Replacing WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP with WEBRTC_ENABLE_PROTOBUF.
Bug: None
Change-Id: I595b094e7fcb12723614df3197a40833932ba0a0
Reviewed-on: https://chromium-review.googlesource.com/533074
Reviewed-by: Michael T <tschumim@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18568}
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 8f87576..b2839be 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -933,7 +933,6 @@
":ana_config_proto",
":ana_debug_dump_proto",
]
- defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
}
if (!build_with_chromium && is_clang) {
@@ -2178,10 +2177,7 @@
defines = audio_coding_defines
if (rtc_enable_protobuf) {
- defines += [
- "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP",
- "WEBRTC_NETEQ_UNITTEST_BITEXACT",
- ]
+ defines += [ "WEBRTC_NETEQ_UNITTEST_BITEXACT" ]
deps += [
":ana_config_proto",
":neteq_unittest_proto",
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index c445ebe..8cb142f 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -23,7 +23,7 @@
#include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
@@ -37,7 +37,7 @@
namespace {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
std::unique_ptr<FecControllerPlrBased> CreateFecControllerPlrBased(
const audio_network_adaptor::config::FecController& config,
@@ -180,7 +180,7 @@
return std::unique_ptr<BitrateController>(new BitrateController(
BitrateController::Config(initial_bitrate_bps, initial_frame_length_ms)));
}
-#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif // WEBRTC_ENABLE_PROTOBUF
} // namespace
@@ -201,7 +201,7 @@
int initial_bitrate_bps,
bool initial_fec_enabled,
bool initial_dtx_enabled) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
audio_network_adaptor::config::ControllerManager controller_manager_config;
controller_manager_config.ParseFromString(config_string);
@@ -270,7 +270,7 @@
#else
RTC_NOTREACHED();
return nullptr;
-#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif // WEBRTC_ENABLE_PROTOBUF
}
ControllerManagerImpl::ControllerManagerImpl(const Config& config)
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index 52f1583..a2deb7b 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -17,7 +17,7 @@
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
#include "webrtc/test/gtest.h"
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
@@ -203,7 +203,7 @@
{kNumControllers - 2, kNumControllers - 1, 0, 1});
}
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
namespace {
@@ -439,6 +439,6 @@
ControllerType::CHANNEL, ControllerType::DTX,
ControllerType::BIT_RATE});
}
-#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif // WEBRTC_ENABLE_PROTOBUF
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
index 8b485e4..fdedf6c 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
@@ -14,7 +14,7 @@
#include "webrtc/base/ignore_wundef.h"
#include "webrtc/base/protobuf_utils.h"
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
@@ -26,7 +26,7 @@
namespace webrtc {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
namespace {
using audio_network_adaptor::debug_dump::Event;
@@ -43,7 +43,7 @@
}
} // namespace
-#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif // WEBRTC_ENABLE_PROTOBUF
class DebugDumpWriterImpl final : public DebugDumpWriter {
public:
@@ -62,17 +62,18 @@
DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle)
: dump_file_(FileWrapper::Create()) {
-#ifndef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
- RTC_NOTREACHED();
-#endif
+#if WEBRTC_ENABLE_PROTOBUF
dump_file_->OpenFromFileHandle(file_handle);
RTC_CHECK(dump_file_->is_open());
+#else
+ RTC_NOTREACHED();
+#endif
}
void DebugDumpWriterImpl::DumpNetworkMetrics(
const Controller::NetworkMetrics& metrics,
int64_t timestamp) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
Event event;
event.set_timestamp(timestamp);
event.set_type(Event::NETWORK_METRICS);
@@ -100,13 +101,13 @@
}
DumpEventToFile(event, dump_file_.get());
-#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif // WEBRTC_ENABLE_PROTOBUF
}
void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
const AudioEncoderRuntimeConfig& config,
int64_t timestamp) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
Event event;
event.set_timestamp(timestamp);
event.set_type(Event::ENCODER_RUNTIME_CONFIG);
@@ -133,7 +134,7 @@
dump_config->set_num_channels(*config.num_channels);
DumpEventToFile(event, dump_file_.get());
-#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif // WEBRTC_ENABLE_PROTOBUF
}
std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {