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Give Audio{De,En}coderIsac* an "Impl" suffix, to free up the original names

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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}
This commit is contained in:
kwiberg
2017-08-17 05:31:02 -07:00
committed by Commit Bot
parent 7c206b5b6c
commit 6ff045f097
13 changed files with 31 additions and 31 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
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@ -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<AudioEncoderIsac> isac_encoder_;
std::unique_ptr<AudioEncoderIsacFloatImpl> isac_encoder_;
std::unique_ptr<SimulatedClock> fake_clock_;
test::AudioLoop audio_loop_;
};

8
webrtc/modules/audio_coding/acm2/rent_a_codec.cc
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@ -154,12 +154,12 @@ std::unique_ptr<AudioEncoder> CreateEncoder(
#if defined(WEBRTC_CODEC_ISACFX)
if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
return std::unique_ptr<AudioEncoder>(
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<AudioEncoder>(
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<AudioDecoder> CreateIsacDecoder(
const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
#if defined(WEBRTC_CODEC_ISACFX)
return std::unique_ptr<AudioDecoder>(
new AudioDecoderIsacFix(sample_rate_hz, bwinfo));
new AudioDecoderIsacFixImpl(sample_rate_hz, bwinfo));
#elif defined(WEBRTC_CODEC_ISAC)
return std::unique_ptr<AudioDecoder>(
new AudioDecoderIsac(sample_rate_hz, bwinfo));
new AudioDecoderIsacFloatImpl(sample_rate_hz, bwinfo));
#else
FATAL() << "iSAC is not supported.";
return std::unique_ptr<AudioDecoder>();

4
webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc
View File

@ -91,7 +91,7 @@ NamedDecoderConstructor decoder_constructors[] = {
[](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* 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 {

4
webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc
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@ -68,9 +68,9 @@ NamedEncoderFactory encoder_factories[] = {
NamedEncoderFactory::ForEncoder<AudioEncoderIlbcImpl>(),
#endif
#if defined(WEBRTC_CODEC_ISACFX)
NamedEncoderFactory::ForEncoder<AudioEncoderIsacFix>(),
NamedEncoderFactory::ForEncoder<AudioEncoderIsacFixImpl>(),
#elif defined(WEBRTC_CODEC_ISAC)
NamedEncoderFactory::ForEncoder<AudioEncoderIsac>(),
NamedEncoderFactory::ForEncoder<AudioEncoderIsacFloatImpl>(),
#endif
#ifdef WEBRTC_CODEC_OPUS

2
webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h
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@ -16,7 +16,7 @@
namespace webrtc {
using AudioDecoderIsacFix = AudioDecoderIsacT<IsacFix>;
using AudioDecoderIsacFixImpl = AudioDecoderIsacT<IsacFix>;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_

2
webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h
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@ -16,7 +16,7 @@
namespace webrtc {
using AudioEncoderIsacFix = AudioEncoderIsacT<IsacFix>;
using AudioEncoderIsacFixImpl = AudioEncoderIsacT<IsacFix>;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_

2
webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h
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@ -16,7 +16,7 @@
namespace webrtc {
using AudioDecoderIsac = AudioDecoderIsacT<IsacFloat>;
using AudioDecoderIsacFloatImpl = AudioDecoderIsacT<IsacFloat>;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_

2
webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h
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@ -16,7 +16,7 @@
namespace webrtc {
using AudioEncoderIsac = AudioEncoderIsacT<IsacFloat>;
using AudioEncoderIsacFloatImpl = AudioEncoderIsacT<IsacFloat>;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_

8
webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
View File

@ -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);

18
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
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@ -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<int>(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<int>(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<int>(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_);
}
};

2
webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc
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@ -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);
}

2
webrtc/test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc
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@ -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

2
webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc
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@ -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);
}