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.

(This was previously landed as revisions 10046 and 10060, and got
reverted because it broke several of the Chromium FYI bots.)

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1368843003

Cr-Commit-Position: refs/heads/master@{#10127}

1 files changed, 22 insertions, 7 deletions

diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 8a560dc200..6dfcac0d3a 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -307,18 +307,20 @@ void NetEqDecodingTest::LoadDecoders() {
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMu, 0));
   // Load PCMa.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMa, 8));
-#ifndef WEBRTC_ANDROID
+#ifdef WEBRTC_CODEC_ILBC
   // Load iLBC.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderILBC, 102));
-#endif  // WEBRTC_ANDROID
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
   // Load iSAC.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, 103));
-#ifndef WEBRTC_ANDROID
+#endif
+#ifdef WEBRTC_CODEC_ISAC
   // Load iSAC SWB.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACswb, 104));
   // Load iSAC FB.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACfb, 105));
-#endif  // WEBRTC_ANDROID
+#endif
   // Load PCM16B nb.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16B, 93));
   // Load PCM16B wb.
@@ -441,8 +443,15 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
   *payload_len = 1;  // Only noise level, no spectral parameters.
 }
 
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISAC)) && \
+    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+#define IF_ALL_CODECS(x) x
+#else
+#define IF_ALL_CODECS(x) DISABLED_##x
+#endif
+
 TEST_F(NetEqDecodingTest,
-       DISABLED_ON_IOS(DISABLED_ON_ANDROID(TestBitExactness))) {
+       DISABLED_ON_IOS(DISABLED_ON_ANDROID(IF_ALL_CODECS(TestBitExactness)))) {
   const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
       "resources/audio_coding/neteq_universal_new.rtp";
   // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
@@ -820,7 +829,13 @@ TEST_F(NetEqDecodingTest, UnknownPayloadType) {
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 }
 
-TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#define IF_ISAC(x) x
+#else
+#define IF_ISAC(x) DISABLED_##x
+#endif
+
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) {
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
@@ -1051,7 +1066,7 @@ TEST_F(NetEqBgnTestFade, RunTest) {
   CheckBgn(32000);
 }
 
-TEST_F(NetEqDecodingTest, SyncPacketInsert) {
+TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
   WebRtcRTPHeader rtp_info;
   uint32_t receive_timestamp = 0;
   // For the readability use the following payloads instead of the defaults of