Fix broken AEC dump in Chrome M33.
Reverts P4 CL 58093675.
R=tommi@webrtc.org
TBR=wu@webrtc.org
BUG= https://code.google.com/p/chromium/issues/detail?id=336171
Review URL: https://webrtc-codereview.appspot.com/7399004
git-svn-id: http://webrtc.googlecode.com/svn/branches/3.48/talk@5412 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/app/webrtc/localaudiosource.cc b/app/webrtc/localaudiosource.cc
index 3dc5c6c..2cd472a 100644
--- a/app/webrtc/localaudiosource.cc
+++ b/app/webrtc/localaudiosource.cc
@@ -54,6 +54,8 @@
const char MediaConstraintsInterface::kTypingNoiseDetection[] =
"googTypingNoiseDetection";
const char MediaConstraintsInterface::kAudioMirroring[] = "googAudioMirroring";
+// TODO(perkj): Remove kInternalAecDump once its not used by Chrome.
+const char MediaConstraintsInterface::kInternalAecDump[] = "deprecatedAecDump";
namespace {
@@ -127,6 +129,8 @@
return;
}
options_.SetAll(audio_options);
+ if (options.enable_aec_dump)
+ options_.aec_dump.Set(true);
source_state_ = kLive;
}
diff --git a/app/webrtc/mediaconstraintsinterface.h b/app/webrtc/mediaconstraintsinterface.h
index 5cf2184..ba6b09b 100644
--- a/app/webrtc/mediaconstraintsinterface.h
+++ b/app/webrtc/mediaconstraintsinterface.h
@@ -117,6 +117,13 @@
// stripped by Chrome before passed down to Libjingle.
static const char kInternalConstraintPrefix[];
+ // These constraints are for internal use only, representing Chrome command
+ // line flags. So they are prefixed with "internal" so JS values will be
+ // removed.
+ // Used by a local audio source.
+ // TODO(perkj): Remove once Chrome use PeerConnectionFactory::SetOptions.
+ static const char kInternalAecDump[]; // internalAecDump
+
protected:
// Dtor protected as objects shouldn't be deleted via this interface
virtual ~MediaConstraintsInterface() {}
diff --git a/app/webrtc/peerconnectionfactory.cc b/app/webrtc/peerconnectionfactory.cc
index ee15b5d..e8b8f63 100644
--- a/app/webrtc/peerconnectionfactory.cc
+++ b/app/webrtc/peerconnectionfactory.cc
@@ -105,21 +105,12 @@
scoped_refptr<webrtc::VideoSourceInterface> source;
};
-struct StartAecDumpParams : public talk_base::MessageData {
- explicit StartAecDumpParams(FILE* aec_dump_file)
- : aec_dump_file(aec_dump_file) {
- }
- FILE* aec_dump_file;
- bool result;
-};
-
enum {
MSG_INIT_FACTORY = 1,
MSG_TERMINATE_FACTORY,
MSG_CREATE_PEERCONNECTION,
MSG_CREATE_AUDIOSOURCE,
MSG_CREATE_VIDEOSOURCE,
- MSG_START_AEC_DUMP,
};
} // namespace
@@ -232,12 +223,6 @@
pdata->source = CreateVideoSource_s(pdata->capturer, pdata->constraints);
break;
}
- case MSG_START_AEC_DUMP: {
- StartAecDumpParams* pdata =
- static_cast<StartAecDumpParams*>(msg->pdata);
- pdata->result = StartAecDump_s(pdata->aec_dump_file);
- break;
- }
}
}
@@ -289,10 +274,6 @@
return VideoSourceProxy::Create(signaling_thread_, source);
}
-bool PeerConnectionFactory::StartAecDump_s(FILE* file) {
- return channel_manager_->StartAecDump(file);
-}
-
scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
const PeerConnectionInterface::IceServers& configuration,
@@ -380,12 +361,6 @@
return AudioTrackProxy::Create(signaling_thread_, track);
}
-bool PeerConnectionFactory::StartAecDump(FILE* file) {
- StartAecDumpParams params(file);
- signaling_thread_->Send(this, MSG_START_AEC_DUMP, ¶ms);
- return params.result;
-}
-
cricket::ChannelManager* PeerConnectionFactory::channel_manager() {
return channel_manager_.get();
}
diff --git a/app/webrtc/peerconnectionfactory.h b/app/webrtc/peerconnectionfactory.h
index 63d37f0..dff885d 100644
--- a/app/webrtc/peerconnectionfactory.h
+++ b/app/webrtc/peerconnectionfactory.h
@@ -78,8 +78,6 @@
CreateAudioTrack(const std::string& id,
AudioSourceInterface* audio_source);
- virtual bool StartAecDump(FILE* file);
-
virtual cricket::ChannelManager* channel_manager();
virtual talk_base::Thread* signaling_thread();
virtual talk_base::Thread* worker_thread();
@@ -95,6 +93,7 @@
cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
virtual ~PeerConnectionFactory();
+
private:
bool Initialize_s();
void Terminate_s();
@@ -109,8 +108,6 @@
PortAllocatorFactoryInterface* allocator_factory,
DTLSIdentityServiceInterface* dtls_identity_service,
PeerConnectionObserver* observer);
- bool StartAecDump_s(FILE* file);
-
// Implements talk_base::MessageHandler.
void OnMessage(talk_base::Message* msg);
diff --git a/app/webrtc/peerconnectioninterface.h b/app/webrtc/peerconnectioninterface.h
index 01f1e1c..a127dad 100644
--- a/app/webrtc/peerconnectioninterface.h
+++ b/app/webrtc/peerconnectioninterface.h
@@ -393,9 +393,11 @@
class Options {
public:
Options() :
+ enable_aec_dump(false),
disable_encryption(false),
disable_sctp_data_channels(false) {
}
+ bool enable_aec_dump;
bool disable_encryption;
bool disable_sctp_data_channels;
};
@@ -440,12 +442,6 @@
CreateAudioTrack(const std::string& label,
AudioSourceInterface* source) = 0;
- // Starts AEC dump using existing file. Takes ownership of |file| and passes
- // it on to VoiceEngine (via other objects) immediately, which will take
- // the ownerhip.
