Bug 932112: Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=jesup

* * *
* * *
Add AndroidAudioManager to the moz.build files.

content/media/webrtc/MediaEngineWebRTC.cpp

@@ -254,7 +254,7 @@ MediaEngineWebRTC::EnumerateAudioDevices(nsTArray<nsRefPtr<MediaEngineAudioSourc
   JNIEnv *env;
   jvm->AttachCurrentThread(&env, nullptr);
 
-  if (webrtc::VoiceEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
+  if (webrtc::VoiceEngine::SetAndroidObjects(jvm, env, (void*)context) != 0) {
     LOG(("VoiceEngine:SetAndroidObjects Failed"));
     return;
   }

content/media/webrtc/MediaEngineWebRTC.h

@@ -45,7 +45,6 @@
 #include "webrtc/video_engine/include/vie_codec.h"
 #include "webrtc/video_engine/include/vie_render.h"
 #include "webrtc/video_engine/include/vie_capture.h"
-#include "webrtc/video_engine/include/vie_file.h"
 #ifdef MOZ_B2G_CAMERA
 #include "CameraPreviewMediaStream.h"
 #include "DOMCameraManager.h"
@@ -118,7 +117,14 @@ public:
 #else
   // ViEExternalRenderer.
   virtual int FrameSizeChange(unsigned int, unsigned int, unsigned int);
-  virtual int DeliverFrame(unsigned char*, int, uint32_t, int64_t);
+  virtual int DeliverFrame(unsigned char*,int, uint32_t , int64_t,
+                           void *handle);
+  /**
+   * Does DeliverFrame() support a null buffer and non-null handle
+   * (video texture)?
+   * XXX Investigate!  Especially for Android/B2G
+   */
+  virtual bool IsTextureSupported() { return false; }
 
   MediaEngineWebRTCVideoSource(webrtc::VideoEngine* aVideoEnginePtr, int aIndex)
     : mVideoEngine(aVideoEnginePtr)

content/media/webrtc/MediaEngineWebRTCVideo.cpp

@@ -40,7 +40,8 @@ MediaEngineWebRTCVideoSource::FrameSizeChange(
 // ViEExternalRenderer Callback. Process every incoming frame here.
 int
 MediaEngineWebRTCVideoSource::DeliverFrame(
-   unsigned char* buffer, int size, uint32_t time_stamp, int64_t render_time)
+   unsigned char* buffer, int size, uint32_t time_stamp, int64_t render_time,
+   void *handle)
 {
   // mInSnapshotMode can only be set before the camera is turned on and
   // the renderer is started, so this amounts to a 1-shot
@@ -399,126 +400,7 @@ MediaEngineWebRTCVideoSource::Stop(SourceMediaStream *aSource, TrackID aID)
 nsresult
 MediaEngineWebRTCVideoSource::Snapshot(uint32_t aDuration, nsIDOMFile** aFile)
 {
-  /**
+  return NS_ERROR_NOT_IMPLEMENTED;
-   * To get a Snapshot we do the following:
-   * - Set a condition variable (mInSnapshotMode) to true
-   * - Attach the external renderer and start the camera
-   * - Wait for the condition variable to change to false
-   *
-   * Starting the camera has the effect of invoking DeliverFrame() when
-   * the first frame arrives from the camera. We only need one frame for
-   * GetCaptureDeviceSnapshot to work, so we immediately set the condition
-   * variable to false and notify this method.
-   *
-   * This causes the current thread to continue (PR_CondWaitVar will return),
-   * at which point we can grab a snapshot, convert it to a file and
-   * return from this function after cleaning up the temporary stream object
-   * and caling Stop() on the media source.
-   */
-#ifdef MOZ_B2G_CAMERA
-  ReentrantMonitorAutoEnter sync(mCallbackMonitor);
-#endif
-  *aFile = nullptr;
-  if (!mInitDone || mState != kAllocated) {
-    return NS_ERROR_FAILURE;
-  }
-#ifdef MOZ_B2G_CAMERA
-  mLastCapture = nullptr;
-
-  NS_DispatchToMainThread(WrapRunnable(this,
-                                       &MediaEngineWebRTCVideoSource::StartImpl,
-                                       mCapability));
-  mCallbackMonitor.Wait();
-  if (mState != kStarted) {
-    return NS_ERROR_FAILURE;
-  }
-
-  NS_DispatchToMainThread(WrapRunnable(this,
-                                       &MediaEngineWebRTCVideoSource::SnapshotImpl));
-  mCallbackMonitor.Wait();
-  if (mLastCapture == nullptr)
-    return NS_ERROR_FAILURE;
-
-  mState = kStopped;
-  NS_DispatchToMainThread(WrapRunnable(this,
-                                       &MediaEngineWebRTCVideoSource::StopImpl));
-
-  // The camera return nsDOMMemoryFile indeed, and the inheritance tree is:
-  // nsIDOMBlob <- nsIDOMFile <- nsDOMFileBase <- nsDOMFile <- nsDOMMemoryFile
-  *aFile = mLastCapture.get();
-  return NS_OK;
-#else
-  {
-    MonitorAutoLock lock(mMonitor);
-    mInSnapshotMode = true;
-  }
-
-  // Start the rendering (equivalent to calling Start(), but without a track).
-  int error = 0;
-  if (!mInitDone || mState != kAllocated) {
-    return NS_ERROR_FAILURE;
-  }
-  error = mViERender->AddRenderer(mCaptureIndex, webrtc::kVideoI420, (webrtc::ExternalRenderer*)this);
-  if (error == -1) {
-    return NS_ERROR_FAILURE;
-  }
-  error = mViERender->StartRender(mCaptureIndex);
-  if (error == -1) {
-    return NS_ERROR_FAILURE;
-  }
-
-  if (mViECapture->StartCapture(mCaptureIndex, mCapability) < 0) {
-    return NS_ERROR_FAILURE;
-  }
-
-  // Wait for the condition variable, will be set in DeliverFrame.
-  // We use a while loop, because even if Wait() returns, it's not
-  // guaranteed that the condition variable changed.
-  // FIX: we need need a way to cancel this and to bail if it appears to not be working
-  // Perhaps a maximum time, though some cameras can take seconds to start.  10 seconds?
-  {
-    MonitorAutoLock lock(mMonitor);
-    while (mInSnapshotMode) {
-      lock.Wait();
-    }
-  }
-
-  // If we get here, DeliverFrame received at least one frame.
-  webrtc::ViEFile* vieFile = webrtc::ViEFile::GetInterface(mVideoEngine);
-  if (!vieFile) {
-    return NS_ERROR_FAILURE;
-  }
-
-  // Create a temporary file on the main thread and put the snapshot in it.
-  // See Run() in MediaEngineWebRTCVideo.h (sets mSnapshotPath).
-  NS_DispatchToMainThread(this, NS_DISPATCH_SYNC);
-
-  if (!mSnapshotPath) {
-    return NS_ERROR_FAILURE;
-  }
-
-  NS_ConvertUTF16toUTF8 path(*mSnapshotPath);
-  if (vieFile->GetCaptureDeviceSnapshot(mCaptureIndex, path.get()) < 0) {
-    delete mSnapshotPath;
-    mSnapshotPath = nullptr;
-    return NS_ERROR_FAILURE;
-  }
-
-  // Stop the camera.
-  mViERender->StopRender(mCaptureIndex);
-  mViERender->RemoveRenderer(mCaptureIndex);
-
-  nsCOMPtr<nsIFile> file;
-  nsresult rv = NS_NewLocalFile(*mSnapshotPath, false, getter_AddRefs(file));
-
-  delete mSnapshotPath;
-  mSnapshotPath = nullptr;
-
-  NS_ENSURE_SUCCESS(rv, rv);
-
-  NS_ADDREF(*aFile = new nsDOMFileFile(file));
-#endif
-  return NS_OK;
 }
 
 /**

layout/media/webrtc/Makefile.in

@@ -9,10 +9,12 @@ SHARED_LIBRARY_LIBS = \
   $(call EXPAND_LIBNAME_PATH,video_capture_module,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_video_capture_module) \
   $(call EXPAND_LIBNAME_PATH,webrtc_utility,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_webrtc_utility) \
   $(call EXPAND_LIBNAME_PATH,audio_coding_module,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_audio_coding_module) \
+  $(call EXPAND_LIBNAME_PATH,acm2,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_acm2) \
   $(call EXPAND_LIBNAME_PATH,CNG,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_CNG) \
   $(call EXPAND_LIBNAME_PATH,G711,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_G711) \
   $(call EXPAND_LIBNAME_PATH,PCM16B,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_PCM16B) \
   $(call EXPAND_LIBNAME_PATH,NetEq,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_NetEq) \
+  $(call EXPAND_LIBNAME_PATH,NetEq4,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_NetEq4) \
   $(call EXPAND_LIBNAME_PATH,system_wrappers,$(DEPTH)/media/webrtc/trunk/webrtc/system_wrappers/source/system_wrappers_system_wrappers) \
   $(call EXPAND_LIBNAME_PATH,webrtc_video_coding,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_webrtc_video_coding) \
   $(call EXPAND_LIBNAME_PATH,video_coding_utility,$(DEPTH)/media/webrtc/trunk/webrtc/modules/video_coding/utility/video_coding_utility_video_coding_utility) \
@@ -26,6 +28,7 @@ SHARED_LIBRARY_LIBS = \
   $(call EXPAND_LIBNAME_PATH,rtp_rtcp,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_rtp_rtcp) \
   $(call EXPAND_LIBNAME_PATH,bitrate_controller,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_bitrate_controller) \
   $(call EXPAND_LIBNAME_PATH,remote_bitrate_estimator,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_remote_bitrate_estimator) \
+  $(call EXPAND_LIBNAME_PATH,rbe_components,$(DEPTH)/media/webrtc/trunk/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_components_rbe_components) \
   $(call EXPAND_LIBNAME_PATH,paced_sender,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_paced_sender) \
   $(call EXPAND_LIBNAME_PATH,video_processing,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_video_processing) \
   $(call EXPAND_LIBNAME_PATH,audio_conference_mixer,$(DEPTH)/media/webrtc/trunk/webrtc/modules/modules_audio_conference_mixer) \

media/webrtc/signaling/src/media-conduit/AudioConduit.cpp

@@ -153,8 +153,9 @@ MediaConduitErrorCode WebrtcAudioConduit::Init(WebrtcAudioConduit *other)
 
       // get the JVM
       JavaVM *jvm = jsjni_GetVM();
+      JNIEnv* jenv = jsjni_GetJNIForThread();
 
-      if (webrtc::VoiceEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
+      if (webrtc::VoiceEngine::SetAndroidObjects(jvm, jenv, (void*)context) != 0) {
         CSFLogError(logTag, "%s Unable to set Android objects", __FUNCTION__);
         return kMediaConduitSessionNotInited;
       }

media/webrtc/signaling/src/media-conduit/VideoConduit.cpp

@@ -972,7 +972,8 @@ int
 WebrtcVideoConduit::DeliverFrame(unsigned char* buffer,
                                  int buffer_size,
                                  uint32_t time_stamp,
-                                 int64_t render_time)
+                                 int64_t render_time,
+                                 void *handle)
 {
   CSFLogDebug(logTag,  "%s Buffer Size %d", __FUNCTION__, buffer_size);
 

media/webrtc/signaling/src/media-conduit/VideoConduit.h

@@ -16,7 +16,6 @@
 #include "webrtc/video_engine/include/vie_codec.h"
 #include "webrtc/video_engine/include/vie_render.h"
 #include "webrtc/video_engine/include/vie_network.h"
-#include "webrtc/video_engine/include/vie_file.h"
 #include "webrtc/video_engine/include/vie_rtp_rtcp.h"
 
 /** This file hosts several structures identifying different aspects
@@ -148,7 +147,15 @@ public:
    */
   virtual int FrameSizeChange(unsigned int, unsigned int, unsigned int);
 
-  virtual int DeliverFrame(unsigned char*,int, uint32_t , int64_t);
+  virtual int DeliverFrame(unsigned char*,int, uint32_t , int64_t,
+                           void *handle);
+
+  /**
+   * Does DeliverFrame() support a null buffer and non-null handle
+   * (video texture)?
+   * XXX Investigate!  Especially for Android/B2G
+   */
+  virtual bool IsTextureSupported() { return false; }
 
   unsigned short SendingWidth() {
     return mSendingWidth;

media/webrtc/signaling/src/media/VcmSIPCCBinding.cpp

@@ -588,7 +588,7 @@ static short vcmRxAllocICE_s(TemporaryRef<NrIceCtx> ctx_in,
   VcmSIPCCBinding::connectCandidateSignal(stream);
 
   std::vector<std::string> candidates = stream->GetCandidates();
-  CSFLogDebug( logTag, "%s: Got %lu candidates", __FUNCTION__, candidates.size());
+  CSFLogDebug( logTag, "%s: Got %lu candidates", __FUNCTION__, (unsigned long) candidates.size());
 
   std::string default_addr;
   int default_port;

media/webrtc/trunk/webrtc/build/arm_neon.gypi

@@ -23,8 +23,30 @@
   'cflags!': [
     '-mfpu=vfpv3-d16',
   ],
+  'cflags_mozilla!': [
+    '-mfpu=vfpv3-d16',
+  ],
   'cflags': [
     '-mfpu=neon',
     '-flax-vector-conversions',
   ],
+  'cflags_mozilla': [
+    '-mfpu=neon',
+    '-flax-vector-conversions',
+  ],
+  'asflags!': [
+    '-mfpu=vfpv3-d16',
+  ],
+  'asflags_mozilla!': [
+    '-mfpu=vfpv3-d16',
+  ],
+  'asflags': [
+    '-mfpu=neon',
+    '-flax-vector-conversions',
+  ],
+  'asflags_mozilla': [
+    '-mfpu=neon',
+    '-flax-vector-conversions',
+  ],
+
 }

media/webrtc/trunk/webrtc/build/common.gypi

@@ -45,7 +45,12 @@
 
       'webrtc_vp8_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp8',
       'rbe_components_path%': '<(webrtc_root)/modules/remote_bitrate_estimator',
+      'include_g711%': 1,
+      'include_g722%': 1,
+      'include_ilbc%': 1,
       'include_opus%': 1,
+      'include_isac%': 1,
+      'include_pcm16b%': 1,
     },
     'build_with_chromium%': '<(build_with_chromium)',
     'build_with_libjingle%': '<(build_with_libjingle)',
@@ -54,7 +59,14 @@
     'import_isolate_path%': '<(import_isolate_path)',
     'modules_java_gyp_path%': '<(modules_java_gyp_path)',
     'webrtc_vp8_dir%': '<(webrtc_vp8_dir)',
+
+    'include_g711%': '<(include_g711)',
+    'include_g722%': '<(include_g722)',
+    'include_ilbc%': '<(include_ilbc)',
     'include_opus%': '<(include_opus)',
+    'include_isac%': '<(include_isac)',
+    'include_pcm16b%': '<(include_pcm16b)',
+
     'rbe_components_path%': '<(rbe_components_path)',
 
