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}

webrtc/api/audio_codecs/isac/BUILD.gn

@@ -37,3 +37,29 @@ rtc_static_library("audio_decoder_isac_fix") {
     "../../../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",
+  ]
+}

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

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_

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

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_

webrtc/api/audio_codecs/test/BUILD.gn

@@ -34,7 +34,9 @@ if (rtc_include_tests) {
       "../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",

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 @@ TEST(AudioDecoderFactoryTemplateTest, IsacFix) {
                   {"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(

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 @@ TEST(AudioEncoderFactoryTemplateTest, IsacFix) {
             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 @@ TEST(AudioEncoderFactoryTemplateTest, IsacFix) {
   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(