Revert of Reimplement the builtin audio codec factories using the new stuff in api/ (patchset #1 id:60001 of https://codereview.webrtc.org/2997713002/ )
Reason for revert:
Speculatively reverting, likely breaks chromium.webrtc.fyi.
Failed to create local offer: Test failed: Error: setSdpDefaultCodec() failed: "Unknown ID for |codec| = 'G722'."
Failing bots:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Tester/builds/42349
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win8%20Tester/builds/1561
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/8517
Original issue's description:
> Reimplement the builtin audio codec factories using the new stuff in api/
>
> The whole point of all the audio codec stuff we've recently published
> in api/ is to function as lego bricks so that building stuff like our
> builtin audio codec factories will be easy.
>
> BUG=webrtc:7821, webrtc:7822
>
> Review-Url: https://codereview.webrtc.org/2997713002
> Cr-Commit-Position: refs/heads/master@{#19446}
> Committed: https://chromium.googlesource.com/external/webrtc/+/417989a8643d0d19052a87e95905b469d5ede173
TBR=ossu@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7821, webrtc:7822
Review-Url: https://codereview.webrtc.org/2996373002
Cr-Original-Commit-Position: refs/heads/master@{#19452}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: bcc655c2c71bb1df2ef94cdbf126b318135f1920
diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index 3ed6396..2174fb1 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -38,46 +38,9 @@
]
deps = [
":audio_codecs_api",
+ "../../modules/audio_coding:builtin_audio_decoder_factory_internal",
"../../rtc_base:rtc_base_approved",
- "L16:audio_decoder_L16",
- "g711:audio_decoder_g711",
]
- defines = []
- if (rtc_include_ilbc) {
- deps += [ "ilbc:audio_decoder_ilbc" ]
- defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
- } else {
- defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
- }
- if (rtc_include_opus) {
- deps += [ "opus:audio_decoder_opus" ]
- defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
- } else {
- defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
- }
- if (build_with_mozilla) {
- defines += [
- "WEBRTC_USE_BUILTIN_G722=0",
- "WEBRTC_USE_BUILTIN_ISAC_FIX=0",
- "WEBRTC_USE_BUILTIN_ISAC_FLOAT=0",
- ]
- } else {
- if (current_cpu == "arm") {
- deps += [ "isac:audio_decoder_isac_fix" ]
- defines += [
- "WEBRTC_USE_BUILTIN_ISAC_FIX=1",
- "WEBRTC_USE_BUILTIN_ISAC_FLOAT=0",
- ]
- } else {
- deps += [ "isac:audio_decoder_isac_float" ]
- defines += [
- "WEBRTC_USE_BUILTIN_ISAC_FIX=0",
- "WEBRTC_USE_BUILTIN_ISAC_FLOAT=1",
- ]
- }
- deps += [ "g722:audio_decoder_g722" ]
- defines += [ "WEBRTC_USE_BUILTIN_G722=1" ]
- }
}
rtc_static_library("builtin_audio_encoder_factory") {
@@ -87,44 +50,7 @@
]
deps = [
":audio_codecs_api",
+ "../../modules/audio_coding:builtin_audio_encoder_factory_internal",
"../../rtc_base:rtc_base_approved",
- "L16:audio_encoder_L16",
- "g711:audio_encoder_g711",
]
- defines = []
- if (rtc_include_ilbc) {
- deps += [ "ilbc:audio_encoder_ilbc" ]
- defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
- } else {
- defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
- }
- if (rtc_include_opus) {
- deps += [ "opus:audio_encoder_opus" ]
- defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
- } else {
- defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
- }
- if (build_with_mozilla) {
- defines += [
- "WEBRTC_USE_BUILTIN_G722=0",
- "WEBRTC_USE_BUILTIN_ISAC_FIX=0",
- "WEBRTC_USE_BUILTIN_ISAC_FLOAT=0",
- ]
- } else {
- if (current_cpu == "arm") {
- deps += [ "isac:audio_encoder_isac_fix" ]
- defines += [
- "WEBRTC_USE_BUILTIN_ISAC_FIX=1",
- "WEBRTC_USE_BUILTIN_ISAC_FLOAT=0",
- ]
- } else {
- deps += [ "isac:audio_encoder_isac_float" ]
- defines += [
- "WEBRTC_USE_BUILTIN_ISAC_FIX=0",
- "WEBRTC_USE_BUILTIN_ISAC_FLOAT=1",
- ]
- }
- deps += [ "g722:audio_encoder_g722" ]
- defines += [ "WEBRTC_USE_BUILTIN_G722=1" ]
- }
}
diff --git a/api/audio_codecs/builtin_audio_decoder_factory.cc b/api/audio_codecs/builtin_audio_decoder_factory.cc
index 69a3e7c..9bd049b 100644
--- a/api/audio_codecs/builtin_audio_decoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_decoder_factory.cc
@@ -10,70 +10,12 @@
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include <memory>
-#include <vector>
-
-#include "webrtc/api/audio_codecs/L16/audio_decoder_L16.h"
-#include "webrtc/api/audio_codecs/audio_decoder_factory_template.h"
-#include "webrtc/api/audio_codecs/g711/audio_decoder_g711.h"
-#if WEBRTC_USE_BUILTIN_G722
-#include "webrtc/api/audio_codecs/g722/audio_decoder_g722.h" // nogncheck
-#endif
-#if WEBRTC_USE_BUILTIN_ILBC
-#include "webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck
-#endif
-#if WEBRTC_USE_BUILTIN_ISAC_FIX
-#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h" // nogncheck
-#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT
-#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h" // nogncheck
-#endif
-#if WEBRTC_USE_BUILTIN_OPUS
-#include "webrtc/api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck
-#endif
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h"
namespace webrtc {
-namespace {
-
-// Modify an audio decoder to not advertise support for anything.
