diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index 5f0f47b043..eaf2e747f8 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -709,8 +709,6 @@ source_set("audio_network_adaptor") { "audio_network_adaptor/controller.h", "audio_network_adaptor/controller_manager.cc", "audio_network_adaptor/controller_manager.h", - "audio_network_adaptor/debug_dump_writer.cc", - "audio_network_adaptor/debug_dump_writer.h", "audio_network_adaptor/dtx_controller.cc", "audio_network_adaptor/dtx_controller.h", "audio_network_adaptor/fec_controller.cc", @@ -723,13 +721,6 @@ source_set("audio_network_adaptor") { ] configs += [ "../..:common_config" ] public_configs = [ "../..:common_inherited_config" ] - - if (rtc_enable_protobuf) { - deps = [ - ":ana_debug_dump_proto", - ] - defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ] - } } config("neteq_config") { @@ -986,13 +977,6 @@ if (rtc_include_tests) { } # audio_decoder_unittests if (rtc_enable_protobuf) { - proto_library("ana_debug_dump_proto") { - sources = [ - "audio_network_adaptor/debug_dump.proto", - ] - proto_out_dir = "webrtc/modules/audio_coding/audio_network_adaptor" - } - proto_library("neteq_unittest_proto") { sources = [ "neteq/neteq_unittest.proto", diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi index 508b62b2e6..af6ba55c6d 100644 --- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi +++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi @@ -22,8 +22,6 @@ 'controller.cc', 'controller_manager.cc', 'controller_manager.h', - 'debug_dump_writer.cc', - 'debug_dump_writer.h', 'dtx_controller.h', 'dtx_controller.cc', 'fec_controller.h', @@ -33,32 +31,7 @@ 'include/audio_network_adaptor.h', 'smoothing_filter.h', 'smoothing_filter.cc', - ], # sources - 'conditions': [ - ['enable_protobuf==1', { - 'dependencies': ['debug_dump_proto'], - 'defines': ['WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP'], - }], - ], # conditions + ], # source }, ], # targets - - 'conditions': [ - ['enable_protobuf==1', { - 'targets': [ - { 'target_name': 'debug_dump_proto', - 'type': 'static_library', - 'sources': ['debug_dump.proto',], - 'variables': { - 'proto_in_dir': '.', - # Workaround to protect against gyp's pathname relativization when - # this file is included by modules.gyp. - 'proto_out_protected': 'webrtc/modules/audio_coding/audio_network_adaptor', - 'proto_out_dir': '<(proto_out_protected)', - }, - 'includes': ['../../../build/protoc.gypi',], - }, - ], # targets - }], - ], # conditions } 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 3d8b2beb8d..0303c84920 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 @@ -8,10 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h" - #include +#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h" + namespace webrtc { AudioNetworkAdaptorImpl::Config::Config() = default; @@ -21,15 +21,7 @@ AudioNetworkAdaptorImpl::Config::~Config() = default; AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl( const Config& config, std::unique_ptr controller_manager) - : AudioNetworkAdaptorImpl(config, std::move(controller_manager), nullptr) {} - -AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl( - const Config& config, - std::unique_ptr controller_manager, - std::unique_ptr debug_dump_writer) - : config_(config), - controller_manager_(std::move(controller_manager)), - debug_dump_writer_(std::move(debug_dump_writer)) { + : config_(config), controller_manager_(std::move(controller_manager)) { RTC_DCHECK(controller_manager_); } @@ -37,19 +29,16 @@ AudioNetworkAdaptorImpl::~AudioNetworkAdaptorImpl() = default; void AudioNetworkAdaptorImpl::SetUplinkBandwidth(int uplink_bandwidth_bps) { last_metrics_.uplink_bandwidth_bps = rtc::Optional(uplink_bandwidth_bps); - DumpNetworkMetrics(); + + // TODO(minyue): Add debug dumping. } void AudioNetworkAdaptorImpl::SetUplinkPacketLossFraction( float uplink_packet_loss_fraction) { last_metrics_.uplink_packet_loss_fraction = rtc::Optional(uplink_packet_loss_fraction); - DumpNetworkMetrics(); -} -void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) { - last_metrics_.rtt_ms = rtc::Optional(rtt_ms); - DumpNetworkMetrics(); + // TODO(minyue): Add debug dumping. } AudioNetworkAdaptor::EncoderRuntimeConfig @@ -60,9 +49,6 @@ AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() { controller->MakeDecision(last_metrics_, &config); // TODO(minyue): Add debug dumping. - if (debug_dump_writer_) - debug_dump_writer_->DumpEncoderRuntimeConfig( - config, config_.clock->TimeInMilliseconds()); return config; } @@ -81,17 +67,7 @@ void AudioNetworkAdaptorImpl::SetReceiverFrameLengthRange( } void AudioNetworkAdaptorImpl::StartDebugDump(FILE* file_handle) { - debug_dump_writer_ = DebugDumpWriter::Create(file_handle); -} - -void AudioNetworkAdaptorImpl::StopDebugDump() { - debug_dump_writer_.reset(nullptr); -} - -void AudioNetworkAdaptorImpl::DumpNetworkMetrics() { - if (debug_dump_writer_) - debug_dump_writer_->DumpNetworkMetrics(last_metrics_, - config_.clock->TimeInMilliseconds()); + // TODO(minyue): Implement this method. } } // namespace webrtc 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 289b6777e5..6f8d348d03 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 @@ -16,9 +16,7 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h" #include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h" -#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h" #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" -#include "webrtc/system_wrappers/include/clock.h" namespace webrtc { @@ -27,27 +25,18 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor { struct Config { Config(); ~Config(); - const Clock* clock; }; AudioNetworkAdaptorImpl( const Config& config, std::unique_ptr controller_manager); - // Dependency injection for testing. - AudioNetworkAdaptorImpl( - const Config& config, - std::unique_ptr controller_manager, - std::unique_ptr debug_dump_writer = nullptr); - ~AudioNetworkAdaptorImpl() override; void SetUplinkBandwidth(int uplink_bandwidth_bps) override; void SetUplinkPacketLossFraction(float uplink_packet_loss_fraction) override; - void SetRtt(int rtt_ms) override; - void SetReceiverFrameLengthRange(int min_frame_length_ms, int max_frame_length_ms) override; @@ -55,17 +44,11 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor { void StartDebugDump(FILE* file_handle) override; - void StopDebugDump() override; - private: - void DumpNetworkMetrics(); - const Config config_; std::unique_ptr controller_manager_; - std::unique_ptr debug_dump_writer_; - Controller::NetworkMetrics last_metrics_; RTC_DISALLOW_COPY_AND_ASSIGN(AudioNetworkAdaptorImpl); 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 aee41da1b0..65626743ad 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 @@ -15,25 +15,20 @@ #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h" #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h" #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h" -#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h" namespace webrtc { using ::testing::_; using ::testing::NiceMock; using ::testing::Return; -using ::testing::SetArgPointee; namespace { constexpr size_t kNumControllers = 2; -constexpr int64_t kClockInitialTimeMs = 12345678; - MATCHER_P(NetworkMetricsIs, metric, "") { return arg.uplink_bandwidth_bps == metric.uplink_bandwidth_bps && arg.target_audio_bitrate_bps == metric.target_audio_bitrate_bps && - arg.rtt_ms == metric.rtt_ms && arg.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction; } @@ -44,21 +39,9 @@ MATCHER_P(ConstraintsReceiverFrameLengthRangeIs, frame_length_range, "") { frame_length_range.max_frame_length_ms; } -MATCHER_P(EncoderRuntimeConfigIs, config, "") { - return arg.bitrate_bps == config.bitrate_bps && - arg.frame_length_ms == config.frame_length_ms && - arg.uplink_packet_loss_fraction == - config.uplink_packet_loss_fraction && - arg.enable_fec == config.enable_fec && - arg.enable_dtx == config.enable_dtx && - arg.num_channels == config.num_channels; -} - struct AudioNetworkAdaptorStates { std::unique_ptr audio_network_adaptor; std::vector> mock_controllers; - std::unique_ptr simulated_clock; - MockDebugDumpWriter* mock_debug_dump_writer; }; AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() { @@ -81,19 +64,9 @@ AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() { EXPECT_CALL(*controller_manager, GetSortedControllers(_)) .WillRepeatedly(Return(controllers)); - states.simulated_clock.reset(new SimulatedClock(kClockInitialTimeMs * 1000)); - - auto debug_dump_writer = - std::unique_ptr(new NiceMock()); - EXPECT_CALL(*debug_dump_writer, Die()); - states.mock_debug_dump_writer = debug_dump_writer.get(); - - AudioNetworkAdaptorImpl::Config config; - config.clock = states.simulated_clock.get(); // AudioNetworkAdaptorImpl governs the lifetime of controller manager. states.audio_network_adaptor.reset(new AudioNetworkAdaptorImpl( - config, - std::move(controller_manager), std::move(debug_dump_writer))); + AudioNetworkAdaptorImpl::Config(), std::move(controller_manager))); return states; } @@ -135,52 +108,4 @@ TEST(AudioNetworkAdaptorImplTest, SetConstraintsIsCalledOnSetFrameLengthRange) { states.audio_network_adaptor->SetReceiverFrameLengthRange(20, 120); } -TEST(AudioNetworkAdaptorImplTest, - DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) { - auto states = CreateAudioNetworkAdaptor(); - - AudioNetworkAdaptor::EncoderRuntimeConfig config; - config.bitrate_bps = rtc::Optional(32000); - config.enable_fec = rtc::Optional(true); - - EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_, _)) - .WillOnce(SetArgPointee<1>(config)); - - EXPECT_CALL(*states.mock_debug_dump_writer, - DumpEncoderRuntimeConfig(EncoderRuntimeConfigIs(config), - kClockInitialTimeMs)); - states.audio_network_adaptor->GetEncoderRuntimeConfig(); -} - -TEST(AudioNetworkAdaptorImplTest, - DumpNetworkMetricsIsCalledOnSetNetworkMetrics) { - auto states = CreateAudioNetworkAdaptor(); - - constexpr int kBandwidth = 16000; - constexpr float kPacketLoss = 0.7f; - constexpr int kRtt = 100; - - Controller::NetworkMetrics check; - check.uplink_bandwidth_bps = rtc::Optional(kBandwidth); - int64_t timestamp_check = kClockInitialTimeMs; - - EXPECT_CALL(*states.mock_debug_dump_writer, - DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check)); - states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth); - - states.simulated_clock->AdvanceTimeMilliseconds(100); - timestamp_check += 100; - check.uplink_packet_loss_fraction = rtc::Optional(kPacketLoss); - EXPECT_CALL(*states.mock_debug_dump_writer, - DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check)); - states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss); - - states.simulated_clock->AdvanceTimeMilliseconds(200); - timestamp_check += 200; - check.rtt_ms = rtc::Optional(kRtt); - EXPECT_CALL(*states.mock_debug_dump_writer, - DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check)); - states.audio_network_adaptor->SetRtt(kRtt); -} - } // namespace webrtc diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h index f27a391ad5..c1b16c72a9 100644 --- a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h +++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h @@ -24,7 +24,6 @@ class Controller { rtc::Optional uplink_bandwidth_bps; rtc::Optional uplink_packet_loss_fraction; rtc::Optional target_audio_bitrate_bps; - rtc::Optional rtt_ms; }; struct Constraints { diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto deleted file mode 100644 index f425244998..0000000000 --- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto +++ /dev/null @@ -1,30 +0,0 @@ -syntax = "proto2"; -option optimize_for = LITE_RUNTIME; -package webrtc.audio_network_adaptor.debug_dump; - -message NetworkMetrics { - optional int32 uplink_bandwidth_bps = 1; - optional float uplink_packet_loss_fraction = 2; - optional int32 target_audio_bitrate_bps = 3; - optional int32 rtt_ms = 4; -} - -message EncoderRuntimeConfig { - optional int32 bitrate_bps = 1; - optional int32 frame_length_ms = 2; - optional float uplink_packet_loss_fraction = 3; - optional bool enable_fec = 4; - optional bool enable_dtx = 5; - optional uint32 num_channels = 6; -} - -message Event { - enum Type { - NETWORK_METRICS = 0; - ENCODER_RUNTIME_CONFIG = 1; - } - required Type type = 1; - required uint32 timestamp = 2; - optional NetworkMetrics network_metrics = 3; - optional EncoderRuntimeConfig encoder_runtime_config = 4; -} 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 deleted file mode 100644 index 9992e2dbd7..0000000000 --- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc +++ /dev/null @@ -1,135 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h" - -#include "webrtc/base/checks.h" - -#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP -#ifdef WEBRTC_ANDROID_PLATFORM_BUILD -#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h" -#else -#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h" -#endif -#endif - -namespace webrtc { - -#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP -namespace { - -using audio_network_adaptor::debug_dump::Event; -using audio_network_adaptor::debug_dump::NetworkMetrics; -using audio_network_adaptor::debug_dump::EncoderRuntimeConfig; - -void DumpEventToFile(const Event& event, FileWrapper* dump_file) { - RTC_CHECK(dump_file->is_open()); - std::string dump_data; - event.SerializeToString(&dump_data); - int32_t size = event.ByteSize(); - dump_file->Write(&size, sizeof(size)); - dump_file->Write(dump_data.data(), dump_data.length()); -} - -} // namespace -#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP - -class DebugDumpWriterImpl final : public DebugDumpWriter { - public: - explicit DebugDumpWriterImpl(FILE* file_handle); - ~DebugDumpWriterImpl() override = default; - - void DumpEncoderRuntimeConfig( - const AudioNetworkAdaptor::EncoderRuntimeConfig& config, - int64_t timestamp) override; - - void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics, - int64_t timestamp) override; - - private: - std::unique_ptr dump_file_; -}; - -DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle) - : dump_file_(FileWrapper::Create()) { -#ifndef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP - RTC_DCHECK(false); -#endif - dump_file_->OpenFromFileHandle(file_handle); - RTC_CHECK(dump_file_->is_open()); -} - -void DebugDumpWriterImpl::DumpNetworkMetrics( - const Controller::NetworkMetrics& metrics, - int64_t timestamp) { -#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP - Event event; - event.set_timestamp(timestamp); - event.set_type(Event::NETWORK_METRICS); - auto dump_metrics = event.mutable_network_metrics(); - - if (metrics.uplink_bandwidth_bps) - dump_metrics->set_uplink_bandwidth_bps(*metrics.uplink_bandwidth_bps); - - if (metrics.uplink_packet_loss_fraction) { - dump_metrics->set_uplink_packet_loss_fraction( - *metrics.uplink_packet_loss_fraction); - } - - if (metrics.target_audio_bitrate_bps) { - dump_metrics->set_target_audio_bitrate_bps( - *metrics.target_audio_bitrate_bps); - } - - if (metrics.rtt_ms) - dump_metrics->set_rtt_ms(*metrics.rtt_ms); - - DumpEventToFile(event, dump_file_.get()); -#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP -} - -void DebugDumpWriterImpl::DumpEncoderRuntimeConfig( - const AudioNetworkAdaptor::EncoderRuntimeConfig& config, - int64_t timestamp) { -#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP - Event event; - event.set_timestamp(timestamp); - event.set_type(Event::ENCODER_RUNTIME_CONFIG); - auto dump_config = event.mutable_encoder_runtime_config(); - - if (config.bitrate_bps) - dump_config->set_bitrate_bps(*config.bitrate_bps); - - if (config.frame_length_ms) - dump_config->set_frame_length_ms(*config.frame_length_ms); - - if (config.uplink_packet_loss_fraction) { - dump_config->set_uplink_packet_loss_fraction( - *config.uplink_packet_loss_fraction); - } - - if (config.enable_fec) - dump_config->set_enable_fec(*config.enable_fec); - - if (config.enable_dtx) - dump_config->set_enable_dtx(*config.enable_dtx); - - if (config.num_channels) - dump_config->set_num_channels(*config.num_channels); - - DumpEventToFile(event, dump_file_.get()); -#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP -} - -std::unique_ptr DebugDumpWriter::Create(FILE* file_handle) { - return std::unique_ptr(new DebugDumpWriterImpl(file_handle)); -} - -} // namespace webrtc 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 deleted file mode 100644 index da4b0317f6..0000000000 --- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_ - -#include -#include - -#include "webrtc/base/constructormagic.h" -#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h" -#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" -#include "webrtc/system_wrappers/include/file_wrapper.h" - -namespace webrtc { - -class DebugDumpWriter { - public: - static std::unique_ptr Create(FILE* file_handle); - - virtual ~DebugDumpWriter() = default; - - virtual void DumpEncoderRuntimeConfig( - const AudioNetworkAdaptor::EncoderRuntimeConfig& config, - int64_t timestamp) = 0; - - virtual void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics, - int64_t timestamp) = 0; -}; - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_ 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 e3c4db033e..4b432763e2 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 @@ -43,16 +43,12 @@ class AudioNetworkAdaptor { virtual void SetUplinkPacketLossFraction( float uplink_packet_loss_fraction) = 0; - virtual void SetRtt(int rtt_ms) = 0; - virtual void SetReceiverFrameLengthRange(int min_frame_length_ms, int max_frame_length_ms) = 0; virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0; virtual void StartDebugDump(FILE* file_handle) = 0; - - virtual void StopDebugDump() = 0; }; } // 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 deleted file mode 100644 index f7e226e47f..0000000000 --- a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h +++ /dev/null @@ -1,34 +0,0 @@ -/* - * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_ - -#include "testing/gmock/include/gmock/gmock.h" -#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h" - -namespace webrtc { - -class MockDebugDumpWriter : public DebugDumpWriter { - public: - virtual ~MockDebugDumpWriter() { Die(); } - MOCK_METHOD0(Die, void()); - - MOCK_METHOD2(DumpEncoderRuntimeConfig, - void(const AudioNetworkAdaptor::EncoderRuntimeConfig& config, - int64_t timestamp)); - MOCK_METHOD2(DumpNetworkMetrics, - void(const Controller::NetworkMetrics& metrics, - int64_t timestamp)); -}; - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_