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Don't link with audio codecs that we don't use

Browse Source 
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}
This commit is contained in:
kwiberg
2015-09-30 21:54:21 -07:00
committed by Commit bot
parent 456696a9c1
commit 98ab3a46d6
19 changed files with 355 additions and 210 deletions
Download Patch File Download Diff File Expand all files Collapse all files

12
webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
View File

@ -38,9 +38,13 @@ AcmReceiveTest::AcmReceiveTest(PacketSource* packet_source,
}
void AcmReceiveTest::RegisterDefaultCodecs() {
#ifdef WEBRTC_CODEC_OPUS
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kOpus, 120));
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
#ifndef WEBRTC_ANDROID
#endif
#ifdef WEBRTC_CODEC_ISAC
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 105));
#endif
@ -56,13 +60,19 @@ void AcmReceiveTest::RegisterDefaultCodecs() {
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU_2ch, 110));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA_2ch, 118));
#ifdef WEBRTC_CODEC_ILBC
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
#endif
#ifdef WEBRTC_CODEC_G722
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722_2ch, 119));
#endif
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
#ifdef WEBRTC_CODEC_RED
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 127));
#endif
}
void AcmReceiveTest::RegisterNetEqTestCodecs() {

8
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
View File

@ -293,7 +293,13 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PostdecodingVad)) {
EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
}
TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) {
#ifdef WEBRTC_CODEC_ISAC
#define IF_ISAC_FLOAT(x) x
#else
#define IF_ISAC_FLOAT(x) DISABLED_##x
#endif
TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(IF_ISAC_FLOAT(LastAudioCodec))) {
const int kCodecId[] = {
ACMCodecDB::kISAC, ACMCodecDB::kPCMA, ACMCodecDB::kISACSWB,
ACMCodecDB::kPCM16Bswb32kHz,

9
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
View File

@ -292,7 +292,14 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
}
TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(LastAudioCodec)) {
#ifdef WEBRTC_CODEC_ISAC
#define IF_ISAC_FLOAT(x) x
#else
#define IF_ISAC_FLOAT(x) DISABLED_##x
#endif
TEST_F(AcmReceiverTestOldApi,
DISABLED_ON_ANDROID(IF_ISAC_FLOAT(LastAudioCodec))) {
const int kCodecId[] = {
ACMCodecDB::kISAC, ACMCodecDB::kPCMA, ACMCodecDB::kISACSWB,
ACMCodecDB::kPCM16Bswb32kHz,

