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Don't select audio codecs depending on GN vars build_with_{chromium|mozilla}

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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}
This commit is contained in:
Karl Wiberg
2017-11-01 15:08:12 +01:00
committed by Commit Bot
parent e0aca5e320
commit eb254b40b3
18 changed files with 34 additions and 135 deletions
Download Patch File Download Diff File Expand all files Collapse all files

34
api/audio_codecs/BUILD.gn
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@ -43,6 +43,8 @@ rtc_static_library("builtin_audio_decoder_factory") {
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
"L16:audio_decoder_L16", "L16:audio_decoder_L16",
"g711:audio_decoder_g711", "g711:audio_decoder_g711",
"g722:audio_decoder_g722",
"isac:audio_decoder_isac",
] ]
defines = [] defines = []
if (rtc_include_ilbc) { if (rtc_include_ilbc) {
@ -57,21 +59,6 @@ rtc_static_library("builtin_audio_decoder_factory") {
} else { } else {
defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ] 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") { rtc_static_library("builtin_audio_encoder_factory") {
@ -84,6 +71,8 @@ rtc_static_library("builtin_audio_encoder_factory") {
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
"L16:audio_encoder_L16", "L16:audio_encoder_L16",
"g711:audio_encoder_g711", "g711:audio_encoder_g711",
"g722:audio_encoder_g722",
"isac:audio_encoder_isac",
] ]
defines = [] defines = []
if (rtc_include_ilbc) { if (rtc_include_ilbc) {
@ -98,19 +87,4 @@ rtc_static_library("builtin_audio_encoder_factory") {
} else { } else {
defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ] 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",
]
}
} }

16
api/audio_codecs/builtin_audio_decoder_factory.cc
View File

@ -16,15 +16,11 @@
#include "api/audio_codecs/L16/audio_decoder_L16.h" #include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/audio_decoder_factory_template.h" #include "api/audio_codecs/audio_decoder_factory_template.h"
#include "api/audio_codecs/g711/audio_decoder_g711.h" #include "api/audio_codecs/g711/audio_decoder_g711.h"
#if WEBRTC_USE_BUILTIN_G722 #include "api/audio_codecs/g722/audio_decoder_g722.h"
#include "api/audio_codecs/g722/audio_decoder_g722.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_ILBC #if WEBRTC_USE_BUILTIN_ILBC
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck #include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck
#endif #endif
#if WEBRTC_USE_BUILTIN_ISAC #include "api/audio_codecs/isac/audio_decoder_isac.h"
#include "api/audio_codecs/isac/audio_decoder_isac.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_OPUS #if WEBRTC_USE_BUILTIN_OPUS
#include "api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck #include "api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck
#endif #endif
@ -57,13 +53,7 @@ rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
AudioDecoderOpus, AudioDecoderOpus,
#endif #endif
#if WEBRTC_USE_BUILTIN_ISAC AudioDecoderIsac, AudioDecoderG722,
AudioDecoderIsac,
#endif
#if WEBRTC_USE_BUILTIN_G722
AudioDecoderG722,
#endif
#if WEBRTC_USE_BUILTIN_ILBC #if WEBRTC_USE_BUILTIN_ILBC
AudioDecoderIlbc, AudioDecoderIlbc,

16
api/audio_codecs/builtin_audio_encoder_factory.cc
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@ -16,15 +16,11 @@
#include "api/audio_codecs/L16/audio_encoder_L16.h" #include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/audio_encoder_factory_template.h" #include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/g711/audio_encoder_g711.h" #include "api/audio_codecs/g711/audio_encoder_g711.h"
#if WEBRTC_USE_BUILTIN_G722 #include "api/audio_codecs/g722/audio_encoder_g722.h"
#include "api/audio_codecs/g722/audio_encoder_g722.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_ILBC #if WEBRTC_USE_BUILTIN_ILBC
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck #include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck
#endif #endif
#if WEBRTC_USE_BUILTIN_ISAC #include "api/audio_codecs/isac/audio_encoder_isac.h"
#include "api/audio_codecs/isac/audio_encoder_isac.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_OPUS #if WEBRTC_USE_BUILTIN_OPUS
#include "api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck #include "api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck
#endif #endif
@ -61,13 +57,7 @@ rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory() {
AudioEncoderOpus, AudioEncoderOpus,
#endif #endif
#if WEBRTC_USE_BUILTIN_ISAC AudioEncoderIsac, AudioEncoderG722,
AudioEncoderIsac,
#endif
#if WEBRTC_USE_BUILTIN_G722
AudioEncoderG722,
#endif
#if WEBRTC_USE_BUILTIN_ILBC #if WEBRTC_USE_BUILTIN_ILBC
AudioEncoderIlbc, AudioEncoderIlbc,

