This issue is related to
https://chromereviews.googleplex.com/9908014/
I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL.
BUG=
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2171004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index b705b4d..12b5a63 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -1017,6 +1017,20 @@
virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
};
+struct AudioCodingModuleFactory {
+ AudioCodingModuleFactory() {}
+ virtual ~AudioCodingModuleFactory() {}
+
+ virtual AudioCodingModule* Create(int id) const;
+};
+
+struct NewAudioCodingModuleFactory : AudioCodingModuleFactory {
+ NewAudioCodingModuleFactory() {}
+ virtual ~NewAudioCodingModuleFactory() {}
+
+ virtual AudioCodingModule* Create(int id) const;
+};
+
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
index 570fec5..4ac40b7 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module.cc
@@ -98,4 +98,14 @@
}
}
+AudioCodingModule* AudioCodingModuleFactory::Create(int id) const {
+ return new AudioCodingModuleImpl(static_cast<int32_t>(id),
+ Clock::GetRealTimeClock());
+}
+
+AudioCodingModule* NewAudioCodingModuleFactory::Create(int id) const {
+ // TODO(minyue): return new AudioCodingModuleImpl (new version).
+ return NULL;
+}
+
} // namespace webrtc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 9655471..b1a2aa6 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -10,6 +10,7 @@
#include "webrtc/voice_engine/channel.h"
+#include "webrtc/common.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
@@ -816,13 +817,14 @@
int32_t
Channel::CreateChannel(Channel*& channel,
int32_t channelId,
- uint32_t instanceId)
+ uint32_t instanceId,
+ const Config& config)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId),
"Channel::CreateChannel(channelId=%d, instanceId=%d)",
channelId, instanceId);
- channel = new Channel(channelId, instanceId);
+ channel = new Channel(channelId, instanceId, config);
if (channel == NULL)
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice,
@@ -900,7 +902,8 @@
}
Channel::Channel(int32_t channelId,
- uint32_t instanceId) :
+ uint32_t instanceId,
+ const Config& config) :
_fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
_callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
_instanceId(instanceId),
@@ -915,7 +918,7 @@
VoEModuleId(instanceId, channelId), Clock::GetRealTimeClock(), this,
this, this, rtp_payload_registry_.get())),
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
- _audioCodingModule(*AudioCodingModule::Create(
+ _audioCodingModule(*config.Get<AudioCodingModuleFactory>().Create(
VoEModuleId(instanceId, channelId))),
_rtpDumpIn(*RtpDump::CreateRtpDump()),
_rtpDumpOut(*RtpDump::CreateRtpDump()),
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index a614707..4d9146e 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -36,6 +36,7 @@
namespace webrtc
{
class AudioDeviceModule;
+class Config;
class CriticalSectionWrapper;
class FileWrapper;
class ProcessThread;
@@ -80,8 +81,9 @@
virtual ~Channel();
static int32_t CreateChannel(Channel*& channel,
int32_t channelId,
- uint32_t instanceId);
- Channel(int32_t channelId, uint32_t instanceId);
+ uint32_t instanceId,
+ const Config& config);
+ Channel(int32_t channelId, uint32_t instanceId, const Config& config);
int32_t Init();
int32_t SetEngineInformation(
Statistics& engineStatistics,
diff --git a/webrtc/voice_engine/channel_manager.cc b/webrtc/voice_engine/channel_manager.cc
index 8110bb6..b56c54a 100644
--- a/webrtc/voice_engine/channel_manager.cc
+++ b/webrtc/voice_engine/channel_manager.cc
@@ -44,14 +44,15 @@
ChannelOwner::ChannelRef::ChannelRef(class Channel* channel)
: channel(channel), ref_count(1) {}
-ChannelManager::ChannelManager(uint32_t instance_id)
+ChannelManager::ChannelManager(uint32_t instance_id, const Config& config)
: instance_id_(instance_id),
last_channel_id_(-1),
- lock_(CriticalSectionWrapper::CreateCriticalSection()) {}
+ lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ config_(config) {}
ChannelOwner ChannelManager::CreateChannel() {
Channel* channel;
- Channel::CreateChannel(channel, ++last_channel_id_, instance_id_);
+ Channel::CreateChannel(channel, ++last_channel_id_, instance_id_, config_);
ChannelOwner channel_owner(channel);
CriticalSectionScoped crit(lock_.get());
diff --git a/webrtc/voice_engine/channel_manager.h b/webrtc/voice_engine/channel_manager.h
index 4efeb15..85e64fa 100644
--- a/webrtc/voice_engine/channel_manager.h
+++ b/webrtc/voice_engine/channel_manager.h
@@ -20,6 +20,9 @@
#include "webrtc/typedefs.h"
namespace webrtc {
+
+class Config;
+
namespace voe {
class Channel;
@@ -66,7 +69,7 @@
class ChannelManager {
public:
- ChannelManager(uint32_t instance_id);
+ ChannelManager(uint32_t instance_id, const Config& config);
// Upon construction of an Iterator it will grab a copy of the channel list of
// the ChannelManager. The iteration will then occur over this state, not the
@@ -110,6 +113,8 @@
scoped_ptr<CriticalSectionWrapper> lock_;
std::vector<ChannelOwner> channels_;
+ const Config& config_;
+
DISALLOW_COPY_AND_ASSIGN(ChannelManager);
};
} // namespace voe
diff --git a/webrtc/voice_engine/include/voe_base.h b/webrtc/voice_engine/include/voe_base.h
index de483b4..6859ddb 100644
--- a/webrtc/voice_engine/include/voe_base.h
+++ b/webrtc/voice_engine/include/voe_base.h
@@ -40,6 +40,7 @@
class AudioDeviceModule;
class AudioProcessing;
+class Config;
const int kVoEDefault = -1;
@@ -63,6 +64,7 @@
// Creates a VoiceEngine object, which can then be used to acquire
// sub-APIs. Returns NULL on failure.
static VoiceEngine* Create();
+ static VoiceEngine* Create(const Config& config);
// Deletes a created VoiceEngine object and releases the utilized resources.