-template <typename T>
-struct NotAdvertised {
- using Config = typename T::Config;
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
- return T::SdpToConfig(audio_format);
- }
- static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
- // Don't advertise support for anything.
- }
- static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Config& config) {
- return T::MakeAudioDecoder(config);
- }
-};
-
-} // namespace
-
rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
- return CreateAudioDecoderFactory<
-
-#if WEBRTC_USE_BUILTIN_OPUS
- AudioDecoderOpus,
-#endif
-
-#if WEBRTC_USE_BUILTIN_ISAC_FIX
- AudioDecoderIsacFix,
-#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT
- AudioDecoderIsacFloat,
-#endif
-
-#if WEBRTC_USE_BUILTIN_G722
- AudioDecoderG722,
-#endif
-
-#if WEBRTC_USE_BUILTIN_ILBC
- AudioDecoderIlbc,
-#endif
-
- AudioDecoderG711, NotAdvertised<AudioDecoderL16>>();
+ return CreateBuiltinAudioDecoderFactoryInternal();
}
} // namespace webrtc
diff --git a/api/audio_codecs/builtin_audio_encoder_factory.cc b/api/audio_codecs/builtin_audio_encoder_factory.cc
index ae1bf4b..bb57a5f 100644
--- a/api/audio_codecs/builtin_audio_encoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_encoder_factory.cc
@@ -10,74 +10,12 @@
#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
-#include <memory>
-#include <vector>
-
-#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h"
-#include "webrtc/api/audio_codecs/audio_encoder_factory_template.h"
-#include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h"
-#if WEBRTC_USE_BUILTIN_G722
-#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h" // nogncheck
-#endif
-#if WEBRTC_USE_BUILTIN_ILBC
-#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck
-#endif
-#if WEBRTC_USE_BUILTIN_ISAC_FIX
-#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h" // nogncheck
-#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT
-#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h" // nogncheck
-#endif
-#if WEBRTC_USE_BUILTIN_OPUS
-#include "webrtc/api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck
-#endif
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.h"
namespace webrtc {
-namespace {
-
-// Modify an audio encoder to not advertise support for anything.
-template <typename T>
-struct NotAdvertised {
- using Config = typename T::Config;
- static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
- return T::SdpToConfig(audio_format);
- }
- static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
- // Don't advertise support for anything.
- }
- static AudioCodecInfo QueryAudioEncoder(const Config& config) {
- return T::QueryAudioEncoder(config);
- }
- static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config,
- int payload_type) {
- return T::MakeAudioEncoder(config, payload_type);
- }
-};
-
-} // namespace
-
rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory() {
- return CreateAudioEncoderFactory<
-
-#if WEBRTC_USE_BUILTIN_OPUS
- AudioEncoderOpus,
-#endif
-
-#if WEBRTC_USE_BUILTIN_ISAC_FIX
- AudioEncoderIsacFix,
-#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT
- AudioEncoderIsacFloat,
-#endif
-
-#if WEBRTC_USE_BUILTIN_G722
- AudioEncoderG722,
-#endif
-
-#if WEBRTC_USE_BUILTIN_ILBC
- AudioEncoderIlbc,
-#endif
-
- AudioEncoderG711, NotAdvertised<AudioEncoderL16>>();
+ return CreateBuiltinAudioEncoderFactoryInternal();
}
} // namespace webrtc
diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
index 658d151..29a68ff 100644
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@@ -36,12 +36,9 @@
":audio_encoder_opus_config",
"..:audio_codecs_api",
"../../../modules/audio_coding:webrtc_opus",
+ "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed?
