From 6ff045f097529ab71fbe9233a712c4bdce3f440d Mon Sep 17 00:00:00 2001 From: kwiberg Date: Thu, 17 Aug 2017 05:31:02 -0700 Subject: [PATCH] Give Audio{De,En}coderIsac* an "Impl" suffix, to free up the original names I want to publish an API for iSAC in webrtc/api/, and I want to use the class names Audio{De,En}coderIsac{Fix,Float}. BUG=webrtc:7835, webrtc:7841 Review-Url: https://codereview.webrtc.org/2996593002 Cr-Commit-Position: refs/heads/master@{#19381} --- .../acm2/audio_coding_module_unittest.cc | 6 +++--- .../modules/audio_coding/acm2/rent_a_codec.cc | 8 ++++---- .../builtin_audio_decoder_factory_internal.cc | 4 ++-- .../builtin_audio_encoder_factory_internal.cc | 4 ++-- .../isac/fix/include/audio_decoder_isacfix.h | 2 +- .../isac/fix/include/audio_encoder_isacfix.h | 2 +- .../isac/main/include/audio_decoder_isac.h | 2 +- .../isac/main/include/audio_encoder_isac.h | 2 +- .../main/source/audio_encoder_isac_unittest.cc | 8 ++++---- .../neteq/audio_decoder_unittest.cc | 18 +++++++++--------- .../test/fuzzers/audio_decoder_isac_fuzzer.cc | 2 +- ...udio_decoder_isac_incoming_packet_fuzzer.cc | 2 +- .../fuzzers/audio_decoder_isacfix_fuzzer.cc | 2 +- 13 files changed, 31 insertions(+), 31 deletions(-) diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc index c16c5b1890..1d8571d017 100644 --- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -753,9 +753,9 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi { receive_packet_count_(0), next_insert_packet_time_ms_(0), fake_clock_(new SimulatedClock(0)) { - AudioEncoderIsac::Config config; + AudioEncoderIsacFloatImpl::Config config; config.payload_type = kPayloadType; - isac_encoder_.reset(new AudioEncoderIsac(config)); + isac_encoder_.reset(new AudioEncoderIsacFloatImpl(config)); clock_ = fake_clock_.get(); } @@ -882,7 +882,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi { bool codec_registered_ GUARDED_BY(crit_sect_); int receive_packet_count_ GUARDED_BY(crit_sect_); int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_); - std::unique_ptr isac_encoder_; + std::unique_ptr isac_encoder_; std::unique_ptr fake_clock_; test::AudioLoop audio_loop_; }; diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc index 491df316ba..3bc1464908 100644 --- a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc +++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc @@ -154,12 +154,12 @@ std::unique_ptr CreateEncoder( #if defined(WEBRTC_CODEC_ISACFX) if (STR_CASE_CMP(speech_inst.plname, "isac") == 0) return std::unique_ptr( - new AudioEncoderIsacFix(speech_inst, bwinfo)); + new AudioEncoderIsacFixImpl(speech_inst, bwinfo)); #endif #if defined(WEBRTC_CODEC_ISAC) if (STR_CASE_CMP(speech_inst.plname, "isac") == 0) return std::unique_ptr( - new AudioEncoderIsac(speech_inst, bwinfo)); + new AudioEncoderIsacFloatImpl(speech_inst, bwinfo)); #endif #ifdef WEBRTC_CODEC_OPUS if (STR_CASE_CMP(speech_inst.plname, "opus") == 0) @@ -229,10 +229,10 @@ std::unique_ptr CreateIsacDecoder( const rtc::scoped_refptr& bwinfo) { #if defined(WEBRTC_CODEC_ISACFX) return std::unique_ptr( - new AudioDecoderIsacFix(sample_rate_hz, bwinfo)); + new AudioDecoderIsacFixImpl(sample_rate_hz, bwinfo)); #elif defined(WEBRTC_CODEC_ISAC) return std::unique_ptr( - new AudioDecoderIsac(sample_rate_hz, bwinfo)); + new AudioDecoderIsacFloatImpl(sample_rate_hz, bwinfo)); #else FATAL() << "iSAC is not supported."; return std::unique_ptr(); diff --git a/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc b/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc index 3624c79f27..f853cbda31 100644 --- a/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc +++ b/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc @@ -91,7 +91,7 @@ NamedDecoderConstructor decoder_constructors[] = { [](const SdpAudioFormat& format, std::unique_ptr* out) { if (format.clockrate_hz == 16000 && format.num_channels == 1) { if (out) { - out->reset(new