Don't select audio codecs depending on GN vars `build_with_{chromium|mozilla}`
BUG=webrtc:8343
Change-Id: I5943006a4da17f72eb88eae9d7ea57574d54f680
Reviewed-on: https://webrtc-review.googlesource.com/9401
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20540}diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index fbb2fc5..8ea533b 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -43,6 +43,8 @@
"../../rtc_base:rtc_base_approved",
"L16:audio_decoder_L16",
"g711:audio_decoder_g711",
+ "g722:audio_decoder_g722",
+ "isac:audio_decoder_isac",
]
defines = []
if (rtc_include_ilbc) {
@@ -57,21 +59,6 @@
} else {
defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
}
- if (build_with_mozilla) {
- defines += [
- "WEBRTC_USE_BUILTIN_G722=0",
- "WEBRTC_USE_BUILTIN_ISAC=0",
- ]
- } else {
- deps += [
- "g722:audio_decoder_g722",
- "isac:audio_decoder_isac",
- ]
- defines += [
- "WEBRTC_USE_BUILTIN_G722=1",
- "WEBRTC_USE_BUILTIN_ISAC=1",
- ]
- }
}
rtc_static_library("builtin_audio_encoder_factory") {
@@ -84,6 +71,8 @@
"../../rtc_base:rtc_base_approved",
"L16:audio_encoder_L16",
"g711:audio_encoder_g711",
+ "g722:audio_encoder_g722",
+ "isac:audio_encoder_isac",
]
defines = []
if (rtc_include_ilbc) {
@@ -98,19 +87,4 @@
} else {
defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
}
- if (build_with_mozilla) {
- defines += [
- "WEBRTC_USE_BUILTIN_G722=0",
- "WEBRTC_USE_BUILTIN_ISAC=0",
- ]
- } else {
- deps += [
- "g722:audio_encoder_g722",
- "isac:audio_encoder_isac",
- ]
- defines += [
- "WEBRTC_USE_BUILTIN_G722=1",
- "WEBRTC_USE_BUILTIN_ISAC=1",
- ]
- }
}
diff --git a/api/audio_codecs/builtin_audio_decoder_factory.cc b/api/audio_codecs/builtin_audio_decoder_factory.cc
index a6eac72..9520d2a 100644
--- a/api/audio_codecs/builtin_audio_decoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_decoder_factory.cc
@@ -16,15 +16,11 @@
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/audio_decoder_factory_template.h"
#include "api/audio_codecs/g711/audio_decoder_g711.h"
-#if WEBRTC_USE_BUILTIN_G722
-#include "api/audio_codecs/g722/audio_decoder_g722.h" // nogncheck
-#endif
+#include "api/audio_codecs/g722/audio_decoder_g722.h"
#if WEBRTC_USE_BUILTIN_ILBC
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck
#endif
-#if WEBRTC_USE_BUILTIN_ISAC
-#include "api/audio_codecs/isac/audio_decoder_isac.h" // nogncheck
-#endif
+#include "api/audio_codecs/isac/audio_decoder_isac.h"
#if WEBRTC_USE_BUILTIN_OPUS
#include "api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck
#endif
@@ -57,13 +53,7 @@
AudioDecoderOpus,
#endif
-#if WEBRTC_USE_BUILTIN_ISAC
- AudioDecoderIsac,
-#endif
-
-#if WEBRTC_USE_BUILTIN_G722
- AudioDecoderG722,
-#endif
+ AudioDecoderIsac, AudioDecoderG722,
#if WEBRTC_USE_BUILTIN_ILBC
AudioDecoderIlbc,
diff --git a/api/audio_codecs/builtin_audio_encoder_factory.cc b/api/audio_codecs/builtin_audio_encoder_factory.cc
index 8654d0d..877f850 100644
--- a/api/audio_codecs/builtin_audio_encoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_encoder_factory.cc
@@ -16,15 +16,11 @@
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/g711/audio_encoder_g711.h"
-#if WEBRTC_USE_BUILTIN_G722
-#include "api/audio_codecs/g722/audio_encoder_g722.h" // nogncheck
-#endif
+#include "api/audio_codecs/g722/audio_encoder_g722.h"
#if WEBRTC_USE_BUILTIN_ILBC
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck
#endif
-#if WEBRTC_USE_BUILTIN_ISAC
-#include "api/audio_codecs/isac/audio_encoder_isac.h" // nogncheck
-#endif
+#include "api/audio_codecs/isac/audio_encoder_isac.h"
#if WEBRTC_USE_BUILTIN_OPUS
#include "api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck
#endif
@@ -61,13 +57,7 @@
