From 7a75415419cbd52d798f9226010e9190e1cbad53 Mon Sep 17 00:00:00 2001 From: Niklas Enbom Date: Fri, 19 Jun 2015 14:30:21 -0700 Subject: [PATCH] Revert "Added ACM_dump protobuf, class for reading/writing and unittest." This reverts commit e9bdfd859c309991b4ea759587f39eecdbd42bd4. This CL makes the GN chrome bot fail, not really sure why... FAILED: /mnt/data/b/build/goma/gomacc ../../third_party/llvm-build/Release+Asserts/bin/clang++ -MMD -MF obj/third_party/webrtc/modules/audio_coding/main/acm2/acm_dump.acm_dump.o.d -DRTC_AUDIOCODING_DEBUG_DUMP -DV8_DEPRECATION_WARNINGS -DCLD_VERSION=2 -DENABLE_MDNS=1 -DENABLE_NOTIFICATIONS -DENABLE_PEPPER_CDMS -DENABLE_PLUGINS=1 -DENABLE_PRINTING=1 -DENABLE_BASIC_PRINTING=1 -DENABLE_PRINT_PREVIEW=1 -DENABLE_SPELLCHECK=1 -DDONT_EMBED_BUILD_METADATA -DUSE_UDEV -DUI_COMPOSITOR_IMAGE_TRANSPORT -DUSE_ASH=1 -DUSE_AURA=1 -DUSE_PANGO=1 -DUSE_CAIRO=1 -DUSE_CLIPBOARD_AURAX11=1 -DUSE_DEFAULT_RENDER_THEME=1 -DUSE_GLIB=1 -DUSE_NSS_CERTS=1 -DUSE_X11=1 -DENABLE_WEBRTC=1 -DENABLE_EXTENSIONS=1 -DENABLE_CONFIGURATION_POLICY -DENABLE_TASK_MANAGER=1 -DENABLE_THEMES=1 -DENABLE_CAPTIVE_PORTAL_DETECTION=1 -DENABLE_SESSION_SERVICE=1 -DENABLE_APP_LIST=1 -DENABLE_SETTINGS_APP=1 -DENABLE_SUPERVISED_USERS=1 -DENABLE_SERVICE_DISCOVERY=1 -DENABLE_AUTOFILL_DIALOG=1 -DENABLE_REMOTING=1 -DENABLE_GOOGLE_NOW=1 -DENABLE_ONE_CLICK_SIGNIN -DENABLE_HIDPI=1 -DV8_USE_EXTERNAL_STARTUP_DATA -DENABLE_BACKGROUND=1 -DENABLE_PRE_SYNC_BACKUP -DFULL_SAFE_BROWSING -DSAFE_BROWSING_CSD -DSAFE_BROWSING_DB_LOCAL -DSAFE_BROWSING_SERVICE -DCHROMIUM_BUILD -DENABLE_MEDIA_ROUTER=1 -DCR_CLANG_REVISION=239765-1 -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_LARGEFILE64_SOURCE -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -DNDEBUG -DNVALGRIND -DDYNAMIC_ANNOTATIONS_ENABLED=0 -DGOOGLE_PROTOBUF_NO_RTTI -DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER -I../.. -Igen -I../../third_party/protobuf/src -Igen/protoc_out -I../../third_party/protobuf/src -I../../third_party/protobuf -fno-strict-aliasing -fstack-protector --param=ssp-buffer-size=4 -m64 -march=x86-64 -funwind-tables -fPIC -pipe -pthread -B../../third_party/binutils/Linux_x64/Release/bin -fcolor-diagnostics -Wall -Wsign-compare -Wendif-labels -Werror -Wno-missing-field-initializers -Wno-unused-parameter -Wno-c++11-narrowing -Wno-char-subscripts -Wno-covered-switch-default -Wno-deprecated-register -Wno-unneeded-internal-declaration -Wno-reserved-user-defined-literal -Wno-inconsistent-missing-override -fvisibility=hidden -Xclang -load -Xclang ../../third_party/llvm-build/Release+Asserts/lib/libFindBadConstructs.so -Xclang -plugin-arg-find-bad-constructs -Xclang check-templates -Xclang -add-plugin -Xclang find-bad-constructs -Wheader-hygiene -Wstring-conversion -O2 -fno-ident -fdata-sections -ffunction-sections -g1 -gsplit-dwarf -fno-threadsafe-statics -fvisibility-inlines-hidden -std=gnu++11 -fno-rtti -fno-exceptions -c ../../third_party/webrtc/modules/audio_coding/main/acm2/acm_dump.cc -o obj/third_party/webrtc/modules/audio_coding/main/acm2/acm_dump.acm_dump.o ../../third_party/webrtc/modules/audio_coding/main/acm2/acm_dump.cc:11:10: fatal error: 'webrtc/modules/audio_coding/main/acm2/acm_dump.h' file not found #include "webrtc/modules/audio_coding/main/acm2/acm_dump.h" ^ 1 error generated. ninja: build stopped: subcommand failed. TBR=ivoc@webrtc.org BUG= Review URL: https://codereview.webrtc.org/1195963002. Cr-Commit-Position: refs/heads/master@{#9474} --- DEPS | 1 - webrtc/modules/audio_coding/BUILD.gn | 25 -- .../audio_coding/main/acm2/acm_dump.cc | 220 ------------------ .../modules/audio_coding/main/acm2/acm_dump.h | 59 ----- .../main/acm2/acm_dump_unittest.cc | 117 ---------- .../main/acm2/audio_coding_module.gypi | 28 --- .../modules/audio_coding/main/acm2/dump.proto | 78 ------- webrtc/modules/modules.gyp | 7 +- 8 files changed, 1 insertion(+), 534 deletions(-) delete mode 100644 webrtc/modules/audio_coding/main/acm2/acm_dump.cc delete mode 100644 webrtc/modules/audio_coding/main/acm2/acm_dump.h delete mode 100644 webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc delete mode 100644 webrtc/modules/audio_coding/main/acm2/dump.proto diff --git a/DEPS b/DEPS index cd7dd3f86..6e8bee792 100644 --- a/DEPS +++ b/DEPS @@ -34,7 +34,6 @@ include_rules = [ # WebRTC production code. '-base', '-chromium', - '+external/webrtc/webrtc', # Android platform build. '+gflags', '+libyuv', '+net', diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index 6185d7a5d..7b7acd34d 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -7,7 +7,6 @@ # be found in the AUTHORS file in the root of the source tree. import("//build/config/arm.gni") -import("//third_party/protobuf/proto_library.gni") import("../../build/webrtc.gni") config("audio_coding_config") { @@ -80,30 +79,6 @@ source_set("audio_coding") { } } -proto_library("acm_dump_proto") { - sources = [ - "main/acm2/dump.proto", - ] - proto_out_dir = "webrtc/audio_coding" -} - -source_set("acm_dump") { - sources = [ - "main/acm2/acm_dump.cc", - "main/acm2/acm_dump.h", - ] - - defines = [] - - deps = [ - ":acm_dump_proto", - ] - - if (rtc_enable_protobuf) { - defines += [ "RTC_AUDIOCODING_DEBUG_DUMP" ] - } -} - source_set("audio_decoder_interface") { sources = [ "codecs/audio_decoder.cc", diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump.cc b/webrtc/modules/audio_coding/main/acm2/acm_dump.cc deleted file mode 100644 index 4454c2594..000000000 --- a/webrtc/modules/audio_coding/main/acm2/acm_dump.cc +++ /dev/null @@ -1,220 +0,0 @@ -/* - * Copyright (c) 2015 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/main/acm2/acm_dump.h" - -#include - -#include "webrtc/base/checks.h" -#include "webrtc/base/thread_annotations.h" -#include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/file_wrapper.h" - -// Files generated at build-time by the protobuf compiler. -#ifdef WEBRTC_ANDROID_PLATFORM_BUILD -#include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h" -#else -#include "webrtc/audio_coding/dump.pb.h" -#endif - -namespace webrtc { - -// Noop implementation if flag is not set -#ifndef RTC_AUDIOCODING_DEBUG_DUMP -class AcmDumpImpl final : public AcmDump { - public: - void StartLogging(const std::string& file_name, int duration_ms) override{}; - void LogRtpPacket(bool incoming, - const uint8_t* packet, - size_t length) override{}; - void LogDebugEvent(DebugEvent event_type, - const std::string& event_message) override{}; - void LogDebugEvent(DebugEvent event_type) override{}; -}; -#else - -class AcmDumpImpl final : public AcmDump { - public: - AcmDumpImpl(); - - void StartLogging(const std::string& file_name, int duration_ms) override; - void LogRtpPacket(bool incoming, - const uint8_t* packet, - size_t length) override; - void LogDebugEvent(DebugEvent event_type, - const std::string& event_message) override; - void LogDebugEvent(DebugEvent event_type) override; - - private: - // Checks if the logging time has expired, and if so stops the logging. - void StopIfNecessary() EXCLUSIVE_LOCKS_REQUIRED(crit_); - // Stops logging and clears the stored data and buffers. - void Clear() EXCLUSIVE_LOCKS_REQUIRED(crit_); - // Returns true if the logging is currently active. - bool