     # The Chromium common.gypi we use treats all gyp files without
@@ -113,6 +125,9 @@
 
         # Include ndk cpu features in Chromium build.
         'include_ndk_cpu_features%': 1,
+
+        # lazily allocate the ~4MB of trace message buffers if set
+        'enable_lazy_trace_alloc%': 0,
       }, {  # Settings for the standalone (not-in-Chromium) build.
         # TODO(andrew): For now, disable the Chrome plugins, which causes a
         # flood of chromium-style warnings. Investigate enabling them:
@@ -136,6 +151,21 @@
         # and Java Implementation
         'enable_android_opensl%': 0,
       }],
+      ['OS=="linux"', {
+        'include_alsa_audio%': 1,
+      }, {
+        'include_alsa_audio%': 0,
+      }],
+      ['OS=="solaris" or os_bsd==1', {
+        'include_pulse_audio%': 1,
+      }, {
+        'include_pulse_audio%': 0,
+      }],
+      ['OS=="linux" or OS=="solaris" or os_bsd==1', {
+        'include_v4l2_video_capture%': 1,
+      }, {
+        'include_v4l2_video_capture%': 0,
+      }],
       ['OS=="ios"', {
         'build_libjpeg%': 0,
         'enable_protobuf%': 0,
@@ -160,10 +190,15 @@
     'defines': [
       # TODO(leozwang): Run this as a gclient hook rather than at build-time:
       # http://code.google.com/p/webrtc/issues/detail?id=687
-      'WEBRTC_SVNREVISION="Unavailable(issue687)"',
+      'WEBRTC_SVNREVISION="\\\"Unavailable_issue687\\\""',
       #'WEBRTC_SVNREVISION="<!(python <(webrtc_root)/build/version.py)"',
     ],
     'conditions': [
+      ['moz_widget_toolkit_gonk==1', {
+        'defines' : [
+          'WEBRTC_GONK',
+        ],
+      }],
       ['enable_tracing==1', {
         'defines': ['WEBRTC_LOGGING',],
       }],
@@ -201,7 +236,8 @@
         ],
         'conditions': [
           ['armv7==1', {
-            'defines': ['WEBRTC_ARCH_ARM_V7',],
+            'defines': ['WEBRTC_ARCH_ARM_V7',
+                        'WEBRTC_BUILD_NEON_LIBS'],
             'conditions': [
               ['arm_neon==1', {
                 'defines': ['WEBRTC_ARCH_ARM_NEON',],
@@ -212,6 +248,19 @@
           }],
         ],
       }],
+      ['os_bsd==1', {
+        'defines': [
+          'WEBRTC_BSD',
+          'WEBRTC_THREAD_RR',
+        ],
+      }],
+      ['OS=="dragonfly" or OS=="netbsd"', {
+        'defines': [
+          # doesn't support pthread_condattr_setclock
+          'WEBRTC_CLOCK_TYPE_REALTIME',
+        ],
+      }],
+      # Mozilla: if we support Mozilla on MIPS, we'll need to mod the cflags entries here
       ['target_arch=="mipsel"', {
         'defines': [
           'MIPS32_LE',
@@ -272,6 +321,13 @@
         ],
       }],
       ['OS=="linux"', {
+#        'conditions': [
+#          ['have_clock_monotonic==1', {
+#            'defines': [
+#              'WEBRTC_CLOCK_TYPE_REALTIME',
+#            ],
+#          }],
+#        ],
         'defines': [
           'WEBRTC_LINUX',
         ],
@@ -295,17 +351,18 @@
         # Re-enable some warnings that Chromium disables.
         'msvs_disabled_warnings!': [4189,],
       }],
+      # used on GONK as well
+      ['enable_android_opensl==1 and (OS=="android" or moz_widget_toolkit_gonk==1)', {
+        'defines': [
+          'WEBRTC_ANDROID_OPENSLES',
+        ],
+      }],
       ['OS=="android"', {
         'defines': [
           'WEBRTC_LINUX',
           'WEBRTC_ANDROID',
          ],
          'conditions': [
-           ['enable_android_opensl==1', {
-             'defines': [
-               'WEBRTC_ANDROID_OPENSLES',
-             ],
-           }],
            ['clang!=1', {
              # The Android NDK doesn't provide optimized versions of these
              # functions. Ensure they are disabled for all compilers.

media/webrtc/trunk/webrtc/build/merge_libs.gyp

@@ -44,5 +44,7 @@
         },
       ],
     },
+#      }],
+#    ],
   ],
 }

media/webrtc/trunk/webrtc/common_audio/common_audio.gyp

@@ -155,6 +155,7 @@
             'resampler/sinc_resampler_sse.cc',
           ],
           'cflags': ['-msse2',],
+          'cflags_mozilla': ['-msse2',],
           'xcode_settings': {
             'OTHER_CFLAGS': ['-msse2',],
           },

media/webrtc/trunk/webrtc/common_audio/resampler/include/resampler.h

@@ -17,98 +17,47 @@
 #define WEBRTC_RESAMPLER_RESAMPLER_H_
 
 #include "webrtc/typedefs.h"
+#include "speex/speex_resampler.h"
 
 namespace webrtc
 {
 
-// TODO(andrew): the implementation depends on the exact values of this enum.
+#define FIXED_RATE_RESAMPLER 0x10
-// It should be rewritten in a less fragile way.
 enum ResamplerType
 {
-    // 4 MSB = Number of channels
+    kResamplerSynchronous            = 0x00,
-    // 4 LSB = Synchronous or asynchronous
+    kResamplerSynchronousStereo      = 0x01,
-
+    kResamplerFixedSynchronous       = 0x00 | FIXED_RATE_RESAMPLER,
-    kResamplerSynchronous = 0x10,
+    kResamplerFixedSynchronousStereo = 0x01 | FIXED_RATE_RESAMPLER,
-    kResamplerAsynchronous = 0x11,
-    kResamplerSynchronousStereo = 0x20,
-    kResamplerAsynchronousStereo = 0x21,
-    kResamplerInvalid = 0xff
-};
-
-// TODO(andrew): doesn't need to be part of the interface.
-enum ResamplerMode
-{
-    kResamplerMode1To1,
-    kResamplerMode1To2,
-    kResamplerMode1To3,
-    kResamplerMode1To4,
-    kResamplerMode1To6,
-    kResamplerMode1To12,
-    kResamplerMode2To3,
-    kResamplerMode2To11,
-    kResamplerMode4To11,
-    kResamplerMode8To11,
-    kResamplerMode11To16,
-    kResamplerMode11To32,
-    kResamplerMode2To1,
-    kResamplerMode3To1,
-    kResamplerMode4To1,
-    kResamplerMode6To1,
-    kResamplerMode12To1,
-    kResamplerMode3To2,
-    kResamplerMode11To2,
-    kResamplerMode11To4,
-    kResamplerMode11To8
 };
 
 class Resampler
 {
-
 public:
     Resampler();
     // TODO(andrew): use an init function instead.
-    Resampler(int inFreq, int outFreq, ResamplerType type);
+    Resampler(int in_freq, int out_freq, ResamplerType type);
     ~Resampler();
 
     // Reset all states
-    int Reset(int inFreq, int outFreq, ResamplerType type);
+    int Reset(int in_freq, int out_freq, ResamplerType type);
 
     // Reset all states if any parameter has changed
-    int ResetIfNeeded(int inFreq, int outFreq, ResamplerType type);
+    int ResetIfNeeded(int in_freq, int out_freq, ResamplerType type);
 
     // Synchronous resampling, all output samples are written to samplesOut
-    int Push(const int16_t* samplesIn, int lengthIn, int16_t* samplesOut,
+    int Push(const int16_t* samples_in, int length_in,
-             int maxLen, int &outLen);
+             int16_t* samples_out, int max_len, int &out_len);
-
-    // Asynchronous resampling, input
-    int Insert(int16_t* samplesIn, int lengthIn);
-
-    // Asynchronous resampling output, remaining samples are buffered
-    int Pull(int16_t* samplesOut, int desiredLen, int &outLen);
 
 private:
-    // Generic pointers since we don't know what states we'll need
+    bool IsFixedRate() { return !!(type_ & FIXED_RATE_RESAMPLER); }
-    void* state1_;
-    void* state2_;
-    void* state3_;
 
-    // Storage if needed
+    SpeexResamplerState* state_;
-    int16_t* in_buffer_;
-    int16_t* out_buffer_;
-    int in_buffer_size_;
-    int out_buffer_size_;
-    int in_buffer_size_max_;
-    int out_buffer_size_max_;
 
-    // State
+    int in_freq_;
-    int my_in_frequency_khz_;
+    int out_freq_;
-    int my_out_frequency_khz_;
+    int channels_;
-    ResamplerMode my_mode_;
+    ResamplerType type_;
-    ResamplerType my_type_;
-
-    // Extra instance for stereo
-    Resampler* slave_left_;
-    Resampler* slave_right_;
 };
 
 }  // namespace webrtc

media/webrtc/trunk/webrtc/common_audio/resampler/push_resampler.cc

@@ -13,7 +13,6 @@
 #include <string.h>
 
 #include "webrtc/common_audio/include/audio_util.h"
-#include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 
 namespace webrtc {

media/webrtc/trunk/webrtc/common_audio/resampler/resampler_unittest.cc

@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <math.h>
+
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
@@ -18,10 +20,7 @@ namespace webrtc {
 namespace {
 const ResamplerType kTypes[] = {
   kResamplerSynchronous,
-  kResamplerAsynchronous,
   kResamplerSynchronousStereo,
-  kResamplerAsynchronousStereo
-  // kResamplerInvalid excluded
 };
 const size_t kTypesSize = sizeof(kTypes) / sizeof(*kTypes);
 
@@ -31,7 +30,7 @@ const int kRates[] = {
   8000,
   16000,
   32000,
-  44000,
+  44100,
   48000,
   kMaxRate
 };
@@ -39,26 +38,19 @@ const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
 const int kMaxChannels = 2;
 const size_t kDataSize = static_cast<size_t> (kMaxChannels * kMaxRate / 100);
 
-// TODO(andrew): should we be supporting these combinations?
-bool ValidRates(int in_rate, int out_rate) {
-  // Not the most compact notation, for clarity.
-  if ((in_rate == 44000 && (out_rate == 48000 || out_rate == 96000)) ||
-      (out_rate == 44000 && (in_rate == 48000 || in_rate == 96000))) {
-    return false;
-  }
-
-  return true;
-}
-
 class ResamplerTest : public testing::Test {
  protected:
   ResamplerTest();
   virtual void SetUp();
   virtual void TearDown();
+  void RunResampleTest(int channels,
+                       int src_sample_rate_hz,
+                       int dst_sample_rate_hz);
 
   Resampler rs_;
   int16_t data_in_[kDataSize];
   int16_t data_out_[kDataSize];
+  int16_t data_reference_[kDataSize];
 };
 
 ResamplerTest::ResamplerTest() {}
@@ -83,34 +75,119 @@ TEST_F(ResamplerTest, Reset) {
         ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j]
             << ", type: " << kTypes[k];
         SCOPED_TRACE(ss.str());
-        if (ValidRates(kRates[i], kRates[j]))
         EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+      }
+    }
+  }
+}
+
+// Sets the signal value to increase by |data| with every sample. Floats are
+// used so non-integer values result in rounding error, but not an accumulating
+// error.
+void SetMonoFrame(int16_t* buffer, float data, int sample_rate_hz) {
+  for (int i = 0; i < sample_rate_hz / 100; i++) {
+    buffer[i] = data * i;
+  }
+}
+
+// Sets the signal value to increase by |left| and |right| with every sample in
+// each channel respectively.
+void SetStereoFrame(int16_t* buffer, float left, float right,
+                    int sample_rate_hz) {
+  for (int i = 0; i < sample_rate_hz / 100; i++) {
+    buffer[i * 2] = left * i;
+    buffer[i * 2 + 1] = right * i;
+  }
+}
+
+// Computes the best SNR based on the error between |ref_frame| and
+// |test_frame|. It allows for a sample delay between the signals to
+// compensate for the resampling delay.
+float ComputeSNR(const int16_t* reference, const int16_t* test,
+                 int sample_rate_hz, int channels, int max_delay) {
+  float best_snr = 0;
+  int best_delay = 0;
+  int samples_per_channel = sample_rate_hz/100;
+  for (int delay = 0; delay < max_delay; delay++) {
+    float mse = 0;
+    float variance = 0;
+    for (int i = 0; i < samples_per_channel * channels - delay; i++) {
+      int error = reference[i] - test[i + delay];
+      mse += error * error;
+      variance += reference[i] * reference[i];
+    }
+    float snr = 100;  // We assign 100 dB to the zero-error case.
+    if (mse > 0)
+      snr = 10 * log10(variance / mse);
+    if (snr > best_snr) {
+      best_snr = snr;
+      best_delay = delay;
+    }
+  }
+  printf("SNR=%.1f dB at delay=%d\n", best_snr, best_delay);
+  return best_snr;
+}
+
+void ResamplerTest::RunResampleTest(int channels,
+                                    int src_sample_rate_hz,
+                                    int dst_sample_rate_hz) {
+  Resampler resampler;  // Create a new one with every test.
+  const int16_t kSrcLeft = 60;  // Shouldn't overflow for any used sample rate.
+  const int16_t kSrcRight = 30;
+  const float kResamplingFactor = (1.0 * src_sample_rate_hz) /
+      dst_sample_rate_hz;
+  const float kDstLeft = kResamplingFactor * kSrcLeft;
+  const float kDstRight = kResamplingFactor * kSrcRight;
+  if (channels == 1)
+    SetMonoFrame(data_in_, kSrcLeft, src_sample_rate_hz);
   else
-          EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+    SetStereoFrame(data_in_, kSrcLeft, kSrcRight, src_sample_rate_hz);
-      }
+
-    }
+  if (channels == 1) {
-  }
+    SetMonoFrame(data_out_, 0, dst_sample_rate_hz);
+    SetMonoFrame(data_reference_, kDstLeft, dst_sample_rate_hz);
+  } else {
+    SetStereoFrame(data_out_, 0, 0, dst_sample_rate_hz);
+    SetStereoFrame(data_reference_, kDstLeft, kDstRight, dst_sample_rate_hz);
   }
 
