AudioCodingModule: Remove support for creating encoders
After this CL, all audio encoders have to be injected by the caller.
This means that there is no special "built-in" set of codecs, and
users won't have to pay the binary size and security costs of codecs
they aren't using.
Bug: webrtc:8396
Change-Id: Idb0959ce395940c8bb3bbb49256cdcd84fc87bb6
Reviewed-on: https://webrtc-review.googlesource.com/c/103821
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25600}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index ec81697..b836646 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -14,31 +14,6 @@
visibility = [ ":*" ]
-audio_codec_deps = [
- ":g711",
- ":pcm16b",
-]
-if (rtc_include_ilbc) {
- audio_codec_deps += [ ":ilbc" ]
-}
-if (rtc_include_opus) {
- audio_codec_deps += [ ":webrtc_opus" ]
-}
-if (current_cpu == "arm") {
- audio_codec_deps += [ ":isac_fix" ]
-} else {
- audio_codec_deps += [ ":isac" ]
-}
-audio_codec_deps += [ ":g722" ]
-if (!build_with_mozilla && !build_with_chromium) {
- audio_codec_deps += [ ":red" ]
-}
-audio_coding_deps = audio_codec_deps + [
- "../..:webrtc_common",
- "../../common_audio",
- "../../system_wrappers",
- ]
-
rtc_static_library("audio_format_conversion") {
visibility += webrtc_default_visibility
sources = [
@@ -63,7 +38,6 @@
# TODO(bugs.webrtc.org/9808): Move to private visibility as soon as that
# client code gets updated.
visibility += [ "*" ]
- allow_poison = [ "audio_codecs" ]
sources = [
"acm2/acm_codec_database.cc",
@@ -72,22 +46,18 @@
"acm2/rent_a_codec.h",
]
deps = [
- "../../rtc_base:checks",
- "../../api:array_view",
- "//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
- "../../api/audio_codecs:audio_codecs_api",
- "../..:webrtc_common",
- "../../rtc_base:protobuf_utils",
- "../../rtc_base:rtc_base_approved",
- "../../system_wrappers",
- ":audio_coding_module_typedefs",
- ":isac_common",
- ":isac_fix_c",
- ":audio_encoder_cng",
- ":neteq_decoder_enum",
- ] + audio_codec_deps
-
+ ":audio_coding_module_typedefs",
+ ":neteq_decoder_enum",
+ "../..:webrtc_common",
+ "../../api:array_view",
+ "../../api/audio_codecs:audio_codecs_api",
+ "../../rtc_base:checks",
+ "../../rtc_base:protobuf_utils",
+ "../../rtc_base:rtc_base_approved",
+ "../../system_wrappers",
+ "//third_party/abseil-cpp/absl/strings",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
defines = audio_codec_defines
}
@@ -102,7 +72,6 @@
rtc_static_library("audio_coding") {
visibility += [ "*" ]
- allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
sources = [
"acm2/acm_receiver.cc",
"acm2/acm_receiver.h",
@@ -111,40 +80,34 @@
"acm2/audio_coding_module.cc",
"acm2/call_statistics.cc",
"acm2/call_statistics.h",
- "acm2/codec_manager.cc",
- "acm2/codec_manager.h",
"include/audio_coding_module.h",
]
defines = []
- if (rtc_include_opus) {
- public_deps = [
- ":webrtc_opus",
- ]
- }
-
- deps = audio_coding_deps + [
- "../../system_wrappers:metrics",
- "../../api/audio:audio_frame_api",
- "..:module_api",
- "..:module_api_public",
- "../../common_audio:common_audio_c",
- "../../rtc_base:deprecation",
- "../../rtc_base:checks",
- "../../api:array_view",
- "../../api/audio_codecs:audio_codecs_api",
- ":audio_coding_module_typedefs",
- ":neteq",
- ":neteq_decoder_enum",
- ":rent_a_codec",
- "../../rtc_base:audio_format_to_string",
- "../../rtc_base:rtc_base_approved",
- "//third_party/abseil-cpp/absl/strings",
- "//third_party/abseil-cpp/absl/types:optional",
- "../../logging:rtc_event_log_api",
- ]
- defines = audio_coding_defines
+ deps = [
+ ":audio_coding_module_typedefs",
+ ":neteq",
+ ":neteq_decoder_enum",
+ ":rent_a_codec",
+ "..:module_api",
+ "..:module_api_public",
+ "../..:webrtc_common",
+ "../../api:array_view",
+ "../../api/audio:audio_frame_api",
+ "../../api/audio_codecs:audio_codecs_api",
+ "../../common_audio:common_audio",
+ "../../common_audio:common_audio_c",
+ "../../logging:rtc_event_log_api",
+ "../../rtc_base:audio_format_to_string",
+ "../../rtc_base:checks",
+ "../../rtc_base:deprecation",
+ "../../rtc_base:rtc_base_approved",
+ "../../system_wrappers",
+ "../../system_wrappers:metrics",
+ "//third_party/abseil-cpp/absl/strings",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
}
rtc_static_library("legacy_encoded_audio_frame") {
@@ -1280,6 +1243,30 @@
}
if (rtc_include_tests) {
+ audio_coding_deps = [
+ "../../common_audio",
+ "../../system_wrappers",
+ "../..:webrtc_common",
+ ":audio_encoder_cng",
+ ":g711",
+ ":g722",
+ ":pcm16b",
+ ]
+ if (rtc_include_ilbc) {
+ audio_coding_deps += [ ":ilbc" ]
+ }
+ if (rtc_include_opus) {
+ audio_coding_deps += [ ":webrtc_opus" ]
+ }
+ if (current_cpu == "arm") {
+ audio_coding_deps += [ ":isac_fix" ]
+ } else {
+ audio_coding_deps += [ ":isac" ]
+ }
+ if (!build_with_mozilla && !build_with_chromium) {
+ audio_coding_deps += [ ":red" ]
+ }
+
rtc_source_set("mocks") {
testonly = true
sources = [
@@ -1368,6 +1355,7 @@
":audio_format_conversion",
":pcm16b_c",
":red",
+ ":webrtc_opus_c",
"..:module_api",
"../..:webrtc_common",
"../../api/audio:audio_frame_api",
@@ -2007,8 +1995,6 @@
"acm2/acm_receiver_unittest.cc",
"acm2/audio_coding_module_unittest.cc",
"acm2/call_statistics_unittest.cc",
- "acm2/codec_manager_unittest.cc",
- "acm2/rent_a_codec_unittest.cc",
"audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
"audio_network_adaptor/bitrate_controller_unittest.cc",
"audio_network_adaptor/channel_controller_unittest.cc",