- // TODO(grunell): Remove when Chromium has started to use AEC in each source.
- virtual bool StartAecDump(FILE* file) = 0;
-
protected:
// Dtor and ctor protected as objects shouldn't be created or deleted via
// this interface.
diff --git a/media/base/fakemediaengine.h b/media/base/fakemediaengine.h
index d71c660..1a4e8ab 100644
--- a/media/base/fakemediaengine.h
+++ b/media/base/fakemediaengine.h
@@ -778,8 +778,6 @@
bool SetLocalMonitor(bool enable) { return true; }
- bool StartAecDump(FILE* file) { return false; }
-
bool RegisterProcessor(uint32 ssrc, VoiceProcessor* voice_processor,
MediaProcessorDirection direction) {
if (direction == MPD_RX) {
diff --git a/media/base/filemediaengine.h b/media/base/filemediaengine.h
index 843806b..dfdb037 100644
--- a/media/base/filemediaengine.h
+++ b/media/base/filemediaengine.h
@@ -133,7 +133,6 @@
virtual bool FindVideoCodec(const VideoCodec& codec) { return true; }
virtual void SetVoiceLogging(int min_sev, const char* filter) {}
virtual void SetVideoLogging(int min_sev, const char* filter) {}
- virtual bool StartAecDump(FILE* file) { return false; }
virtual bool RegisterVideoProcessor(VideoProcessor* processor) {
return true;
diff --git a/media/base/mediaengine.h b/media/base/mediaengine.h
index c04df9f..f916572 100644
--- a/media/base/mediaengine.h
+++ b/media/base/mediaengine.h
@@ -135,9 +135,6 @@
virtual void SetVoiceLogging(int min_sev, const char* filter) = 0;
virtual void SetVideoLogging(int min_sev, const char* filter) = 0;
- // Starts AEC dump using existing file.
- virtual bool StartAecDump(FILE* file) = 0;
-
// Voice processors for effects.
virtual bool RegisterVoiceProcessor(uint32 ssrc,
VoiceProcessor* video_processor,
@@ -256,10 +253,6 @@
video_.SetLogging(min_sev, filter);
}
- virtual bool StartAecDump(FILE* file) {
- return voice_.StartAecDump(file);
- }
-
virtual bool RegisterVoiceProcessor(uint32 ssrc,
VoiceProcessor* processor,
MediaProcessorDirection direction) {
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index 2aa6b8c..51da9ac 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -1433,22 +1433,6 @@
return true;
}
-bool WebRtcVoiceEngine::StartAecDump(FILE* file) {
-#ifdef USE_WEBRTC_DEV_BRANCH
- StopAecDump();
- if (voe_wrapper_->processing()->StartDebugRecording(file) !=
- webrtc::AudioProcessing::kNoError) {
- LOG_RTCERR1(StartDebugRecording, "FILE*");
- fclose(file);
- return false;
- }
- is_dumping_aec_ = true;
- return true;
-#else
- return false;
-#endif
-}
-
bool WebRtcVoiceEngine::RegisterProcessor(
uint32 ssrc,
VoiceProcessor* voice_processor,
@@ -1606,7 +1590,7 @@
// Start dumping AEC when we are not dumping.
if (voe_wrapper_->processing()->StartDebugRecording(
filename.c_str()) != webrtc::AudioProcessing::kNoError) {
- LOG_RTCERR1(StartDebugRecording, filename.c_str());
+ LOG_RTCERR0(StartDebugRecording);
} else {
is_dumping_aec_ = true;
}
diff --git a/media/webrtc/webrtcvoiceengine.h b/media/webrtc/webrtcvoiceengine.h
index adf4853..23d97f5 100644
--- a/media/webrtc/webrtcvoiceengine.h
+++ b/media/webrtc/webrtcvoiceengine.h
@@ -174,9 +174,6 @@
bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc);
- // Starts AEC dump using existing file.
- bool StartAecDump(FILE* file);
-
// Check whether the supplied trace should be ignored.
bool ShouldIgnoreTrace(const std::string& trace);
diff --git a/session/media/channelmanager.cc b/session/media/channelmanager.cc
index 4d5d8fc..d4fcc79 100644
--- a/session/media/channelmanager.cc
+++ b/session/media/channelmanager.cc
@@ -947,9 +947,4 @@
return true;
}
-bool ChannelManager::StartAecDump(FILE* file) {
- return worker_thread_->Invoke<bool>(
- Bind(&MediaEngineInterface::StartAecDump, media_engine_.get(), file));
-}
-
} // namespace cricket
diff --git a/session/media/channelmanager.h b/session/media/channelmanager.h
index f19d3d0..fdb8f73 100644
--- a/session/media/channelmanager.h
+++ b/session/media/channelmanager.h
@@ -214,9 +214,6 @@
void SetVideoCaptureDeviceMaxFormat(const std::string& usb_id,
const VideoFormat& max_format);
- // Starts AEC dump using existing file.
- bool StartAecDump(FILE* file);
-
sigslot::repeater0<> SignalDevicesChange;
sigslot::signal2<VideoCapturer*, CaptureState> SignalVideoCaptureStateChange;
