Resolve cyclic dependency between audio network adaptor and event log api
BUG=webrtc:7257
Review-Url: https://codereview.webrtc.org/2745473003
Cr-Commit-Position: refs/heads/master@{#17565}
diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn
index 0cf5731..aedbd9e 100644
--- a/webrtc/logging/BUILD.gn
+++ b/webrtc/logging/BUILD.gn
@@ -87,6 +87,7 @@
":rtc_event_log_proto",
"..:webrtc_common",
"../call:call_interfaces",
+ "../modules/audio_coding:audio_network_adaptor",
"../modules/rtp_rtcp:rtp_rtcp",
"../system_wrappers",
]
@@ -114,6 +115,7 @@
"../base:rtc_base_approved",
"../base:rtc_base_tests_utils",
"../call",
+ "../modules/audio_coding:audio_network_adaptor",
"../modules/rtp_rtcp",
"../system_wrappers:metrics_default",
"../test:test_support",
diff --git a/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h b/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
index fcc5700..154882f 100644
--- a/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
+++ b/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
@@ -14,6 +14,7 @@
#include <string>
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/test/gmock.h"
namespace webrtc {
@@ -70,7 +71,7 @@
void(int32_t bitrate_bps, BandwidthUsage detector_state));
MOCK_METHOD1(LogAudioNetworkAdaptation,
- void(const AudioNetworkAdaptor::EncoderRuntimeConfig& config));
+ void(const AudioEncoderRuntimeConfig& config));
MOCK_METHOD4(LogProbeClusterCreated,
void(int id, int bitrate_bps, int min_probes, int min_bytes));
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc
index 970d0df..d10dc98 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc
@@ -22,6 +22,7 @@
#include "webrtc/base/timeutils.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
@@ -29,8 +30,8 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
@@ -85,7 +86,7 @@
void LogDelayBasedBweUpdate(int32_t bitrate_bps,
BandwidthUsage detector_state) override;
void LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override;
+ const AudioEncoderRuntimeConfig& config) override;
void LogProbeClusterCreated(int id,
int bitrate_bps,
int min_probes,
@@ -504,7 +505,7 @@
}
void RtcEventLogImpl::LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+ const AudioEncoderRuntimeConfig& config) {
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
event->set_timestamp_us(rtc::TimeMicros());
event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.h b/webrtc/logging/rtc_event_log/rtc_event_log.h
index ccb37b3..f842252 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.h
@@ -17,7 +17,6 @@
#include "webrtc/base/platform_file.h"
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/audio_send_stream.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@@ -32,6 +31,7 @@
class Clock;
class RtcEventLogImpl;
+struct AudioEncoderRuntimeConfig;
enum class MediaType;
@@ -135,7 +135,7 @@
// Logs audio encoder re-configuration driven by audio network adaptor.
virtual void LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) = 0;
+ const AudioEncoderRuntimeConfig& config) = 0;
// Logs when a probe cluster is created.
virtual void LogProbeClusterCreated(int id,
@@ -199,7 +199,7 @@
void LogDelayBasedBweUpdate(int32_t bitrate_bps,
BandwidthUsage detector_state) override {}
void LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override {}
+ const AudioEncoderRuntimeConfig& config) override {}
void LogProbeClusterCreated(int id,
int bitrate_bps,
int min_probes,
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
index 815308d..ec10396 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -22,6 +22,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
@@ -511,7 +512,7 @@
void ParsedRtcEventLog::GetAudioNetworkAdaptation(
size_t index,
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) const {
+ AudioEncoderRuntimeConfig* config) const {
RTC_CHECK_LT(index, GetNumberOfEvents());
const rtclog::Event& event = events_[index];
RTC_CHECK(event.has_type());
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.h b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
index bb3c406..d711739 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
@@ -151,11 +151,10 @@
BandwidthUsage* detector_state) const;
// Reads a audio network adaptation event to a (non-NULL)
- // AudioNetworkAdaptor::EncoderRuntimeConfig struct. Only the fields that are
+ // AudioEncoderRuntimeConfig struct. Only the fields that are
// stored in the protobuf will be written.
