iSAC fixed-point implementation of the Audio{En,De}coderFactoryTemplate APIs

BUG=webrtc:7835, webrtc:7841

Review-Url: https://codereview.webrtc.org/2996693002
Cr-Commit-Position: refs/heads/master@{#19409}

webrtc/api/audio_codecs/isac/BUILD.gn

@@ -0,0 +1,39 @@
+# 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_static_library("audio_encoder_isac_fix") {
+  sources = [
+    "audio_encoder_isac_fix.cc",
+    "audio_encoder_isac_fix.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../..:webrtc_common",
+    "../../../modules/audio_coding:isac_fix",
+    "../../../rtc_base:rtc_base_approved",
+  ]
+}
+
+rtc_static_library("audio_decoder_isac_fix") {
+  sources = [
+    "audio_decoder_isac_fix.cc",
+    "audio_decoder_isac_fix.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../..:webrtc_common",
+    "../../../modules/audio_coding:isac_fix",
+    "../../../rtc_base:rtc_base_approved",
+  ]
+}

webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.cc

@@ -0,0 +1,37 @@
+/*
+ *  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_fix.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
+#include "webrtc/rtc_base/ptr_util.h"
+
+namespace webrtc {
+
+rtc::Optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
+    const SdpAudioFormat& format) {
+  return STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
+                 format.clockrate_hz == 16000 && format.num_channels == 1
+             ? rtc::Optional<Config>(Config())
+             : rtc::Optional<Config>();
+}
+
+void AudioDecoderIsacFix::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  specs->push_back({{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderIsacFix::MakeAudioDecoder(
+    Config config) {
+  return rtc::MakeUnique<AudioDecoderIsacFixImpl>(16000);
+}
+
+}  // namespace webrtc

webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h

@@ -0,0 +1,36 @@
+/*
+ *  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_FIX_H_
+#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_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 (fixed-point implementation) for use as a template
+// parameter to CreateAudioDecoderFactory<...>().
+//
+// NOTE: This struct is still under development and may change without notice.
+struct AudioDecoderIsacFix {
+  struct Config {};  // Empty---no config values needed!
+  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_FIX_H_

webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc

@@ -0,0 +1,61 @@
+/*
+ *  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_fix.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+rtc::Optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
+      format.clockrate_hz == 16000 && format.num_channels == 1) {
+    Config config;
+    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 AudioEncoderIsacFix::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"ISAC", 16000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderIsacFix::QueryAudioEncoder(
+    AudioEncoderIsacFix::Config config) {
+  RTC_DCHECK(config.IsOk());
+  return {16000, 1, 32000, 10000, 32000};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderIsacFix::MakeAudioEncoder(
+    AudioEncoderIsacFix::Config config,
+    int payload_type) {
+  RTC_DCHECK(config.IsOk());
+  AudioEncoderIsacFixImpl::Config c;
+  c.frame_size_ms = config.frame_size_ms;
+  c.payload_type = payload_type;
+  return rtc::MakeUnique<AudioEncoderIsacFixImpl>(c);
+}
+
+}  // namespace webrtc

webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.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_ENCODER_ISAC_FIX_H_
+#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_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 (fixed-point implementation) for use as a template
+// parameter to CreateAudioEncoderFactory<...>().
+//
+// NOTE: This struct is still under development and may change without notice.
+struct AudioEncoderIsacFix {
+  struct Config {
+    bool IsOk() const { return frame_size_ms == 30 || frame_size_ms == 60; }
+    int frame_size_ms = 30;
+  };
+  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(Config config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(Config config,
+                                                        int payload_type);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_

webrtc/api/audio_codecs/test/BUILD.gn

@@ -33,6 +33,8 @@ if (rtc_include_tests) {
       "../g722:audio_encoder_g722",
       "../ilbc:audio_decoder_ilbc",
       "../ilbc:audio_encoder_ilbc",
+      "../isac:audio_decoder_isac_fix",
+      "../isac:audio_encoder_isac_fix",
       "../opus:audio_decoder_opus",
       "../opus:audio_encoder_opus",
       "//testing/gmock",

webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc

@@ -13,6 +13,7 @@
 #include "webrtc/api/audio_codecs/g711/audio_decoder_g711.h"
 #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/opus/audio_decoder_opus.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/test/gmock.h"
@@ -166,6 +167,19 @@ TEST(AudioDecoderFactoryTemplateTest, Ilbc) {
   EXPECT_EQ(8000, dec->SampleRateHz());
 }
 
+TEST(AudioDecoderFactoryTemplateTest, IsacFix) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderIsacFix>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              testing::ElementsAre(AudioCodecSpec{
+                  {"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 16000, 2}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 16000, 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, L16) {
   auto factory = CreateAudioDecoderFactory<AudioDecoderL16>();
   EXPECT_THAT(

webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc

@@ -13,6 +13,7 @@
 #include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h"
 #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/opus/audio_encoder_opus.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/test/gmock.h"
@@ -171,6 +172,26 @@ TEST(AudioEncoderFactoryTemplateTest, Ilbc) {
   EXPECT_EQ(8000, enc->SampleRateHz());
 }
 
+TEST(AudioEncoderFactoryTemplateTest, IsacFix) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderIsacFix>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              testing::ElementsAre(AudioCodecSpec{
+                  {"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}}));
+  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(nullptr, factory->MakeAudioEncoder(17, {"isac", 8000, 1}));
+  auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1});
+  ASSERT_NE(nullptr, enc1);
+  EXPECT_EQ(16000, enc1->SampleRateHz());
+  EXPECT_EQ(3u, enc1->Num10MsFramesInNextPacket());
+  auto enc2 =
+      factory->MakeAudioEncoder(17, {"isac", 16000, 1, {{"ptime", "60"}}});
+  ASSERT_NE(nullptr, enc2);
+  EXPECT_EQ(6u, enc2->Num10MsFramesInNextPacket());
+}
+
 TEST(AudioEncoderFactoryTemplateTest, L16) {
   auto factory = CreateAudioEncoderFactory<AudioEncoderL16>();
   EXPECT_THAT(