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}
diff --git a/DEPS b/DEPS
index cd7dd3f..6e8bee7 100644
--- a/DEPS
+++ b/DEPS
@@ -34,7 +34,6 @@
# 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 6185d7a..7b7acd3 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 @@
}
}
-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 4454c25..0000000
--- 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 <sstream>
-
-#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<webrtc::CriticalSectionWrapper> crit_;
- rtc::scoped_ptr<webrtc::FileWrapper> file_ GUARDED_BY(crit_);
- rtc::scoped_ptr<ACMDumpEventStream> 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<int64_t>(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> AcmDump::Create() {
- return rtc::scoped_ptr<AcmDump>(new AcmDumpImpl());
-}
-
-bool AcmDump::ParseAcmDump(const std::string& file_name,
- ACMDumpEventStream* result) {
- char tmp_buffer[1024];
- int bytes_read = 0;
- rtc::scoped_ptr<FileWrapper> 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 c72c387..0000000
--- 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 <string>
-
-#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<AcmDump> 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 55c948e..0000000
--- 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 <stdio.h>
-#include <string>
-#include <vector>
-
-#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<uint8_t>(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<uint8_t> rtp_packet_;
- rtc::scoped_ptr<AcmDump> 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 c78bcd7..9a38fac 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 416bb7a..0000000
--- 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 150ee8e..e29f683 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',