AudioEncoderOpus,
#endif
-#if WEBRTC_USE_BUILTIN_ISAC
- AudioEncoderIsac,
-#endif
-
-#if WEBRTC_USE_BUILTIN_G722
- AudioEncoderG722,
-#endif
+ AudioEncoderIsac, AudioEncoderG722,
#if WEBRTC_USE_BUILTIN_ILBC
AudioEncoderIlbc,
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 941f13c..e5854ef 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -22,14 +22,12 @@
if (rtc_include_opus) {
audio_codec_deps += [ ":webrtc_opus" ]
}
-if (!build_with_mozilla) {
- if (current_cpu == "arm") {
- audio_codec_deps += [ ":isac_fix" ]
- } else {
- audio_codec_deps += [ ":isac" ]
- }
- audio_codec_deps += [ ":g722" ]
+if (current_cpu == "arm") {
+ audio_codec_deps += [ ":isac_fix" ]
+} else {
+ audio_codec_deps += [ ":isac" ]
}
+audio_codec_deps += [ ":g722" ]
if (!build_with_mozilla && !build_with_chromium) {
audio_codec_deps += [ ":red" ]
}
diff --git a/modules/audio_coding/acm2/acm_codec_database.cc b/modules/audio_coding/acm2/acm_codec_database.cc
index 7b3b1d2..4553b52 100644
--- a/modules/audio_coding/acm2/acm_codec_database.cc
+++ b/modules/audio_coding/acm2/acm_codec_database.cc
@@ -85,12 +85,10 @@
#ifdef WEBRTC_CODEC_ILBC
{102, "ILBC", 8000, 240, 1, 13300},
#endif
-#ifdef WEBRTC_CODEC_G722
// Mono
{9, "G722", 16000, 320, 1, 64000},
// Stereo
{119, "G722", 16000, 320, 2, 64000},
-#endif
#ifdef WEBRTC_CODEC_OPUS
// Opus internally supports 48, 24, 16, 12, 8 kHz.
// Mono and stereo.
@@ -143,12 +141,10 @@
#ifdef WEBRTC_CODEC_ILBC
{4, {160, 240, 320, 480}, 0, 1},
#endif
-#ifdef WEBRTC_CODEC_G722
// Mono
{6, {160, 320, 480, 640, 800, 960}, 0, 2},
// Stereo
{6, {160, 320, 480, 640, 800, 960}, 0, 2},
-#endif
#ifdef WEBRTC_CODEC_OPUS
// Opus supports frames shorter than 10ms,
// but it doesn't help us to use them.
@@ -200,12 +196,10 @@
#ifdef WEBRTC_CODEC_ILBC
NetEqDecoder::kDecoderILBC,
#endif
-#ifdef WEBRTC_CODEC_G722
// Mono
NetEqDecoder::kDecoderG722,
// Stereo
NetEqDecoder::kDecoderG722_2ch,
-#endif
#ifdef WEBRTC_CODEC_OPUS
// Mono and stereo.
NetEqDecoder::kDecoderOpus,
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 1d5b954..ca59b31 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -980,7 +980,7 @@
};
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
- defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+ defined(WEBRTC_CODEC_ILBC)
TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
Run(8000, PlatformChecksum("2adede965c6f87de7142c51552111d08",
"028c0fc414b1c9ab7e582dccdf381e98",
@@ -1438,7 +1438,6 @@
#else
#define MAYBE_G722_20ms G722_20ms
#endif
-#if defined(WEBRTC_CODEC_G722)
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
@@ -1451,14 +1450,12 @@
"android_arm64_payload", "android_arm64_clang_payload"),
50, test::AcmReceiveTestOldApi::kMonoOutput);
}
-#endif
#if defined(WEBRTC_ANDROID)
#define MAYBE_G722_stereo_20ms DISABLED_G722_stereo_20ms
#else
#define MAYBE_G722_stereo_20ms G722_stereo_20ms
#endif
-#if defined(WEBRTC_CODEC_G722)
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
@@ -1471,7 +1468,6 @@
"android_arm64_payload", "android_arm64_clang_payload"),
50, test::AcmReceiveTestOldApi::kStereoOutput);
}
-#endif
TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
diff --git a/modules/audio_coding/acm2/rent_a_codec.cc b/modules/audio_coding/acm2/rent_a_codec.cc
index ff66890..120d54c 100644
--- a/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/modules/audio_coding/acm2/rent_a_codec.cc
@@ -16,9 +16,7 @@
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "rtc_base/logging.h"
-#ifdef WEBRTC_CODEC_G722
#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
-#endif
#ifdef WEBRTC_CODEC_ILBC
#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#endif
@@ -175,10 +173,8 @@