"../../../rtc_base:rtc_base_approved",
]
- public_deps = [
- # TODO(kwiberg): Remove this public_dep when bug 7847 has been fixed.
- "../../../rtc_base:protobuf_utils",
- ]
}
rtc_static_library("audio_decoder_opus") {
diff --git a/api/audio_codecs/test/BUILD.gn b/api/audio_codecs/test/BUILD.gn
index 0f742f5..16fdeb9 100644
--- a/api/audio_codecs/test/BUILD.gn
+++ b/api/audio_codecs/test/BUILD.gn
@@ -21,6 +21,7 @@
]
deps = [
"..:audio_codecs_api",
+ "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed?
"../../../rtc_base:rtc_base_approved",
"../../../test:audio_codec_mocks",
"../../../test:test_support",
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index a3964c9..3e6a183 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -51,6 +51,34 @@
]
}
+rtc_static_library("builtin_audio_decoder_factory_internal") {
+ sources = [
+ "codecs/builtin_audio_decoder_factory_internal.cc",
+ "codecs/builtin_audio_decoder_factory_internal.h",
+ ]
+ deps = [
+ "../..:webrtc_common",
+ "../../rtc_base:protobuf_utils",
+ "../../rtc_base:rtc_base_approved",
+ "../../api/audio_codecs:audio_codecs_api",
+ ] + audio_codec_deps
+ defines = audio_codec_defines
+}
+
+rtc_static_library("builtin_audio_encoder_factory_internal") {
+ sources = [
+ "codecs/builtin_audio_encoder_factory_internal.cc",
+ "codecs/builtin_audio_encoder_factory_internal.h",
+ ]
+ deps = [
+ "../..:webrtc_common",
+ "../../rtc_base:protobuf_utils",
+ "../../rtc_base:rtc_base_approved",
+ "../../api/audio_codecs:audio_codecs_api",
+ ] + audio_codec_deps
+ defines = audio_codec_defines
+}
+
rtc_static_library("rent_a_codec") {
sources = [
"acm2/acm_codec_database.cc",
@@ -812,13 +840,13 @@
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs/opus:audio_encoder_opus_config",
"../../common_audio",
+ "../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../system_wrappers",
]
public_deps = [
":webrtc_opus_c",
- "../../rtc_base:protobuf_utils",
]
defines = audio_codec_defines
diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc b/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc
new file mode 100644
index 0000000..f853cbd
--- /dev/null
+++ b/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
diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h b/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h
new file mode 100644
index 0000000..c856954
--- /dev/null
+++ b/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h
@@ -0,0 +1,24 @@
+/*
+ * 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_INTERNAL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_INTERNAL_H_
+
+#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDecoderFactory>
+CreateBuiltinAudioDecoderFactoryInternal();
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_INTERNAL_H_
diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc b/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc
new file mode 100644
index 0000000..b44268a
--- /dev/null
+++ b/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc
@@ -0,0 +1,144 @@
+/*
+ * 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_internal.h"
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/optional.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" // nogncheck
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" // nogncheck
+#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<AudioEncoderG722Impl>(),
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+ NamedEncoderFactory::ForEncoder<AudioEncoderIlbcImpl>(),
+#endif
+#if defined(WEBRTC_CODEC_ISACFX)
+ NamedEncoderFactory::ForEncoder<AudioEncoderIsacFixImpl>(),
+#elif defined(WEBRTC_CODEC_ISAC)
+ NamedEncoderFactory::ForEncoder<AudioEncoderIsacFloatImpl>(),
+#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>
+CreateBuiltinAudioEncoderFactoryInternal() {
+ return rtc::scoped_refptr<AudioEncoderFactory>(
+ new rtc::RefCountedObject<BuiltinAudioEncoderFactory>());
+}
+
+} // namespace webrtc
diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.h b/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.h
new file mode 100644
index 0000000..327e6ca
--- /dev/null
+++ b/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.h
@@ -0,0 +1,26 @@
+/*
+ * 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_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_INTERNAL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_INTERNAL_H_
+
+#include "webrtc/api/audio_codecs/audio_encoder_factory.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create the built-in types of audio encoders.
+// NOTE: This function is still under development and may change without notice.
+rtc::scoped_refptr<AudioEncoderFactory>
+CreateBuiltinAudioEncoderFactoryInternal();
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_INTERNAL_H_