CurrentlyLogging() const EXCLUSIVE_LOCKS_REQUIRED(crit_) { - return active_ && - (clock_->TimeInMicroseconds() <= start_time_us_ + duration_us_); - } - // This function is identical to LogDebugEvent, but requires holding the lock. - void LogDebugEventLocked(DebugEvent event_type, - const std::string& event_message) - EXCLUSIVE_LOCKS_REQUIRED(crit_); - - rtc::scoped_ptr crit_; - rtc::scoped_ptr file_ GUARDED_BY(crit_); - rtc::scoped_ptr stream_ GUARDED_BY(crit_); - bool active_ GUARDED_BY(crit_); - int64_t start_time_us_ GUARDED_BY(crit_); - int64_t duration_us_ GUARDED_BY(crit_); - const webrtc::Clock* clock_ GUARDED_BY(crit_); -}; - -namespace { - -// Convert from AcmDump's debug event enum (runtime format) to the corresponding -// protobuf enum (serialized format). -ACMDumpDebugEvent_EventType convertDebugEvent(AcmDump::DebugEvent event_type) { - switch (event_type) { - case AcmDump::DebugEvent::kLogStart: - return ACMDumpDebugEvent::LOG_START; - case AcmDump::DebugEvent::kLogEnd: - return ACMDumpDebugEvent::LOG_END; - case AcmDump::DebugEvent::kAudioPlayout: - return ACMDumpDebugEvent::AUDIO_PLAYOUT; - } - return ACMDumpDebugEvent::UNKNOWN_EVENT; -} - -} // Anonymous namespace. - -// AcmDumpImpl member functions. -AcmDumpImpl::AcmDumpImpl() - : crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), - file_(webrtc::FileWrapper::Create()), - stream_(new webrtc::ACMDumpEventStream()), - active_(false), - start_time_us_(0), - duration_us_(0), - clock_(webrtc::Clock::GetRealTimeClock()) { -} - -void AcmDumpImpl::StartLogging(const std::string& file_name, int duration_ms) { - CriticalSectionScoped lock(crit_.get()); - Clear(); - if (file_->OpenFile(file_name.c_str(), false) != 0) { - return; - } - // Add a single object to the stream that is reused at every log event. - stream_->add_stream(); - active_ = true; - start_time_us_ = clock_->TimeInMicroseconds(); - duration_us_ = static_cast(duration_ms) * 1000; - // Log the start event. - std::stringstream log_msg; - log_msg << "Initial timestamp: " << start_time_us_; - LogDebugEventLocked(DebugEvent::kLogStart, log_msg.str()); -} - -void AcmDumpImpl::LogRtpPacket(bool incoming, - const uint8_t* packet, - size_t length) { - CriticalSectionScoped lock(crit_.get()); - if (!CurrentlyLogging()) { - StopIfNecessary(); - return; - } - // Reuse the same object at every log event. - auto rtp_event = stream_->mutable_stream(0); - rtp_event->clear_debug_event(); - const int64_t timestamp = clock_->TimeInMicroseconds() - start_time_us_; - rtp_event->set_timestamp_us(timestamp); - rtp_event->set_type(webrtc::ACMDumpEvent::RTP_EVENT); - rtp_event->mutable_packet()->set_direction( - incoming ? ACMDumpRTPPacket::INCOMING : ACMDumpRTPPacket::OUTGOING); - rtp_event->mutable_packet()->set_rtp_data(packet, length); - std::string dump_buffer; - stream_->SerializeToString(&dump_buffer); - file_->Write(dump_buffer.data(), dump_buffer.size()); - file_->Flush(); -} - -void AcmDumpImpl::LogDebugEvent(DebugEvent event_type, - const std::string& event_message) { - CriticalSectionScoped lock(crit_.get()); - LogDebugEventLocked(event_type, event_message); -} - -void AcmDumpImpl::LogDebugEvent(DebugEvent event_type) { - CriticalSectionScoped lock(crit_.get()); - LogDebugEventLocked(event_type, ""); -} - -void AcmDumpImpl::StopIfNecessary() { - if (active_) { - DCHECK_GT(clock_->TimeInMicroseconds(), start_time_us_ + duration_us_); - LogDebugEventLocked(DebugEvent::kLogEnd, ""); - Clear(); - } -} - -void AcmDumpImpl::Clear() { - if (active_ || file_->Open()) { - file_->CloseFile(); - } - active_ = false; - stream_->Clear(); -} - -void