-// TODO(tlegrand): Replace code inside the two tests below with a function
+  // The speex resampler has a known delay dependent on quality and rates,
-// with number of channels and ResamplerType as input.
+  // which we approximate here. Multiplying by two gives us a crude maximum
-TEST_F(ResamplerTest, Synchronous) {
+  // for any resampling, as the old resampler typically (but not always)
-  for (size_t i = 0; i < kRatesSize; ++i) {
+  // has lower delay.  The actual delay is calculated internally based on the
-    for (size_t j = 0; j < kRatesSize; ++j) {
+  // filter length in the QualityMap.
-      std::ostringstream ss;
+  static const int kInputKernelDelaySamples = 16*3;
-      ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
+  const int max_delay = std::min(1.0f, 1/kResamplingFactor) *
-      SCOPED_TRACE(ss.str());
+                        kInputKernelDelaySamples * channels * 2;
+  printf("(%d, %d Hz) -> (%d, %d Hz) ",  // SNR reported on the same line later.
+      channels, src_sample_rate_hz, channels, dst_sample_rate_hz);
 
-      if (ValidRates(kRates[i], kRates[j])) {
+  int in_length = channels * src_sample_rate_hz / 100;
-        int in_length = kRates[i] / 100;
   int out_length = 0;
-        EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+  EXPECT_EQ(0, rs_.Reset(src_sample_rate_hz, dst_sample_rate_hz,
+                         (channels == 1 ?
+                          kResamplerSynchronous :
+                          kResamplerSynchronousStereo)));
   EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
                         out_length));
-        EXPECT_EQ(kRates[j] / 100, out_length);
+  EXPECT_EQ(channels * dst_sample_rate_hz / 100, out_length);
-      } else {
+
-        EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+  //  EXPECT_EQ(0, Resample(src_frame_, &resampler, &dst_frame_));
+  EXPECT_GT(ComputeSNR(data_reference_, data_out_, dst_sample_rate_hz,
+                       channels, max_delay), 40.0f);
 }
+
+TEST_F(ResamplerTest, Synchronous) {
+  // Number of channels is 1, mono mode.
+  const int kChannels = 1;
+  // We don't attempt to be exhaustive here, but just get good coverage. Some
+  // combinations of rates will not be resampled, and some give an odd
+  // resampling factor which makes it more difficult to evaluate.
+  const int kSampleRates[] = {16000, 32000, 44100, 48000};
+  const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
+  for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
+    for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
+      RunResampleTest(kChannels, kSampleRates[src_rate], kSampleRates[dst_rate]);
     }
   }
 }
@@ -118,24 +195,14 @@ TEST_F(ResamplerTest, Synchronous) {
 TEST_F(ResamplerTest, SynchronousStereo) {
   // Number of channels is 2, stereo mode.
   const int kChannels = 2;
-  for (size_t i = 0; i < kRatesSize; ++i) {
+  // We don't attempt to be exhaustive here, but just get good coverage. Some
-    for (size_t j = 0; j < kRatesSize; ++j) {
+  // combinations of rates will not be resampled, and some give an odd
-      std::ostringstream ss;
+  // resampling factor which makes it more difficult to evaluate.
-      ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
+  const int kSampleRates[] = {16000, 32000, 44100, 48000};
-      SCOPED_TRACE(ss.str());
+  const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
-
+  for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
-      if (ValidRates(kRates[i], kRates[j])) {
+    for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
-        int in_length = kChannels * kRates[i] / 100;
+      RunResampleTest(kChannels, kSampleRates[src_rate], kSampleRates[dst_rate]);
-        int out_length = 0;
-        EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j],
-                               kResamplerSynchronousStereo));
-        EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
-                              out_length));
-        EXPECT_EQ(kChannels * kRates[j] / 100, out_length);
-      } else {
-        EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j],
-                                kResamplerSynchronousStereo));
-      }
     }
   }
 }

media/webrtc/trunk/webrtc/common_audio/resampler/sinc_resampler_neon.cc

@@ -26,11 +26,11 @@ float SincResampler::Convolve_NEON(const float* input_ptr, const float* k1,
 
   const float* upper = input_ptr + kKernelSize;
   for (; input_ptr < upper; ) {
-    m_input = vld1q_f32(input_ptr);
+    m_input = vld1q_f32((const float32_t *) input_ptr);
     input_ptr += 4;
-    m_sums1 = vmlaq_f32(m_sums1, m_input, vld1q_f32(k1));
+    m_sums1 = vmlaq_f32(m_sums1, m_input, vld1q_f32((const float32_t *) k1));
     k1 += 4;
-    m_sums2 = vmlaq_f32(m_sums2, m_input, vld1q_f32(k2));
+    m_sums2 = vmlaq_f32(m_sums2, m_input, vld1q_f32((const float32_t *) k2));
     k2 += 4;
   }
 

media/webrtc/trunk/webrtc/common_types.h

@@ -341,7 +341,7 @@ enum NsModes    // type of Noise Suppression
     kNsLowSuppression,  // lowest suppression
     kNsModerateSuppression,
     kNsHighSuppression,
-    kNsVeryHighSuppression,     // highest suppression
+    kNsVeryHighSuppression     // highest suppression
 };
 
 enum AgcModes                  // type of Automatic Gain Control
@@ -366,7 +366,7 @@ enum EcModes                   // type of Echo Control
     kEcDefault,                // platform default
     kEcConference,             // conferencing default (aggressive AEC)
     kEcAec,                    // Acoustic Echo Cancellation
-    kEcAecm,                   // AEC mobile
+    kEcAecm                    // AEC mobile
 };
 
 // AECM modes
@@ -418,7 +418,7 @@ enum NetEqModes             // NetEQ playout configurations
     kNetEqFax = 2,
     // Minimal buffer management. Inserts zeros for lost packets and during
     // buffer increases.
-    kNetEqOff = 3,
+    kNetEqOff = 3
 };
 
 enum OnHoldModes            // On Hold direction
@@ -432,7 +432,7 @@ enum AmrMode
 {
     kRfc3267BwEfficient = 0,
     kRfc3267OctetAligned = 1,
-    kRfc3267FileStorage = 2,
+    kRfc3267FileStorage = 2
 };
 
 // ==================================================================

media/webrtc/trunk/webrtc/engine_configurations.h

@@ -35,7 +35,9 @@
 #define WEBRTC_CODEC_AVT
 
 // PCM16 is useful for testing and incurs only a small binary size cost.
+#ifndef WEBRTC_CODEC_PCM16
 #define WEBRTC_CODEC_PCM16
+#endif
 
 // iLBC, G.722, and Redundancy coding are excluded from Chromium and Mozilla
 // builds to reduce binary size.

media/webrtc/trunk/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c

@@ -15,10 +15,6 @@
 
 #include "typedefs.h"
 
-#ifdef WEBRTC_BIG_ENDIAN
-#include "signal_processing_library.h"
-#endif
-
 #define HIGHEND 0xFF00
 #define LOWEND    0xFF
 
@@ -30,7 +26,7 @@ int16_t WebRtcPcm16b_EncodeW16(int16_t *speechIn16b,
                                int16_t *speechOut16b)
 {
 #ifdef WEBRTC_BIG_ENDIAN
-    WEBRTC_SPL_MEMCPY_W16(speechOut16b, speechIn16b, len);
+    memcpy(speechOut16b, speechIn16b, len * sizeof(int16_t));
 #else
     int i;
     for (i=0;i<len;i++) {
@@ -69,7 +65,7 @@ int16_t WebRtcPcm16b_DecodeW16(void *inst,
                                int16_t* speechType)
 {
 #ifdef WEBRTC_BIG_ENDIAN
-    WEBRTC_SPL_MEMCPY_W8(speechOut16b, speechIn16b, ((len*sizeof(int16_t)+1)>>1));
+    memcpy(speechOut16b, speechIn16b, ((len*sizeof(int16_t)+1)>>1));
 #else
     int i;
     int samples=len>>1;

media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi

@@ -44,12 +44,6 @@
         'acm_common_defs.h',
         'acm_dtmf_playout.cc',
         'acm_dtmf_playout.h',
-        'acm_g722.cc',
-        'acm_g722.h',
-        'acm_g7221.cc',
-        'acm_g7221.h',
-        'acm_g7221c.cc',
-        'acm_g7221c.h',
         'acm_g729.cc',
         'acm_g729.h',
         'acm_g7291.cc',
@@ -58,11 +52,6 @@
         'acm_generic_codec.h',
         'acm_gsmfr.cc',
         'acm_gsmfr.h',
-        'acm_ilbc.cc',
-        'acm_ilbc.h',
-        'acm_isac.cc',
-        'acm_isac.h',
-        'acm_isac_macros.h',
         'acm_opus.cc',
         'acm_opus.h',
         'acm_speex.cc',

media/webrtc/trunk/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi

@@ -10,12 +10,6 @@
   'variables': {
     'audio_coding_dependencies': [
       'CNG',
-      'G711',
-      'G722',
-      'iLBC',
-      'iSAC',
-      'iSACFix',
-      'PCM16B',
       'NetEq',
       '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
       '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
@@ -25,6 +19,57 @@
       ['include_opus==1', {
         'audio_coding_dependencies': ['webrtc_opus',],
         'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
+        'audio_coding_sources': [
+          'acm_opus.cc',
+          'acm_opus.h',
+        ],
+      }],
+      ['include_g711==1', {
+        'audio_coding_dependencies': ['G711',],
+        'audio_coding_defines': ['WEBRTC_CODEC_G711',],
+        'audio_coding_sources': [
+          'acm_pcma.cc',
+          'acm_pcma.h',
+          'acm_pcmu.cc',
+          'acm_pcmu.h',
+        ],
+      }],
+      ['include_g722==1', {
+        'audio_coding_dependencies': ['G722',],
+        'audio_coding_defines': ['WEBRTC_CODEC_G722',],
+        'audio_coding_sources': [
+          'acm_g722.cc',
+          'acm_g722.h',
+          'acm_g7221.cc',
+          'acm_g7221.h',
+          'acm_g7221c.cc',
+          'acm_g7221c.h',
+        ],
+      }],
+      ['include_ilbc==1', {
+        'audio_coding_dependencies': ['iLBC',],
+        'audio_coding_defines': ['WEBRTC_CODEC_ILBC',],
+        'audio_coding_sources': [
+          'acm_ilbc.cc',
+          'acm_ilbc.h',
+        ],
+      }],
+      ['include_isac==1', {
+        'audio_coding_dependencies': ['iSAC', 'iSACFix',],
+        'audio_coding_defines': ['WEBRTC_CODEC_ISAC', 'WEBRTC_CODEC_ISACFX',],
+        'audio_coding_sources': [
+          'acm_isac.cc',
+          'acm_isac.h',
+          'acm_isac_macros.h',
+        ],
+      }],
+      ['include_pcm16b==1', {
+        'audio_coding_dependencies': ['PCM16B',],
+        'audio_coding_defines': ['WEBRTC_CODEC_PCM16',],
+        'audio_coding_sources': [
+          'acm_pcm16b.cc',
+          'acm_pcm16b.h',
+        ],
       }],
     ],
   },
@@ -50,14 +95,9 @@
         ],
       },
       'sources': [
+#        '<@(audio_coding_sources)',
         '../interface/audio_coding_module.h',
         '../interface/audio_coding_module_typedefs.h',
-        'acm_amr.cc',
-        'acm_amr.h',
-        'acm_amrwb.cc',
-        'acm_amrwb.h',
-        'acm_celt.cc',
-        'acm_celt.h',
         'acm_cng.cc',
         'acm_cng.h',
         'acm_codec_database.cc',
@@ -66,31 +106,13 @@
         'acm_dtmf_detection.h',
         'acm_dtmf_playout.cc',
         'acm_dtmf_playout.h',
-        'acm_g722.cc',
-        'acm_g722.h',
-        'acm_g7221.cc',
-        'acm_g7221.h',
-        'acm_g7221c.cc',
-        'acm_g7221c.h',
-        'acm_g729.cc',
-        'acm_g729.h',
-        'acm_g7291.cc',
-        'acm_g7291.h',
         'acm_generic_codec.cc',
         'acm_generic_codec.h',
-        'acm_gsmfr.cc',
-        'acm_gsmfr.h',
-        'acm_ilbc.cc',
-        'acm_ilbc.h',
-        'acm_isac.cc',
-        'acm_isac.h',
-        'acm_isac_macros.h',
         'acm_neteq.cc',
         'acm_neteq.h',
+# cheat until I get audio_coding_sources to work
         'acm_opus.cc',
         'acm_opus.h',
-        'acm_speex.cc',
-        'acm_speex.h',
         'acm_pcm16b.cc',
         'acm_pcm16b.h',
         'acm_pcma.cc',

media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_defines.h

@@ -69,6 +69,8 @@
  * decoded signal is at 32 kHz.
  * NETEQ_ISAC_FB_CODEC            Enable iSAC-FB
  *
+ * NETEQ_OPUS_CODEC               Enable Opus
+ *
  * NETEQ_G722_CODEC               Enable G.722
  *
  * NETEQ_G729_CODEC               Enable G.729
@@ -321,6 +323,7 @@
     #define NETEQ_PCM16B_CODEC
     #define NETEQ_G711_CODEC
     #define NETEQ_ILBC_CODEC
+    #define NETEQ_OPUS_CODEC
     #define NETEQ_G729_CODEC
     #define NETEQ_G726_CODEC
     #define NETEQ_GSMFR_CODEC
@@ -329,6 +332,7 @@
     /* Wideband codecs */
     #define NETEQ_WIDEBAND
     #define NETEQ_ISAC_CODEC
+    /*#define NETEQ_OPUS_CODEC define only once */
     #define NETEQ_G722_CODEC
     #define NETEQ_G722_1_CODEC
     #define NETEQ_G729_1_CODEC
@@ -337,13 +341,18 @@
 
