AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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12
webrtc/modules/audio_processing/aec_dump/BUILD.gn
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@ -14,10 +14,10 @@ rtc_source_set("aec_dump") {
deps = [
"..:aec_dump_interface",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:file_wrapper",
"../../../rtc_base/system:rtc_export",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
if (rtc_include_tests) {
@ -46,7 +46,6 @@ if (rtc_include_tests) {
"..:api",
"..:audioproc_test_utils",
"../",
"../../../rtc_base:rtc_base_approved",
"//testing/gtest",
]
}
@ -59,8 +58,6 @@ if (rtc_enable_protobuf) {
"aec_dump_impl.h",
"capture_stream_info.cc",
"capture_stream_info.h",
"write_to_file_task.cc",
"write_to_file_task.h",
]
deps = [
@ -70,12 +67,16 @@ if (rtc_enable_protobuf) {
"../../../api/task_queue",
"../../../rtc_base:checks",
"../../../rtc_base:ignore_wundef",
"../../../rtc_base:logging",
"../../../rtc_base:macromagic",
"../../../rtc_base:protobuf_utils",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:race_checker",
"../../../rtc_base:rtc_event",
"../../../rtc_base:rtc_task_queue",
"../../../rtc_base/system:file_wrapper",
"../../../system_wrappers",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
deps += [ "../:audioproc_debug_proto" ]
}
@ -107,4 +108,5 @@ rtc_library("null_aec_dump_factory") {
":aec_dump",
"..:aec_dump_interface",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}

12
webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
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@ -12,8 +12,8 @@
#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
#include <memory>
#include <string>
#include "absl/strings/string_view.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "rtc_base/system/file_wrapper.h"
#include "rtc_base/system/rtc_export.h"
@ -26,16 +26,16 @@ namespace webrtc {
class RTC_EXPORT AecDumpFactory {
public:
// The |worker_queue| may not be null and must outlive the created
// AecDump instance. |max_log_size_bytes == -1| means the log size
// will be unlimited. |handle| may not be null. The AecDump takes
// responsibility for |handle| and closes it in the destructor. A
// The `worker_queue` may not be null and must outlive the created
// AecDump instance. `max_log_size_bytes == -1` means the log size
// will be unlimited. `handle` may not be null. The AecDump takes
// responsibility for `handle` and closes it in the destructor. A
// non-null return value indicates that the file has been
// sucessfully opened.
static std::unique_ptr<AecDump> Create(webrtc::FileWrapper file,
int64_t max_log_size_bytes,
rtc::TaskQueue* worker_queue);
static std::unique_ptr<AecDump> Create(std::string file_name,
static std::unique_ptr<AecDump> Create(absl::string_view file_name,
int64_t max_log_size_bytes,
rtc::TaskQueue* worker_queue);
static std::unique_ptr<AecDump> Create(FILE* handle,

3
webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
View File

@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "absl/strings/string_view.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/include/aec_dump.h"
@ -19,7 +20,7 @@ std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
return nullptr;
}
std::unique_ptr<AecDump> AecDumpFactory::Create(std::string file_name,
std::unique_ptr<AecDump> AecDumpFactory::Create(absl::string_view file_name,
int64_t max_log_size_bytes,
rtc::TaskQueue* worker_queue) {
return nullptr;