8
modules/audio_coding/BUILD.gn
View File

@ -22,14 +22,12 @@ if (rtc_include_ilbc) {
if (rtc_include_opus) { if (rtc_include_opus) {
audio_codec_deps += [ ":webrtc_opus" ] audio_codec_deps += [ ":webrtc_opus" ]
} }
if (!build_with_mozilla) { if (current_cpu == "arm") {
if (current_cpu == "arm") {
audio_codec_deps += [ ":isac_fix" ] audio_codec_deps += [ ":isac_fix" ]
} else { } else {
audio_codec_deps += [ ":isac" ] audio_codec_deps += [ ":isac" ]
}
audio_codec_deps += [ ":g722" ]
} }
audio_codec_deps += [ ":g722" ]
if (!build_with_mozilla && !build_with_chromium) { if (!build_with_mozilla && !build_with_chromium) {
audio_codec_deps += [ ":red" ] audio_codec_deps += [ ":red" ]
} }

6
modules/audio_coding/acm2/acm_codec_database.cc
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@ -85,12 +85,10 @@ const CodecInst ACMCodecDB::database_[] = {
#ifdef WEBRTC_CODEC_ILBC #ifdef WEBRTC_CODEC_ILBC
{102, "ILBC", 8000, 240, 1, 13300}, {102, "ILBC", 8000, 240, 1, 13300},
#endif #endif
#ifdef WEBRTC_CODEC_G722
// Mono // Mono
{9, "G722", 16000, 320, 1, 64000}, {9, "G722", 16000, 320, 1, 64000},
// Stereo // Stereo
{119, "G722", 16000, 320, 2, 64000}, {119, "G722", 16000, 320, 2, 64000},
#endif
#ifdef WEBRTC_CODEC_OPUS #ifdef WEBRTC_CODEC_OPUS
// Opus internally supports 48, 24, 16, 12, 8 kHz. // Opus internally supports 48, 24, 16, 12, 8 kHz.
// Mono and stereo. // Mono and stereo.
@ -143,12 +141,10 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
#ifdef WEBRTC_CODEC_ILBC #ifdef WEBRTC_CODEC_ILBC
{4, {160, 240, 320, 480}, 0, 1}, {4, {160, 240, 320, 480}, 0, 1},
#endif #endif
#ifdef WEBRTC_CODEC_G722
// Mono // Mono
{6, {160, 320, 480, 640, 800, 960}, 0, 2}, {6, {160, 320, 480, 640, 800, 960}, 0, 2},
// Stereo // Stereo
{6, {160, 320, 480, 640, 800, 960}, 0, 2}, {6, {160, 320, 480, 640, 800, 960}, 0, 2},
#endif
#ifdef WEBRTC_CODEC_OPUS #ifdef WEBRTC_CODEC_OPUS
// Opus supports frames shorter than 10ms, // Opus supports frames shorter than 10ms,
// but it doesn't help us to use them. // but it doesn't help us to use them.
@ -200,12 +196,10 @@ const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
#ifdef WEBRTC_CODEC_ILBC #ifdef WEBRTC_CODEC_ILBC
NetEqDecoder::kDecoderILBC, NetEqDecoder::kDecoderILBC,
#endif #endif
#ifdef WEBRTC_CODEC_G722
// Mono // Mono
NetEqDecoder::kDecoderG722, NetEqDecoder::kDecoderG722,
// Stereo // Stereo
NetEqDecoder::kDecoderG722_2ch, NetEqDecoder::kDecoderG722_2ch,
#endif
#ifdef WEBRTC_CODEC_OPUS #ifdef WEBRTC_CODEC_OPUS
// Mono and stereo. // Mono and stereo.
NetEqDecoder::kDecoderOpus, NetEqDecoder::kDecoderOpus,

