iSAC floating-point implementation of the Audio{En,De}coderFactoryTemplate APIs
BUG=webrtc:7835, webrtc:7841
Review-Url: https://codereview.webrtc.org/3001483002
Cr-Commit-Position: refs/heads/master@{#19427}
diff --git a/webrtc/api/audio_codecs/isac/BUILD.gn b/webrtc/api/audio_codecs/isac/BUILD.gn
index 164babf..3340e28 100644
--- a/webrtc/api/audio_codecs/isac/BUILD.gn
+++ b/webrtc/api/audio_codecs/isac/BUILD.gn
@@ -37,3 +37,29 @@
"../../../rtc_base:rtc_base_approved",
]
}
+
+rtc_static_library("audio_encoder_isac_float") {
+ sources = [
+ "audio_encoder_isac_float.cc",
+ "audio_encoder_isac_float.h",
+ ]
+ deps = [
+ "..:audio_codecs_api",
+ "../../..:webrtc_common",
+ "../../../modules/audio_coding:isac",
+ "../../../rtc_base:rtc_base_approved",
+ ]
+}
+
+rtc_static_library("audio_decoder_isac_float") {
+ sources = [
+ "audio_decoder_isac_float.cc",
+ "audio_decoder_isac_float.h",
+ ]
+ deps = [
+ "..:audio_codecs_api",
+ "../../..:webrtc_common",
+ "../../../modules/audio_coding:isac",
+ "../../../rtc_base:rtc_base_approved",
+ ]
+}
diff --git a/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc b/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc
new file mode 100644
index 0000000..e26e651
--- /dev/null
+++ b/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc
@@ -0,0 +1,44 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#include "webrtc/rtc_base/ptr_util.h"
+
+namespace webrtc {
+
+rtc::Optional<AudioDecoderIsacFloat::Config> AudioDecoderIsacFloat::SdpToConfig(
+ const SdpAudioFormat& format) {
+ if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
+ (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
+ format.num_channels == 1) {
+ Config config;
+ config.sample_rate_hz = format.clockrate_hz;
+ return rtc::Optional<Config>(config);
+ } else {
+ return rtc::Optional<Config>();
+ }
+}
+
+void AudioDecoderIsacFloat::AppendSupportedDecoders(
+ std::vector<AudioCodecSpec>* specs) {
+ specs->push_back({{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}});
+ specs->push_back({{"ISAC", 32000, 1}, {32000, 1, 56000, 10000, 56000}});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderIsacFloat::MakeAudioDecoder(
+ Config config) {
+ RTC_DCHECK(config.IsOk());
+ return rtc::MakeUnique<AudioDecoderIsacFloatImpl>(config.sample_rate_hz);
+}
+
+} // namespace webrtc
diff --git a/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h b/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
new file mode 100644
index 0000000..c0dc880
--- /dev/null
+++ b/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
+#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/api/audio_codecs/audio_format.h"
+#include "webrtc/rtc_base/optional.h"
+
+namespace webrtc {
+
+// iSAC decoder API (floating-point implementation) for use as a template
+// parameter to CreateAudioDecoderFactory<...>().
+//
+// NOTE: This struct is still under development and may change without notice.
+struct AudioDecoderIsacFloat {
+ struct Config {
+ bool IsOk() const {
+ return sample_rate_hz == 16000 || sample_rate_hz == 32000;
+ }
+ int sample_rate_hz = 16000;
+ };
+ static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+ static std::unique_ptr<AudioDecoder> MakeAudioDecoder(Config config);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
diff --git a/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc b/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
new file mode 100644
index 0000000..500cfd1
--- /dev/null
+++ b/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
@@ -0,0 +1,73 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+rtc::Optional<AudioEncoderIsacFloat::Config> AudioEncoderIsacFloat::SdpToConfig(
+ const SdpAudioFormat& format) {
+ if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
+ (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
+ format.num_channels == 1) {
+ Config config;
+ config.sample_rate_hz = format.clockrate_hz;
+ if (config.sample_rate_hz == 16000) {
+ // For sample rate 16 kHz, optionally use 60 ms frames, instead of the
+ // default 30 ms.