AcmDumpImpl::LogDebugEventLocked(DebugEvent event_type, - const std::string& event_message) { - if (!CurrentlyLogging()) { - StopIfNecessary(); - return; - } - - // Reuse the same object at every log event. - auto event = stream_->mutable_stream(0); - int64_t timestamp = clock_->TimeInMicroseconds() - start_time_us_; - event->set_timestamp_us(timestamp); - event->set_type(webrtc::ACMDumpEvent::DEBUG_EVENT); - event->clear_packet(); - auto debug_event = event->mutable_debug_event(); - debug_event->set_type(convertDebugEvent(event_type)); - debug_event->set_message(event_message); - std::string dump_buffer; - stream_->SerializeToString(&dump_buffer); - file_->Write(dump_buffer.data(), dump_buffer.size()); -} - -#endif // RTC_AUDIOCODING_DEBUG_DUMP - -// AcmDump member functions. -rtc::scoped_ptr AcmDump::Create() { - return rtc::scoped_ptr(new AcmDumpImpl()); -} - -bool AcmDump::ParseAcmDump(const std::string& file_name, - ACMDumpEventStream* result) { - char tmp_buffer[1024]; - int bytes_read = 0; - rtc::scoped_ptr dump_file(FileWrapper::Create()); - if (dump_file->OpenFile(file_name.c_str(), true) != 0) { - return false; - } - std::string dump_buffer; - while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) { - dump_buffer.append(tmp_buffer, bytes_read); - } - dump_file->CloseFile(); - return result->ParseFromString(dump_buffer); -} - -} // namespace webrtc diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump.h b/webrtc/modules/audio_coding/main/acm2/acm_dump.h deleted file mode 100644 index c72c38709..000000000 --- a/webrtc/modules/audio_coding/main/acm2/acm_dump.h +++ /dev/null @@ -1,59 +0,0 @@ -/* - * Copyright (c) 2015 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_MAIN_ACM2_ACM_DUMP_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DUMP_H_ - -#include - -#include "webrtc/base/scoped_ptr.h" - -namespace webrtc { - -// Forward declaration of storage class that is automatically generated from -// the protobuf file. -class ACMDumpEventStream; - -class AcmDumpImpl; - -class AcmDump { - public: - // The types of debug events that are currently supported for logging. - enum class DebugEvent { kLogStart, kLogEnd, kAudioPlayout }; - - virtual ~AcmDump() {} - - static rtc::scoped_ptr Create(); - - // Starts logging for the specified duration to the specified file. - // The logging will stop automatically after the specified duration. - // If the file already exists it will be overwritten. - // The function will return false on failure. - virtual void StartLogging(const std::string& file_name, int duration_ms) = 0; - - // Logs an incoming or outgoing RTP packet. - virtual void LogRtpPacket(bool incoming, - const uint8_t* packet, - size_t length) = 0; - - // Logs a debug event, with optional message. - virtual void LogDebugEvent(DebugEvent event_type, - const std::string& event_message) = 0; - virtual void LogDebugEvent(DebugEvent event_type) = 0; - - // Reads an AcmDump file and returns true when reading was successful. - // The result is stored in the given ACMDumpEventStream object. - static bool ParseAcmDump(const std::string& file_name, - ACMDumpEventStream* result); -}; - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DUMP_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc deleted file mode 100644 index 55c948ebf..000000000 --- a/webrtc/modules/audio_coding/main/acm2/acm_dump_unittest.cc +++ /dev/null @@ -1,117 +0,0 @@ -/* - * Copyright (c) 2015 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. - */ - -#ifdef RTC_AUDIOCODING_DEBUG_DUMP - -#include -#include -#include - -#include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_dump.h" -#include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/test/test_suite.h" -#include "webrtc/test/testsupport/fileutils.h" -#include "webrtc/test/testsupport/gtest_disable.h" - -// Files generated at build-time by the protobuf compiler. -#ifdef WEBRTC_ANDROID_PLATFORM_BUILD -#include "external/webrtc/webrtc/modules/audio_coding/dump.pb.h" -#else -#include "webrtc/audio_coding/dump.pb.h" -#endif - -namespace webrtc { - -// Test for the acm dump class. Dumps some RTP packets to disk, then reads them -// back to see if they match. -class AcmDumpTest : public ::testing::Test { - public: - AcmDumpTest() : log_dumper_(AcmDump::Create()) {} - void VerifyResults(const ACMDumpEventStream& parsed_stream, - size_t packet_size) { - // Verify the result. - EXPECT_EQ(3, parsed_stream.stream_size()); - const ACMDumpEvent& start_event = parsed_stream.stream(0); - ASSERT_TRUE(start_event.has_type()); - EXPECT_EQ(ACMDumpEvent::DEBUG_EVENT, start_event.type()); - EXPECT_TRUE(start_event.has_timestamp_us()); - EXPECT_FALSE(start_event.has_packet()); - ASSERT_TRUE(start_event.has_debug_event()); - auto start_debug_event = start_event.debug_event(); - ASSERT_TRUE(start_debug_event.has_type()); - EXPECT_EQ(ACMDumpDebugEvent::LOG_START, start_debug_event.type()); - ASSERT_TRUE(start_debug_event.has_message()); - - for (int i = 1; i < parsed_stream.stream_size(); i++) { - const ACMDumpEvent& test_event = parsed_stream.stream(i); - ASSERT_TRUE(test_event.has_type()); - EXPECT_EQ(ACMDumpEvent::RTP_EVENT, test_event.type()); - EXPECT_TRUE(test_event.has_timestamp_us()); - EXPECT_FALSE(test_event.has_debug_event()); - ASSERT_TRUE(test_event.has_packet()); - const ACMDumpRTPPacket& test_packet = test_event.packet(); - ASSERT_TRUE(test_packet.has_direction()); - if (i == 1) { - EXPECT_EQ(ACMDumpRTPPacket::INCOMING, test_packet.direction()); - } else if (i == 2) { - EXPECT_EQ(ACMDumpRTPPacket::OUTGOING, test_packet.direction()); - } - ASSERT_TRUE(test_packet.has_rtp_data()); - ASSERT_EQ(packet_size, test_packet.rtp_data().size()); - for (size_t i = 0; i < packet_size; i++) { - EXPECT_EQ(rtp_packet_[i], - static_cast(test_packet.rtp_data()[i])); - } - } - } - - void Run(int packet_size, int random_seed) { - rtp_packet_.clear(); - rtp_packet_.reserve(packet_size); - srand(random_seed); - // Fill the packet vector with random data. - for (int i = 0; i < packet_size; i++) { - rtp_packet_.push_back(rand()); - } - // Find the name of the current test, in order to use it as a temporary - // filename. - auto test_info = ::testing::UnitTest::GetInstance()->current_test_info(); - const std::string temp_filename = - test::OutputPath() + test_info->test_case_name() + test_info->name(); - - log_dumper_->StartLogging(temp_filename, 10000000); - log_dumper_->LogRtpPacket(true, rtp_packet_.data(), rtp_packet_.size()); - log_dumper_->LogRtpPacket(false, rtp_packet_.data(), rtp_packet_.size()); - - // Read the generated file from disk. - ACMDumpEventStream parsed_stream; - - ASSERT_EQ(true, AcmDump::ParseAcmDump(temp_filename, &parsed_stream)); - - VerifyResults(parsed_stream, packet_size); - - // Clean up temporary file - can be pretty slow. - remove(temp_filename.c_str()); - } - - std::vector rtp_packet_; - rtc::scoped_ptr log_dumper_; -}; - -TEST_F(AcmDumpTest, DumpAndRead) { - Run(256, 321); - Run(256, 123); -} - -} // namespace webrtc - -#endif // RTC_AUDIOCODING_DEBUG_DUMP diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi index c78bcd74f..9a38faca8 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi @@ -78,34 +78,6 @@ 