commit 98ab3a46d6b98bd6626ab741092f7cbf104d127b
author kwiberg <kwiberg@webrtc.org> Wed Sep 30 21:54:21 2015 -0700
committer Commit bot <commit-bot@chromium.org> Thu Oct 01 04:54:29 2015 +0000
tree aeb35ee96119d07d88d32cf67eec873174fa60df
parent 456696a9c1bbd586701dcca3e4b2695e419a10ba

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}

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 8a560dc..6dfcac0 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -307,18 +307,20 @@
   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 @@
   *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 @@
   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 @@
   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