMIUM_BUILD && !WEBRTC_MOZILLA_BUILD
+
 // ----------------------------------------------------------------------------
 //  [Video] Codec settings
 // ----------------------------------------------------------------------------

webrtc/modules/audio_coding/BUILD.gn

@@ -67,8 +67,13 @@ source_set("audio_coding") {
   deps = [
     ":cng",
     ":g711",
+    ":g722",
+    ":ilbc",
+    ":isac",
+    ":isac_fix",
     ":neteq",
     ":pcm16b",
+    ":red",
     "../..:rtc_event_log",
     "../..:webrtc_common",
     "../../common_audio",
@@ -79,27 +84,6 @@ source_set("audio_coding") {
     defines += [ "WEBRTC_CODEC_OPUS" ]
     deps += [ ":webrtc_opus" ]
   }
-  if (!build_with_mozilla) {
-    if (current_cpu == "arm") {
-      defines += [ "WEBRTC_CODEC_ISACFX" ]
-      deps += [ ":isac_fix" ]
-    } else {
-      defines += [ "WEBRTC_CODEC_ISAC" ]
-      deps += [ ":isac" ]
-    }
-    defines += [ "WEBRTC_CODEC_G722" ]
-    deps += [ ":g722" ]
-  }
-  if (!build_with_mozilla && !build_with_chromium) {
-    defines += [
-      "WEBRTC_CODEC_ILBC",
-      "WEBRTC_CODEC_RED",
-    ]
-    deps += [
-      ":ilbc",
-      ":red",
-    ]
-  }
 }
 
 source_set("audio_decoder_interface") {
@@ -804,6 +788,10 @@ source_set("neteq") {
     ":audio_decoder_interface",
     ":cng",
     ":g711",
+    ":g722",
+    ":ilbc",
+    ":isac",
+    ":isac_fix",
     ":pcm16b",
     "../..:webrtc_common",
     "../../common_audio",
@@ -816,19 +804,4 @@ source_set("neteq") {
     defines += [ "WEBRTC_CODEC_OPUS" ]
     deps += [ ":webrtc_opus" ]
   }
-  if (!build_with_mozilla) {
-    if (current_cpu == "arm") {
-      defines += [ "WEBRTC_CODEC_ISACFX" ]
-      deps += [ ":isac_fix" ]
-    } else {
-      defines += [ "WEBRTC_CODEC_ISAC" ]
-      deps += [ ":isac" ]
-    }
-    defines += [ "WEBRTC_CODEC_G722" ]
-    deps += [ ":g722" ]
-  }
-  if (!build_with_mozilla && !build_with_chromium) {
-    defines += [ "WEBRTC_CODEC_ILBC" ]
-    deps += [ ":ilbc" ]
-  }
 }

webrtc/modules/audio_coding/main/acm2/codec_owner.cc

@@ -135,18 +135,15 @@ rtc::scoped_ptr<AudioEncoder> CreateSpeechEncoder(
 AudioEncoder* CreateRedEncoder(int red_payload_type,
                                AudioEncoder* encoder,
                                rtc::scoped_ptr<AudioEncoder>* red_encoder) {
-#ifdef WEBRTC_CODEC_RED
+  if (red_payload_type == -1) {
-  if (red_payload_type != -1) {
+    red_encoder->reset();
+    return encoder;
+  }
   AudioEncoderCopyRed::Config config;
   config.payload_type = red_payload_type;
   config.speech_encoder = encoder;
   red_encoder->reset(new AudioEncoderCopyRed(config));
   return red_encoder->get();
-  }
-#endif
-
-  red_encoder->reset();
-  return encoder;
 }
 
 void CreateCngEncoder(int cng_payload_type,

webrtc/modules/audio_coding/main/audio_coding_module.gypi

@@ -11,7 +11,12 @@
     'audio_coding_dependencies': [
       'cng',
       'g711',
+      'g722',
+      'ilbc',
+      'isac',
+      'isac_fix',
       'pcm16b',
+      'red',
       '<(webrtc_root)/common.gyp:webrtc_common',
       '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
@@ -22,23 +27,6 @@
         'audio_coding_dependencies': ['webrtc_opus',],
         'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
       }],
-      ['build_with_mozilla==0', {
-        'conditions': [
-          ['target_arch=="arm"', {
-            'audio_coding_dependencies': ['isac_fix',],
-            'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',],
-          }, {
-            'audio_coding_dependencies': ['isac',],
-            'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
-          }],
-        ],
-        'audio_coding_dependencies': ['g722',],
-        'audio_coding_defines': ['WEBRTC_CODEC_G722',],
-      }],
-      ['build_with_mozilla==0 and build_with_chromium==0', {
-        'audio_coding_dependencies': ['ilbc', 'red',],
-        'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
-      }],
     ],
   },
   'targets': [

webrtc/modules/audio_coding/neteq/audio_decoder_impl.h

@@ -13,7 +13,11 @@
 
 #include <assert.h>
 
+#ifndef AUDIO_DECODER_UNITTEST
+// If this is compiled as a part of the audio_deoder_unittest, the codec
+// selection is made in the gypi file instead of in engine_configurations.h.
 #include "webrtc/engine_configurations.h"
+#endif
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"

webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

@@ -562,6 +562,7 @@ TEST_F(AudioDecoderIsacSwbTest, EncodeDecode) {
   int tolerance = 19757;
   double mse = 8.18e6;
   int delay = 160;  // Delay from input to output.
+  EXPECT_TRUE(CodecSupported(kDecoderISACswb));
   EncodeDecodeTest(0, tolerance, mse, delay);
   ReInitTest();
   EXPECT_FALSE(decoder_->HasDecodePlc());
@@ -675,10 +676,8 @@ TEST(AudioDecoder, CodecSampleRateHz) {
   EXPECT_EQ(8000, CodecSampleRateHz(kDecoderPCMa_2ch));
   EXPECT_EQ(8000, CodecSampleRateHz(kDecoderILBC));
   EXPECT_EQ(16000, CodecSampleRateHz(kDecoderISAC));
-#ifndef WEBRTC_ARCH_ARM
   EXPECT_EQ(32000, CodecSampleRateHz(kDecoderISACswb));
   EXPECT_EQ(32000, CodecSampleRateHz(kDecoderISACfb));
-#endif
   EXPECT_EQ(8000, CodecSampleRateHz(kDecoderPCM16B));
   EXPECT_EQ(16000, CodecSampleRateHz(kDecoderPCM16Bwb));
   EXPECT_EQ(32000, CodecSampleRateHz(kDecoderPCM16Bswb32kHz));
@@ -703,19 +702,14 @@ TEST(AudioDecoder, CodecSampleRateHz) {
 }
 
 TEST(AudioDecoder, CodecSupported) {
-#ifdef WEBRTC_ARCH_ARM
-  static const bool has_isac_swb = false;
-#else
-  static const bool has_isac_swb = true;
-#endif
   EXPECT_TRUE(CodecSupported(kDecoderPCMu));
   EXPECT_TRUE(CodecSupported(kDecoderPCMa));
   EXPECT_TRUE(CodecSupported(kDecoderPCMu_2ch));
   EXPECT_TRUE(CodecSupported(kDecoderPCMa_2ch));
   EXPECT_TRUE(CodecSupported(kDecoderILBC));
   EXPECT_TRUE(CodecSupported(kDecoderISAC));
-  EXPECT_EQ(has_isac_swb, CodecSupported(kDecoderISACswb));
+  EXPECT_TRUE(CodecSupported(kDecoderISACswb));
-  EXPECT_EQ(has_isac_swb, CodecSupported(kDecoderISACfb));
+  EXPECT_TRUE(CodecSupported(kDecoderISACfb));
   EXPECT_TRUE(CodecSupported(kDecoderPCM16B));
   EXPECT_TRUE(CodecSupported(kDecoderPCM16Bwb));
   EXPECT_TRUE(CodecSupported(kDecoderPCM16Bswb32kHz));

webrtc/modules/audio_coding/neteq/neteq.gypi

@@ -11,6 +11,10 @@
     'codecs': [
       'cng',
       'g711',
+      'g722',
+      'ilbc',
+      'isac',
+      'isac_fix',
       'pcm16b',
     ],
     'neteq_defines': [],
@@ -19,23 +23,6 @@
         'codecs': ['webrtc_opus',],
         'neteq_defines': ['WEBRTC_CODEC_OPUS',],
       }],
-      ['build_with_mozilla==0', {
-        'conditions': [
-          ['target_arch=="arm"', {
-            'codecs': ['isac_fix',],
-            'neteq_defines': ['WEBRTC_CODEC_ISACFX',],
-          }, {
-            'codecs': ['isac',],
-            'neteq_defines': ['WEBRTC_CODEC_ISAC',],
-          }],
-        ],
-        'codecs': ['g722',],
-        'neteq_defines': ['WEBRTC_CODEC_G722',],
-      }],
-      ['build_with_mozilla==0 and build_with_chromium==0', {
-        'codecs': ['ilbc',],
-        'neteq_defines': ['WEBRTC_CODEC_ILBC',],
-      }],
     ],
     'neteq_dependencies': [
       '<@(codecs)',
@@ -133,10 +120,6 @@
           'type': '<(gtest_target_type)',
           'dependencies': [
             '<@(codecs)',
-            'g722',
-            'ilbc',
-            'isac',
-            'isac_fix',
             'audio_decoder_interface',
             'neteq_unittest_tools',
             '<(DEPTH)/testing/gtest.gyp:gtest',
@@ -144,6 +127,11 @@
             '<(webrtc_root)/test/test.gyp:test_support_main',
           ],
           'defines': [
+            'AUDIO_DECODER_UNITTEST',
+            'WEBRTC_CODEC_G722',
+            'WEBRTC_CODEC_ILBC',
+            'WEBRTC_CODEC_ISACFX',
+            'WEBRTC_CODEC_ISAC',
             '<@(neteq_defines)',
           ],
           'sources': [