+ const auto ptime_iter = format.parameters.find("ptime");
+ if (ptime_iter != format.parameters.end()) {
+ const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+ if (ptime && *ptime >= 60) {
+ config.frame_size_ms = 60;
+ }
+ }
+ }
+ return rtc::Optional<Config>(config);
+ } else {
+ return rtc::Optional<Config>();
+ }
+}
+
+void AudioEncoderIsacFloat::AppendSupportedEncoders(
+ std::vector<AudioCodecSpec>* specs) {
+ for (int sample_rate_hz : {16000, 32000}) {
+ const SdpAudioFormat fmt = {"ISAC", sample_rate_hz, 1};
+ const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+ specs->push_back({fmt, info});
+ }
+}
+
+AudioCodecInfo AudioEncoderIsacFloat::QueryAudioEncoder(
+ const AudioEncoderIsacFloat::Config& config) {
+ RTC_DCHECK(config.IsOk());
+ constexpr int min_bitrate = 10000;
+ const int max_bitrate = config.sample_rate_hz == 16000 ? 32000 : 56000;
+ const int default_bitrate = max_bitrate;
+ return {config.sample_rate_hz, 1, default_bitrate, min_bitrate, max_bitrate};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder(
+ const AudioEncoderIsacFloat::Config& config,
+ int payload_type) {
+ RTC_DCHECK(config.IsOk());
+ AudioEncoderIsacFloatImpl::Config c;
+ c.sample_rate_hz = config.sample_rate_hz;
+ c.frame_size_ms = config.frame_size_ms;
+ c.payload_type = payload_type;
+ return rtc::MakeUnique<AudioEncoderIsacFloatImpl>(c);
+}
+
+} // namespace webrtc
diff --git a/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h b/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
new file mode 100644
index 0000000..35bc94b
--- /dev/null
+++ b/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
+#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/api/audio_codecs/audio_encoder.h"
+#include "webrtc/api/audio_codecs/audio_format.h"
+#include "webrtc/rtc_base/optional.h"
+
+namespace webrtc {
+
+// iSAC encoder API (floating-point implementation) for use as a template
+// parameter to CreateAudioEncoderFactory<...>().
+//
+// NOTE: This struct is still under development and may change without notice.
+struct AudioEncoderIsacFloat {
+ struct Config {
+ bool IsOk() const {
+ return (sample_rate_hz == 16000 &&
+ (frame_size_ms == 30 || frame_size_ms == 60)) ||
+ (sample_rate_hz == 32000 && frame_size_ms == 30);
+ }
+ int sample_rate_hz = 16000;
+ int frame_size_ms = 30;
+ };
+ static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+ static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+ static AudioCodecInfo QueryAudioEncoder(const Config& config);
+ static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config,
+ int payload_type);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
diff --git a/webrtc/api/audio_codecs/test/BUILD.gn b/webrtc/api/audio_codecs/test/BUILD.gn
index ce6ed92..16fdeb9 100644
--- a/webrtc/api/audio_codecs/test/BUILD.gn
+++ b/webrtc/api/audio_codecs/test/BUILD.gn
@@ -34,7 +34,9 @@
"../ilbc:audio_decoder_ilbc",
"../ilbc:audio_encoder_ilbc",
"../isac:audio_decoder_isac_fix",
+ "../isac:audio_decoder_isac_float",
"../isac:audio_encoder_isac_fix",
+ "../isac:audio_encoder_isac_float",
"../opus:audio_decoder_opus",
"../opus:audio_encoder_opus",
"//testing/gmock",
diff --git a/webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc b/webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
index c27f242..8d65a65 100644
--- a/webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
+++ b/webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
@@ -14,6 +14,7 @@
#include "webrtc/api/audio_codecs/g722/audio_decoder_g722.h"
#include "webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h"
#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h"
+#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h"
#include "webrtc/api/audio_codecs/opus/audio_decoder_opus.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/test/gmock.h"
