(Auto)update libjingle 65729829-> 65752960

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6004 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/media/base/mediachannel.h

@@ -171,7 +171,6 @@ struct AudioOptions {
     experimental_aec.SetFrom(change.experimental_aec);
     experimental_ns.SetFrom(change.experimental_ns);
     aec_dump.SetFrom(change.aec_dump);
-    experimental_acm.SetFrom(change.experimental_acm);
     tx_agc_target_dbov.SetFrom(change.tx_agc_target_dbov);
     tx_agc_digital_compression_gain.SetFrom(
         change.tx_agc_digital_compression_gain);
@@ -200,7 +199,6 @@ struct AudioOptions {
         experimental_ns == o.experimental_ns &&
         adjust_agc_delta == o.adjust_agc_delta &&
         aec_dump == o.aec_dump &&
-        experimental_acm == o.experimental_acm &&
         tx_agc_target_dbov == o.tx_agc_target_dbov &&
         tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
         tx_agc_limiter == o.tx_agc_limiter &&
@@ -229,7 +227,6 @@ struct AudioOptions {
     ost << ToStringIfSet("experimental_aec", experimental_aec);
     ost << ToStringIfSet("experimental_ns", experimental_ns);
     ost << ToStringIfSet("aec_dump", aec_dump);
-    ost << ToStringIfSet("experimental_acm", experimental_acm);
     ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
     ost << ToStringIfSet("tx_agc_digital_compression_gain",
         tx_agc_digital_compression_gain);
@@ -267,7 +264,6 @@ struct AudioOptions {
   Settable<bool> experimental_aec;
   Settable<bool> experimental_ns;
   Settable<bool> aec_dump;
-  Settable<bool> experimental_acm;
   // Note that tx_agc_* only applies to non-experimental AGC.
   Settable<uint16> tx_agc_target_dbov;
   Settable<uint16> tx_agc_digital_compression_gain;

talk/media/webrtc/fakewebrtcvoiceengine.h

@@ -40,8 +40,6 @@
 #include "talk/media/base/voiceprocessor.h"
 #include "talk/media/webrtc/fakewebrtccommon.h"
 #include "talk/media/webrtc/webrtcvoe.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/common.h"
 
 namespace webrtc {
 class ViENetwork;
@@ -91,7 +89,7 @@ class FakeWebRtcVoiceEngine
     int dtmf_length_ms;
   };
   struct Channel {
-    explicit Channel(bool use_experimental_acm)
+    explicit Channel()
         : external_transport(false),
           send(false),
           playout(false),
@@ -115,8 +113,7 @@ class FakeWebRtcVoiceEngine
           send_ssrc(0),
           send_audio_level_ext_(-1),
           send_absolute_sender_time_ext_(-1),
-          receive_absolute_sender_time_ext_(-1),
+          receive_absolute_sender_time_ext_(-1) {
-          using_experimental_acm(use_experimental_acm) {
       memset(&send_codec, 0, sizeof(send_codec));
       memset(&rx_agc_config, 0, sizeof(rx_agc_config));
     }
@@ -150,7 +147,6 @@ class FakeWebRtcVoiceEngine
     webrtc::CodecInst send_codec;
     webrtc::PacketTime last_rtp_packet_time;
     std::list<std::string> packets;
-    bool using_experimental_acm;
   };
 