if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
return std::unique_ptr<AudioEncoder>(new AudioEncoderIlbcImpl(speech_inst));
#endif
-#ifdef WEBRTC_CODEC_G722
if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
return std::unique_ptr<AudioEncoder>(new AudioEncoderG722Impl(speech_inst));
-#endif
LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname;
return std::unique_ptr<AudioEncoder>();
}
diff --git a/modules/audio_coding/acm2/rent_a_codec.h b/modules/audio_coding/acm2/rent_a_codec.h
index f1de72d..cecb914 100644
--- a/modules/audio_coding/acm2/rent_a_codec.h
+++ b/modules/audio_coding/acm2/rent_a_codec.h
@@ -58,10 +58,8 @@
#ifdef WEBRTC_CODEC_ILBC
kILBC,
#endif
-#ifdef WEBRTC_CODEC_G722
kG722, // Mono
kG722_2ch, // Stereo
-#endif
#ifdef WEBRTC_CODEC_OPUS
kOpus, // Mono and stereo
#endif
@@ -92,10 +90,6 @@
#ifndef WEBRTC_CODEC_ILBC
kILBC = -1,
#endif
-#ifndef WEBRTC_CODEC_G722
- kG722 = -1, // Mono
- kG722_2ch = -1, // Stereo
-#endif
#ifndef WEBRTC_CODEC_OPUS
kOpus = -1, // Mono and stereo
#endif
diff --git a/modules/audio_coding/audio_coding.gni b/modules/audio_coding/audio_coding.gni
index 41dcf00..9b0aba8 100644
--- a/modules/audio_coding/audio_coding.gni
+++ b/modules/audio_coding/audio_coding.gni
@@ -20,13 +20,10 @@
} else {
audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
}
-if (!build_with_mozilla) {
- if (current_cpu == "arm") {
- audio_codec_defines += [ "WEBRTC_CODEC_ISACFX" ]
- } else {
- audio_codec_defines += [ "WEBRTC_CODEC_ISAC" ]
- }
- audio_codec_defines += [ "WEBRTC_CODEC_G722" ]
+if (current_cpu == "arm") {
+ audio_codec_defines += [ "WEBRTC_CODEC_ISACFX" ]
+} else {
+ audio_codec_defines += [ "WEBRTC_CODEC_ISAC" ]
}
if (!build_with_mozilla && !build_with_chromium) {
audio_codec_defines += [ "WEBRTC_CODEC_RED" ]
diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
index ec79c28..58bfaed 100644
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
@@ -131,9 +131,7 @@
#ifdef WEBRTC_CODEC_ISAC
{"isac", 32000, 1},
#endif
-#ifdef WEBRTC_CODEC_G722
{"G722", 8000, 1},
-#endif
#ifdef WEBRTC_CODEC_ILBC
{"ilbc", 8000, 1},
#endif
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index bb8c4e9..b0f3a39 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -450,8 +450,7 @@
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
- defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) && \
- !defined(WEBRTC_ARCH_ARM64)
+ defined(WEBRTC_CODEC_ILBC) && !defined(WEBRTC_ARCH_ARM64)
#define MAYBE_TestBitExactness TestBitExactness
#else
#define MAYBE_TestBitExactness DISABLED_TestBitExactness
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index ff28a28..f7f220f 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -151,7 +151,6 @@
// All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes.
-#ifdef WEBRTC_CODEC_G722
if (test_mode_ != 0) {
printf("===============================================================\n");
}
@@ -171,7 +170,6 @@
RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
-#endif
#ifdef WEBRTC_CODEC_ILBC
if (test_mode_ != 0) {
printf("===============================================================\n");
@@ -324,9 +322,6 @@
/* Print out all codecs that were not tested in the run */
printf("The following codecs was not included in the test:\n");
-#ifndef WEBRTC_CODEC_G722
- printf(" G.722\n");
-#endif
#ifndef WEBRTC_CODEC_ILBC
printf(" iLBC\n");
#endif
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index 58561c6..034c595 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -38,13 +38,9 @@
const char kNamePCMU[] = "PCMU";
const char kNameCN[] = "CN";
const char kNameRED[] = "RED";
-
- // These three are only used by code #ifdeffed on WEBRTC_CODEC_G722.
-#ifdef WEBRTC_CODEC_G722
const char kNameISAC[] = "ISAC";
const char kNameG722[] = "G722";
const char kNameOPUS[] = "opus";
-#endif
}
TestRedFec::TestRedFec()
@@ -104,11 +100,6 @@
Run();
_outFileB.Close();
-#ifndef WEBRTC_CODEC_G722
- EXPECT_TRUE(false);
- printf("G722 needs to be activated to run this test\n");
- return;
-#else
EXPECT_EQ(0, RegisterSendCodec('A', kNameG722, 16000));
EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000));
@@ -412,8 +403,6 @@
EXPECT_FALSE(_acmA->REDStatus());
EXPECT_EQ(0, _acmA->SetCodecFEC(false));
EXPECT_FALSE(_acmA->CodecFEC());
-
-#endif // defined(WEBRTC_CODEC_G722)
}
int32_t TestRedFec::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode) {
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index eca81f8..ba86719 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -114,13 +114,11 @@
test_cntr_(0),
pack_size_samp_(0),
pack_size_bytes_(0),
- counter_(0)
-#ifdef WEBRTC_CODEC_G722
- , g722_pltype_(0)
-#endif
- , l16_8khz_pltype_(-1)
- , l16_16khz_pltype_(-1)
- , l16_32khz_pltype_(-1)
+ counter_(0),
+ g722_pltype_(0),
+ l16_8khz_pltype_(-1),
+ l16_16khz_pltype_(-1),
+ l16_32khz_pltype_(-1)
#ifdef PCMA_AND_PCMU
, pcma_pltype_(-1)
, pcmu_pltype_(-1)
@@ -128,7 +126,7 @@
#ifdef WEBRTC_CODEC_OPUS
, opus_pltype_(-1)
#endif
- {
+{
// test_mode = 0 for silent test (auto test)
test_mode_ = test_mode;
}
@@ -217,7 +215,6 @@
// All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes.
-#ifdef WEBRTC_CODEC_G722
if (test_mode_ != 0) {
printf("===========================================================\n");
printf("Test number: %d\n", test_cntr_ + 1);
@@ -246,7 +243,7 @@
g722_pltype_);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
-#endif
+
if (test_mode_ != 0) {
printf("===========================================================\n");
printf("Test number: %d\n", test_cntr_ + 1);
@@ -419,7 +416,6 @@
audio_channels = 1;
codec_channels = 2;
-#ifdef WEBRTC_CODEC_G722
if (test_mode_ != 0) {
printf("===============================================================\n");
printf("Test number: %d\n", test_cntr_ + 1);
@@ -432,7 +428,7 @@
g722_pltype_);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
-#endif
+
if (test_mode_ != 0) {
printf("===============================================================\n");
printf("Test number: %d\n", test_cntr_ + 1);
@@ -512,7 +508,6 @@
codec_channels = 1;
channel_a2b_->set_codec_mode(kMono);
-#ifdef WEBRTC_CODEC_G722
// Run stereo audio and mono codec.
if (test_mode_ != 0) {
printf("===============================================================\n");
@@ -533,7 +528,7 @@
EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
-#endif
+
if (test_mode_ != 0) {
printf("===============================================================\n");
printf("Test number: %d\n", test_cntr_ + 1);
@@ -659,9 +654,7 @@
// Print out which codecs were tested, and which were not, in the run.
if (test_mode_ != 0) {
printf("\nThe following codecs was INCLUDED in the test:\n");
-#ifdef WEBRTC_CODEC_G722
printf(" G.722\n");
-#endif
printf(" PCM16\n");
printf(" G.711\n");
#ifdef WEBRTC_CODEC_OPUS
diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
index a27d8d7..a454f25 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -98,9 +98,7 @@
char* send_codec_name_;
// Payload types for stereo codecs and CNG
-#ifdef WEBRTC_CODEC_G722
int g722_pltype_;
-#endif
int l16_8khz_pltype_;
int l16_16khz_pltype_;
int l16_32khz_pltype_;
diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc
index 73625f1..9cd774d 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -63,7 +63,7 @@
#endif
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
- defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+ defined(WEBRTC_CODEC_ILBC)
#if defined(WEBRTC_ANDROID)
TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
#else
diff --git a/modules/media_file/media_file_utility.cc b/modules/media_file/media_file_utility.cc
index a80d4de..b21509f 100644
--- a/modules/media_file/media_file_utility.cc
+++ b/modules/media_file/media_file_utility.cc
@@ -1388,12 +1388,10 @@
}
}
#endif
-#ifdef WEBRTC_CODEC_G722
else if(STR_CASE_CMP(codecInst.plname, "G722") == 0)
{
_codecId = kCodecG722;
}
-#endif
if(_codecId == kCodecNoCodec)
{
return -1;
diff --git a/webrtc.gni b/webrtc.gni
index 7154b87..a4b7c18 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -33,6 +33,9 @@
}
declare_args() {
+ # Include the iLBC audio codec?
+ rtc_include_ilbc = true
+
# Disable this to avoid building the Opus audio codec.
rtc_include_opus = true
@@ -173,9 +176,6 @@
# depend on the possibly overridden variables in the first
# declare_args block.
declare_args() {
- # Include the iLBC audio codec?
- rtc_include_ilbc = !(build_with_chromium || build_with_mozilla)
-
rtc_restrict_logging = build_with_chromium
# Excluded in Chromium since its prerequisites don't require Pulse Audio.