'nack.h', ], }, - { - 'target_name': 'acm_dump_proto', - 'type': 'static_library', - 'sources': ['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/audio_coding', - 'proto_out_dir': '<(proto_out_protected)', - }, - 'includes': ['../../../../build/protoc.gypi',], - }, - { - 'target_name': 'acm_dump', - 'type': 'static_library', - 'conditions': [ - ['enable_protobuf==1', { - 'defines': ['RTC_AUDIOCODING_DEBUG_DUMP'], - } - ], - ], - 'sources': [ - 'acm_dump.h', - 'acm_dump.cc' - ], - 'dependencies': ['acm_dump_proto'], - }, ], 'conditions': [ ['include_tests==1', { diff --git a/webrtc/modules/audio_coding/main/acm2/dump.proto b/webrtc/modules/audio_coding/main/acm2/dump.proto deleted file mode 100644 index 416bb7a61..000000000 --- a/webrtc/modules/audio_coding/main/acm2/dump.proto +++ /dev/null @@ -1,78 +0,0 @@ -syntax = "proto2"; -option optimize_for = LITE_RUNTIME; -package webrtc; - -// This is the main message to dump to a file, it can contain multiple event -// messages, but it is possible to append multiple EventStreams (each with a -// single event) to a file. -// This has the benefit that there's no need to keep all data in memory. -message ACMDumpEventStream { - repeated ACMDumpEvent stream = 1; -} - -message ACMDumpEvent { - // required - Elapsed wallclock time in us since the start of the log. - optional int64 timestamp_us = 1; - - // The different types of events that can occur, the UNKNOWN_EVENT entry - // is added in case future EventTypes are added, in that case old code will - // receive the new events as UNKNOWN_EVENT. - enum EventType { - UNKNOWN_EVENT = 0; - RTP_EVENT = 1; - DEBUG_EVENT = 2; - } - - // required - Indicates the type of this event - optional EventType type = 2; - - // optional - but required if type == RTP_EVENT - optional ACMDumpRTPPacket packet = 3; - - // optional - but required if type == DEBUG_EVENT - optional ACMDumpDebugEvent debug_event = 4; -} - -message ACMDumpRTPPacket { - // Indicates if the packet is incoming or outgoing with respect to the user - // that is logging the data. - enum Direction { - UNKNOWN_DIRECTION = 0; - OUTGOING = 1; - INCOMING = 2; - } - enum PayloadType { - UNKNOWN_TYPE = 0; - AUDIO = 1; - VIDEO = 2; - RTX = 3; - } - - // required - optional Direction direction = 1; - - // required - optional PayloadType type = 2; - - // required - Contains the whole RTP packet (header+payload). - optional bytes RTP_data = 3; -} - -message ACMDumpDebugEvent { - // Indicates the type of the debug event. - // LOG_START and LOG_END indicate the start and end of the log respectively. - // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM. - enum EventType { - UNKNOWN_EVENT = 0; - LOG_START = 1; - LOG_END = 2; - AUDIO_PLAYOUT = 3; - } - - // required - optional EventType type = 1; - - // An optional message that can be used to store additional information about - // the debug event. - optional string message = 2; -} \ No newline at end of file diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 150ee8e57..e29f68328 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -310,17 +310,12 @@ 'defines': [ 'WEBRTC_AUDIOPROC_FLOAT_PROFILE' ], }], ['enable_protobuf==1', { - 'defines': [ - 'WEBRTC_AUDIOPROC_DEBUG_DUMP', - 'RTC_AUDIOCODING_DEBUG_DUMP', - ], + 'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ], 'dependencies': [ - 'acm_dump', 'audioproc_protobuf_utils', 'audioproc_unittest_proto', ], 'sources': [ - 'audio_coding/main/acm2/acm_dump_unittest.cc', 'audio_processing/audio_processing_impl_unittest.cc', 'audio_processing/test/audio_processing_unittest.cc', 'audio_processing/test/test_utils.h',