third_party/libjingle/overrides/init_webrtc.cc

// Copyright 2013 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "init_webrtc.h"

#include "base/command_line.h"
#include "base/debug/trace_event.h"
#include "base/files/file_path.h"
#include "base/files/file_util.h"
#include "base/metrics/field_trial.h"
#include "base/native_library.h"
#include "base/path_service.h"
#include "third_party/webrtc/common.h"
#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/logging.h"

const unsigned char* GetCategoryGroupEnabled(const char* category_group) {
  return TRACE_EVENT_API_GET_CATEGORY_GROUP_ENABLED(category_group);
}

void AddTraceEvent(char phase,
                   const unsigned char* category_group_enabled,
                   const char* name,
                   unsigned long long id,
                   int num_args,
                   const char** arg_names,
                   const unsigned char* arg_types,
                   const unsigned long long* arg_values,
                   unsigned char flags) {
  TRACE_EVENT_API_ADD_TRACE_EVENT(phase, category_group_enabled, name, id,
                                  num_args, arg_names, arg_types, arg_values,
                                  NULL, flags);
}

// Define webrtc:field_trial::FindFullName to provide webrtc with a field trial
// implementation.
namespace webrtc {
namespace field_trial {
std::string FindFullName(const std::string& trial_name) {
  return base::FieldTrialList::FindFullName(trial_name);
}
}  // namespace field_trial
}  // namespace webrtc

#if defined(LIBPEERCONNECTION_LIB)

// libpeerconnection is being compiled as a static lib.  In this case
// we don't need to do any initializing but to keep things simple we
// provide an empty intialization routine so that this #ifdef doesn't
// have to be in other places.
bool InitializeWebRtcModule() {
  webrtc::SetupEventTracer(&GetCategoryGroupEnabled, &AddTraceEvent);
  return true;
}

webrtc::AudioProcessing* CreateWebRtcAudioProcessing(
    const webrtc::Config& config) {
  // libpeerconnection is being compiled as a static lib, use
  // webrtc::AudioProcessing directly.
  return webrtc::AudioProcessing::Create(config);
}

#else  // !LIBPEERCONNECTION_LIB

// When being compiled as a shared library, we need to bridge the gap between
// the current module and the libpeerconnection module, so things get a tad
// more complicated.

// Global function pointers to the factory functions in the shared library.
CreateWebRtcMediaEngineFunction g_create_webrtc_media_engine = NULL;
DestroyWebRtcMediaEngineFunction g_destroy_webrtc_media_engine = NULL;
CreateWebRtcAudioProcessingFunction g_create_webrtc_audio_processing = NULL;

// Returns the full or relative path to the libpeerconnection module depending
// on what platform we're on.
static base::FilePath GetLibPeerConnectionPath() {
  base::FilePath path;
  CHECK(PathService::Get(base::DIR_MODULE, &path));
#if defined(OS_WIN)
  path = path.Append(FILE_PATH_LITERAL("libpeerconnection.dll"));
#elif defined(OS_MACOSX)
  // Simulate '@loader_path/Libraries'.
  path = path.Append(FILE_PATH_LITERAL("Libraries"))
             .Append(FILE_PATH_LITERAL("libpeerconnection.so"));
#elif defined(OS_ANDROID)
  path = path.Append(FILE_PATH_LITERAL("libpeerconnection.so"));
#else
  path = path.Append(FILE_PATH_LITERAL("lib"))
             .Append(FILE_PATH_LITERAL("libpeerconnection.so"));
#endif
  return path;
}

bool InitializeWebRtcModule() {
  TRACE_EVENT0("webrtc", "InitializeWebRtcModule");

  if (g_create_webrtc_media_engine)
    return true;  // InitializeWebRtcModule has already been called.

  base::FilePath path(GetLibPeerConnectionPath());
  DVLOG(1) << "Loading WebRTC module: " << path.value();

  base::NativeLibraryLoadError error;
  static base::NativeLibrary lib = base::LoadNativeLibrary(path, &error);
#if defined(OS_WIN)
  // We've been seeing problems on Windows with loading the DLL and we're
  // not sure exactly why.  It could be that AV programs are quarantining the
  // file or disallowing loading the DLL. To get a better picture of the errors
  // we're checking these specific error codes.
  if (error.code == ERROR_MOD_NOT_FOUND) {
    // It's possible that we get this error due to failure to load other
    // dependencies, so check first that libpeerconnection actually exists.
    CHECK(base::PathExists(path));  // libpeerconnection itself is missing.
    CHECK(lib);  // If we hit this, a dependency is missing.
  } else if (error.code == ERROR_ACCESS_DENIED) {
    CHECK(lib);  // AV blocking access?
  }
#endif

  // Catch-all error handler for all other sorts of errors.
  CHECK(lib) << error.ToString();

  InitializeModuleFunction initialize_module =
      reinterpret_cast<InitializeModuleFunction>(
          base::GetFunctionPointerFromNativeLibrary(
              lib, "InitializeModule"));

  // Initialize the proxy by supplying it with a pointer to our
  // allocator/deallocator routines.
  // On mac we use malloc zones, which are global, so we provide NULLs for
  // the alloc/dealloc functions.
  // PS: This function is actually implemented in allocator_proxy.cc with the
  // new/delete overrides.
  InitDiagnosticLoggingDelegateFunctionFunction init_diagnostic_logging = NULL;
  bool init_ok = initialize_module(*CommandLine::ForCurrentProcess(),
#if !defined(OS_MACOSX) && !defined(OS_ANDROID)
                                   &Allocate,
                                   &Dellocate,
#endif
                                   &webrtc::field_trial::FindFullName,
                                   logging::GetLogMessageHandler(),
                                   &GetCategoryGroupEnabled,
                                   &AddTraceEvent,
                                   &g_create_webrtc_media_engine,
                                   &g_destroy_webrtc_media_engine,
                                   &init_diagnostic_logging,
                                   &g_create_webrtc_audio_processing);
  if (init_ok)
    rtc::SetExtraLoggingInit(init_diagnostic_logging);
  return init_ok;
}

cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
    webrtc::AudioDeviceModule* adm,
    webrtc::AudioDeviceModule* adm_sc,
    cricket::WebRtcVideoEncoderFactory* encoder_factory,
    cricket::WebRtcVideoDecoderFactory* decoder_factory) {
  // For convenience of tests etc, we call InitializeWebRtcModule here.
  // For Chrome however, InitializeWebRtcModule must be called
  // explicitly before the sandbox is initialized.  In that case, this call is
  // effectively a noop.
  InitializeWebRtcModule();
  return g_create_webrtc_media_engine(adm, adm_sc, encoder_factory,
      decoder_factory);
}

void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) {
  g_destroy_webrtc_media_engine(media_engine);
}

webrtc::AudioProcessing* CreateWebRtcAudioProcessing(
    const webrtc::Config& config) {
  // The same as CreateWebRtcMediaEngine(), we call InitializeWebRtcModule here
  // for convenience of tests.
  InitializeWebRtcModule();
  return g_create_webrtc_audio_processing(config);
}

#endif  // LIBPEERCONNECTION_LIB