   FakeWebRtcVoiceEngine(const cricket::AudioCodec* const* codecs,
@@ -239,10 +235,6 @@ class FakeWebRtcVoiceEngine
     WEBRTC_ASSERT_CHANNEL(channel);
     return channels_[channel]->last_rtp_packet_time;
   }
-  bool IsUsingExperimentalAcm(int channel) {
-    WEBRTC_ASSERT_CHANNEL(channel);
-    return channels_[channel]->using_experimental_acm;
-  }
   int GetSendCNPayloadType(int channel, bool wideband) {
     return (wideband) ?
         channels_[channel]->cn16_type :
@@ -296,11 +288,11 @@ class FakeWebRtcVoiceEngine
                                 true);
     }
   }
-  int AddChannel(bool use_experimental_acm) {
+  int AddChannel() {
     if (fail_create_channel_) {
       return -1;
     }
-    Channel* ch = new Channel(use_experimental_acm);
+    Channel* ch = new Channel();
     for (int i = 0; i < NumOfCodecs(); ++i) {
       webrtc::CodecInst codec;
       GetCodec(i, codec);
@@ -343,13 +335,10 @@ class FakeWebRtcVoiceEngine
     return NULL;
   }
   WEBRTC_FUNC(CreateChannel, ()) {
-    return AddChannel(false);
+    return AddChannel();
   }
-  WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
+  WEBRTC_FUNC(CreateChannel, (const webrtc::Config& /*config*/)) {
-    talk_base::scoped_ptr<webrtc::AudioCodingModule> acm(
+    return AddChannel();
-        config.Get<webrtc::AudioCodingModuleFactory>().Create(0));
-    return AddChannel(strcmp(acm->Version(), webrtc::kExperimentalAcmVersion)
-                      == 0);
   }
   WEBRTC_FUNC(DeleteChannel, (int channel)) {
     WEBRTC_CHECK_CHANNEL(channel);

talk/media/webrtc/webrtcvoiceengine.cc

@@ -253,7 +253,6 @@ static AudioOptions GetDefaultEngineOptions() {
   options.experimental_aec.Set(false);
   options.experimental_ns.Set(false);
   options.aec_dump.Set(false);
-  options.experimental_acm.Set(false);
   return options;
 }
 
@@ -357,7 +356,6 @@ WebRtcVoiceEngine::WebRtcVoiceEngine()
       log_filter_(SeverityToFilter(kDefaultLogSeverity)),
       is_dumping_aec_(false),
       desired_local_monitor_enable_(false),
-      use_experimental_acm_(false),
       tx_processor_ssrc_(0),
       rx_processor_ssrc_(0) {
   Construct();
@@ -375,7 +373,6 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
       log_filter_(SeverityToFilter(kDefaultLogSeverity)),
       is_dumping_aec_(false),
       desired_local_monitor_enable_(false),
-      use_experimental_acm_(false),
       tx_processor_ssrc_(0),
       rx_processor_ssrc_(0) {
   Construct();
@@ -408,10 +405,6 @@ void WebRtcVoiceEngine::Construct() {
                          kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
 #endif
   options_ = GetDefaultEngineOptions();
-
-  // Initialize the VoE Configuration to the new ACM.
-  voe_config_.Set<webrtc::AudioCodingModuleFactory>(
-      new webrtc::NewAudioCodingModuleFactory);
 }
 
 static bool IsOpus(const AudioCodec& codec) {
@@ -750,13 +743,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
 
   LOG(LS_INFO) << "Applying audio options: " << options.ToString();
 
-  // Configure whether ACM1 or ACM2 is used.
-  bool enable_acm2 = false;
-  if (options.experimental_acm.Get(&enable_acm2)) {
-    EnableExperimentalAcm(enable_acm2);
-  }
-  LOG(LS_INFO) << "ACM2 enabled? " << enable_acm2;
-
   webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
 
   bool echo_cancellation;
@@ -1318,21 +1304,6 @@ bool WebRtcVoiceEngine::ShouldIgnoreTrace(const std::string& trace) {
   return false;
 }
 
-void WebRtcVoiceEngine::EnableExperimentalAcm(bool enable) {
-  if (enable == use_experimental_acm_)
-    return;
-  if (enable) {
-    LOG(LS_INFO) << "VoiceEngine is set to use new ACM (ACM2 + NetEq4).";
-    voe_config_.Set<webrtc::AudioCodingModuleFactory>(
-        new webrtc::NewAudioCodingModuleFactory());
-  } else {
-    LOG(LS_INFO) << "VoiceEngine is set to use legacy ACM (ACM1 + Neteq3).";
-    voe_config_.Set<webrtc::AudioCodingModuleFactory>(
-        new webrtc::AudioCodingModuleFactory());
-  }
-  use_experimental_acm_ = enable;
-}
-
 void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
                               int length) {
   talk_base::LoggingSeverity sev = talk_base::LS_VERBOSE;

