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Replacing WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP with WEBRTC_ENABLE_PROTOBUF.

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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}
This commit is contained in:
minyue-webrtc 2017-06-13 11:49:29 +02:00 committed by Commit Bot
parent 10e1f7548a
commit 7ed35f4643
4 changed files with 20 additions and 23 deletions
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6
webrtc/modules/audio_coding/BUILD.gn
View File

@ -933,7 +933,6 @@ rtc_static_library("audio_network_adaptor") {
":ana_config_proto",
":ana_debug_dump_proto",
]
defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
}
if (!build_with_chromium && is_clang) {
@ -2178,10 +2177,7 @@ if (rtc_include_tests) {
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",

10
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
View File

@ -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 webrtc {
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 @@ std::unique_ptr<BitrateController> CreateBitrateController(
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 @@ std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
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 @@ std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
#else
RTC_NOTREACHED();
return nullptr;
#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
#endif // WEBRTC_ENABLE_PROTOBUF
}
ControllerManagerImpl::ControllerManagerImpl(const Config& config)

6
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
View File

@ -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 @@ TEST(ControllerManagerTest, DoNotReorderIfNetworkMetricsChangeTooSmall) {
{kNumControllers - 2, kNumControllers - 1, 0, 1});
}
#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
#if WEBRTC_ENABLE_PROTOBUF
namespace {
@ -439,6 +439,6 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
ControllerType::CHANNEL, ControllerType::DTX,
ControllerType::BIT_RATE});
}
#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
#endif // WEBRTC_ENABLE_PROTOBUF
} // namespace webrtc

21
webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
View File

@ -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 @@ RTC_POP_IGNORING_WUNDEF()
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 @@ void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
}
} // namespace
#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
#endif // WEBRTC_ENABLE_PROTOBUF
class DebugDumpWriterImpl final : public DebugDumpWriter {
public:
@ -62,17 +62,18 @@ class DebugDumpWriterImpl final : public DebugDumpWriter {
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 @@ void DebugDumpWriterImpl::DumpNetworkMetrics(
}
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 @@ void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
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) {