From eb254b40b33380fcec43028dd89f3f6bab3d08a7 Mon Sep 17 00:00:00 2001 From: Karl Wiberg Date: Wed, 1 Nov 2017 15:08:12 +0100 Subject: [PATCH] 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 Reviewed-by: Oskar Sundbom Cr-Commit-Position: refs/heads/master@{#20540} --- api/audio_codecs/BUILD.gn | 34 +++---------------- .../builtin_audio_decoder_factory.cc | 16 ++------- .../builtin_audio_encoder_factory.cc | 16 ++------- modules/audio_coding/BUILD.gn | 12 +++---- .../audio_coding/acm2/acm_codec_database.cc | 6 ---- .../acm2/audio_coding_module_unittest.cc | 6 +--- modules/audio_coding/acm2/rent_a_codec.cc | 4 --- modules/audio_coding/acm2/rent_a_codec.h | 6 ---- modules/audio_coding/audio_coding.gni | 11 +++--- .../builtin_audio_encoder_factory_unittest.cc | 2 -- modules/audio_coding/neteq/neteq_unittest.cc | 3 +- modules/audio_coding/test/TestAllCodecs.cc | 5 --- modules/audio_coding/test/TestRedFec.cc | 11 ------ modules/audio_coding/test/TestStereo.cc | 25 +++++--------- modules/audio_coding/test/TestStereo.h | 2 -- modules/audio_coding/test/Tester.cc | 2 +- modules/media_file/media_file_utility.cc | 2 -- webrtc.gni | 6 ++-- 18 files changed, 34 insertions(+), 135 deletions(-) diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn index fbb2fc5072..8ea533b034 100644 --- a/api/audio_codecs/BUILD.gn +++ b/api/audio_codecs/BUILD.gn @@ -43,6 +43,8 @@ rtc_static_library("builtin_audio_decoder_factory") { "../../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 @@ rtc_static_library("builtin_audio_decoder_factory") { } 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_static_library("builtin_audio_encoder_factory") { "../../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 @@ rtc_static_library("builtin_audio_encoder_factory") { } 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 a6eac7246b..9520d2a9e7 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 @@ rtc::scoped_refptr CreateBuiltinAudioDecoderFactory() { 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 8654d0d2ae..877f85026f 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 @@ rtc::scoped_refptr CreateBuiltinAudioEncoderFactory() { 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 941f13ce86..e5854ef1fc 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -22,14 +22,12 @@ if (rtc_include_ilbc) { 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 7b3b1d2327..4553b52c62 100644 --- a/modules/audio_coding/acm2/acm_codec_database.cc +++ b/modules/audio_coding/acm2/acm_codec_database.cc @@ -85,12 +85,10 @@ const CodecInst ACMCodecDB::database_[] = { #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 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = { #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 @@ const NetEqDecoder ACMCodecDB::neteq_decoders_[] = { #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 1d5b9542d8..ca59b31efa 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 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test { }; #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 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Ilbc_30ms) { #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 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) { "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 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) { "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 ff6689028a..120d54cfe5 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 @@ std::unique_ptr CreateEncoder( if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0) return std::unique_ptr(new AudioEncoderIlbcImpl(speech_inst)); #endif -#ifdef WEBRTC_CODEC_G722 if (STR_CASE_CMP(speech_inst.plname, "g722") == 0) return std::unique_ptr(new AudioEncoderG722Impl(speech_inst)); -#endif LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname; return std::unique_ptr(); } diff --git a/modules/audio_coding/acm2/rent_a_codec.h b/modules/audio_coding/acm2/rent_a_codec.h index f1de72d2cb..cecb914e72 100644 --- a/modules/audio_coding/acm2/rent_a_codec.h +++ b/modules/audio_coding/acm2/rent_a_codec.h @@ -58,10 +58,8 @@ class RentACodec { #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 @@ class RentACodec { #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 41dcf0044c..9b0aba856a 100644 --- a/modules/audio_coding/audio_coding.gni +++ b/modules/audio_coding/audio_coding.gni @@ -20,13 +20,10 @@ if (rtc_opus_support_120ms_ptime) { } 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 ec79c28efe..58bfaed6a8 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 @@ TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) { #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 bb8c4e9b4c..b0f3a39de8 100644 --- a/modules/audio_coding/neteq/neteq_unittest.cc +++ b/modules/audio_coding/neteq/neteq_unittest.cc @@ -450,8 +450,7 @@ void NetEqDecodingTest::PopulateCng(int frame_index, #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 ff28a28807..f7f220f21f 100644 --- a/modules/audio_coding/test/TestAllCodecs.cc +++ b/modules/audio_coding/test/TestAllCodecs.cc @@ -151,7 +151,6 @@ void TestAllCodecs::Perform() { // 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 @@ void TestAllCodecs::Perform() { 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 @@ void TestAllCodecs::Perform() { /* 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 58561c6fe1..034c5954af 100644 --- a/modules/audio_coding/test/TestRedFec.cc +++ b/modules/audio_coding/test/TestRedFec.cc @@ -38,13 +38,9 @@ namespace { 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 @@ void TestRedFec::Perform() { 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 @@ void TestRedFec::Perform() { 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 eca81f81cf..ba86719fe8 100644 --- a/modules/audio_coding/test/TestStereo.cc +++ b/modules/audio_coding/test/TestStereo.cc @@ -114,13 +114,11 @@ TestStereo::TestStereo(int test_mode) 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 @@ TestStereo::TestStereo(int test_mode) #ifdef WEBRTC_CODEC_OPUS , opus_pltype_(-1) #endif - { +{ // test_mode = 0 for silent test (auto test) test_mode_ = test_mode; } @@ -217,7 +215,6 @@ void TestStereo::Perform() { // 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 @@ void TestStereo::Perform() { 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 @@ void TestStereo::Perform() { 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 @@ void TestStereo::Perform() { 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 @@ void TestStereo::Perform() { 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 @@ void TestStereo::Perform() { 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 @@ void TestStereo::Perform() { // 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 a27d8d7206..a454f25639 100644 --- a/modules/audio_coding/test/TestStereo.h +++ b/modules/audio_coding/test/TestStereo.h @@ -98,9 +98,7 @@ class TestStereo : public ACMTest { 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 73625f1708..9cd774dd15 100644 --- a/modules/audio_coding/test/Tester.cc +++ b/modules/audio_coding/test/Tester.cc @@ -63,7 +63,7 @@ TEST(AudioCodingModuleTest, TestIsac) { #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 a80d4ded16..b21509fb13 100644 --- a/modules/media_file/media_file_utility.cc +++ b/modules/media_file/media_file_utility.cc @@ -1388,12 +1388,10 @@ int32_t ModuleFileUtility::set_codec_info(const CodecInst& codecInst) } } #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 7154b87605..a4b7c1881f 100644 --- a/webrtc.gni +++ b/webrtc.gni @@ -33,6 +33,9 @@ if (is_ios) { } 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 @@ declare_args() { # 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.