talk/media/webrtc/webrtcvoiceengine.h

@@ -44,7 +44,6 @@
 #include "talk/media/webrtc/webrtcvoe.h"
 #include "talk/session/media/channel.h"
 #include "webrtc/common.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 
 #if !defined(LIBPEERCONNECTION_LIB) && \
     !defined(LIBPEERCONNECTION_IMPLEMENTATION)
@@ -201,9 +200,6 @@ class WebRtcVoiceEngine
   // allows us to selectively turn on and off different options easily
   // at any time.
   bool ApplyOptions(const AudioOptions& options);
-  // Configure for using ACM2, if |enable| is true, otherwise configure for
-  // ACM1.
-  void EnableExperimentalAcm(bool enable);
   virtual void Print(webrtc::TraceLevel level, const char* trace, int length);
   virtual void CallbackOnError(int channel, int errCode);
   // Given the device type, name, and id, find device id. Return true and
@@ -261,7 +257,6 @@ class WebRtcVoiceEngine
   webrtc::AgcConfig default_agc_config_;
 
   webrtc::Config voe_config_;
-  bool use_experimental_acm_;
 
   bool initialized_;
   // See SetOptions and SetOptionOverrides for a description of the

talk/media/webrtc/webrtcvoiceengine_unittest.cc

@@ -3085,39 +3085,3 @@ TEST(WebRtcVoiceEngineTest, CoInitialize) {
 }
 #endif
 
-
-TEST_F(WebRtcVoiceEngineTestFake, SetExperimentalAcm) {
-  EXPECT_TRUE(SetupEngine());
-
-  // By default the new ACM should be used.
-  int media_channel = engine_.CreateMediaVoiceChannel();
-  ASSERT_GE(media_channel, 0);
-  EXPECT_TRUE(voe_.IsUsingExperimentalAcm(media_channel));
-
-  int soundclip_channel = engine_.CreateSoundclipVoiceChannel();
-  ASSERT_GE(soundclip_channel, 0);
-  EXPECT_TRUE(voe_sc_.IsUsingExperimentalAcm(soundclip_channel));
-
-  // Set options to use experimental ACM.
-  cricket::AudioOptions options;
-  options.experimental_acm.Set(true);
-  ASSERT_TRUE(engine_.SetOptions(options));
-  media_channel = engine_.CreateMediaVoiceChannel();
-  ASSERT_GE(media_channel, 0);
-  EXPECT_TRUE(voe_.IsUsingExperimentalAcm(media_channel));
-
-  soundclip_channel = engine_.CreateSoundclipVoiceChannel();
-  ASSERT_GE(soundclip_channel, 0);
-  EXPECT_TRUE(voe_sc_.IsUsingExperimentalAcm(soundclip_channel));
-
-  // Set option to use legacy ACM.
-  options.experimental_acm.Set(false);
-  ASSERT_TRUE(engine_.SetOptions(options));
-  media_channel = engine_.CreateMediaVoiceChannel();
-  ASSERT_GE(media_channel, 0);
-  EXPECT_FALSE(voe_.IsUsingExperimentalAcm(media_channel));
-
-  soundclip_channel = engine_.CreateSoundclipVoiceChannel();
-  ASSERT_GE(soundclip_channel, 0);
-  EXPECT_FALSE(voe_sc_.IsUsingExperimentalAcm(soundclip_channel));
-}

talk/session/media/channelmanager.cc

@@ -939,14 +939,6 @@ VideoFormat ChannelManager::GetStartCaptureFormat() {
       Bind(&MediaEngineInterface::GetStartCaptureFormat, media_engine_.get()));
 }
 
-bool ChannelManager::SetAudioOptions(const AudioOptions& options) {
-  if (!media_engine_->SetAudioOptions(options)) {
-    return false;
-  }
-  audio_options_ = options;
-  return true;
-}
-
 bool ChannelManager::StartAecDump(talk_base::PlatformFile file) {
   return worker_thread_->Invoke<bool>(
       Bind(&MediaEngineInterface::StartAecDump, media_engine_.get(), file));

talk/session/media/channelmanager.h

@@ -230,10 +230,6 @@ class ChannelManager : public talk_base::MessageHandler,
   // removed.
   VideoFormat GetStartCaptureFormat();
 
-  // TODO(turajs): Remove this function when ACM2 is in use. Used mainly to
-  // choose between ACM1 and ACM2.
-  bool SetAudioOptions(const AudioOptions& options);
-
  protected:
   // Adds non-transient parameters which can only be changed through the
   // options store.