53
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
View File

@ -504,7 +504,13 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest {
test::AudioLoop audio_loop_;
};
TEST_F(AcmIsacMtTest, DoTest) {
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
#define IF_ISAC(x) x
#else
#define IF_ISAC(x) DISABLED_##x
#endif
TEST_F(AcmIsacMtTest, IF_ISAC(DoTest)) {
EXPECT_EQ(kEventSignaled, RunTest());
}
@ -559,13 +565,20 @@ class AcmReceiverBitExactness : public ::testing::Test {
}
};
#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
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_8kHzOutput DISABLED_8kHzOutput
#else
#define MAYBE_8kHzOutput 8kHzOutput
#endif
TEST_F(AcmReceiverBitExactness, MAYBE_8kHzOutput) {
TEST_F(AcmReceiverBitExactness, IF_ALL_CODECS(MAYBE_8kHzOutput)) {
Run(8000,
PlatformChecksum("dcee98c623b147ebe1b40dd30efa896e",
"adc92e173f908f93b96ba5844209815a",
@ -578,7 +591,7 @@ TEST_F(AcmReceiverBitExactness, MAYBE_8kHzOutput) {
#else
#define MAYBE_16kHzOutput 16kHzOutput
#endif
TEST_F(AcmReceiverBitExactness, MAYBE_16kHzOutput) {
TEST_F(AcmReceiverBitExactness, IF_ALL_CODECS(MAYBE_16kHzOutput)) {
Run(16000,
PlatformChecksum("f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
"8cffa6abcb3e18e33b9d857666dff66a",
@ -591,7 +604,7 @@ TEST_F(AcmReceiverBitExactness, MAYBE_16kHzOutput) {
#else
#define MAYBE_32kHzOutput 32kHzOutput
#endif
TEST_F(AcmReceiverBitExactness, MAYBE_32kHzOutput) {
TEST_F(AcmReceiverBitExactness, IF_ALL_CODECS(MAYBE_32kHzOutput)) {
Run(32000,
PlatformChecksum("306e0d990ee6e92de3fbecc0123ece37",
"3e126fe894720c3f85edadcc91964ba5",
@ -604,7 +617,7 @@ TEST_F(AcmReceiverBitExactness, MAYBE_32kHzOutput) {
#else
#define MAYBE_48kHzOutput 48kHzOutput
#endif
TEST_F(AcmReceiverBitExactness, MAYBE_48kHzOutput) {
TEST_F(AcmReceiverBitExactness, IF_ALL_CODECS(MAYBE_48kHzOutput)) {
Run(48000,
PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
"0155665e93067c4e89256b944dd11999",
@ -770,7 +783,7 @@ class AcmSenderBitExactness : public ::testing::Test,
#else
#define MAYBE_IsacWb30ms IsacWb30ms
#endif
TEST_F(AcmSenderBitExactness, MAYBE_IsacWb30ms) {
TEST_F(AcmSenderBitExactness, IF_ISAC(MAYBE_IsacWb30ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 480, 480));
Run(AcmReceiverBitExactness::PlatformChecksum(
"c7e5bdadfa2871df95639fcc297cf23d",
@ -790,7 +803,7 @@ TEST_F(AcmSenderBitExactness, MAYBE_IsacWb30ms) {
#else
#define MAYBE_IsacWb60ms IsacWb60ms
#endif
TEST_F(AcmSenderBitExactness, MAYBE_IsacWb60ms) {
TEST_F(AcmSenderBitExactness, IF_ISAC(MAYBE_IsacWb60ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 960, 960));
Run(AcmReceiverBitExactness::PlatformChecksum(
"14d63c5f08127d280e722e3191b73bdd",
@ -804,7 +817,13 @@ TEST_F(AcmSenderBitExactness, MAYBE_IsacWb60ms) {
test::AcmReceiveTest::kMonoOutput);
}
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IsacSwb30ms)) {
#ifdef WEBRTC_CODEC_ISAC
#define IF_ISAC_FLOAT(x) x
#else
#define IF_ISAC_FLOAT(x) DISABLED_##x
#endif
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IF_ISAC_FLOAT(IsacSwb30ms))) {
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kISACSWB, 1, 104, 960, 960));
Run(AcmReceiverBitExactness::PlatformChecksum(
@ -905,7 +924,13 @@ TEST_F(AcmSenderBitExactness, Pcma_stereo_20ms) {
test::AcmReceiveTest::kStereoOutput);
}
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(Ilbc_30ms)) {
#ifdef WEBRTC_CODEC_ILBC
#define IF_ILBC(x) x
#else
#define IF_ILBC(x) DISABLED_##x
#endif
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IF_ILBC(Ilbc_30ms))) {
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kILBC, 1, 102, 240, 240));
Run(AcmReceiverBitExactness::PlatformChecksum(
"7b6ec10910debd9af08011d3ed5249f7",
@ -919,7 +944,13 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(Ilbc_30ms)) {
test::AcmReceiveTest::kMonoOutput);
}
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_20ms)) {
#ifdef WEBRTC_CODEC_G722
#define IF_G722(x) x
#else
#define IF_G722(x) DISABLED_##x
#endif
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kG722, 1, 9, 320, 160));
Run(AcmReceiverBitExactness::PlatformChecksum(
"7d759436f2533582950d148b5161a36c",
@ -933,7 +964,7 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_20ms)) {
test::AcmReceiveTest::kMonoOutput);
}
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_stereo_20ms)) {
TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IF_G722(G722_stereo_20ms))) {
ASSERT_NO_FATAL_FAILURE(
SetUpTest(acm2::ACMCodecDB::kG722_2ch, 2, 119, 320, 160));
Run(AcmReceiverBitExactness::PlatformChecksum(

