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.
(This has landed once before, but got reverted because of Chromium test
failures---apparently, someone isn't ignoring the case of the codec names
like they're supposed to. The quick fix was to preserve the same case
used by the old implementation.)
BUG=webrtc:7821, webrtc:7822
Review-Url: https://codereview.webrtc.org/2998263002
Cr-Original-Commit-Position: refs/heads/master@{#19512}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: e5eb7247ffaefade251d9c8b4cc1df1530564482
diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index 2174fb1..3ed6396 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -38,9 +38,46 @@
]
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") {
@@ -50,7 +87,44 @@
]
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 9bd049b..69a3e7c 100644
--- a/api/audio_codecs/builtin_audio_decoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_decoder_factory.cc
@@ -10,12 +10,70 @@
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.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
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 CreateBuiltinAudioDecoderFactoryInternal();
+ 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>>();
}
} // namespace webrtc
diff --git a/api/audio_codecs/builtin_audio_encoder_factory.cc b/api/audio_codecs/builtin_audio_encoder_factory.cc
index bb57a5f..ae1bf4b 100644
--- a/api/audio_codecs/builtin_audio_encoder_factory.cc
+++ b/api/audio_codecs/builtin_audio_encoder_factory.cc
@@ -10,12 +10,74 @@
#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.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
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 CreateBuiltinAudioEncoderFactoryInternal();
+ 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>>();
}
} // namespace webrtc
diff --git a/api/audio_codecs/g722/audio_decoder_g722.cc b/api/audio_codecs/g722/audio_decoder_g722.cc
index 9c4bff7..6f3ce97 100644
--- a/api/audio_codecs/g722/audio_decoder_g722.cc
+++ b/api/audio_codecs/g722/audio_decoder_g722.cc
@@ -22,7 +22,7 @@
rtc::Optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
const SdpAudioFormat& format) {
- return STR_CASE_CMP(format.name.c_str(), "g722") == 0 &&
+ return STR_CASE_CMP(format.name.c_str(), "G722") == 0 &&
format.clockrate_hz == 8000 &&
(format.num_channels == 1 || format.num_channels == 2)
? rtc::Optional<Config>(
@@ -32,7 +32,7 @@
void AudioDecoderG722::AppendSupportedDecoders(
std::vector<AudioCodecSpec>* specs) {
- specs->push_back({{"g722", 8000, 1}, {16000, 1, 64000}});
+ specs->push_back({{"G722", 8000, 1}, {16000, 1, 64000}});
}
std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
diff --git a/api/audio_codecs/g722/audio_encoder_g722.cc b/api/audio_codecs/g722/audio_encoder_g722.cc
index b3e4063..09b3faf 100644
--- a/api/audio_codecs/g722/audio_encoder_g722.cc
+++ b/api/audio_codecs/g722/audio_encoder_g722.cc
@@ -26,7 +26,7 @@
void AudioEncoderG722::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
- const SdpAudioFormat fmt = {"g722", 8000, 1};
+ const SdpAudioFormat fmt = {"G722", 8000, 1};
const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
specs->push_back({fmt, info});
}
diff --git a/api/audio_codecs/ilbc/audio_decoder_ilbc.cc b/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
index b3053ef..dc17751 100644
--- a/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
+++ b/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
@@ -21,7 +21,7 @@
rtc::Optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
const SdpAudioFormat& format) {
- return STR_CASE_CMP(format.name.c_str(), "ilbc") == 0 &&
+ return STR_CASE_CMP(format.name.c_str(), "ILBC") == 0 &&
format.clockrate_hz == 8000 && format.num_channels == 1
? rtc::Optional<Config>(Config())
: rtc::Optional<Config>();
diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
index 29a68ff..658d151 100644
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@@ -36,9 +36,12 @@
":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 16fdeb9..0f742f5 100644
--- a/api/audio_codecs/test/BUILD.gn
+++ b/api/audio_codecs/test/BUILD.gn
@@ -21,7 +21,6 @@
]
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/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc b/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
index 8d65a65..0b1135c 100644
--- a/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
+++ b/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
@@ -123,7 +123,7 @@
testing::ElementsAre(
AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}},
AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}}));
- EXPECT_FALSE(factory->IsSupportedDecoder({"g711", 8000, 1}));
+ EXPECT_FALSE(factory->IsSupportedDecoder({"G711", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"PCMU", 8000, 1}));
EXPECT_TRUE(factory->IsSupportedDecoder({"pcma", 8000, 1}));
EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"pcmu", 16000, 1}));
@@ -139,19 +139,19 @@
auto factory = CreateAudioDecoderFactory<AudioDecoderG722>();
EXPECT_THAT(factory->GetSupportedDecoders(),
testing::ElementsAre(
- AudioCodecSpec{{"g722", 8000, 1}, {16000, 1, 64000}}));
+ AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
- EXPECT_TRUE(factory->IsSupportedDecoder({"g722", 8000, 1}));
+ EXPECT_TRUE(factory->IsSupportedDecoder({"G722", 8000, 1}));
EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"bar", 16000, 1}));
- auto dec1 = factory->MakeAudioDecoder({"g722", 8000, 1});
+ auto dec1 = factory->MakeAudioDecoder({"G722", 8000, 1});
ASSERT_NE(nullptr, dec1);
EXPECT_EQ(16000, dec1->SampleRateHz());
EXPECT_EQ(1u, dec1->Channels());
- auto dec2 = factory->MakeAudioDecoder({"g722", 8000, 2});
+ auto dec2 = factory->MakeAudioDecoder({"G722", 8000, 2});
ASSERT_NE(nullptr, dec2);
EXPECT_EQ(16000, dec2->SampleRateHz());
EXPECT_EQ(2u, dec2->Channels());
- auto dec3 = factory->MakeAudioDecoder({"g722", 8000, 3});
+ auto dec3 = factory->MakeAudioDecoder({"G722", 8000, 3});
ASSERT_EQ(nullptr, dec3);
}
diff --git a/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc b/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
index 7d8f3a8..891821a 100644
--- a/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
+++ b/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
@@ -147,13 +147,13 @@
auto factory = CreateAudioEncoderFactory<AudioEncoderG722>();
EXPECT_THAT(factory->GetSupportedEncoders(),
testing::ElementsAre(
- AudioCodecSpec{{"g722", 8000, 1}, {16000, 1, 64000}}));
+ AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
EXPECT_EQ(rtc::Optional<AudioCodecInfo>(),
factory->QueryAudioEncoder({"foo", 8000, 1}));
EXPECT_EQ(rtc::Optional<AudioCodecInfo>({16000, 1, 64000}),
- factory->QueryAudioEncoder({"g722", 8000, 1}));
+ factory->QueryAudioEncoder({"G722", 8000, 1}));
EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"bar", 16000, 1}));
- auto enc = factory->MakeAudioEncoder(17, {"g722", 8000, 1});
+ auto enc = factory->MakeAudioEncoder(17, {"G722", 8000, 1});
ASSERT_NE(nullptr, enc);
EXPECT_EQ(16000, enc->SampleRateHz());
}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 3e6a183..a3964c9 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -51,34 +51,6 @@
]
}
-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",
@@ -840,13 +812,13 @@
"../../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
deleted file mode 100644
index f853cbd..0000000
--- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc
+++ /dev/null
@@ -1,257 +0,0 @@
-/*
- * 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
deleted file mode 100644
index c856954..0000000
--- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h
+++ /dev/null
@@ -1,24 +0,0 @@
-/*
- * 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
deleted file mode 100644
index b44268a..0000000
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 327e6ca..0000000
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * 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_