6
modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -980,7 +980,7 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
}; };
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ #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) { TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
Run(8000, PlatformChecksum("2adede965c6f87de7142c51552111d08", Run(8000, PlatformChecksum("2adede965c6f87de7142c51552111d08",
"028c0fc414b1c9ab7e582dccdf381e98", "028c0fc414b1c9ab7e582dccdf381e98",
@ -1438,7 +1438,6 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Ilbc_30ms) {
#else #else
#define MAYBE_G722_20ms G722_20ms #define MAYBE_G722_20ms G722_20ms
#endif #endif
#if defined(WEBRTC_CODEC_G722)
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) { TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160)); ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
@ -1451,14 +1450,12 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_20ms) {
"android_arm64_payload", "android_arm64_clang_payload"), "android_arm64_payload", "android_arm64_clang_payload"),
50, test::AcmReceiveTestOldApi::kMonoOutput); 50, test::AcmReceiveTestOldApi::kMonoOutput);
} }
#endif
#if defined(WEBRTC_ANDROID) #if defined(WEBRTC_ANDROID)
#define MAYBE_G722_stereo_20ms DISABLED_G722_stereo_20ms #define MAYBE_G722_stereo_20ms DISABLED_G722_stereo_20ms
#else #else
#define MAYBE_G722_stereo_20ms G722_stereo_20ms #define MAYBE_G722_stereo_20ms G722_stereo_20ms
#endif #endif
#if defined(WEBRTC_CODEC_G722)
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) { TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160)); ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
@ -1471,7 +1468,6 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
"android_arm64_payload", "android_arm64_clang_payload"), "android_arm64_payload", "android_arm64_clang_payload"),
50, test::AcmReceiveTestOldApi::kStereoOutput); 50, test::AcmReceiveTestOldApi::kStereoOutput);
} }
#endif
TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) { TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960)); ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));

4
modules/audio_coding/acm2/rent_a_codec.cc
View File

@ -16,9 +16,7 @@
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "rtc_base/logging.h" #include "rtc_base/logging.h"
#ifdef WEBRTC_CODEC_G722
#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h" #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
#endif
#ifdef WEBRTC_CODEC_ILBC #ifdef WEBRTC_CODEC_ILBC
#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h" #include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#endif #endif
@ -175,10 +173,8 @@ std::unique_ptr<AudioEncoder> CreateEncoder(
if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0) if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
return std::unique_ptr<AudioEncoder>(new AudioEncoderIlbcImpl(speech_inst)); return std::unique_ptr<AudioEncoder>(new AudioEncoderIlbcImpl(speech_inst));
#endif #endif
#ifdef WEBRTC_CODEC_G722
if (STR_CASE_CMP(speech_inst.plname, "g722") == 0) if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
return std::unique_ptr<AudioEncoder>(new AudioEncoderG722Impl(speech_inst)); return std::unique_ptr<AudioEncoder>(new AudioEncoderG722Impl(speech_inst));
#endif
LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname; LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname;
return std::unique_ptr<AudioEncoder>(); return std::unique_ptr<AudioEncoder>();
} }

6
modules/audio_coding/acm2/rent_a_codec.h
View File

@ -58,10 +58,8 @@ class RentACodec {
#ifdef WEBRTC_CODEC_ILBC #ifdef WEBRTC_CODEC_ILBC
kILBC, kILBC,
#endif #endif
#ifdef WEBRTC_CODEC_G722
kG722, // Mono kG722, // Mono
kG722_2ch, // Stereo kG722_2ch, // Stereo
#endif
#ifdef WEBRTC_CODEC_OPUS #ifdef WEBRTC_CODEC_OPUS
kOpus, // Mono and stereo kOpus, // Mono and stereo
#endif #endif
@ -92,10 +90,6 @@ class RentACodec {
#ifndef WEBRTC_CODEC_ILBC #ifndef WEBRTC_CODEC_ILBC
kILBC = -1, kILBC = -1,
#endif #endif
#ifndef WEBRTC_CODEC_G722
kG722 = -1, // Mono
kG722_2ch = -1, // Stereo
#endif
#ifndef WEBRTC_CODEC_OPUS #ifndef WEBRTC_CODEC_OPUS
kOpus = -1, // Mono and stereo kOpus = -1, // Mono and stereo
#endif #endif