63
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
View File

@ -342,9 +342,16 @@ TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
EXPECT_EQ(kAudioFrameSpeech, packet_cb_.last_frame_type());
}
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
#define IF_ISAC(x) x
#else
#define IF_ISAC(x) DISABLED_##x
#endif
// Verifies that the RTP timestamp series is not reset when the codec is
// changed.
TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
TEST_F(AudioCodingModuleTestOldApi,
IF_ISAC(TimestampSeriesContinuesWhenCodecChanges)) {
RegisterCodec(); // This registers the default codec.
uint32_t expected_ts = input_frame_.timestamp_;
int blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
@ -700,7 +707,7 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
test::AudioLoop audio_loop_;
};
TEST_F(AcmIsacMtTestOldApi, DISABLED_ON_IOS(DoTest)) {
TEST_F(AcmIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
EXPECT_EQ(kEventSignaled, RunTest());
}
@ -854,7 +861,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
test::AudioLoop audio_loop_;
};
TEST_F(AcmReRegisterIsacMtTestOldApi, DISABLED_ON_IOS(DoTest)) {
TEST_F(AcmReRegisterIsacMtTestOldApi, DISABLED_ON_IOS(IF_ISAC(DoTest))) {
EXPECT_EQ(kEventSignaled, RunTest());
}
@ -926,13 +933,20 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
}
};
#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
// Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
#define MAYBE_8kHzOutput DISABLED_8kHzOutput
#else
#define MAYBE_8kHzOutput 8kHzOutput
#endif
TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_8kHzOutput) {
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_8kHzOutput)) {
Run(8000, PlatformChecksum("dcee98c623b147ebe1b40dd30efa896e",
"adc92e173f908f93b96ba5844209815a",
"908002dc01fc4eb1d2be24eb1d3f354b"),
@ -945,7 +959,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_8kHzOutput) {
#else
#define MAYBE_16kHzOutput 16kHzOutput
#endif
TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_16kHzOutput) {
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_16kHzOutput)) {
Run(16000, PlatformChecksum("f790e7a8cce4e2c8b7bb5e0e4c5dac0d",
"8cffa6abcb3e18e33b9d857666dff66a",
"a909560b5ca49fa472b17b7b277195e9"),
@ -958,7 +972,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_16kHzOutput) {
#else
#define MAYBE_32kHzOutput 32kHzOutput
#endif
TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_32kHzOutput) {
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_32kHzOutput)) {
Run(32000, PlatformChecksum("306e0d990ee6e92de3fbecc0123ece37",
"3e126fe894720c3f85edadcc91964ba5",
"441aab4b347fb3db4e9244337aca8d8e"),
@ -971,7 +985,7 @@ TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_32kHzOutput) {
#else
#define MAYBE_48kHzOutput 48kHzOutput
#endif
TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_48kHzOutput) {
TEST_F(AcmReceiverBitExactnessOldApi, IF_ALL_CODECS(MAYBE_48kHzOutput)) {
Run(48000, PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
"0155665e93067c4e89256b944dd11999",
"4ee2730fa1daae755e8a8fd3abd779ec"),
@ -984,7 +998,8 @@ TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_48kHzOutput) {
#else
#define MAYBE_48kHzOutputExternalDecoder 48kHzOutputExternalDecoder
#endif
TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_48kHzOutputExternalDecoder) {
TEST_F(AcmReceiverBitExactnessOldApi,
IF_ALL_CODECS(MAYBE_48kHzOutputExternalDecoder)) {
AudioDecoderPcmU decoder(1);
MockAudioDecoder mock_decoder;
// Set expectations on the mock decoder and also delegate the calls to the
@ -1200,7 +1215,7 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
#else
#define MAYBE_IsacWb30ms IsacWb30ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacWb30ms) {
TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(MAYBE_IsacWb30ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"c7e5bdadfa2871df95639fcc297cf23d",
@ -1220,7 +1235,7 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacWb30ms) {
#else
#define MAYBE_IsacWb60ms IsacWb60ms
#endif
TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacWb60ms) {
TEST_F(AcmSenderBitExactnessOldApi, IF_ISAC(MAYBE_IsacWb60ms)) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"14d63c5f08127d280e722e3191b73bdd",
@ -1234,7 +1249,14 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacWb60ms) {
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IsacSwb30ms)) {
#ifdef WEBRTC_CODEC_ISAC
#define IF_ISAC_FLOAT(x) x
#else
#define IF_ISAC_FLOAT(x) DISABLED_##x
#endif
TEST_F(AcmSenderBitExactnessOldApi,
DISABLED_ON_ANDROID(IF_ISAC_FLOAT(IsacSwb30ms))) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"2b3c387d06f00b7b7aad4c9be56fb83d",
@ -1327,7 +1349,13 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
test::AcmReceiveTestOldApi::kStereoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(Ilbc_30ms)) {
#ifdef WEBRTC_CODEC_ILBC
#define IF_ILBC(x) x
#else
#define IF_ILBC(x) DISABLED_##x
#endif
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_ILBC(Ilbc_30ms))) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"7b6ec10910debd9af08011d3ed5249f7",
@ -1341,7 +1369,13 @@ TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(Ilbc_30ms)) {
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_20ms)) {
#ifdef WEBRTC_CODEC_G722
#define IF_G722(x) x
#else
#define IF_G722(x) DISABLED_##x
#endif
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IF_G722(G722_20ms))) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"7d759436f2533582950d148b5161a36c",
@ -1355,7 +1389,8 @@ TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_20ms)) {
test::AcmReceiveTestOldApi::kMonoOutput);
}
TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_stereo_20ms)) {
TEST_F(AcmSenderBitExactnessOldApi,
DISABLED_ON_ANDROID(IF_G722(G722_stereo_20ms))) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"7190ee718ab3d80eca181e5f7140c210",