- void GetAudioNetworkAdaptation(
- size_t index,
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) const;
+ void GetAudioNetworkAdaptation(size_t index,
+ AudioEncoderRuntimeConfig* config) const;
ParsedRtcEventLog::BweProbeClusterCreatedEvent GetBweProbeClusterCreated(
size_t index) const;
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
index 71c588c..d41a883 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -22,6 +22,7 @@
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
@@ -227,10 +228,9 @@
}
}
-void GenerateAudioNetworkAdaptation(
- uint32_t extensions_bitvector,
- AudioNetworkAdaptor::EncoderRuntimeConfig* config,
- Random* prng) {
+void GenerateAudioNetworkAdaptation(uint32_t extensions_bitvector,
+ AudioEncoderRuntimeConfig* config,
+ Random* prng) {
config->bitrate_bps = rtc::Optional<int>(prng->Rand(0, 3000000));
config->enable_fec = rtc::Optional<bool>(prng->Rand<bool>());
config->enable_dtx = rtc::Optional<bool>(prng->Rand<bool>());
@@ -859,7 +859,7 @@
RtcEventLogTestHelper::VerifyAudioNetworkAdaptation(parsed_log, index,
config);
}
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
};
TEST(RtcEventLogTest, LogAudioReceiveConfig) {
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
index e66c090..7519ee5 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
@@ -15,6 +15,7 @@
#include <string>
#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
@@ -548,8 +549,8 @@
void RtcEventLogTestHelper::VerifyAudioNetworkAdaptation(
const ParsedRtcEventLog& parsed_log,
size_t index,
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
- AudioNetworkAdaptor::EncoderRuntimeConfig parsed_config;
+ const AudioEncoderRuntimeConfig& config) {
+ AudioEncoderRuntimeConfig parsed_config;
parsed_log.GetAudioNetworkAdaptation(index, &parsed_config);
EXPECT_EQ(config.bitrate_bps, parsed_config.bitrate_bps);
EXPECT_EQ(config.enable_dtx, parsed_config.enable_dtx);
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
index 0ca2d62..235d112 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
@@ -63,7 +63,7 @@
static void VerifyAudioNetworkAdaptation(
const ParsedRtcEventLog& parsed_log,
size_t index,
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config);
+ const AudioEncoderRuntimeConfig& config);
static void VerifyLogStartEvent(const ParsedRtcEventLog& parsed_log,
size_t index);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.cc
index bf79d78..ce1e250 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.cc
@@ -12,11 +12,11 @@
namespace webrtc {
-AudioNetworkAdaptor::EncoderRuntimeConfig::EncoderRuntimeConfig() = default;
+AudioEncoderRuntimeConfig::AudioEncoderRuntimeConfig() = default;
-AudioNetworkAdaptor::EncoderRuntimeConfig::~EncoderRuntimeConfig() = default;
+AudioEncoderRuntimeConfig::~AudioEncoderRuntimeConfig() = default;
-AudioNetworkAdaptor::EncoderRuntimeConfig::EncoderRuntimeConfig(
- const EncoderRuntimeConfig& other) = default;
+AudioEncoderRuntimeConfig::AudioEncoderRuntimeConfig(
+ const AudioEncoderRuntimeConfig& other) = default;
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index e1952f4..7408df2 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -112,9 +112,8 @@
UpdateNetworkMetrics(network_metrics);
}
-AudioNetworkAdaptor::EncoderRuntimeConfig
-AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
- EncoderRuntimeConfig config;
+AudioEncoderRuntimeConfig AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
+ AudioEncoderRuntimeConfig config;
for (auto& controller :
controller_manager_->GetSortedControllers(last_metrics_))
controller->MakeDecision(&config);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index 3713bda..f7bf70d 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -54,7 +54,7 @@
void SetOverhead(size_t overhead_bytes_per_packet) override;
- EncoderRuntimeConfig GetEncoderRuntimeConfig() override;
+ AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() override;
void StartDebugDump(FILE* file_handle) override;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index c434be3..53334c6 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -182,7 +182,7 @@
DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) {
auto states = CreateAudioNetworkAdaptor();
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
config.bitrate_bps = rtc::Optional<int>(32000);
config.enable_fec = rtc::Optional<bool>(true);
@@ -255,7 +255,7 @@
TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) {
auto states = CreateAudioNetworkAdaptor();
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
config.bitrate_bps = rtc::Optional<int>(32000);
config.enable_fec = rtc::Optional<bool>(true);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
index d8c74cd..92a9fad 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
@@ -43,8 +43,7 @@
overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet;
}
-void BitrateController::MakeDecision(
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+void BitrateController::MakeDecision(AudioEncoderRuntimeConfig* config) {
// Decision on |bitrate_bps| should not have been made.
RTC_DCHECK(!config->bitrate_bps);
if (target_audio_bitrate_bps_ && overhead_bytes_per_packet_) {
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
index 5e03b45..ac13c50 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
@@ -32,7 +32,7 @@
void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
- void MakeDecision(AudioNetworkAdaptor::EncoderRuntimeConfig* config) override;
+ void MakeDecision(AudioEncoderRuntimeConfig* config) override;
private:
const Config config_;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
index a90cb9a..9fab781 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
@@ -39,7 +39,7 @@
void CheckDecision(BitrateController* controller,
const rtc::Optional<int>& frame_length_ms,
int expected_bitrate_bps) {
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
config.frame_length_ms = frame_length_ms;
controller->MakeDecision(&config);
EXPECT_EQ(rtc::Optional<int>(expected_bitrate_bps), config.bitrate_bps);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
index 77217a3..90c1e56 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
@@ -41,8 +41,7 @@
uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
}
-void ChannelController::MakeDecision(
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+void ChannelController::MakeDecision(AudioEncoderRuntimeConfig* config) {
// Decision on |num_channels| should not have been made.
RTC_DCHECK(!config->num_channels);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
index 0bcb4fd..9355d30 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
@@ -39,7 +39,7 @@
void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
- void MakeDecision(AudioNetworkAdaptor::EncoderRuntimeConfig* config) override;
+ void MakeDecision(AudioEncoderRuntimeConfig* config) override;
private:
const Config config_;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
index def2e51..980292c 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
@@ -39,7 +39,7 @@
network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
controller->UpdateNetworkMetrics(network_metrics);
}
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
controller->MakeDecision(&config);
EXPECT_EQ(rtc::Optional<size_t>(expected_num_channels), config.num_channels);
}
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
index 0ed23c8..4ae7951 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
@@ -35,8 +35,7 @@
// indicates an update on the corresponding network metric.
virtual void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) = 0;
- virtual void MakeDecision(
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) = 0;
+ virtual void MakeDecision(AudioEncoderRuntimeConfig* config) = 0;
};
} // namespace webrtc
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 ed96e1b..4a2a57b 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
@@ -299,10 +299,10 @@
ASSERT_EQ(expected_types.size(), controllers.size());
// We also check that the controllers follow the initial settings.
- AudioNetworkAdaptor::EncoderRuntimeConfig encoder_config;
+ AudioEncoderRuntimeConfig encoder_config;
for (size_t i = 0; i < controllers.size(); ++i) {
- AudioNetworkAdaptor::EncoderRuntimeConfig encoder_config;
+ AudioEncoderRuntimeConfig encoder_config;
// We check the order of |controllers| by judging their decisions.
controllers[i]->MakeDecision(&encoder_config);
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 e0af336..2e4757a 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
@@ -49,9 +49,8 @@
explicit DebugDumpWriterImpl(FILE* file_handle);
~DebugDumpWriterImpl() override = default;
- void DumpEncoderRuntimeConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
- int64_t timestamp) override;
+ void DumpEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& config,
+ int64_t timestamp) override;
void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
int64_t timestamp) override;
@@ -104,7 +103,7 @@
}
void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
+ const AudioEncoderRuntimeConfig& config,
int64_t timestamp) {
#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
Event event;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
index da4b031..1661cd3 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
@@ -27,9 +27,8 @@
virtual ~DebugDumpWriter() = default;
- virtual void DumpEncoderRuntimeConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
- int64_t timestamp) = 0;
+ virtual void DumpEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& config,
+ int64_t timestamp) = 0;
virtual void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
int64_t timestamp) = 0;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
index d03bd39..fc1f44d 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
@@ -31,8 +31,7 @@
uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
}
-void DtxController::MakeDecision(
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+void DtxController::MakeDecision(AudioEncoderRuntimeConfig* config) {
// Decision on |enable_dtx| should not have been made.
RTC_DCHECK(!config->enable_dtx);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
index 1bf2ce7..583ef3c 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
@@ -35,7 +35,7 @@
void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
- void MakeDecision(AudioNetworkAdaptor::EncoderRuntimeConfig* config) override;
+ void MakeDecision(AudioEncoderRuntimeConfig* config) override;
private:
const Config config_;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
index 7b60e8f..73527ee 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
@@ -37,7 +37,7 @@
network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
controller->UpdateNetworkMetrics(network_metrics);
}
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
controller->MakeDecision(&config);
EXPECT_EQ(rtc::Optional<bool>(expected_dtx_enabled), config.enable_dtx);
}
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
index 619a247..b4fcbfd 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
@@ -30,7 +30,7 @@
EventLogWriter::~EventLogWriter() = default;
void EventLogWriter::MaybeLogEncoderConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+ const AudioEncoderRuntimeConfig& config) {
if (last_logged_config_.num_channels != config.num_channels)
return LogEncoderConfig(config);
if (last_logged_config_.enable_dtx != config.enable_dtx)
@@ -59,8 +59,7 @@
}
}
-void EventLogWriter::LogEncoderConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+void EventLogWriter::LogEncoderConfig(const AudioEncoderRuntimeConfig& config) {
event_log_->LogAudioNetworkAdaptation(config);
last_logged_config_ = config;
}
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
index 740da8c..d0b38bd 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
@@ -24,18 +24,16 @@
float min_bitrate_change_fraction,
float min_packet_loss_change_fraction);
~EventLogWriter();
- void MaybeLogEncoderConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config);
+ void MaybeLogEncoderConfig(const AudioEncoderRuntimeConfig& config);
private:
- void LogEncoderConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config);
+ void LogEncoderConfig(const AudioEncoderRuntimeConfig& config);
RtcEventLog* const event_log_;
const int min_bitrate_change_bps_;
const float min_bitrate_change_fraction_;
const float min_packet_loss_change_fraction_;
- AudioNetworkAdaptor::EncoderRuntimeConfig last_logged_config_;
+ AudioEncoderRuntimeConfig last_logged_config_;
RTC_DISALLOW_COPY_AND_ASSIGN(EventLogWriter);
};
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
index 289b8e2..443e4d1 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
@@ -43,7 +43,7 @@
struct EventLogWriterStates {
std::unique_ptr<EventLogWriter> event_log_writer;
std::unique_ptr<testing::StrictMock<MockRtcEventLog>> event_log;
- AudioNetworkAdaptor::EncoderRuntimeConfig runtime_config;
+ AudioEncoderRuntimeConfig runtime_config;
};
EventLogWriterStates CreateEventLogWriter() {
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
index 835970f..c39457d 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
@@ -79,8 +79,7 @@
}
}
-void FecControllerPlrBased::MakeDecision(
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+void FecControllerPlrBased::MakeDecision(AudioEncoderRuntimeConfig* config) {
RTC_DCHECK(!config->enable_fec);
RTC_DCHECK(!config->uplink_packet_loss_fraction);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
index 98d8543..52d0265 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
@@ -55,7 +55,7 @@
void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
- void MakeDecision(AudioNetworkAdaptor::EncoderRuntimeConfig* config) override;
+ void MakeDecision(AudioEncoderRuntimeConfig* config) override;
private:
bool FecEnablingDecision(const rtc::Optional<float>& packet_loss) const;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
index f55a443..0830479 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
@@ -96,7 +96,7 @@
void CheckDecision(FecControllerPlrBasedTestStates* states,
bool expected_enable_fec,
float expected_uplink_packet_loss_fraction) {
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
states->controller->MakeDecision(&config);
EXPECT_EQ(rtc::Optional<bool>(expected_enable_fec), config.enable_fec);
EXPECT_EQ(rtc::Optional<float>(expected_uplink_packet_loss_fraction),
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
index 1cab719..d21e8be 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
@@ -42,8 +42,7 @@
}
}
-void FecControllerRplrBased::MakeDecision(
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+void FecControllerRplrBased::MakeDecision(AudioEncoderRuntimeConfig* config) {
RTC_DCHECK(!config->enable_fec);
RTC_DCHECK(!config->uplink_packet_loss_fraction);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
index 849f43c..b2904b3 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
@@ -47,7 +47,7 @@
void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
- void MakeDecision(AudioNetworkAdaptor::EncoderRuntimeConfig* config) override;
+ void MakeDecision(AudioEncoderRuntimeConfig* config) override;
private:
bool FecEnablingDecision() const;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
index 6cb63cd..0376b9a 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
@@ -113,7 +113,7 @@
void CheckDecision(FecControllerRplrBased* controller,
bool expected_enable_fec,
float expected_uplink_packet_loss_fraction) {
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
controller->MakeDecision(&config);
// Less compact than comparing optionals, but yields more readable errors.
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
index 580d080..5111b8a 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
@@ -65,8 +65,7 @@
overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet;
}
-void FrameLengthController::MakeDecision(
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+void FrameLengthController::MakeDecision(AudioEncoderRuntimeConfig* config) {
// Decision on |frame_length_ms| should not have been made.
RTC_DCHECK(!config->frame_length_ms);
@@ -92,7 +91,7 @@
}
bool FrameLengthController::FrameLengthIncreasingDecision(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) const {
+ const AudioEncoderRuntimeConfig& config) const {
// Increase frame length if
// 1. |uplink_bandwidth_bps| is known to be smaller or equal than
// |min_encoder_bitrate_bps| plus |prevent_overuse_margin_bps| plus the
@@ -129,7 +128,7 @@
}
bool FrameLengthController::FrameLengthDecreasingDecision(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) const {
+ const AudioEncoderRuntimeConfig& config) const {
// Decrease frame length if
// 1. shorter frame length is available AND
// 2. |uplink_bandwidth_bps| is known to be bigger than
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
index 74cbb56..4589382 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
@@ -54,14 +54,14 @@
void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
- void MakeDecision(AudioNetworkAdaptor::EncoderRuntimeConfig* config) override;
+ void MakeDecision(AudioEncoderRuntimeConfig* config) override;
private:
bool FrameLengthIncreasingDecision(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) const;
+ const AudioEncoderRuntimeConfig& config) const;
bool FrameLengthDecreasingDecision(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) const;
+ const AudioEncoderRuntimeConfig& config) const;
const Config config_;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
index ac888b6..beff4ed 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
@@ -101,7 +101,7 @@
void CheckDecision(FrameLengthController* controller,
const rtc::Optional<bool>& enable_fec,
int expected_frame_length_ms) {
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
config.enable_fec = enable_fec;
controller->MakeDecision(&config);
EXPECT_EQ(rtc::Optional<int>(expected_frame_length_ms),
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
index 0ad4a1e..2ef8854 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
@@ -15,28 +15,29 @@
namespace webrtc {
+struct AudioEncoderRuntimeConfig {
+ AudioEncoderRuntimeConfig();
+ AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
+ ~AudioEncoderRuntimeConfig();
+ rtc::Optional<int> bitrate_bps;
+ rtc::Optional<int> frame_length_ms;
+ // Note: This is what we tell the encoder. It doesn't have to reflect
+ // the actual NetworkMetrics; it's subject to our decision.
+ rtc::Optional<float> uplink_packet_loss_fraction;
+ rtc::Optional<bool> enable_fec;
+ rtc::Optional<bool> enable_dtx;
+
+ // Some encoders can encode fewer channels than the actual input to make
+ // better use of the bandwidth. |num_channels| sets the number of channels
+ // to encode.
+ rtc::Optional<size_t> num_channels;
+};
+
// An AudioNetworkAdaptor optimizes the audio experience by suggesting a
// suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
// encoder based on network metrics.
class AudioNetworkAdaptor {
public:
- struct EncoderRuntimeConfig {
- EncoderRuntimeConfig();
- EncoderRuntimeConfig(const EncoderRuntimeConfig& other);
- ~EncoderRuntimeConfig();
- rtc::Optional<int> bitrate_bps;
- rtc::Optional<int> frame_length_ms;
- // Note: This is what we tell the encoder. It doesn't have to reflect
- // the actual NetworkMetrics; it's subject to our decision.
- rtc::Optional<float> uplink_packet_loss_fraction;
- rtc::Optional<bool> enable_fec;
- rtc::Optional<bool> enable_dtx;
-
- // Some encoders can encode fewer channels than the actual input to make
- // better use of the bandwidth. |num_channels| sets the number of channels
- // to encode.