7
modules/audio_coding/audio_coding.gni
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@ -20,13 +20,10 @@ if (rtc_opus_support_120ms_ptime) {
} else { } else {
audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ] audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
} }
if (!build_with_mozilla) { if (current_cpu == "arm") {
if (current_cpu == "arm") {
audio_codec_defines += [ "WEBRTC_CODEC_ISACFX" ] audio_codec_defines += [ "WEBRTC_CODEC_ISACFX" ]
} else { } else {
audio_codec_defines += [ "WEBRTC_CODEC_ISAC" ] audio_codec_defines += [ "WEBRTC_CODEC_ISAC" ]
}
audio_codec_defines += [ "WEBRTC_CODEC_G722" ]
} }
if (!build_with_mozilla && !build_with_chromium) { if (!build_with_mozilla && !build_with_chromium) {
audio_codec_defines += [ "WEBRTC_CODEC_RED" ] audio_codec_defines += [ "WEBRTC_CODEC_RED" ]

2
modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
View File

@ -131,9 +131,7 @@ TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) {
#ifdef WEBRTC_CODEC_ISAC #ifdef WEBRTC_CODEC_ISAC
{"isac", 32000, 1}, {"isac", 32000, 1},
#endif #endif
#ifdef WEBRTC_CODEC_G722
{"G722", 8000, 1}, {"G722", 8000, 1},
#endif
#ifdef WEBRTC_CODEC_ILBC #ifdef WEBRTC_CODEC_ILBC
{"ilbc", 8000, 1}, {"ilbc", 8000, 1},
#endif #endif

3
modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -450,8 +450,7 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ #if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) && \ defined(WEBRTC_CODEC_ILBC) && !defined(WEBRTC_ARCH_ARM64)
!defined(WEBRTC_ARCH_ARM64)
#define MAYBE_TestBitExactness TestBitExactness #define MAYBE_TestBitExactness TestBitExactness
#else #else
#define MAYBE_TestBitExactness DISABLED_TestBitExactness #define MAYBE_TestBitExactness DISABLED_TestBitExactness

5
modules/audio_coding/test/TestAllCodecs.cc
View File

@ -151,7 +151,6 @@ void TestAllCodecs::Perform() {
// All codecs are tested for all allowed sampling frequencies, rates and // All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes. // packet sizes.
#ifdef WEBRTC_CODEC_G722
if (test_mode_ != 0) { if (test_mode_ != 0) {
printf("===============================================================\n"); printf("===============================================================\n");
} }
@ -171,7 +170,6 @@ void TestAllCodecs::Perform() {
RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0); RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0);
Run(channel_a_to_b_); Run(channel_a_to_b_);
outfile_b_.Close(); outfile_b_.Close();
#endif
#ifdef WEBRTC_CODEC_ILBC #ifdef WEBRTC_CODEC_ILBC
if (test_mode_ != 0) { if (test_mode_ != 0) {
printf("===============================================================\n"); printf("===============================================================\n");
@ -324,9 +322,6 @@ void TestAllCodecs::Perform() {
/* Print out all codecs that were not tested in the run */ /* Print out all codecs that were not tested in the run */
printf("The following codecs was not included in the test:\n"); printf("The following codecs was not included in the test:\n");
#ifndef WEBRTC_CODEC_G722
printf(" G.722\n");
#endif
#ifndef WEBRTC_CODEC_ILBC #ifndef WEBRTC_CODEC_ILBC
printf(" iLBC\n"); printf(" iLBC\n");
#endif #endif

11
modules/audio_coding/test/TestRedFec.cc
View File

@ -38,13 +38,9 @@ namespace {
const char kNamePCMU[] = "PCMU"; const char kNamePCMU[] = "PCMU";
const char kNameCN[] = "CN"; const char kNameCN[] = "CN";
const char kNameRED[] = "RED"; 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 kNameISAC[] = "ISAC";
const char kNameG722[] = "G722"; const char kNameG722[] = "G722";
const char kNameOPUS[] = "opus"; const char kNameOPUS[] = "opus";
#endif
} }
TestRedFec::TestRedFec() TestRedFec::TestRedFec()
@ -104,11 +100,6 @@ void TestRedFec::Perform() {
Run(); Run();
_outFileB.Close(); _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', kNameG722, 16000));
EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000)); EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000));
@ -412,8 +403,6 @@ void TestRedFec::Perform() {
EXPECT_FALSE(_acmA->REDStatus()); EXPECT_FALSE(_acmA->REDStatus());
EXPECT_EQ(0, _acmA->SetCodecFEC(false)); EXPECT_EQ(0, _acmA->SetCodecFEC(false));
EXPECT_FALSE(_acmA->CodecFEC()); EXPECT_FALSE(_acmA->CodecFEC());
#endif // defined(WEBRTC_CODEC_G722)
} }
int32_t TestRedFec::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode) { int32_t TestRedFec::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode) {

25
modules/audio_coding/test/TestStereo.cc
View File

@ -114,13 +114,11 @@ TestStereo::TestStereo(int test_mode)
test_cntr_(0), test_cntr_(0),
pack_size_samp_(0), pack_size_samp_(0),
pack_size_bytes_(0), pack_size_bytes_(0),
counter_(0) counter_(0),
#ifdef WEBRTC_CODEC_G722 g722_pltype_(0),
, g722_pltype_(0) l16_8khz_pltype_(-1),
#endif l16_16khz_pltype_(-1),
, l16_8khz_pltype_(-1) l16_32khz_pltype_(-1)
, l16_16khz_pltype_(-1)
, l16_32khz_pltype_(-1)
#ifdef PCMA_AND_PCMU #ifdef PCMA_AND_PCMU
, pcma_pltype_(-1) , pcma_pltype_(-1)
, pcmu_pltype_(-1) , pcmu_pltype_(-1)
@ -128,7 +126,7 @@ TestStereo::TestStereo(int test_mode)
#ifdef WEBRTC_CODEC_OPUS #ifdef WEBRTC_CODEC_OPUS
, opus_pltype_(-1) , opus_pltype_(-1)
#endif #endif
{ {
// test_mode = 0 for silent test (auto test) // test_mode = 0 for silent test (auto test)
test_mode_ = test_mode; test_mode_ = test_mode;
} }
@ -217,7 +215,6 @@ void TestStereo::Perform() {
// All codecs are tested for all allowed sampling frequencies, rates and // All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes. // packet sizes.
#ifdef WEBRTC_CODEC_G722
if (test_mode_ != 0) { if (test_mode_ != 0) {
printf("===========================================================\n"); printf("===========================================================\n");
printf("Test number: %d\n", test_cntr_ + 1); printf("Test number: %d\n", test_cntr_ + 1);
@ -246,7 +243,7 @@ void TestStereo::Perform() {
g722_pltype_); g722_pltype_);
Run(channel_a2b_, audio_channels, codec_channels); Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close(); out_file_.Close();
#endif
if (test_mode_ != 0) { if (test_mode_ != 0) {
printf("===========================================================\n"); printf("===========================================================\n");
printf("Test number: %d\n", test_cntr_ + 1); printf("Test number: %d\n", test_cntr_ + 1);
@ -419,7 +416,6 @@ void TestStereo::Perform() {
audio_channels = 1; audio_channels = 1;
codec_channels = 2; codec_channels = 2;
#ifdef WEBRTC_CODEC_G722
if (test_mode_ != 0) { if (test_mode_ != 0) {
printf("===============================================================\n"); printf("===============================================================\n");
printf("Test number: %d\n", test_cntr_ + 1); printf("Test number: %d\n", test_cntr_ + 1);
@ -432,7 +428,7 @@ void TestStereo::Perform() {
g722_pltype_); g722_pltype_);
Run(channel_a2b_, audio_channels, codec_channels); Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close(); out_file_.Close();
#endif
if (test_mode_ != 0) { if (test_mode_ != 0) {
printf("===============================================================\n"); printf("===============================================================\n");
printf("Test number: %d\n", test_cntr_ + 1); printf("Test number: %d\n", test_cntr_ + 1);
@ -512,7 +508,6 @@ void TestStereo::Perform() {
codec_channels = 1; codec_channels = 1;
channel_a2b_->set_codec_mode(kMono); channel_a2b_->set_codec_mode(kMono);
#ifdef WEBRTC_CODEC_G722
// Run stereo audio and mono codec. // Run stereo audio and mono codec.
if (test_mode_ != 0) { if (test_mode_ != 0) {
printf("===============================================================\n"); printf("===============================================================\n");
@ -533,7 +528,7 @@ void TestStereo::Perform() {
EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal)); EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
Run(channel_a2b_, audio_channels, codec_channels); Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close(); out_file_.Close();
#endif
if (test_mode_ != 0) { if (test_mode_ != 0) {
printf("===============================================================\n"); printf("===============================================================\n");
printf("Test number: %d\n", test_cntr_ + 1); 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. // Print out which codecs were tested, and which were not, in the run.
if (test_mode_ != 0) { if (test_mode_ != 0) {
printf("\nThe following codecs was INCLUDED in the test:\n"); printf("\nThe following codecs was INCLUDED in the test:\n");
#ifdef WEBRTC_CODEC_G722
printf(" G.722\n"); printf(" G.722\n");
#endif
printf(" PCM16\n"); printf(" PCM16\n");
printf(" G.711\n"); printf(" G.711\n");
#ifdef WEBRTC_CODEC_OPUS #ifdef WEBRTC_CODEC_OPUS