AudioDecoderIsacFix(format.clockrate_hz)); + out->reset(new AudioDecoderIsacFixImpl(format.clockrate_hz)); } return true; } else { @@ -104,7 +104,7 @@ NamedDecoderConstructor decoder_constructors[] = { if ((format.clockrate_hz == 16000 || format.clockrate_hz == 32000) && format.num_channels == 1) { if (out) { - out->reset(new AudioDecoderIsac(format.clockrate_hz)); + out->reset(new AudioDecoderIsacFloatImpl(format.clockrate_hz)); } return true; } else { diff --git a/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc b/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc index 2876440525..b44268ab07 100644 --- a/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc +++ b/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc @@ -68,9 +68,9 @@ NamedEncoderFactory encoder_factories[] = { NamedEncoderFactory::ForEncoder(), #endif #if defined(WEBRTC_CODEC_ISACFX) - NamedEncoderFactory::ForEncoder(), + NamedEncoderFactory::ForEncoder(), #elif defined(WEBRTC_CODEC_ISAC) - NamedEncoderFactory::ForEncoder(), + NamedEncoderFactory::ForEncoder(), #endif #ifdef WEBRTC_CODEC_OPUS diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h index e78eb786ad..4ddc3bb05c 100644 --- a/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h +++ b/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h @@ -16,7 +16,7 @@ namespace webrtc { -using AudioDecoderIsacFix = AudioDecoderIsacT; +using AudioDecoderIsacFixImpl = AudioDecoderIsacT; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_ diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h index b97f04bbf2..aefad78eb6 100644 --- a/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h +++ b/webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h @@ -16,7 +16,7 @@ namespace webrtc { -using AudioEncoderIsacFix = AudioEncoderIsacT; +using AudioEncoderIsacFixImpl = AudioEncoderIsacT; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_ diff --git a/webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h index dcd4852a68..06821c0447 100644 --- a/webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h +++ b/webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h @@ -16,7 +16,7 @@ namespace webrtc { -using AudioDecoderIsac = AudioDecoderIsacT; +using AudioDecoderIsacFloatImpl = AudioDecoderIsacT; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ diff --git a/webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h b/webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h index cc8665d6b7..06bef4c032 100644 --- a/webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h +++ b/webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h @@ -16,7 +16,7 @@ namespace webrtc { -using AudioEncoderIsac = AudioEncoderIsacT; +using AudioEncoderIsacFloatImpl = AudioEncoderIsacT; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc index e9ca570012..3fe8c1a709 100644 --- a/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc +++ b/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc @@ -17,13 +17,13 @@ namespace webrtc { namespace { -void TestBadConfig(const AudioEncoderIsac::Config& config) { +void TestBadConfig(const AudioEncoderIsacFloatImpl::Config& config) { EXPECT_FALSE(config.IsOk()); } -void TestGoodConfig(const AudioEncoderIsac::Config& config) { +void TestGoodConfig(const AudioEncoderIsacFloatImpl::Config& config) { EXPECT_TRUE(config.IsOk()); - AudioEncoderIsac aei(config); + AudioEncoderIsacFloatImpl aei(config); } // Wrap subroutine calls that test things in this, so that the error messages @@ -34,7 +34,7 @@ void TestGoodConfig(const AudioEncoderIsac::Config& config) { } // namespace TEST(AudioEncoderIsacTest, TestConfigBitrate) { - AudioEncoderIsac::Config config; + AudioEncoderIsacFloatImpl::Config config; // The default value is some real, positive value. EXPECT_GT(config.bit_rate, 1); diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc index a34594a004..f6ae2ddaac 100644 --- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -350,14 +350,14 @@ class AudioDecoderIsacFloatTest : public AudioDecoderTest { codec_input_rate_hz_ = 16000; frame_size_ = 480; data_length_ = 10 * frame_size_; - AudioEncoderIsac::Config config; + AudioEncoderIsacFloatImpl::Config config; config.payload_type = payload_type_; config.sample_rate_hz = codec_input_rate_hz_; config.adaptive_mode = false; config.frame_size_ms = 1000 * static_cast(frame_size_) / codec_input_rate_hz_; - audio_encoder_.reset(new AudioEncoderIsac(config)); - decoder_ = new AudioDecoderIsac(codec_input_rate_hz_); + audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config)); + decoder_ = new AudioDecoderIsacFloatImpl(codec_input_rate_hz_); } }; @@ -367,14 +367,14 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest { codec_input_rate_hz_ = 32000; frame_size_ = 960; data_length_ = 10 * frame_size_; - AudioEncoderIsac::Config config; + AudioEncoderIsacFloatImpl::Config config; config.payload_type = payload_type_; config.sample_rate_hz = codec_input_rate_hz_; config.adaptive_mode = false; config.frame_size_ms = 1000 * static_cast(frame_size_) / codec_input_rate_hz_; - audio_encoder_.reset(new AudioEncoderIsac(config)); - decoder_ = new AudioDecoderIsac(codec_input_rate_hz_); + audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config)); + decoder_ = new AudioDecoderIsacFloatImpl(codec_input_rate_hz_); } }; @@ -384,14 +384,14 @@ class AudioDecoderIsacFixTest : public AudioDecoderTest { codec_input_rate_hz_ = 16000; frame_size_ = 480; data_length_ = 10 * frame_size_; - AudioEncoderIsacFix::Config config; + AudioEncoderIsacFixImpl::Config config; config.payload_type = payload_type_; config.sample_rate_hz = codec_input_rate_hz_; config.adaptive_mode = false; config.frame_size_ms = 1000 * static_cast(frame_size_) / codec_input_rate_hz_; - audio_encoder_.reset(new AudioEncoderIsacFix(config)); - decoder_ = new AudioDecoderIsacFix(codec_input_rate_hz_); + audio_encoder_.reset(new AudioEncoderIsacFixImpl(config)); + decoder_ = new AudioDecoderIsacFixImpl(codec_input_rate_hz_); } }; diff --git a/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc index b013a34617..10227763f7 100644 --- a/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc +++ b/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc @@ -16,7 +16,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) { const int sample_rate_hz = size % 2 == 0 ? 16000 : 32000; // 16 or 32 kHz. static const size_t kAllocatedOuputSizeSamples = 32000 / 10; // 100 ms. int16_t output[kAllocatedOuputSizeSamples]; - AudioDecoderIsac dec(sample_rate_hz); + AudioDecoderIsacFloatImpl dec(sample_rate_hz); FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec, sample_rate_hz, sizeof(output), output); } diff --git a/webrtc/test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc index 9e490d3d4a..9acebbbaae 100644 --- a/webrtc/test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc +++ b/webrtc/test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc @@ -13,7 +13,7 @@ namespace webrtc { void FuzzOneInput(const uint8_t* data, size_t size) { - AudioDecoderIsac dec(16000); + AudioDecoderIsacFloatImpl dec(16000); FuzzAudioDecoderIncomingPacket(data, size, &dec); } } // namespace webrtc diff --git a/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc index 25ab3f454d..64cdccb2fc 100644 --- a/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc +++ b/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc @@ -16,7 +16,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) { static const int kSampleRateHz = 16000; static const size_t kAllocatedOuputSizeSamples = 16000 / 10; // 100 ms. int16_t output[kAllocatedOuputSizeSamples]; - AudioDecoderIsacFix dec(kSampleRateHz); + AudioDecoderIsacFixImpl dec(kSampleRateHz); FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec, kSampleRateHz, sizeof(output), output); }