// Note that if there are outstanding references held via other interfaces,
diff --git a/webrtc/voice_engine/shared_data.cc b/webrtc/voice_engine/shared_data.cc
index 843d5db..2d485ae 100644
--- a/webrtc/voice_engine/shared_data.cc
+++ b/webrtc/voice_engine/shared_data.cc
@@ -23,10 +23,10 @@
static int32_t _gInstanceCounter = 0;
-SharedData::SharedData() :
+SharedData::SharedData(const Config& config) :
_instanceId(++_gInstanceCounter),
_apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
- _channelManager(_gInstanceCounter),
+ _channelManager(_gInstanceCounter, config),
_engineStatistics(_gInstanceCounter),
_audioDevicePtr(NULL),
audioproc_(NULL),
diff --git a/webrtc/voice_engine/shared_data.h b/webrtc/voice_engine/shared_data.h
index a841547..7c7ad5c 100644
--- a/webrtc/voice_engine/shared_data.h
+++ b/webrtc/voice_engine/shared_data.h
@@ -22,6 +22,7 @@
class ProcessThread;
namespace webrtc {
+class Config;
class CriticalSectionWrapper;
namespace voe {
@@ -79,7 +80,7 @@
AudioDeviceModule::AudioLayer _audioDeviceLayer;
- SharedData();
+ SharedData(const Config& config);
virtual ~SharedData();
};
diff --git a/webrtc/voice_engine/voice_engine_impl.cc b/webrtc/voice_engine/voice_engine_impl.cc
index 4c923eb..703b6a4 100644
--- a/webrtc/voice_engine/voice_engine_impl.cc
+++ b/webrtc/voice_engine/voice_engine_impl.cc
@@ -16,6 +16,7 @@
#endif
#endif
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
@@ -28,13 +29,26 @@
// improvement here.
static int32_t gVoiceEngineInstanceCounter = 0;
-extern "C"
+VoiceEngine* GetVoiceEngine(const Config* config, bool owns_config)
{
-WEBRTC_DLLEXPORT VoiceEngine* GetVoiceEngine();
+#if (defined _WIN32)
+ HMODULE hmod = LoadLibrary(TEXT("VoiceEngineTestingDynamic.dll"));
-VoiceEngine* GetVoiceEngine()
-{
- VoiceEngineImpl* self = new VoiceEngineImpl();
+ if (hmod) {
+ typedef VoiceEngine* (*PfnGetVoiceEngine)(void);
+ PfnGetVoiceEngine pfn = (PfnGetVoiceEngine)GetProcAddress(
+ hmod,"GetVoiceEngine");
+ if (pfn) {
+ VoiceEngine* self = pfn();
+ if (owns_config) {
+ delete config;
+ }
+ return (self);
+ }
+ }
+#endif
+
+ VoiceEngineImpl* self = new VoiceEngineImpl(config, owns_config);
if (self != NULL)
{
self->AddRef(); // First reference. Released in VoiceEngine::Delete.
@@ -42,7 +56,6 @@
}
return self;
}
-} // extern "C"
int VoiceEngineImpl::AddRef() {
return ++_ref_count;
@@ -63,25 +76,15 @@
return new_ref;
}
-VoiceEngine* VoiceEngine::Create()
-{
-#if (defined _WIN32)
- HMODULE hmod_ = LoadLibrary(TEXT("VoiceEngineTestingDynamic.dll"));
+VoiceEngine* VoiceEngine::Create() {
+ Config* config = new Config();
+ config->Set<AudioCodingModuleFactory>(new AudioCodingModuleFactory());
- if (hmod_)
- {
- typedef VoiceEngine* (*PfnGetVoiceEngine)(void);
- PfnGetVoiceEngine pfn = (PfnGetVoiceEngine)GetProcAddress(
- hmod_,"GetVoiceEngine");
- if (pfn)
- {
- VoiceEngine* self = pfn();
- return (self);
- }
- }
-#endif
+ return GetVoiceEngine(config, true);
+}
- return GetVoiceEngine();
+VoiceEngine* VoiceEngine::Create(const Config& config) {
+ return GetVoiceEngine(&config, false);
}
int VoiceEngine::SetTraceFilter(unsigned int filter)
diff --git a/webrtc/voice_engine/voice_engine_impl.h b/webrtc/voice_engine/voice_engine_impl.h
index 55c4209..fe6a791 100644
--- a/webrtc/voice_engine/voice_engine_impl.h
+++ b/webrtc/voice_engine/voice_engine_impl.h
@@ -98,7 +98,8 @@
public VoEBaseImpl
{
public:
- VoiceEngineImpl() :
+ VoiceEngineImpl(const Config* config, bool owns_config) :
+ SharedData(*config),
#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
VoEAudioProcessingImpl(this),
#endif
@@ -137,7 +138,8 @@
VoEVolumeControlImpl(this),
#endif
VoEBaseImpl(this),
- _ref_count(0)
+ _ref_count(0),
+ own_config_(owns_config ? config : NULL)
{
}
virtual ~VoiceEngineImpl()
@@ -152,6 +154,7 @@
private:
Atomic32 _ref_count;
+ scoped_ptr<const Config> own_config_;
};
} // namespace webrtc
