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

Review URL: https://codereview.webrtc.org/1349393003

Cr-Commit-Position: refs/heads/master@{#10046}

webrtc/build/common.gypi

@@ -42,9 +42,13 @@
       '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)',
@@ -98,9 +102,6 @@
     # 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

@@ -17,27 +17,6 @@
 //                              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

@@ -67,13 +67,8 @@ source_set("audio_coding") {
   deps = [
     ":cng",
     ":g711",
-    ":g722",
-    ":ilbc",
-    ":isac",
-    ":isac_fix",
     ":neteq",
     ":pcm16b",
-    ":red",
     "../..:rtc_event_log",
     "../..:webrtc_common",
     "../../common_audio",
@@ -84,6 +79,27 @@ 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") {
@@ -788,10 +804,6 @@ source_set("neteq") {
     ":audio_decoder_interface",
     ":cng",
     ":g711",
-    ":g722",
-    ":ilbc",
-    ":isac",
-    ":isac_fix",
     ":pcm16b",
     "../..:webrtc_common",
     "../../common_audio",
@@ -804,4 +816,19 @@ 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

@@ -131,15 +131,18 @@ rtc::scoped_ptr<AudioEncoder> CreateSpeechEncoder(
 AudioEncoder* CreateRedEncoder(int red_payload_type,
                                AudioEncoder* encoder,
                                rtc::scoped_ptr<AudioEncoder>* red_encoder) {
-  if (red_payload_type == -1) {
+#ifdef WEBRTC_CODEC_RED
-    red_encoder->reset();
+  if (red_payload_type != -1) {
-    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();
   }
-  AudioEncoderCopyRed::Config config;
+#endif
-  config.payload_type = red_payload_type;
+
-  config.speech_encoder = encoder;
+  red_encoder->reset();
-  red_encoder->reset(new AudioEncoderCopyRed(config));
+  return encoder;
-  return red_encoder->get();
 }
 
 void CreateCngEncoder(int cng_payload_type,

webrtc/modules/audio_coding/main/audio_coding_module.gypi

@@ -11,12 +11,7 @@
     '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',
@@ -27,6 +22,23 @@
         '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

@@ -13,11 +13,7 @@
 
 #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

@@ -562,7 +562,6 @@ 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());
@@ -676,8 +675,10 @@ 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));
@@ -702,14 +703,19 @@ 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_TRUE(CodecSupported(kDecoderISACswb));
+  EXPECT_EQ(has_isac_swb, CodecSupported(kDecoderISACswb));
-  EXPECT_TRUE(CodecSupported(kDecoderISACfb));
+  EXPECT_EQ(has_isac_swb, CodecSupported(kDecoderISACfb));
   EXPECT_TRUE(CodecSupported(kDecoderPCM16B));
   EXPECT_TRUE(CodecSupported(kDecoderPCM16Bwb));
   EXPECT_TRUE(CodecSupported(kDecoderPCM16Bswb32kHz));

webrtc/modules/audio_coding/neteq/neteq.gypi

@@ -11,10 +11,6 @@
     'codecs': [
       'cng',
       'g711',
-      'g722',
-      'ilbc',
-      'isac',
-      'isac_fix',
       'pcm16b',
     ],
     'neteq_defines': [],
@@ -23,6 +19,23 @@
         '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)',
@@ -120,6 +133,10 @@
           'type': '<(gtest_target_type)',
           'dependencies': [
             '<@(codecs)',
+            'g722',
+            'ilbc',
+            'isac',
+            'isac_fix',
             'audio_decoder_interface',
             'neteq_unittest_tools',
             '<(DEPTH)/testing/gtest.gyp:gtest',
@@ -127,11 +144,6 @@
             '<(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': [