Revert of Reimplement the builtin audio codec factories using the new stuff in api/

webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc

Index: webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc
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
new file mode 100644
index 0000000000000000000000000000000000000000..f853cbda31d4a03bd161d48d12b268855a0e848a
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc
@@ -0,0 +1,257 @@
+/*
+ *  Copyright (c) 2016 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/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h"
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/optional.h"
+#ifdef WEBRTC_CODEC_G722
+#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#endif
+#ifdef WEBRTC_CODEC_ISACFX
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"  // nogncheck
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"  // nogncheck
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#endif
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+
+namespace webrtc {
+
+namespace {
+
+struct NamedDecoderConstructor {
+  const char* name;
+
+  // If |format| is good, return true and (if |out| isn't null) reset |*out| to
+  // a new decoder object. If the |format| is not good, return false.
+  bool (*constructor)(const SdpAudioFormat& format,
+                      std::unique_ptr<AudioDecoder>* out);
+};
+
+// TODO(kwiberg): These factory functions should probably be moved to each
+// decoder.
+NamedDecoderConstructor decoder_constructors[] = {
+    {"pcmu",
+     [](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
+       if (format.clockrate_hz == 8000 && format.num_channels >= 1) {
+         if (out) {
+           out->reset(new AudioDecoderPcmU(format.num_channels));
+         }
+         return true;
+       } else {
+         return false;
+       }
+     }},
+    {"pcma",
+     [](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
+       if (format.clockrate_hz == 8000 && format.num_channels >= 1) {
+         if (out) {
+           out->reset(new AudioDecoderPcmA(format.num_channels));
+         }
+         return true;
+       } else {
+         return false;
+       }
+     }},
+#ifdef WEBRTC_CODEC_ILBC
+    {"ilbc",
+     [](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
+       if (format.clockrate_hz == 8000 && format.num_channels == 1) {
+         if (out) {
+           out->reset(new AudioDecoderIlbcImpl);
+         }
+         return true;
+       } else {
+         return false;
+       }
+     }},
+#endif
+#if defined(WEBRTC_CODEC_ISACFX)
+    {"isac",
+     [](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
+       if (format.clockrate_hz == 16000 && format.num_channels == 1) {
+         if (out) {
+           out->reset(new AudioDecoderIsacFixImpl(format.clockrate_hz));
+         }
+         return true;
+       } else {
+         return false;
+       }
+     }},
+#elif defined(WEBRTC_CODEC_ISAC)
+    {"isac",
+     [](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
+       if ((format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
+           format.num_channels == 1) {
+         if (out) {
+           out->reset(new AudioDecoderIsacFloatImpl(format.clockrate_hz));
+         }
+         return true;
+       } else {
+         return false;
+       }
+     }},
+#endif
+    {"l16",
+     [](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
+       if (format.num_channels >= 1) {
+         if (out) {
+           out->reset(new AudioDecoderPcm16B(format.clockrate_hz,
+                                             format.num_channels));
+         }
+         return true;
+       } else {
+         return false;
+       }
+     }},
+#ifdef WEBRTC_CODEC_G722
+    {"g722",
+     [](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
+       if (format.clockrate_hz == 8000) {
+         if (format.num_channels == 1) {
+           if (out) {
+             out->reset(new AudioDecoderG722Impl);
+           }
+           return true;
+         } else if (format.num_channels == 2) {
+           if (out) {
+             out->reset(new AudioDecoderG722StereoImpl);
+           }
+           return true;
+         }
+       }
+       return false;
+     }},
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    {"opus",
+     [](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
+       const rtc::Optional<int> num_channels = [&] {
+         auto stereo = format.parameters.find("stereo");
+         if (stereo != format.parameters.end()) {
+           if (stereo->second == "0") {
+             return rtc::Optional<int>(1);
+           } else if (stereo->second == "1") {
+             return rtc::Optional<int>(2);
+           } else {
+             return rtc::Optional<int>();  // Bad stereo parameter.
+           }
+         }
+         return rtc::Optional<int>(1);  // Default to mono.
+       }();
+       if (format.clockrate_hz == 48000 && format.num_channels == 2 &&
+           num_channels) {
+         if (out) {
+           out->reset(new AudioDecoderOpusImpl(*num_channels));
+         }
+         return true;
+       } else {
+         return false;
+       }
+     }},
+#endif
+};
+
+class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
+ public:
+  std::vector<AudioCodecSpec> GetSupportedDecoders() override {
+    // Although this looks a bit strange, it means specs need only be
+    // initialized once, and that that initialization is thread-safe.
+    static std::vector<AudioCodecSpec> specs = [] {
+      std::vector<AudioCodecSpec> specs;
+#ifdef WEBRTC_CODEC_OPUS
+      AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
+      opus_info.allow_comfort_noise = false;
+      opus_info.supports_network_adaption = true;
+      // clang-format off
+      SdpAudioFormat opus_format({"opus", 48000, 2, {
+                                    {"minptime", "10"},
+                                    {"useinbandfec", "1"}
+                                  }});
+      // clang-format on
+      specs.push_back({std::move(opus_format), opus_info});
+#endif
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+      specs.push_back(AudioCodecSpec{{"ISAC", 16000, 1},
+                                     {16000, 1, 32000, 10000, 56000}});
+#endif
+#if (defined(WEBRTC_CODEC_ISAC))
+      specs.push_back(AudioCodecSpec{{"ISAC", 32000, 1},
+                                     {32000, 1, 56000, 10000, 56000}});
+#endif
+#ifdef WEBRTC_CODEC_G722
+      specs.push_back(AudioCodecSpec{{"G722", 8000, 1},
+                                     {16000, 1, 64000}});
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+      specs.push_back(AudioCodecSpec{{"ILBC", 8000, 1},
+                                     {8000, 1, 13300}});
+#endif
+      specs.push_back(AudioCodecSpec{{"PCMU", 8000, 1},
+                                     {8000, 1, 64000}});
+      specs.push_back(AudioCodecSpec{{"PCMA", 8000, 1},
+                                     {8000, 1, 64000}});
+      return specs;
+    }();
+    return specs;
+  }
+
+  bool IsSupportedDecoder(const SdpAudioFormat& format) override {
+    for (const auto& dc : decoder_constructors) {
+      if (STR_CASE_CMP(format.name.c_str(), dc.name) == 0) {
+        return dc.constructor(format, nullptr);
+      }
+    }
+    return false;
+  }
+
+  std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format) override {
+    for (const auto& dc : decoder_constructors) {
+      if (STR_CASE_CMP(format.name.c_str(), dc.name) == 0) {
+        std::unique_ptr<AudioDecoder> decoder;
+        bool ok = dc.constructor(format, &decoder);
+        RTC_DCHECK_EQ(ok, decoder != nullptr);
+        if (decoder) {
+          const int expected_sample_rate_hz =
+              STR_CASE_CMP(format.name.c_str(), "g722") == 0
+                  ? 2 * format.clockrate_hz
+                  : format.clockrate_hz;
+          RTC_CHECK_EQ(expected_sample_rate_hz, decoder->SampleRateHz());
+        }
+        return decoder;
+      }
+    }
+    return nullptr;
+  }
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioDecoderFactory>
+CreateBuiltinAudioDecoderFactoryInternal() {
+  return rtc::scoped_refptr<AudioDecoderFactory>(
+      new rtc::RefCountedObject<BuiltinAudioDecoderFactory>);
+}
+
+}  // namespace webrtc