2
modules/audio_coding/test/TestStereo.h
View File

@ -98,9 +98,7 @@ class TestStereo : public ACMTest {
char* send_codec_name_; char* send_codec_name_;
// Payload types for stereo codecs and CNG // Payload types for stereo codecs and CNG
#ifdef WEBRTC_CODEC_G722
int g722_pltype_; int g722_pltype_;
#endif
int l16_8khz_pltype_; int l16_8khz_pltype_;
int l16_16khz_pltype_; int l16_16khz_pltype_;
int l16_32khz_pltype_; int l16_32khz_pltype_;

2
modules/audio_coding/test/Tester.cc
View File

@ -63,7 +63,7 @@ TEST(AudioCodingModuleTest, TestIsac) {
#endif #endif
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ #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) #if defined(WEBRTC_ANDROID)
TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) { TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
#else #else

2
modules/media_file/media_file_utility.cc
View File

@ -1388,12 +1388,10 @@ int32_t ModuleFileUtility::set_codec_info(const CodecInst& codecInst)
} }
} }
#endif #endif
#ifdef WEBRTC_CODEC_G722
else if(STR_CASE_CMP(codecInst.plname, "G722") == 0) else if(STR_CASE_CMP(codecInst.plname, "G722") == 0)
{ {
_codecId = kCodecG722; _codecId = kCodecG722;
} }
#endif
if(_codecId == kCodecNoCodec) if(_codecId == kCodecNoCodec)
{ {
return -1; return -1;

6
webrtc.gni
View File

@ -33,6 +33,9 @@ if (is_ios) {
} }
declare_args() { declare_args() {
# Include the iLBC audio codec?
rtc_include_ilbc = true
# Disable this to avoid building the Opus audio codec. # Disable this to avoid building the Opus audio codec.
rtc_include_opus = true rtc_include_opus = true
@ -173,9 +176,6 @@ declare_args() {
# depend on the possibly overridden variables in the first # depend on the possibly overridden variables in the first
# declare_args block. # declare_args block.
declare_args() { declare_args() {
# Include the iLBC audio codec?
rtc_include_ilbc = !(build_with_chromium || build_with_mozilla)
rtc_restrict_logging = build_with_chromium rtc_restrict_logging = build_with_chromium
# Excluded in Chromium since its prerequisites don't require Pulse Audio. # Excluded in Chromium since its prerequisites don't require Pulse Audio.