125
webrtc/modules/audio_coding/main/acm2/codec_owner.cc
View File

@ -15,65 +15,31 @@
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
#ifdef WEBRTC_CODEC_G722
#include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
#endif
#ifdef WEBRTC_CODEC_ILBC
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_encoder_ilbc.h"
#endif
#ifdef WEBRTC_CODEC_ISACFX
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_decoder_isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
#endif
#ifdef WEBRTC_CODEC_ISAC
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_decoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
#endif
#ifdef WEBRTC_CODEC_OPUS
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
#endif
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
#ifdef WEBRTC_CODEC_RED
#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#endif
namespace webrtc {
namespace acm2 {
namespace {
bool IsIsac(const CodecInst& codec) {
return
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
!STR_CASE_CMP(codec.plname, "isac") ||
#endif
false;
}
bool IsOpus(const CodecInst& codec) {
return
#ifdef WEBRTC_CODEC_OPUS
!STR_CASE_CMP(codec.plname, "opus") ||
#endif
false;
}
bool IsPcmU(const CodecInst& codec) {
return !STR_CASE_CMP(codec.plname, "pcmu");
}
bool IsPcmA(const CodecInst& codec) {
return !STR_CASE_CMP(codec.plname, "pcma");
}
bool IsPcm16B(const CodecInst& codec) {
return !STR_CASE_CMP(codec.plname, "l16");
}
bool IsIlbc(const CodecInst& codec) {
return
#ifdef WEBRTC_CODEC_ILBC
!STR_CASE_CMP(codec.plname, "ilbc") ||
#endif
false;
}
bool IsG722(const CodecInst& codec) {
return
#ifdef WEBRTC_CODEC_G722
!STR_CASE_CMP(codec.plname, "g722") ||
#endif
false;
}
} // namespace
CodecOwner::CodecOwner() : external_speech_encoder_(nullptr) {
}
@ -93,57 +59,56 @@ rtc::scoped_ptr<AudioDecoder> CreateIsacDecoder(
#endif
}
rtc::scoped_ptr<AudioEncoder> CreateIsacEncoder(
const CodecInst& speech_inst,
LockedIsacBandwidthInfo* bwinfo) {
#if defined(WEBRTC_CODEC_ISACFX)
return rtc_make_scoped_ptr(new AudioEncoderIsacFix(speech_inst, bwinfo));
#elif defined(WEBRTC_CODEC_ISAC)
return rtc_make_scoped_ptr(new AudioEncoderIsac(speech_inst, bwinfo));
#else
FATAL() << "iSAC is not supported.";
return rtc::scoped_ptr<AudioEncoderMutable>();
#endif
}
// Returns a new speech encoder, or null on error.
// TODO(kwiberg): Don't handle errors here (bug 5033)
rtc::scoped_ptr<AudioEncoder> CreateSpeechEncoder(
const CodecInst& speech_inst,
LockedIsacBandwidthInfo* bwinfo) {
if (IsIsac(speech_inst)) {
return CreateIsacEncoder(speech_inst, bwinfo);
} else if (IsOpus(speech_inst)) {
#if defined(WEBRTC_CODEC_ISACFX)
if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
return rtc_make_scoped_ptr(new AudioEncoderIsacFix(speech_inst, bwinfo));
#endif
#if defined(WEBRTC_CODEC_ISAC)
if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
return rtc_make_scoped_ptr(new AudioEncoderIsac(speech_inst, bwinfo));
#endif
#ifdef WEBRTC_CODEC_OPUS
if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
return rtc_make_scoped_ptr(new AudioEncoderOpus(speech_inst));
} else if (IsPcmU(speech_inst)) {
#endif
if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
return rtc_make_scoped_ptr(new AudioEncoderPcmU(speech_inst));
} else if (IsPcmA(speech_inst)) {
if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
return rtc_make_scoped_ptr(new AudioEncoderPcmA(speech_inst));
} else if (IsPcm16B(speech_inst)) {
if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
return rtc_make_scoped_ptr(new AudioEncoderPcm16B(speech_inst));
} else if (IsIlbc(speech_inst)) {
#ifdef WEBRTC_CODEC_ILBC
if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
return rtc_make_scoped_ptr(new AudioEncoderIlbc(speech_inst));
} else if (IsG722(speech_inst)) {
#endif
#ifdef WEBRTC_CODEC_G722
if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
return rtc_make_scoped_ptr(new AudioEncoderG722(speech_inst));
} else {
LOG_F(LS_ERROR) << "Could not create encoder of type "
<< speech_inst.plname;
return rtc::scoped_ptr<AudioEncoder>();
}
#endif
LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname;
return rtc::scoped_ptr<AudioEncoder>();
}
AudioEncoder* CreateRedEncoder(int red_payload_type,
AudioEncoder* encoder,
rtc::scoped_ptr<AudioEncoder>* red_encoder) {
if (red_payload_type == -1) {
red_encoder->reset();
return encoder;
#ifdef WEBRTC_CODEC_RED
if (red_payload_type != -1) {
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type;
config.speech_encoder = encoder;
red_encoder->reset(new AudioEncoderCopyRed(config));
return red_encoder->get();
}
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type;
config.speech_encoder = encoder;
red_encoder->reset(new AudioEncoderCopyRed(config));
return red_encoder->get();
#endif
red_encoder->reset();
return encoder;
}
void CreateCngEncoder(int cng_payload_type,

