Index: webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.cc |
diff --git a/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.cc b/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..54345fd28b0e1c0034fbda243a3126f170dbd5cb |
--- /dev/null |
+++ b/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.cc |
@@ -0,0 +1,143 @@ |
+/* |
+ * 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/modules/audio_coding/codecs/builtin_audio_encoder_factory.h" |
+ |
+#include <memory> |
+#include <vector> |
+ |
+#include "webrtc/base/checks.h" |
+#include "webrtc/base/logging.h" |
+#include "webrtc/base/optional.h" |
+#include "webrtc/common_types.h" |
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h" |
+#ifdef WEBRTC_CODEC_G722 |
+#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h" |
+#endif |
+#ifdef WEBRTC_CODEC_ILBC |
+#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h" |
+#endif |
+#ifdef WEBRTC_CODEC_ISACFX |
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h" |
+#endif |
+#ifdef WEBRTC_CODEC_ISAC |
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" |
+#endif |
+#ifdef WEBRTC_CODEC_OPUS |
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h" |
+#endif |
+#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" |
+ |
+namespace webrtc { |
+ |
+namespace { |
+ |
+struct NamedEncoderFactory { |
+ const char* name; |
+ rtc::Optional<AudioCodecInfo> (*QueryAudioEncoder)( |
+ const SdpAudioFormat& format); |
+ std::unique_ptr<AudioEncoder> ( |
+ *MakeAudioEncoder)(int payload_type, const SdpAudioFormat& format); |
+ |
+ template <typename T> |
+ static NamedEncoderFactory ForEncoder() { |
+ auto constructor = [](int payload_type, const SdpAudioFormat& format) { |
+ auto opt_info = T::QueryAudioEncoder(format); |
+ if (opt_info) { |
+ return std::unique_ptr<AudioEncoder>(new T(payload_type, format)); |
+ } |
+ return std::unique_ptr<AudioEncoder>(); |
+ }; |
+ |
+ return {T::GetPayloadName(), T::QueryAudioEncoder, constructor}; |
+ } |
+}; |
+ |
+NamedEncoderFactory encoder_factories[] = { |
+#ifdef WEBRTC_CODEC_G722 |
+ NamedEncoderFactory::ForEncoder<AudioEncoderG722>(), |
+#endif |
+#ifdef WEBRTC_CODEC_ILBC |
+ NamedEncoderFactory::ForEncoder<AudioEncoderIlbc>(), |
+#endif |
+#if defined(WEBRTC_CODEC_ISACFX) |
+ NamedEncoderFactory::ForEncoder<AudioEncoderIsacFix>(), |
+#elif defined(WEBRTC_CODEC_ISAC) |
+ NamedEncoderFactory::ForEncoder<AudioEncoderIsac>(), |
+#endif |
+ |
+#ifdef WEBRTC_CODEC_OPUS |
+ NamedEncoderFactory::ForEncoder<AudioEncoderOpus>(), |
+#endif |
+ NamedEncoderFactory::ForEncoder<AudioEncoderPcm16B>(), |
+ NamedEncoderFactory::ForEncoder<AudioEncoderPcmA>(), |
+ NamedEncoderFactory::ForEncoder<AudioEncoderPcmU>(), |
+}; |
+} // namespace |
+ |
+class BuiltinAudioEncoderFactory : public AudioEncoderFactory { |
+ public: |
+ std::vector<AudioCodecSpec> GetSupportedEncoders() override { |
+ static const SdpAudioFormat desired_encoders[] = { |
+ {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}, |
+ {"isac", 16000, 1}, |
+ {"isac", 32000, 1}, |
+ {"G722", 8000, 1}, |
+ {"iLBC", 8000, 1}, |
+ {"PCMU", 8000, 1}, |
+ {"PCMA", 8000, 1}, |
+ }; |
+ |
+ // Initialize thread-safely, once, on first use. |
+ static const std::vector<AudioCodecSpec> specs = [] { |
+ std::vector<AudioCodecSpec> specs; |
+ for (const auto& format : desired_encoders) { |
+ for (const auto& ef : encoder_factories) { |
+ if (STR_CASE_CMP(format.name.c_str(), ef.name) == 0) { |
+ auto opt_info = ef.QueryAudioEncoder(format); |
+ if (opt_info) { |
+ specs.push_back({format, *opt_info}); |
+ } |
+ } |
+ } |
+ } |
+ return specs; |
+ }(); |
+ return specs; |
+ } |
+ |
+ rtc::Optional<AudioCodecInfo> QueryAudioEncoder( |
+ const SdpAudioFormat& format) override { |
+ for (const auto& ef : encoder_factories) { |
+ if (STR_CASE_CMP(format.name.c_str(), ef.name) == 0) { |
+ return ef.QueryAudioEncoder(format); |
+ } |
+ } |
+ return rtc::Optional<AudioCodecInfo>(); |
+ } |
+ |
+ std::unique_ptr<AudioEncoder> MakeAudioEncoder( |
+ int payload_type, |
+ const SdpAudioFormat& format) override { |
+ for (const auto& ef : encoder_factories) { |
+ if (STR_CASE_CMP(format.name.c_str(), ef.name) == 0) { |
+ return ef.MakeAudioEncoder(payload_type, format); |
+ } |
+ } |
+ return nullptr; |
+ } |
+}; |
+ |
+rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory() { |
+ return rtc::scoped_refptr<AudioEncoderFactory>( |
+ new rtc::RefCountedObject<BuiltinAudioEncoderFactory>()); |
+} |
+ |
+} // namespace webrtc |