@@ -174,12 +175,32 @@
{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}}));
EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 16000, 2}));
EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 16000, 1}));
+ EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 32000, 1}));
EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"isac", 8000, 1}));
auto dec = factory->MakeAudioDecoder({"isac", 16000, 1});
ASSERT_NE(nullptr, dec);
EXPECT_EQ(16000, dec->SampleRateHz());
}
+TEST(AudioDecoderFactoryTemplateTest, IsacFloat) {
+ auto factory = CreateAudioDecoderFactory<AudioDecoderIsacFloat>();
+ EXPECT_THAT(
+ factory->GetSupportedDecoders(),
+ testing::ElementsAre(
+ AudioCodecSpec{{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}},
+ AudioCodecSpec{{"ISAC", 32000, 1}, {32000, 1, 56000, 10000, 56000}}));
+ EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 16000, 2}));
+ EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 16000, 1}));
+ EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 32000, 1}));
+ EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"isac", 8000, 1}));
+ auto dec1 = factory->MakeAudioDecoder({"isac", 16000, 1});
+ ASSERT_NE(nullptr, dec1);
+ EXPECT_EQ(16000, dec1->SampleRateHz());
+ auto dec2 = factory->MakeAudioDecoder({"isac", 32000, 1});
+ ASSERT_NE(nullptr, dec2);
+ EXPECT_EQ(32000, dec2->SampleRateHz());
+}
+
TEST(AudioDecoderFactoryTemplateTest, L16) {
auto factory = CreateAudioDecoderFactory<AudioDecoderL16>();
EXPECT_THAT(
diff --git a/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc b/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
index 4120978..7d8f3a8 100644
--- a/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
+++ b/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
@@ -14,6 +14,7 @@
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h"
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h"
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h"
+#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "webrtc/api/audio_codecs/opus/audio_encoder_opus.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/test/gmock.h"
@@ -181,6 +182,8 @@
factory->QueryAudioEncoder({"isac", 16000, 2}));
EXPECT_EQ(rtc::Optional<AudioCodecInfo>({16000, 1, 32000, 10000, 32000}),
factory->QueryAudioEncoder({"isac", 16000, 1}));
+ EXPECT_EQ(rtc::Optional<AudioCodecInfo>(),
+ factory->QueryAudioEncoder({"isac", 32000, 1}));
EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"isac", 8000, 1}));
auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1});
ASSERT_NE(nullptr, enc1);
@@ -192,6 +195,28 @@
EXPECT_EQ(6u, enc2->Num10MsFramesInNextPacket());
}
+TEST(AudioEncoderFactoryTemplateTest, IsacFloat) {
+ auto factory = CreateAudioEncoderFactory<AudioEncoderIsacFloat>();
+ EXPECT_THAT(
+ factory->GetSupportedEncoders(),
+ testing::ElementsAre(
+ AudioCodecSpec{{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}},
+ AudioCodecSpec{{"ISAC", 32000, 1}, {32000, 1, 56000, 10000, 56000}}));
+ EXPECT_EQ(rtc::Optional<AudioCodecInfo>(),
+ factory->QueryAudioEncoder({"isac", 16000, 2}));
+ EXPECT_EQ(rtc::Optional<AudioCodecInfo>({16000, 1, 32000, 10000, 32000}),
+ factory->QueryAudioEncoder({"isac", 16000, 1}));
+ EXPECT_EQ(rtc::Optional<AudioCodecInfo>({32000, 1, 56000, 10000, 56000}),
+ factory->QueryAudioEncoder({"isac", 32000, 1}));
+ EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"isac", 8000, 1}));
+ auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1});
+ ASSERT_NE(nullptr, enc1);
+ EXPECT_EQ(16000, enc1->SampleRateHz());
+ auto enc2 = factory->MakeAudioEncoder(17, {"isac", 32000, 1});
+ ASSERT_NE(nullptr, enc2);
+ EXPECT_EQ(32000, enc2->SampleRateHz());
+}
+
TEST(AudioEncoderFactoryTemplateTest, L16) {
auto factory = CreateAudioEncoderFactory<AudioEncoderL16>();
EXPECT_THAT(