12
webrtc/modules/audio_coding/main/acm2/codec_owner.h
View File

@ -15,13 +15,19 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
#else
// Dummy implementation, for when we don't have iSAC.
namespace webrtc {
class LockedIsacBandwidthInfo {};
}
#endif
class AudioDecoder;
namespace webrtc {
namespace acm2 {
class CodecOwner {

22
webrtc/modules/audio_coding/main/audio_coding_module.gypi
View File

@ -11,12 +11,7 @@
'audio_coding_dependencies': [
'cng',
'g711',
'g722',
'ilbc',
'isac',
'isac_fix',
'pcm16b',
'red',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
@ -27,6 +22,23 @@
'audio_coding_dependencies': ['webrtc_opus',],
'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
}],
['build_with_mozilla==0', {
'conditions': [
['target_arch=="arm"', {
'audio_coding_dependencies': ['isac_fix',],
'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',],
}, {
'audio_coding_dependencies': ['isac',],
'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
}],
],
'audio_coding_dependencies': ['g722',],
'audio_coding_defines': ['WEBRTC_CODEC_G722',],
}],
['build_with_mozilla==0 and build_with_chromium==0', {
'audio_coding_dependencies': ['ilbc', 'red',],
'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
}],
],
},
'targets': [

12
webrtc/modules/audio_coding/main/test/TestRedFec.cc
View File

@ -37,11 +37,15 @@ namespace webrtc {
namespace {
const char kNameL16[] = "L16";
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";
const char kNameCN[] = "CN";
const char kNameRED[] = "RED";
#endif
}
TestRedFec::TestRedFec()
@ -104,7 +108,7 @@ void TestRedFec::Perform() {
EXPECT_TRUE(false);
printf("G722 needs to be activated to run this test\n");
return;
#endif
#else
EXPECT_EQ(0, RegisterSendCodec('A', kNameG722, 16000));
EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000));
@ -408,6 +412,8 @@ 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) {

26
webrtc/modules/audio_coding/main/test/Tester.cc
View File

@ -50,7 +50,13 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestRedFec)) {
#ifdef WEBRTC_CODEC_RED
#define IF_RED(x) x
#else
#define IF_RED(x) DISABLED_##x
#endif
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_RED(TestRedFec))) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_fec_trace.txt").c_str());
@ -58,7 +64,13 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestRedFec)) {
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestIsac)) {
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
#define IF_ISAC(x) x
#else
#define IF_ISAC(x) DISABLED_##x
#endif
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_ISAC(TestIsac))) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_isac_trace.txt").c_str());
@ -66,7 +78,15 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestIsac)) {
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TwoWayCommunication)) {
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
#define IF_ALL_CODECS(x) x
#else
#define IF_ALL_CODECS(x) DISABLED_##x
#endif
TEST(AudioCodingModuleTest,
DISABLED_ON_ANDROID(IF_ALL_CODECS(TwoWayCommunication))) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_twowaycom_trace.txt").c_str());