- rtc::Optional<size_t> num_channels;
- };
virtual ~AudioNetworkAdaptor() = default;
@@ -54,7 +55,7 @@
virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0;
- virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
+ virtual AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
virtual void StartDebugDump(FILE* file_handle) = 0;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
index 104dde6..4b9a477 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
@@ -35,7 +35,7 @@
MOCK_METHOD1(SetOverhead, void(size_t overhead_bytes_per_packet));
- MOCK_METHOD0(GetEncoderRuntimeConfig, EncoderRuntimeConfig());
+ MOCK_METHOD0(GetEncoderRuntimeConfig, AudioEncoderRuntimeConfig());
MOCK_METHOD1(StartDebugDump, void(FILE* file_handle));
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h
index 2b8dc9e..e856601 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h
@@ -22,8 +22,7 @@
MOCK_METHOD0(Die, void());
MOCK_METHOD1(UpdateNetworkMetrics,
void(const NetworkMetrics& network_metrics));
- MOCK_METHOD1(MakeDecision,
- void(AudioNetworkAdaptor::EncoderRuntimeConfig* config));
+ MOCK_METHOD1(MakeDecision, void(AudioEncoderRuntimeConfig* config));
};
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
index 6a20f7a..a276b81 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
@@ -22,7 +22,7 @@
MOCK_METHOD0(Die, void());
MOCK_METHOD2(DumpEncoderRuntimeConfig,
- void(const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
+ void(const AudioEncoderRuntimeConfig& config,
int64_t timestamp));
MOCK_METHOD2(DumpNetworkMetrics,
void(const Controller::NetworkMetrics& metrics,
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index c5fefc4..04c0cf1 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -87,14 +87,14 @@
return states;
}
-AudioNetworkAdaptor::EncoderRuntimeConfig CreateEncoderRuntimeConfig() {
+AudioEncoderRuntimeConfig CreateEncoderRuntimeConfig() {
constexpr int kBitrate = 40000;
constexpr int kFrameLength = 60;
constexpr bool kEnableFec = true;
constexpr bool kEnableDtx = false;
constexpr size_t kNumChannels = 1;
constexpr float kPacketLossFraction = 0.1f;
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
config.bitrate_bps = rtc::Optional<int>(kBitrate);
config.frame_length_ms = rtc::Optional<int>(kFrameLength);
config.enable_fec = rtc::Optional<bool>(kEnableFec);
@@ -105,9 +105,8 @@
return config;
}
-void CheckEncoderRuntimeConfig(
- const AudioEncoderOpus* encoder,
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+void CheckEncoderRuntimeConfig(const AudioEncoderOpus* encoder,
+ const AudioEncoderRuntimeConfig& config) {
EXPECT_EQ(*config.bitrate_bps, encoder->GetTargetBitrate());
EXPECT_EQ(*config.frame_length_ms, encoder->next_frame_length_ms());
EXPECT_EQ(*config.enable_fec, encoder->fec_enabled());
@@ -472,7 +471,7 @@
states.encoder->EnableAudioNetworkAdaptor("", nullptr, nullptr);
auto config = CreateEncoderRuntimeConfig();
- AudioNetworkAdaptor::EncoderRuntimeConfig empty_config;
+ AudioEncoderRuntimeConfig empty_config;
EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
.WillOnce(Return(config))
diff --git a/webrtc/tools/DEPS b/webrtc/tools/DEPS
index ac56340..84bf153 100644
--- a/webrtc/tools/DEPS
+++ b/webrtc/tools/DEPS
@@ -4,6 +4,7 @@
"+webrtc/common_video",
"+webrtc/logging/rtc_event_log",
"+webrtc/modules/audio_device",
+ "+webrtc/modules/audio_coding/audio_network_adaptor",
"+webrtc/modules/audio_processing",
"+webrtc/modules/bitrate_controller",
"+webrtc/modules/congestion_controller",
diff --git a/webrtc/tools/event_log_visualizer/analyzer.h b/webrtc/tools/event_log_visualizer/analyzer.h
index 1acf756..d72ad31 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.h
+++ b/webrtc/tools/event_log_visualizer/analyzer.h
@@ -20,6 +20,7 @@
#include "webrtc/base/function_view.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/tools/event_log_visualizer/plot_base.h"
@@ -55,7 +56,7 @@
struct AudioNetworkAdaptationEvent {
uint64_t timestamp;
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
+ AudioEncoderRuntimeConfig config;
};
class EventLogAnalyzer {
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 38bc45a..3a4a970 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -163,7 +163,7 @@
}
void LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override {
+ const AudioEncoderRuntimeConfig& config) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
event_log_->LogAudioNetworkAdaptation(config);