     /* Super wideband 32kHz codecs */
     #define NETEQ_ISAC_SWB_CODEC
+    /*#define NETEQ_OPUS_CODEC*/
     #define NETEQ_32KHZ_WIDEBAND
     #define NETEQ_G722_1C_CODEC
     #define NETEQ_CELT_CODEC
+    /*#define NETEQ_OPUS_CODEC*/
+
+    /* hack in 48 kHz support */
+    #define NETEQ_48KHZ_WIDEBAND
 
     /* Super wideband 48kHz codecs */
     #define NETEQ_48KHZ_WIDEBAND
-    #define NETEQ_OPUS_CODEC
+    /*#define NETEQ_OPUS_CODEC*/
     #define NETEQ_ISAC_FB
 #endif
 

media/webrtc/trunk/webrtc/modules/audio_coding/neteq/packet_buffer.c

@@ -678,6 +678,11 @@ int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID,
             codecBytes = 15300; /* 240ms @ 510kbps (60ms frames) */
             codecBuffers = 30;  /* Replicating the value for PCMu/a */
         }
+        else if (codecID[i] == kDecoderOpus)
+        {
+            codecBytes = 15300; /* 240ms @ 510kbps (60ms frames) */
+            codecBuffers = 30;  /* ?? Codec supports down to 2.5-60 ms frames */
+        }
         else if ((codecID[i] == kDecoderPCM16B) ||
             (codecID[i] == kDecoderPCM16B_2ch))
         {

media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_jni_android.cc

@@ -26,12 +26,13 @@
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
+#include "AndroidJNIWrapper.h"
+
 namespace webrtc
 {
 // TODO(leozwang): Refactor jni and the following global variables, a
 // good example is jni_helper in Chromium.
 JavaVM* AudioDeviceAndroidJni::globalJvm = NULL;
-JNIEnv* AudioDeviceAndroidJni::globalJNIEnv = NULL;
 jobject AudioDeviceAndroidJni::globalContext = NULL;
 jclass AudioDeviceAndroidJni::globalScClass = NULL;
 
@@ -45,62 +46,62 @@ jclass AudioDeviceAndroidJni::globalScClass = NULL;
 
 int32_t AudioDeviceAndroidJni::SetAndroidAudioDeviceObjects(
     void* javaVM,
-    void* env,
     void* context) {
-  __android_log_print(ANDROID_LOG_DEBUG, "WEBRTC", "JNI:%s", __FUNCTION__);
+  return SetAndroidAudioDeviceObjects(javaVM, NULL, context);
+}
+
+int32_t AudioDeviceAndroidJni::SetAndroidAudioDeviceObjects(
+    void* javaVM,
+    void* null_env,
+    void* context) {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, -1,
+               "%s called", __FUNCTION__);
 
   // TODO(leozwang): Make this function thread-safe.
   globalJvm = reinterpret_cast<JavaVM*>(javaVM);
 
-  if (env) {
+  JNIEnv* env = NULL;
-    globalJNIEnv = reinterpret_cast<JNIEnv*>(env);
+
+  // Check if we already got a reference
+  if (globalJvm && !globalScClass) {
+      if (globalJvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK) {
+      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioDevice, -1,
+                   "%s: could not get Java environment", __FUNCTION__);
+      return -1;
+    }
+    globalJvm->AttachCurrentThread(&env, NULL);
+
     // Get java class type (note path to class packet).
-    jclass javaScClassLocal = globalJNIEnv->FindClass(
+    globalScClass = jsjni_GetGlobalClassRef(AudioCaptureClass);
-        "org/webrtc/voiceengine/WebRTCAudioDevice");
+    if (!globalScClass) {
-    if (!javaScClassLocal) {
       WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
                    "%s: could not find java class", __FUNCTION__);
       return -1; // exception thrown
     }
 
-    // Create a global reference to the class (to tell JNI that we are
+    globalContext = env->NewGlobalRef(
-    // referencing it after this function has returned).
-    globalScClass = reinterpret_cast<jclass> (
-        globalJNIEnv->NewGlobalRef(javaScClassLocal));
-    if (!globalScClass) {
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
-                   "%s: could not create reference", __FUNCTION__);
-      return -1;
-    }
-
-    globalContext = globalJNIEnv->NewGlobalRef(
         reinterpret_cast<jobject>(context));
     if (!globalContext) {
       WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
                    "%s: could not create context reference", __FUNCTION__);
       return -1;
     }
-
-    // Delete local class ref, we only use the global ref
-    globalJNIEnv->DeleteLocalRef(javaScClassLocal);
   }
   else { // User is resetting the env variable
     WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
                  "%s: env is NULL, assuming deinit", __FUNCTION__);
 
-    if (!globalJNIEnv) {
+    if (!env) {
       WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
                    "%s: saved env already NULL", __FUNCTION__);
       return 0;
     }
 
-    globalJNIEnv->DeleteGlobalRef(globalScClass);
+    env->DeleteGlobalRef(globalScClass);
     globalScClass = reinterpret_cast<jclass>(NULL);
 
-    globalJNIEnv->DeleteGlobalRef(globalContext);
+    env->DeleteGlobalRef(globalContext);
     globalContext = reinterpret_cast<jobject>(NULL);
-
-    globalJNIEnv = reinterpret_cast<JNIEnv*>(NULL);
   }
 
   return 0;
@@ -140,8 +141,8 @@ AudioDeviceAndroidJni::AudioDeviceAndroidJni(const int32_t id) :
             _playError(0), _recWarning(0), _recError(0), _delayPlayout(0),
             _delayRecording(0),
             _AGC(false),
-            _samplingFreqIn((N_REC_SAMPLES_PER_SEC/1000)),
+            _samplingFreqIn((N_REC_SAMPLES_PER_SEC)),
-            _samplingFreqOut((N_PLAY_SAMPLES_PER_SEC/1000)),
+            _samplingFreqOut((N_PLAY_SAMPLES_PER_SEC)),
             _maxSpeakerVolume(0),
             _loudSpeakerOn(false),
             _recAudioSource(1), // 1 is AudioSource.MIC which is our default
@@ -1385,17 +1386,10 @@ int32_t AudioDeviceAndroidJni::InitPlayout()
     // get the method ID
     jmethodID initPlaybackID = env->GetMethodID(_javaScClass, "InitPlayback",
                                                 "(I)I");
-
-    int samplingFreq = 44100;
-    if (_samplingFreqOut != 44)
-    {
-        samplingFreq = _samplingFreqOut * 1000;
-    }
-
     int retVal = -1;
 
     // Call java sc object method
-    jint res = env->CallIntMethod(_javaScObj, initPlaybackID, samplingFreq);
+    jint res = env->CallIntMethod(_javaScObj, initPlaybackID, _samplingFreqOut);
     if (res < 0)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -1404,7 +1398,7 @@ int32_t AudioDeviceAndroidJni::InitPlayout()
     else
     {
         // Set the audio device buffer sampling rate
-        _ptrAudioBuffer->SetPlayoutSampleRate(_samplingFreqOut * 1000);
+        _ptrAudioBuffer->SetPlayoutSampleRate(_samplingFreqOut);
         _playIsInitialized = true;
         retVal = 0;
     }
@@ -1490,18 +1484,11 @@ int32_t AudioDeviceAndroidJni::InitRecording()
     // get the method ID
     jmethodID initRecordingID = env->GetMethodID(_javaScClass, "InitRecording",
                                                  "(II)I");
-
-    int samplingFreq = 44100;
-    if (_samplingFreqIn != 44)
-    {
-        samplingFreq = _samplingFreqIn * 1000;
-    }
-
     int retVal = -1;
 
     // call java sc object method
     jint res = env->CallIntMethod(_javaScObj, initRecordingID, _recAudioSource,
-                                  samplingFreq);
+                                  _samplingFreqIn);
     if (res < 0)
     {
         WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -1510,10 +1497,10 @@ int32_t AudioDeviceAndroidJni::InitRecording()
     else
     {
         // Set the audio device buffer sampling rate
-        _ptrAudioBuffer->SetRecordingSampleRate(_samplingFreqIn * 1000);
+        _ptrAudioBuffer->SetRecordingSampleRate(_samplingFreqIn);
 
         // the init rec function returns a fixed delay
-        _delayRecording = res / _samplingFreqIn;
+        _delayRecording = (res * 1000) / _samplingFreqIn;
 
         _recIsInitialized = true;
         retVal = 0;
@@ -2031,14 +2018,7 @@ int32_t AudioDeviceAndroidJni::SetRecordingSampleRate(
     }
 
     // set the recording sample rate to use
-    if (samplesPerSec == 44100)
+    _samplingFreqIn = samplesPerSec;
-    {
-        _samplingFreqIn = 44;
-    }
-    else
-    {
-        _samplingFreqIn = samplesPerSec / 1000;
-    }
 
     // Update the AudioDeviceBuffer
     _ptrAudioBuffer->SetRecordingSampleRate(samplesPerSec);
@@ -2062,14 +2042,7 @@ int32_t AudioDeviceAndroidJni::SetPlayoutSampleRate(
     }
 
     // set the playout sample rate to use
-    if (samplesPerSec == 44100)
+    _samplingFreqOut = samplesPerSec;
-    {
-        _samplingFreqOut = 44;
-    }
-    else
-    {
-        _samplingFreqOut = samplesPerSec / 1000;
-    }
 
     // Update the AudioDeviceBuffer
     _ptrAudioBuffer->SetPlayoutSampleRate(samplesPerSec);
@@ -2211,7 +2184,7 @@ int32_t AudioDeviceAndroidJni::InitJavaResources()
     }
 
     WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "construct object", __FUNCTION__);
+                 "%s: construct object", __FUNCTION__);
 
     // construct the object
     jobject javaScObjLocal = env->NewObject(_javaScClass, cid);
@@ -2423,11 +2396,7 @@ int32_t AudioDeviceAndroidJni::InitSampleRate()
     if (_samplingFreqIn > 0)
     {
         // read the configured sampling rate
-        samplingFreq = 44100;
+        samplingFreq = _samplingFreqIn;
-        if (_samplingFreqIn != 44)
-        {
-            samplingFreq = _samplingFreqIn * 1000;
-        }
         WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
                      "  Trying configured recording sampling rate %d",
                      samplingFreq);
@@ -2468,14 +2437,7 @@ int32_t AudioDeviceAndroidJni::InitSampleRate()
     }
 
     // set the recording sample rate to use
-    if (samplingFreq == 44100)
+    _samplingFreqIn = samplingFreq;
-    {
-        _samplingFreqIn = 44;
-    }
-    else
-    {
-        _samplingFreqIn = samplingFreq / 1000;
-    }
 
     WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
                  "Recording sample rate set to (%d)", _samplingFreqIn);
@@ -2499,11 +2461,7 @@ int32_t AudioDeviceAndroidJni::InitSampleRate()
     if (_samplingFreqOut > 0)
     {
         // read the configured sampling rate
-        samplingFreq = 44100;
+        samplingFreq = _samplingFreqOut;
-        if (_samplingFreqOut != 44)
-        {
-            samplingFreq = _samplingFreqOut * 1000;
-        }
         WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
                      "  Trying configured playback sampling rate %d",
                      samplingFreq);
@@ -2557,15 +2515,7 @@ int32_t AudioDeviceAndroidJni::InitSampleRate()
     }
 
     // set the playback sample rate to use
-    if (samplingFreq == 44100)
+    _samplingFreqOut = samplingFreq;
-    {
-        _samplingFreqOut = 44;
-    }
-    else
-    {
-        _samplingFreqOut = samplingFreq / 1000;
-    }
-
     WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
                  "Playback sample rate set to (%d)", _samplingFreqOut);
 
@@ -2678,7 +2628,7 @@ bool AudioDeviceAndroidJni::PlayThreadProcess()
     if (_playing)
     {
         int8_t playBuffer[2 * 480]; // Max 10 ms @ 48 kHz / 16 bit
-        uint32_t samplesToPlay = _samplingFreqOut * 10;
+        uint32_t samplesToPlay = _samplingFreqOut / 100;
 
         // ask for new PCM data to be played out using the AudioDeviceBuffer
         // ensure that this callback is executed without taking the
@@ -2723,7 +2673,7 @@ bool AudioDeviceAndroidJni::PlayThreadProcess()
         else if (res > 0)
         {
             // we are not recording and have got a delay value from playback
-            _delayPlayout = res / _samplingFreqOut;
+            _delayPlayout = (res * 1000) / _samplingFreqOut;
         }
         // If 0 is returned we are recording and then play delay is updated
         // in RecordProcess
@@ -2821,7 +2771,7 @@ bool AudioDeviceAndroidJni::RecThreadProcess()
 
     if (_recording)
     {
-        uint32_t samplesToRec = _samplingFreqIn * 10;
+        uint32_t samplesToRec = _samplingFreqIn / 100;
 
         // Call java sc object method to record data to direct buffer
         // Will block until data has been recorded (see java sc class),
@@ -2838,7 +2788,7 @@ bool AudioDeviceAndroidJni::RecThreadProcess()
         }
         else
         {
-            _delayPlayout = playDelayInSamples / _samplingFreqOut;
+            _delayPlayout = (playDelayInSamples * 1000) / _samplingFreqOut;
         }
         Lock();
 

media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_jni_android.h

@@ -20,19 +20,12 @@
 
 #include <jni.h> // For accessing AudioDeviceAndroid java class
 
+#define AudioCaptureClass "org/webrtc/voiceengine/WebRTCAudioDevice"
+
 namespace webrtc
 {
 class EventWrapper;
 
-const uint32_t N_REC_SAMPLES_PER_SEC = 16000; // Default is 16 kHz
-const uint32_t N_PLAY_SAMPLES_PER_SEC = 16000; // Default is 16 kHz
-
-const uint32_t N_REC_CHANNELS = 1; // default is mono recording
-const uint32_t N_PLAY_CHANNELS = 1; // default is mono playout
-
-const uint32_t REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz
-
-
 class ThreadWrapper;
 
 class AudioDeviceAndroidJni : public AudioDeviceGeneric {
@@ -40,6 +33,9 @@ class AudioDeviceAndroidJni : public AudioDeviceGeneric {
   AudioDeviceAndroidJni(const int32_t id);
   ~AudioDeviceAndroidJni();
 
+  static int32_t SetAndroidAudioDeviceObjects(void* javaVM,
+                                              void* context);
+
   static int32_t SetAndroidAudioDeviceObjects(void* javaVM,
                                               void* env,
                                               void* context);
@@ -158,6 +154,14 @@ class AudioDeviceAndroidJni : public AudioDeviceGeneric {
   virtual int32_t SetLoudspeakerStatus(bool enable);
   virtual int32_t GetLoudspeakerStatus(bool& enable) const;
 
+  static const uint32_t N_REC_SAMPLES_PER_SEC = 16000; // Default is 16 kHz
+  static const uint32_t N_PLAY_SAMPLES_PER_SEC = 16000; // Default is 16 kHz
+
+  static const uint32_t N_REC_CHANNELS = 1; // default is mono recording
+  static const uint32_t N_PLAY_CHANNELS = 1; // default is mono playout
+
+  static const uint32_t REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz
+
  private:
   // Lock
   void Lock() {

media/webrtc/trunk/webrtc/modules/audio_device/android/audio_manager_jni.h

@@ -36,6 +36,7 @@ class AudioManagerJni {
   // It has to be called for this class' APIs to be successful. Calling
   // ClearAndroidAudioDeviceObjects will prevent this class' APIs to be called
   // successfully if SetAndroidAudioDeviceObjects is not called after it.
+  static void SetAndroidAudioDeviceObjects(void* jvm, void* context);
   static void SetAndroidAudioDeviceObjects(void* jvm, void* env,
                                            void* context);
   // This function must be called when the AudioManagerJni class is no

media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.cc

@@ -11,6 +11,7 @@
 #include "webrtc/modules/audio_device/android/opensles_input.h"
 
 #include <assert.h>
+#include <dlfcn.h>
 
 #include "webrtc/modules/audio_device/android/single_rw_fifo.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
@@ -65,7 +66,8 @@ OpenSlesInput::OpenSlesInput(
       active_queue_(0),
       rec_sampling_rate_(0),
       agc_enabled_(false),
-      recording_delay_(0) {
+      recording_delay_(0),
+      opensles_lib_(NULL) {
 }
 
 OpenSlesInput::~OpenSlesInput() {
@@ -74,15 +76,41 @@ OpenSlesInput::~OpenSlesInput() {
 int32_t OpenSlesInput::Init() {
   assert(!initialized_);
 
+  /* Try to dynamically open the OpenSLES library */
+  opensles_lib_ = dlopen("libOpenSLES.so", RTLD_LAZY);
+  if (!opensles_lib_) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_,
+                   "  failed to dlopen OpenSLES library");
+      return -1;
+  }
+
+  f_slCreateEngine = (slCreateEngine_t)dlsym(opensles_lib_, "slCreateEngine");
+  SL_IID_ENGINE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ENGINE");
+  SL_IID_BUFFERQUEUE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_BUFFERQUEUE");
+  SL_IID_ANDROIDCONFIGURATION_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ANDROIDCONFIGURATION");
+  SL_IID_ANDROIDSIMPLEBUFFERQUEUE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ANDROIDSIMPLEBUFFERQUEUE");
+  SL_IID_RECORD_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_RECORD");
+
+  if (!f_slCreateEngine ||
+      !SL_IID_ENGINE_ ||
+      !SL_IID_BUFFERQUEUE_ ||
+      !SL_IID_ANDROIDCONFIGURATION_ ||
+      !SL_IID_ANDROIDSIMPLEBUFFERQUEUE_ ||
+      !SL_IID_RECORD_) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_,
+                   "  failed to find OpenSLES function");
+      return -1;
+  }
+
   // Set up OpenSL engine.
-  OPENSL_RETURN_ON_FAILURE(slCreateEngine(&sles_engine_, 1, kOption, 0,
+  OPENSL_RETURN_ON_FAILURE(f_slCreateEngine(&sles_engine_, 1, kOption, 0,
                                           NULL, NULL),
                            -1);
   OPENSL_RETURN_ON_FAILURE((*sles_engine_)->Realize(sles_engine_,
                                                     SL_BOOLEAN_FALSE),
                            -1);
   OPENSL_RETURN_ON_FAILURE((*sles_engine_)->GetInterface(sles_engine_,
-                                                         SL_IID_ENGINE,
+                                                         SL_IID_ENGINE_,
                                                          &sles_engine_itf_),
                            -1);
 
@@ -101,6 +129,7 @@ int32_t OpenSlesInput::Terminate() {
   initialized_ = false;
   mic_initialized_ = false;
   rec_initialized_ = false;
+  dlclose(opensles_lib_);
   return 0;
 }
 
@@ -270,8 +299,12 @@ void OpenSlesInput::UpdateRecordingDelay() {
 }
 
 void OpenSlesInput::UpdateSampleRate() {
+#if !defined(WEBRTC_GONK)
   rec_sampling_rate_ = audio_manager_.low_latency_supported() ?
       audio_manager_.native_output_sample_rate() : kDefaultSampleRate;
+#else
+  rec_sampling_rate_ = kDefaultSampleRate;
+#endif
 }
 
 void OpenSlesInput::CalculateNumFifoBuffersNeeded() {
@@ -345,7 +378,7 @@ bool OpenSlesInput::CreateAudioRecorder() {
   // Note the interfaces still need to be initialized. This only tells OpenSl
   // that the interfaces will be needed at some point.
   const SLInterfaceID id[kNumInterfaces] = {
-    SL_IID_ANDROIDSIMPLEBUFFERQUEUE, SL_IID_ANDROIDCONFIGURATION };
+    SL_IID_ANDROIDSIMPLEBUFFERQUEUE_, SL_IID_ANDROIDCONFIGURATION_ };
   const SLboolean req[kNumInterfaces] = {
     SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE };
   OPENSL_RETURN_ON_FAILURE(
@@ -363,13 +396,13 @@ bool OpenSlesInput::CreateAudioRecorder() {
                                                       SL_BOOLEAN_FALSE),
                            false);
   OPENSL_RETURN_ON_FAILURE(
-      (*sles_recorder_)->GetInterface(sles_recorder_, SL_IID_RECORD,
+      (*sles_recorder_)->GetInterface(sles_recorder_, SL_IID_RECORD_,
                                       static_cast<void*>(&sles_recorder_itf_)),
       false);
   OPENSL_RETURN_ON_FAILURE(
       (*sles_recorder_)->GetInterface(
           sles_recorder_,
-          SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+          SL_IID_ANDROIDSIMPLEBUFFERQUEUE_,
           static_cast<void*>(&sles_recorder_sbq_itf_)),
       false);
   return true;

media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.h

@@ -15,7 +15,9 @@
 #include <SLES/OpenSLES_Android.h>
 #include <SLES/OpenSLES_AndroidConfiguration.h>
 
+#if !defined(WEBRTC_GONK)
 #include "webrtc/modules/audio_device/android/audio_manager_jni.h"
+#endif
 #include "webrtc/modules/audio_device/android/low_latency_event.h"
 #include "webrtc/modules/audio_device/android/opensles_common.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
@@ -165,8 +167,10 @@ class OpenSlesInput {
   // Thread-compatible.
   bool CbThreadImpl();
 
+#if !defined(WEBRTC_GONK)
   // Java API handle
   AudioManagerJni audio_manager_;
+#endif
 
   int id_;
   webrtc_opensl::PlayoutDelayProvider* delay_provider_;
@@ -212,6 +216,21 @@ class OpenSlesInput {
 
   // Audio status
   uint16_t recording_delay_;
+
+  // dlopen for OpenSLES
+  void *opensles_lib_;
+  typedef SLresult (*slCreateEngine_t)(SLObjectItf *,
+                                       SLuint32,
+                                       const SLEngineOption *,
+                                       SLuint32,
+                                       const SLInterfaceID *,
+                                       const SLboolean *);
+  slCreateEngine_t f_slCreateEngine;
+  SLInterfaceID SL_IID_ENGINE_;
+  SLInterfaceID SL_IID_BUFFERQUEUE_;
+  SLInterfaceID SL_IID_ANDROIDCONFIGURATION_;
+  SLInterfaceID SL_IID_ANDROIDSIMPLEBUFFERQUEUE_;
+  SLInterfaceID SL_IID_RECORD_;
 };
 
 }  // namespace webrtc

media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_output.cc

@@ -11,6 +11,7 @@
 #include "webrtc/modules/audio_device/android/opensles_output.h"
 
 #include <assert.h>
+#include <dlfcn.h>
 
 #include "webrtc/modules/audio_device/android/fine_audio_buffer.h"
 #include "webrtc/modules/audio_device/android/single_rw_fifo.h"
@@ -65,7 +66,8 @@ OpenSlesOutput::OpenSlesOutput(const int32_t id)
       speaker_sampling_rate_(kDefaultSampleRate),
       buffer_size_samples_(0),
       buffer_size_bytes_(0),
-      playout_delay_(0) {
+      playout_delay_(0),
+      opensles_lib_(NULL) {
 }
 
 OpenSlesOutput::~OpenSlesOutput() {
@@ -74,15 +76,43 @@ OpenSlesOutput::~OpenSlesOutput() {
 int32_t OpenSlesOutput::Init() {
   assert(!initialized_);
 
+  /* Try to dynamically open the OpenSLES library */
+  opensles_lib_ = dlopen("libOpenSLES.so", RTLD_LAZY);
+  if (!opensles_lib_) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_,
+                   "  failed to dlopen OpenSLES library");
+      return -1;
+  }
+
+  f_slCreateEngine = (slCreateEngine_t)dlsym(opensles_lib_, "slCreateEngine");
+  SL_IID_ENGINE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ENGINE");
+  SL_IID_BUFFERQUEUE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_BUFFERQUEUE");
+  SL_IID_ANDROIDCONFIGURATION_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ANDROIDCONFIGURATION");
+  SL_IID_PLAY_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_PLAY");
+  SL_IID_ANDROIDSIMPLEBUFFERQUEUE_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_ANDROIDSIMPLEBUFFERQUEUE");
+  SL_IID_VOLUME_ = *(SLInterfaceID *)dlsym(opensles_lib_, "SL_IID_VOLUME");
+
+  if (!f_slCreateEngine ||
+      !SL_IID_ENGINE_ ||
+      !SL_IID_BUFFERQUEUE_ ||
+      !SL_IID_ANDROIDCONFIGURATION_ ||
+      !SL_IID_PLAY_ ||
+      !SL_IID_ANDROIDSIMPLEBUFFERQUEUE_ ||
+      !SL_IID_VOLUME_) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_,
+                   "  failed to find OpenSLES function");
+      return -1;
+  }
+
   // Set up OpenSl engine.
-  OPENSL_RETURN_ON_FAILURE(slCreateEngine(&sles_engine_, 1, kOption, 0,
+  OPENSL_RETURN_ON_FAILURE(f_slCreateEngine(&sles_engine_, 1, kOption, 0,
                                           NULL, NULL),
                            -1);
   OPENSL_RETURN_ON_FAILURE((*sles_engine_)->Realize(sles_engine_,
                                                     SL_BOOLEAN_FALSE),
                            -1);
   OPENSL_RETURN_ON_FAILURE((*sles_engine_)->GetInterface(sles_engine_,
-                                                         SL_IID_ENGINE,
+                                                         SL_IID_ENGINE_,
                                                          &sles_engine_itf_),
                            -1);
   // Set up OpenSl output mix.
@@ -114,6 +144,7 @@ int32_t OpenSlesOutput::Terminate() {
   initialized_ = false;
   speaker_initialized_ = false;
   play_initialized_ = false;
+  dlclose(opensles_lib_);
   return 0;
 }
 
@@ -302,6 +333,7 @@ void OpenSlesOutput::UpdatePlayoutDelay() {
 }
 
 bool OpenSlesOutput::SetLowLatency() {
+#if !defined(WEBRTC_GONK)
   if (!audio_manager_.low_latency_supported()) {
     return false;
   }
@@ -310,6 +342,9 @@ bool OpenSlesOutput::SetLowLatency() {
   speaker_sampling_rate_ = audio_manager_.native_output_sample_rate();
   assert(speaker_sampling_rate_ > 0);
   return true;
+#else
+  return false;
+#endif
 }
 
 void OpenSlesOutput::CalculateNumFifoBuffersNeeded() {
@@ -395,7 +430,7 @@ bool OpenSlesOutput::CreateAudioPlayer() {
   // Note the interfaces still need to be initialized. This only tells OpenSl
   // that the interfaces will be needed at some point.
   SLInterfaceID ids[kNumInterfaces] = {
-    SL_IID_BUFFERQUEUE, SL_IID_VOLUME, SL_IID_ANDROIDCONFIGURATION };
+    SL_IID_BUFFERQUEUE_, SL_IID_VOLUME_, SL_IID_ANDROIDCONFIGURATION_ };
   SLboolean req[kNumInterfaces] = {
     SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE };
   OPENSL_RETURN_ON_FAILURE(
@@ -408,11 +443,11 @@ bool OpenSlesOutput::CreateAudioPlayer() {
                                                     SL_BOOLEAN_FALSE),
                            false);
   OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_)->GetInterface(sles_player_, SL_IID_PLAY,
+      (*sles_player_)->GetInterface(sles_player_, SL_IID_PLAY_,
                                     &sles_player_itf_),
       false);
   OPENSL_RETURN_ON_FAILURE(
-      (*sles_player_)->GetInterface(sles_player_, SL_IID_BUFFERQUEUE,
+      (*sles_player_)->GetInterface(sles_player_, SL_IID_BUFFERQUEUE_,
                                     &sles_player_sbq_itf_),
       false);
   return true;

media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_output.h

@@ -15,7 +15,9 @@
 #include <SLES/OpenSLES_Android.h>
 #include <SLES/OpenSLES_AndroidConfiguration.h>
 
+#if !defined(WEBRTC_GONK)
 #include "webrtc/modules/audio_device/android/audio_manager_jni.h"
+#endif
 #include "webrtc/modules/audio_device/android/low_latency_event.h"
 #include "webrtc/modules/audio_device/android/opensles_common.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
@@ -181,8 +183,10 @@ class OpenSlesOutput : public webrtc_opensl::PlayoutDelayProvider {
   // Thread-compatible.
   bool CbThreadImpl();
 
+#if !defined(WEBRTC_GONK)
   // Java API handle
   AudioManagerJni audio_manager_;
+#endif
 
   int id_;
   bool initialized_;
@@ -229,6 +233,22 @@ class OpenSlesOutput : public webrtc_opensl::PlayoutDelayProvider {
 
   // Audio status
   uint16_t playout_delay_;
+
+  // dlopen for OpenSLES
+  void *opensles_lib_;
+  typedef SLresult (*slCreateEngine_t)(SLObjectItf *,
+                                       SLuint32,
+                                       const SLEngineOption *,
+                                       SLuint32,
+                                       const SLInterfaceID *,
+                                       const SLboolean *);
+  slCreateEngine_t f_slCreateEngine;
+  SLInterfaceID SL_IID_ENGINE_;
+  SLInterfaceID SL_IID_BUFFERQUEUE_;
+  SLInterfaceID SL_IID_ANDROIDCONFIGURATION_;
+  SLInterfaceID SL_IID_PLAY_;
+  SLInterfaceID SL_IID_ANDROIDSIMPLEBUFFERQUEUE_;
+  SLInterfaceID SL_IID_VOLUME_;
 };
 
 }  // namespace webrtc

media/webrtc/trunk/webrtc/modules/audio_device/audio_device.gypi

@@ -46,11 +46,19 @@
         'dummy/audio_device_utility_dummy.h',
       ],
       'conditions': [
-        ['OS=="linux"', {
+        ['build_with_mozilla==1', {
+          'include_dirs': [
+            '$(DIST)/include',
+          ],
+          'cflags_mozilla': [
+            '$(NSPR_CFLAGS)',
+          ],
+        }],
+        ['OS=="linux" or include_alsa_audio==1 or include_pulse_audio==1', {
           'include_dirs': [
             'linux',
           ],
-        }], # OS==linux
+        }], # OS=="linux" or include_alsa_audio==1 or include_pulse_audio==1
         ['OS=="ios"', {
           'include_dirs': [
             'ios',
@@ -68,9 +76,22 @@
         }],
         ['OS=="android"', {
           'include_dirs': [
+            '$(topsrcdir)/widget/android',
             'android',
           ],
         }], # OS==android
+        ['moz_widget_toolkit_gonk==1', {
+          'include_dirs': [
+            '$(ANDROID_SOURCE)/frameworks/wilhelm/include',
+            '$(ANDROID_SOURCE)/system/media/wilhelm/include',
+            'android',
+          ],
+        }], # moz_widget_toolkit_gonk==1
+        ['enable_android_opensl==1', {
+          'include_dirs': [
+	    'opensl',
+          ],
+        }], # enable_android_opensl
         ['include_internal_audio_device==0', {
           'defines': [
             'WEBRTC_DUMMY_AUDIO_BUILD',
@@ -78,14 +99,8 @@
         }],
         ['include_internal_audio_device==1', {
           'sources': [
-            'linux/alsasymboltable_linux.cc',
-            'linux/alsasymboltable_linux.h',
-            'linux/audio_device_alsa_linux.cc',
-            'linux/audio_device_alsa_linux.h',
             'linux/audio_device_utility_linux.cc',
             'linux/audio_device_utility_linux.h',
-            'linux/audio_mixer_manager_alsa_linux.cc',
-            'linux/audio_mixer_manager_alsa_linux.h',
             'linux/latebindingsymboltable_linux.cc',
             'linux/latebindingsymboltable_linux.h',
             'ios/audio_device_ios.cc',
@@ -109,11 +124,17 @@
             'win/audio_device_utility_win.h',
             'win/audio_mixer_manager_win.cc',
             'win/audio_mixer_manager_win.h',
-            'android/audio_device_utility_android.cc',
-            'android/audio_device_utility_android.h',
           ],
           'conditions': [
             ['OS=="android"', {
+	      'sources': [
+                'opensl/audio_manager_jni.cc',
+                'opensl/audio_manager_jni.h',
+		'android/audio_device_jni_android.cc',
+		'android/audio_device_jni_android.h',
+               ],
+	    }],
+            ['OS=="android" or moz_widget_toolkit_gonk==1', {
               'link_settings': {
                 'libraries': [
                   '-llog',
@@ -123,25 +144,27 @@
               'conditions': [
                 ['enable_android_opensl==1', {
                   'sources': [
-                    'android/audio_device_opensles_android.cc',
+                    'opensl/audio_device_opensles.cc',
-                    'android/audio_device_opensles_android.h',
+                    'opensl/audio_device_opensles.h',
-                    'android/audio_manager_jni.cc',
+                    'opensl/fine_audio_buffer.cc',
-                    'android/audio_manager_jni.h',
+                    'opensl/fine_audio_buffer.h',
-                    'android/fine_audio_buffer.cc',
+                    'opensl/low_latency_event_posix.cc',
-                    'android/fine_audio_buffer.h',
+                    'opensl/low_latency_event.h',
-                    'android/low_latency_event_posix.cc',
+                    'opensl/opensles_common.cc',
-                    'android/low_latency_event.h',
+                    'opensl/opensles_common.h',
-                    'android/opensles_common.cc',
+                    'opensl/opensles_input.cc',
-                    'android/opensles_common.h',
+                    'opensl/opensles_input.h',
-                    'android/opensles_input.cc',
+                    'opensl/opensles_output.cc',
-                    'android/opensles_input.h',
+                    'opensl/opensles_output.h',
-                    'android/opensles_output.cc',
+                    'opensl/single_rw_fifo.cc',
-                    'android/opensles_output.h',
+                    'opensl/single_rw_fifo.h',
-                    'android/single_rw_fifo.cc',
+		    'shared/audio_device_utility_shared.cc',
-                    'android/single_rw_fifo.h',
+		    'shared/audio_device_utility_shared.h',
                   ],
                 }, {
                   'sources': [
+		    'shared/audio_device_utility_shared.cc',
+		    'shared/audio_device_utility_shared.h',
 		    'android/audio_device_jni_android.cc',
 		    'android/audio_device_jni_android.h',
                   ],
@@ -149,16 +172,32 @@
               ],
             }],
             ['OS=="linux"', {
-              'defines': [
-                'LINUX_ALSA',
-              ],
               'link_settings': {
                 'libraries': [
                   '-ldl','-lX11',
                 ],
               },
-              'conditions': [
+            }],
+            ['include_alsa_audio==1', {
+              'cflags_mozilla': [
+                '$(MOZ_ALSA_CFLAGS)',
+              ],
+              'defines': [
+                'LINUX_ALSA',
+              ],
+              'sources': [
+                'linux/alsasymboltable_linux.cc',
+                'linux/alsasymboltable_linux.h',
+                'linux/audio_device_alsa_linux.cc',
+                'linux/audio_device_alsa_linux.h',
+                'linux/audio_mixer_manager_alsa_linux.cc',
+                'linux/audio_mixer_manager_alsa_linux.h',
+              ],
+            }],
             ['include_pulse_audio==1', {
+              'cflags_mozilla': [
+                '$(MOZ_PULSEAUDIO_CFLAGS)',
+              ],
               'defines': [
                 'LINUX_PULSE',
               ],
@@ -171,8 +210,6 @@
                 'linux/pulseaudiosymboltable_linux.h',
               ],
             }],
-              ],
-            }],
             ['OS=="mac" or OS=="ios"', {
               'link_settings': {
                 'libraries': [
@@ -279,4 +316,3 @@
     }], # include_tests
   ],
 }
-

media/webrtc/trunk/webrtc/modules/audio_device/audio_device_impl.cc

@@ -16,21 +16,29 @@
 #include <assert.h>
 #include <string.h>
 
-#if defined(_WIN32)
+#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
+// do not include platform specific headers
+#elif defined(_WIN32)
     #include "audio_device_utility_win.h"
     #include "audio_device_wave_win.h"
  #if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
     #include "audio_device_core_win.h"
  #endif
 #elif defined(WEBRTC_ANDROID_OPENSLES)
+// ANDROID and GONK
     #include <stdlib.h>
+    #include <dlfcn.h>
     #include "audio_device_utility_android.h"
     #include "audio_device_opensles_android.h"
+#if !defined(WEBRTC_GONK)
+    #include "audio_device_jni_android.h"
+#endif
 #elif defined(WEBRTC_ANDROID)
+// GONK only supports opensles; android can use that or jni
     #include <stdlib.h>
     #include "audio_device_utility_android.h"
     #include "audio_device_jni_android.h"
-#elif defined(WEBRTC_LINUX)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
     #include "audio_device_utility_linux.h"
 #if defined(LINUX_ALSA)
     #include "audio_device_alsa_linux.h"
@@ -159,7 +167,7 @@ int32_t AudioDeviceModuleImpl::CheckPlatform()
 #elif defined(WEBRTC_ANDROID)
     platform = kPlatformAndroid;
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is ANDROID");
-#elif defined(WEBRTC_LINUX)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
     platform = kPlatformLinux;
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is LINUX");
 #elif defined(WEBRTC_IOS)
@@ -259,41 +267,41 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
     // Create the *Android OpenSLES* implementation of the Audio Device
     //
 #if defined(WEBRTC_ANDROID_OPENSLES)
+    // Check if the OpenSLES library is available before going further.
+    void* opensles_lib = dlopen("libOpenSLES.so", RTLD_LAZY);
+    if (opensles_lib) {
+        // That worked, close for now and proceed normally.
+        dlclose(opensles_lib);
         if (audioLayer == kPlatformDefaultAudio)
         {
-        // Create *Android OpenELSE Audio* implementation
+            // Create *Android OpenSLES Audio* implementation
             ptrAudioDevice = new AudioDeviceAndroidOpenSLES(Id());
             WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                          "Android OpenSLES Audio APIs will be utilized");
         }
-
-    if (ptrAudioDevice != NULL)
-    {
-        // Create the Android implementation of the Device Utility.
-        ptrAudioDeviceUtility = new AudioDeviceUtilityAndroid(Id());
     }
-    // END #if defined(WEBRTC_ANDROID_OPENSLES)
 
+#if !defined(WEBRTC_GONK)
+    // Fall back to this case if on Android 2.2/OpenSLES not available.
+    if (ptrAudioDevice == NULL) {
         // Create the *Android Java* implementation of the Audio Device
-    //
-#elif defined(WEBRTC_ANDROID)
         if (audioLayer == kPlatformDefaultAudio)
         {
             // Create *Android JNI Audio* implementation
             ptrAudioDevice = new AudioDeviceAndroidJni(Id());
             WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Android JNI Audio APIs will be utilized");
         }
+    }
 
     if (ptrAudioDevice != NULL)
     {
       // Create the Android implementation of the Device Utility.
       ptrAudioDeviceUtility = new AudioDeviceUtilityAndroid(Id());
     }
-    // END #if defined(WEBRTC_ANDROID)
+#endif
-
     // Create the *Linux* implementation of the Audio Device
     //
-#elif defined(WEBRTC_LINUX)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
     if ((audioLayer == kLinuxPulseAudio) || (audioLayer == kPlatformDefaultAudio))
     {
 #if defined(LINUX_PULSE)
@@ -339,7 +347,7 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
         //
         ptrAudioDeviceUtility = new AudioDeviceUtilityLinux(Id());
     }
-#endif  // #if defined(WEBRTC_LINUX)
+#endif  // #if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
 
     // Create the *iPhone* implementation of the Audio Device
     //

media/webrtc/trunk/webrtc/modules/audio_device/audio_device_utility.cc

@@ -46,7 +46,7 @@ bool AudioDeviceUtility::StringCompare(
 
 }  // namespace webrtc
 
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
 
 // ============================================================================
 //                                 Linux & Mac
@@ -109,4 +109,4 @@ bool AudioDeviceUtility::StringCompare(
 
 }  // namespace webrtc
 
-#endif  // defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#endif  // defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)

media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.cc

@@ -1332,7 +1332,7 @@ int32_t AudioDeviceIPhone::InitPlayOrRecord() {
     // todo: Add 48 kHz (increase buffer sizes). Other fs?
     if ((playoutDesc.mSampleRate > 44090.0)
         && (playoutDesc.mSampleRate < 44110.0)) {
-        _adbSampFreq = 44000;
+        _adbSampFreq = 44100;
     } else if ((playoutDesc.mSampleRate > 15990.0)
                && (playoutDesc.mSampleRate < 16010.0)) {
         _adbSampFreq = 16000;

media/webrtc/trunk/webrtc/modules/audio_device/ios/audio_device_ios.h

@@ -19,8 +19,8 @@
 namespace webrtc {
 class ThreadWrapper;
 
-const uint32_t N_REC_SAMPLES_PER_SEC = 44000;
+const uint32_t N_REC_SAMPLES_PER_SEC = 44100;
-const uint32_t N_PLAY_SAMPLES_PER_SEC = 44000;
+const uint32_t N_PLAY_SAMPLES_PER_SEC = 44100;
 
 const uint32_t N_REC_CHANNELS = 1;  // default is mono recording
 const uint32_t N_PLAY_CHANNELS = 1;  // default is mono playout

media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc

@@ -19,6 +19,13 @@
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
+#include "Latency.h"
+
+#define LOG_FIRST_CAPTURE(x) LogTime(AsyncLatencyLogger::AudioCaptureBase, \
+                                     reinterpret_cast<uint64_t>(x), 0)
+#define LOG_CAPTURE_FRAMES(x, frames) LogLatency(AsyncLatencyLogger::AudioCapture, \
+                                                 reinterpret_cast<uint64_t>(x), frames)
+
 webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
 
 // Accesses ALSA functions through our late-binding symbol table instead of
@@ -96,6 +103,7 @@ AudioDeviceLinuxALSA::AudioDeviceLinuxALSA(const int32_t id) :
     _playBufType(AudioDeviceModule::kFixedBufferSize),
     _initialized(false),
     _recording(false),
+    _firstRecord(true),
     _playing(false),
     _recIsInitialized(false),
     _playIsInitialized(false),
@@ -986,7 +994,8 @@ int32_t AudioDeviceLinuxALSA::RecordingDeviceName(
         memset(guid, 0, kAdmMaxGuidSize);
     }
     
-    return GetDevicesInfo(1, false, index, name, kAdmMaxDeviceNameSize);
+    return GetDevicesInfo(1, false, index, name, kAdmMaxDeviceNameSize,
+                          guid, kAdmMaxGuidSize);
 }
 
 int16_t AudioDeviceLinuxALSA::RecordingDevices()
@@ -1448,6 +1457,7 @@ int32_t AudioDeviceLinuxALSA::StartRecording()
     }
     // RECORDING
     const char* threadName = "webrtc_audio_module_capture_thread";
+    _firstRecord = true;
     _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc,
                                                 this,
                                                 kRealtimePriority,
@@ -1634,6 +1644,17 @@ int32_t AudioDeviceLinuxALSA::StartPlayout()
         return -1;
     }
 
+    int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
+    if (errVal < 0)
+    {
+        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                     "     playout snd_pcm_prepare failed (%s)\n",
+                     LATE(snd_strerror)(errVal));
+        // just log error
+        // if snd_pcm_open fails will return -1
+    }
+
+
     unsigned int threadID(0);
     if (!_ptrThreadPlay->Start(threadID))
     {
@@ -1648,16 +1669,6 @@ int32_t AudioDeviceLinuxALSA::StartPlayout()
     }
     _playThreadID = threadID;
 
-    int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
-    if (errVal < 0)
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     "     playout snd_pcm_prepare failed (%s)\n",
-                     LATE(snd_strerror)(errVal));
-        // just log error
-        // if snd_pcm_open fails will return -1
-    }
-
     return 0;
 }
 
@@ -1829,7 +1840,9 @@ int32_t AudioDeviceLinuxALSA::GetDevicesInfo(
     const bool playback,
     const int32_t enumDeviceNo,
     char* enumDeviceName,
-    const int32_t ednLen) const
+    const int32_t ednLen,
+    char* enumDeviceId,
+    const int32_t ediLen) const
 {
     
     // Device enumeration based on libjingle implementation
@@ -1868,6 +1881,8 @@ int32_t AudioDeviceLinuxALSA::GetDevicesInfo(
             function == FUNC_GET_DEVICE_NAME_FOR_AN_ENUM) && enumDeviceNo == 0)
         {
             strcpy(enumDeviceName, "default");
+            if (enumDeviceId)
+                memset(enumDeviceId, 0, ediLen);
 
             err = LATE(snd_device_name_free_hint)(hints);
             if (err != 0)
@@ -1930,6 +1945,11 @@ int32_t AudioDeviceLinuxALSA::GetDevicesInfo(
                     // We have found the enum device, copy the name to buffer.
                     strncpy(enumDeviceName, desc, ednLen);
                     enumDeviceName[ednLen-1] = '\0';
+                    if (enumDeviceId)
+                    {
+                        strncpy(enumDeviceId, name, ediLen);
+                        enumDeviceId[ediLen-1] = '\0';
+                    }
                     keepSearching = false;
                     // Replace '\n' with '-'.
                     char * pret = strchr(enumDeviceName, '\n'/*0xa*/); //LF
@@ -1942,6 +1962,11 @@ int32_t AudioDeviceLinuxALSA::GetDevicesInfo(
                     // We have found the enum device, copy the name to buffer.
                     strncpy(enumDeviceName, name, ednLen);
                     enumDeviceName[ednLen-1] = '\0';
+                    if (enumDeviceId)
+                    {
+                        strncpy(enumDeviceId, name, ediLen);
+                        enumDeviceId[ediLen-1] = '\0';
+                    }
                     keepSearching = false;
                 }
 
@@ -2251,6 +2276,11 @@ bool AudioDeviceLinuxALSA::RecThreadProcess()
         { // buf is full
             _recordingFramesLeft = _recordingFramesIn10MS;
 
+            if (_firstRecord) {
+              LOG_FIRST_CAPTURE(this);
+              _firstRecord = false;
+            }
+            LOG_CAPTURE_FRAMES(this, _recordingFramesIn10MS);
             // store the recorded buffer (no action will be taken if the
             // #recorded samples is not a full buffer)
             _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,

media/webrtc/trunk/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h

@@ -167,7 +167,9 @@ private:
                            const bool playback,
                            const int32_t enumDeviceNo = 0,
                            char* enumDeviceName = NULL,
-                           const int32_t ednLen = 0) const;
+                           const int32_t ednLen = 0,
+                           char* enumDeviceID = NULL,
+                           const int32_t ediLen = 0) const;
     int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
 
 private:
@@ -233,6 +235,7 @@ private:
 private:
     bool _initialized;
     bool _recording;
+    bool _firstRecord;
     bool _playing;
     bool _recIsInitialized;
     bool _playIsInitialized;

media/webrtc/trunk/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc

@@ -27,7 +27,7 @@
 
 #include "webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h"
 
-#ifdef WEBRTC_LINUX
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
 #include <dlfcn.h>
 #endif
 
@@ -37,8 +37,8 @@ using namespace webrtc;
 namespace webrtc_adm_linux {
 
 inline static const char *GetDllError() {
-#ifdef WEBRTC_LINUX
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
-  char *err = dlerror();
+  const char *err = dlerror();
   if (err) {
     return err;
   } else {
@@ -50,7 +50,7 @@ inline static const char *GetDllError() {
 }
 
 DllHandle InternalLoadDll(const char dll_name[]) {
-#ifdef WEBRTC_LINUX
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
   DllHandle handle = dlopen(dll_name, RTLD_NOW);
 #else
 #error Not implemented
@@ -63,7 +63,7 @@ DllHandle InternalLoadDll(const char dll_name[]) {
 }
 
 void InternalUnloadDll(DllHandle handle) {
-#ifdef WEBRTC_LINUX
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
   if (dlclose(handle) != 0) {
     WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
                "%s", GetDllError());
@@ -76,9 +76,9 @@ void InternalUnloadDll(DllHandle handle) {
 static bool LoadSymbol(DllHandle handle,
                        const char *symbol_name,
                        void **symbol) {
-#ifdef WEBRTC_LINUX
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
   *symbol = dlsym(handle, symbol_name);
-  char *err = dlerror();
+  const char *err = dlerror();
   if (err) {
     WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
                "Error loading symbol %s : %d", symbol_name, err);
@@ -101,7 +101,7 @@ bool InternalLoadSymbols(DllHandle handle,
                          int num_symbols,
                          const char *const symbol_names[],
                          void *symbols[]) {
-#ifdef WEBRTC_LINUX
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
   // Clear any old errors.
   dlerror();
 #endif

media/webrtc/trunk/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h

@@ -42,7 +42,7 @@
 
 namespace webrtc_adm_linux {
 
-#ifdef WEBRTC_LINUX
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
 typedef void *DllHandle;
 
 const DllHandle kInvalidDllHandle = NULL;

media/webrtc/trunk/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc

@@ -29,7 +29,11 @@
 
 namespace webrtc_adm_linux_pulse {
 
+#if defined(__OpenBSD__) || defined(WEBRTC_GONK)
+LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so")
+#else
 LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so.0")
+#endif
 #define X(sym) \
     LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(PulseAudioSymbolTable, sym)
 PULSE_AUDIO_SYMBOLS_LIST

media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_device_opensles.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/audio_device_opensles_android.cc"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_device_opensles.h

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/audio_device_opensles_android.h"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_manager_jni.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/audio_manager_jni.cc"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/audio_manager_jni.h

@@ -0,0 +1,6 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/audio_manager_jni.h"
+

media/webrtc/trunk/webrtc/modules/audio_device/opensl/fine_audio_buffer.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/fine_audio_buffer.cc"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/fine_audio_buffer.h

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/fine_audio_buffer.h"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/low_latency_event_posix.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/low_latency_event_posix.cc"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/low_latency_event_posix.h

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/low_latency_event_posix.h"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_common.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/opensles_common.cc"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_common.h

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/opensles_common.h"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_input.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/opensles_input.cc"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_input.h

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/opensles_input.h"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_output.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/opensles_output.cc"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/opensles_output.h

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/opensles_output.h"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/single_rw_fifo.cc"

media/webrtc/trunk/webrtc/modules/audio_device/opensl/single_rw_fifo.h

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/single_rw_fifo.h"

media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_jni_android.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/audio_device_jni_android.cc"

media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_jni_android.h

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/audio_device_jni_android.h"

media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_utility_shared.cc

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/audio_device_utility_android.cc"

media/webrtc/trunk/webrtc/modules/audio_device/shared/audio_device_utility_shared.h

@@ -0,0 +1,5 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "../android/audio_device_utility_android.h"

media/webrtc/trunk/webrtc/modules/audio_device/test/audio_device_test_api.cc

@@ -210,7 +210,7 @@ class AudioDeviceAPITest: public testing::Test {
     // Create default implementation instance
     EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
                 kId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
-#elif defined(WEBRTC_LINUX)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
     EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
                 kId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
     EXPECT_TRUE((audio_device_ = AudioDeviceModuleImpl::Create(
@@ -1703,7 +1703,7 @@ TEST_F(AudioDeviceAPITest, CPULoad) {
 
 // TODO(kjellander): Fix flakiness causing failures on Windows.
 // TODO(phoglund):  Fix flakiness causing failures on Linux.
-#if !defined(_WIN32) && !defined(WEBRTC_LINUX)
+#if !defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_BSD)
 TEST_F(AudioDeviceAPITest, StartAndStopRawOutputFileRecording) {
   // NOTE: this API is better tested in a functional test
   CheckInitialPlayoutStates();
@@ -1772,7 +1772,7 @@ TEST_F(AudioDeviceAPITest, StartAndStopRawInputFileRecording) {
   // - size of raw_input_not_recording.pcm shall be 0
   // - size of raw_input_not_recording.pcm shall be > 0
 }
-#endif  // !WIN32 && !WEBRTC_LINUX
+#endif  // !WIN32 && !WEBRTC_LINUX && !defined(WEBRTC_BSD)
 
 TEST_F(AudioDeviceAPITest, RecordingSampleRate) {
   uint32_t sampleRate(0);
@@ -1783,10 +1783,10 @@ TEST_F(AudioDeviceAPITest, RecordingSampleRate) {
   EXPECT_EQ(48000, sampleRate);
 #elif defined(ANDROID)
   TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
-  EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000));
+  EXPECT_TRUE((sampleRate == 44100) || (sampleRate == 16000));
 #elif defined(WEBRTC_IOS)
   TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
-  EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000) ||
+  EXPECT_TRUE((sampleRate == 44100) || (sampleRate == 16000) ||
               (sampleRate == 8000));
 #endif
 
@@ -1802,10 +1802,10 @@ TEST_F(AudioDeviceAPITest, PlayoutSampleRate) {
   EXPECT_EQ(48000, sampleRate);
 #elif defined(ANDROID)
   TEST_LOG("Playout sample rate is %u\n\n", sampleRate);
-  EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000));
+  EXPECT_TRUE((sampleRate == 44100) || (sampleRate == 16000));
 #elif defined(WEBRTC_IOS)
   TEST_LOG("Playout sample rate is %u\n\n", sampleRate);
-  EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000) ||
+  EXPECT_TRUE((sampleRate == 44100) || (sampleRate == 16000) ||
               (sampleRate == 8000));
 #endif
 }

media/webrtc/trunk/webrtc/modules/audio_device/test/func_test_manager.cc

@@ -348,12 +348,6 @@ int32_t AudioTransportImpl::NeedMorePlayData(
                 int32_t fsInHz(samplesPerSecIn);
                 int32_t fsOutHz(samplesPerSec);
 
-                if (fsInHz == 44100)
-                    fsInHz = 44000;
-
-                if (fsOutHz == 44100)
-                    fsOutHz = 44000;
-
                 if (nChannelsIn == 2 && nBytesPerSampleIn == 4)
                 {
                     // input is stereo => we will resample in stereo
@@ -1258,7 +1252,7 @@ int32_t FuncTestManager::TestAudioTransport()
         if (samplesPerSec == 48000) {
             _audioTransport->SetFilePlayout(
                 true, GetResource(_playoutFile48.c_str()));
-        } else if (samplesPerSec == 44100 || samplesPerSec == 44000) {
+        } else if (samplesPerSec == 44100) {
             _audioTransport->SetFilePlayout(
                 true, GetResource(_playoutFile44.c_str()));
         } else if (samplesPerSec == 16000) {
@@ -1491,7 +1485,7 @@ int32_t FuncTestManager::TestSpeakerVolume()
         if (48000 == samplesPerSec) {
             _audioTransport->SetFilePlayout(
                 true, GetResource(_playoutFile48.c_str()));
-        } else if (44100 == samplesPerSec || samplesPerSec == 44000) {
+        } else if (44100 == samplesPerSec) {
             _audioTransport->SetFilePlayout(
                 true, GetResource(_playoutFile44.c_str()));
         } else if (samplesPerSec == 16000) {
@@ -1592,7 +1586,7 @@ int32_t FuncTestManager::TestSpeakerMute()
         EXPECT_EQ(0, audioDevice->PlayoutSampleRate(&samplesPerSec));
         if (48000 == samplesPerSec)
             _audioTransport->SetFilePlayout(true, _playoutFile48.c_str());
-        else if (44100 == samplesPerSec || 44000 == samplesPerSec)
+        else if (44100 == samplesPerSec)
             _audioTransport->SetFilePlayout(true, _playoutFile44.c_str());
         else
         {

media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.c

@@ -109,7 +109,17 @@ const float WebRtcAec_overDriveCurve[65] = {
 // Target suppression levels for nlp modes.
 // log{0.001, 0.00001, 0.00000001}
 static const float kTargetSupp[3] = { -6.9f, -11.5f, -18.4f };
-static const float kMinOverDrive[3] = { 1.0f, 2.0f, 5.0f };
+
+// Two sets of parameters, one for the extended filter mode.
+static const float kExtendedMinOverDrive[3] = { 3.0f, 6.0f, 15.0f };
+static const float kNormalMinOverDrive[3] = { 1.0f, 2.0f, 5.0f };
+static const float kExtendedSmoothingCoefficients[2][2] =
+    { { 0.9f, 0.1f }, { 0.92f, 0.08f } };
+static const float kNormalSmoothingCoefficients[2][2] =
+    { { 0.9f, 0.1f }, { 0.93f, 0.07f } };
+
+// Number of partitions forming the NLP's "preferred" bands.
+enum { kPrefBandSize = 24 };
 
 #ifdef WEBRTC_AEC_DEBUG_DUMP
 extern int webrtc_aec_instance_count;
@@ -281,13 +291,13 @@ int WebRtcAec_FreeAec(AecCore* aec)
 static void FilterFar(AecCore* aec, float yf[2][PART_LEN1])
 {
   int i;
-  for (i = 0; i < NR_PART; i++) {
+  for (i = 0; i < aec->num_partitions; i++) {
     int j;
     int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
+    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-      xPos -= NR_PART*(PART_LEN1);
+      xPos -= aec->num_partitions*(PART_LEN1);
     }
 
     for (j = 0; j < PART_LEN1; j++) {
@@ -301,22 +311,25 @@ static void FilterFar(AecCore* aec, float yf[2][PART_LEN1])
 
 static void ScaleErrorSignal(AecCore* aec, float ef[2][PART_LEN1])
 {
+  const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
+  const float error_threshold = aec->extended_filter_enabled ?
+      kExtendedErrorThreshold : aec->normal_error_threshold;
   int i;
-  float absEf;
+  float abs_ef;
   for (i = 0; i < (PART_LEN1); i++) {
     ef[0][i] /= (aec->xPow[i] + 1e-10f);
     ef[1][i] /= (aec->xPow[i] + 1e-10f);
-    absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
+    abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
 
-    if (absEf > aec->errThresh) {
+    if (abs_ef > error_threshold) {
-      absEf = aec->errThresh / (absEf + 1e-10f);
+      abs_ef = error_threshold / (abs_ef + 1e-10f);
-      ef[0][i] *= absEf;
+      ef[0][i] *= abs_ef;
-      ef[1][i] *= absEf;
+      ef[1][i] *= abs_ef;
     }
 
     // Stepsize factor
-    ef[0][i] *= aec->mu;
+    ef[0][i] *= mu;
-    ef[1][i] *= aec->mu;
+    ef[1][i] *= mu;
   }
 }
 
@@ -325,35 +338,35 @@ static void ScaleErrorSignal(AecCore* aec, float ef[2][PART_LEN1])
 //static void FilterAdaptationUnconstrained(AecCore* aec, float *fft,
 //                                          float ef[2][PART_LEN1]) {
 //  int i, j;
-//  for (i = 0; i < NR_PART; i++) {
+//  for (i = 0; i < aec->num_partitions; i++) {
 //    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
 //    int pos;
 //    // Check for wrap
-//    if (i + aec->xfBufBlockPos >= NR_PART) {
+//    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-//      xPos -= NR_PART * PART_LEN1;
+//      xPos -= aec->num_partitions * PART_LEN1;
 //    }
 //
 //    pos = i * PART_LEN1;
 //
 //    for (j = 0; j < PART_LEN1; j++) {
-//      aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
+//      aec->wfBuf[0][pos + j] += MulRe(aec->xfBuf[0][xPos + j],
-//                                      -aec->xfBuf[xPos + j][1],
+//                                      -aec->xfBuf[1][xPos + j],
-//                                      ef[j][0], ef[j][1]);
+//                                      ef[0][j], ef[1][j]);
-//      aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
+//      aec->wfBuf[1][pos + j] += MulIm(aec->xfBuf[0][xPos + j],
-//                                      -aec->xfBuf[xPos + j][1],
+//                                      -aec->xfBuf[1][xPos + j],
-//                                      ef[j][0], ef[j][1]);
+//                                      ef[0][j], ef[1][j]);
 //    }
 //  }
 //}
 
 static void FilterAdaptation(AecCore* aec, float *fft, float ef[2][PART_LEN1]) {
   int i, j;
-  for (i = 0; i < NR_PART; i++) {
+  for (i = 0; i < aec->num_partitions; i++) {
     int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
     int pos;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
+    if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-      xPos -= NR_PART * PART_LEN1;
+      xPos -= aec->num_partitions * PART_LEN1;
     }
 
     pos = i * PART_LEN1;
@@ -427,12 +440,12 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq)
     aec->sampFreq = sampFreq;
 
     if (sampFreq == 8000) {
-        aec->mu = 0.6f;
+        aec->normal_mu = 0.6f;
-        aec->errThresh = 2e-6f;
+        aec->normal_error_threshold = 2e-6f;
     }
     else {
-        aec->mu = 0.5f;
+        aec->normal_mu = 0.5f;
-        aec->errThresh = 1.5e-6f;
+        aec->normal_error_threshold = 1.5e-6f;
     }
 
     if (WebRtc_InitBuffer(aec->nearFrBuf) == -1) {
@@ -474,6 +487,9 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq)
     aec->delay_logging_enabled = 0;
     memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram));
 
+    aec->extended_filter_enabled = 0;
+    aec->num_partitions = kNormalNumPartitions;
+
     // Default target suppression mode.
     aec->nlp_mode = 1;
 
@@ -514,11 +530,14 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq)
     aec->xfBufBlockPos = 0;
     // TODO: Investigate need for these initializations. Deleting them doesn't
     //       change the output at all and yields 0.4% overall speedup.
-    memset(aec->xfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
+    memset(aec->xfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions *
-    memset(aec->wfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
+        PART_LEN1);
+    memset(aec->wfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions *
+        PART_LEN1);
     memset(aec->sde, 0, sizeof(complex_t) * PART_LEN1);
     memset(aec->sxd, 0, sizeof(complex_t) * PART_LEN1);
-    memset(aec->xfwBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
+    memset(aec->xfwBuf, 0, sizeof(complex_t) * kExtendedNumPartitions *
+        PART_LEN1);
     memset(aec->se, 0, sizeof(float) * PART_LEN1);
 
     // To prevent numerical instability in the first block.
@@ -734,13 +753,11 @@ int WebRtcAec_GetDelayMetricsCore(AecCore* self, int* median, int* std) {
 }
 
 int WebRtcAec_echo_state(AecCore* self) {
-  assert(self != NULL);
   return self->echoState;
 }
 
 void WebRtcAec_GetEchoStats(AecCore* self, Stats* erl, Stats* erle,
                             Stats* a_nlp) {
-  assert(self != NULL);
   assert(erl != NULL);
   assert(erle != NULL);
   assert(a_nlp != NULL);
@@ -751,14 +768,12 @@ void WebRtcAec_GetEchoStats(AecCore* self, Stats* erl, Stats* erle,
 
 #ifdef WEBRTC_AEC_DEBUG_DUMP
 void* WebRtcAec_far_time_buf(AecCore* self) {
-  assert(self != NULL);
   return self->far_time_buf;
 }
 #endif
 
 void WebRtcAec_SetConfigCore(AecCore* self, int nlp_mode, int metrics_mode,
                              int delay_logging) {
-  assert(self != NULL);
   assert(nlp_mode >= 0 && nlp_mode < 3);
   self->nlp_mode = nlp_mode;
   self->metricsMode = metrics_mode;
@@ -771,13 +786,20 @@ void WebRtcAec_SetConfigCore(AecCore* self, int nlp_mode, int metrics_mode,
   }
 }
 
+void WebRtcAec_enable_delay_correction(AecCore* self, int enable) {
+  self->extended_filter_enabled = enable;
+  self->num_partitions = enable ? kExtendedNumPartitions : kNormalNumPartitions;
+}
+
+int WebRtcAec_delay_correction_enabled(AecCore* self) {
+  return self->extended_filter_enabled;
+}
+
 int WebRtcAec_system_delay(AecCore* self) {
-  assert(self != NULL);
   return self->system_delay;
 }
 
 void WebRtcAec_SetSystemDelay(AecCore* self, int delay) {
-  assert(self != NULL);
   assert(delay >= 0);
   self->system_delay = delay;
 }
@@ -853,7 +875,8 @@ static void ProcessBlock(AecCore* aec) {
     for (i = 0; i < PART_LEN1; i++) {
       far_spectrum = (xf_ptr[i] * xf_ptr[i]) +
           (xf_ptr[PART_LEN1 + i] * xf_ptr[PART_LEN1 + i]);
-      aec->xPow[i] = gPow[0] * aec->xPow[i] + gPow[1] * NR_PART * far_spectrum;
+      aec->xPow[i] = gPow[0] * aec->xPow[i] + gPow[1] * aec->num_partitions *
+          far_spectrum;
       // Calculate absolute spectra
       abs_far_spectrum[i] = sqrtf(far_spectrum);
 
@@ -913,7 +936,7 @@ static void ProcessBlock(AecCore* aec) {
     // Update the xfBuf block position.
     aec->xfBufBlockPos--;
     if (aec->xfBufBlockPos == -1) {
-        aec->xfBufBlockPos = NR_PART - 1;
+        aec->xfBufBlockPos = aec->num_partitions - 1;
     }
 
     // Buffer xf
@@ -1014,18 +1037,21 @@ static void NonLinearProcessing(AecCore* aec, short *output, short *outputH)
     float cohde[PART_LEN1], cohxd[PART_LEN1];
     float hNlDeAvg, hNlXdAvg;
     float hNl[PART_LEN1];
-    float hNlPref[PREF_BAND_SIZE];
+    float hNlPref[kPrefBandSize];
     float hNlFb = 0, hNlFbLow = 0;
     const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
-    const int prefBandSize = PREF_BAND_SIZE / aec->mult;
+    const int prefBandSize = kPrefBandSize / aec->mult;
     const int minPrefBand = 4 / aec->mult;
 
     // Near and error power sums
     float sdSum = 0, seSum = 0;
 
-    // Power estimate smoothing coefficients
+    // Power estimate smoothing coefficients.
-    const float gCoh[2][2] = {{0.9f, 0.1f}, {0.93f, 0.07f}};
+    const float *ptrGCoh = aec->extended_filter_enabled ?
-    const float *ptrGCoh = gCoh[aec->mult - 1];
+        kExtendedSmoothingCoefficients[aec->mult - 1] :
+        kNormalSmoothingCoefficients[aec->mult - 1];
+    const float* min_overdrive = aec->extended_filter_enabled ?
+        kExtendedMinOverDrive : kNormalMinOverDrive;
 
     // Filter energy
     float wfEnMax = 0, wfEn = 0;
@@ -1048,7 +1074,7 @@ static void NonLinearProcessing(AecCore* aec, short *output, short *outputH)
     if (aec->delayEstCtr == 0) {
         wfEnMax = 0;
         aec->delayIdx = 0;
-        for (i = 0; i < NR_PART; i++) {
+        for (i = 0; i < aec->num_partitions; i++) {
             pos = i * PART_LEN1;
             wfEn = 0;
             for (j = 0; j < PART_LEN1; j++) {
@@ -1189,7 +1215,7 @@ static void NonLinearProcessing(AecCore* aec, short *output, short *outputH)
 
     if (aec->hNlXdAvgMin == 1) {
         aec->echoState = 0;
-        aec->overDrive = kMinOverDrive[aec->nlp_mode];
+        aec->overDrive = min_overdrive[aec->nlp_mode];
 
         if (aec->stNearState == 1) {
             memcpy(hNl, cohde, sizeof(hNl));
@@ -1245,7 +1271,7 @@ static void NonLinearProcessing(AecCore* aec, short *output, short *outputH)
         aec->hNlMinCtr = 0;
         aec->overDrive = WEBRTC_SPL_MAX(kTargetSupp[aec->nlp_mode] /
             ((float)log(aec->hNlFbMin + 1e-10f) + 1e-10f),
-            kMinOverDrive[aec->nlp_mode]);
+            min_overdrive[aec->nlp_mode]);
     }
 
     // Smooth the overdrive.
@@ -1465,7 +1491,6 @@ static void InitStats(Stats* stats) {
 }
 
 static void InitMetrics(AecCore* self) {
-  assert(self != NULL);
   self->stateCounter = 0;
   InitLevel(&self->farlevel);
   InitLevel(&self->nearlevel);
@@ -1687,3 +1712,4 @@ static void TimeToFrequency(float time_data[PART_LEN2],
     freq_data[1][i] = time_data[2 * i + 1];
   }
 }
+

media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.h

@@ -70,23 +70,38 @@ void WebRtcAec_ProcessFrame(AecCore* aec,
 // Returns the number of elements moved, and adjusts |system_delay| by the
 // corresponding amount in ms.
 int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements);
+
 // Calculates the median and standard deviation among the delay estimates
 // collected since the last call to this function.
 int WebRtcAec_GetDelayMetricsCore(AecCore* self, int* median, int* std);
+
 // Returns the echo state (1: echo, 0: no echo).
 int WebRtcAec_echo_state(AecCore* self);
+
 // Gets statistics of the echo metrics ERL, ERLE, A_NLP.
 void WebRtcAec_GetEchoStats(AecCore* self, Stats* erl, Stats* erle,
                             Stats* a_nlp);
 #ifdef WEBRTC_AEC_DEBUG_DUMP
 void* WebRtcAec_far_time_buf(AecCore* self);
 #endif
+
 // Sets local configuration modes.
 void WebRtcAec_SetConfigCore(AecCore* self, int nlp_mode, int metrics_mode,
                              int delay_logging);
+
+// We now interpret delay correction to mean an extended filter length feature.
+// We reuse the delay correction infrastructure to avoid changes through to
+// libjingle. See details along with |DelayCorrection| in
+// echo_cancellation_impl.h. Non-zero enables, zero disables.
+void WebRtcAec_enable_delay_correction(AecCore* self, int enable);
+
+// Returns non-zero if delay correction is enabled and zero if disabled.
+int WebRtcAec_delay_correction_enabled(AecCore* self);
+
 // Returns the current |system_delay|, i.e., the buffered difference between
 // far-end and near-end.
 int WebRtcAec_system_delay(AecCore* self);
+
 // Sets the |system_delay| to |value|.  Note that if the value is changed
 // improperly, there can be a performance regression.  So it should be used with
 // care.

media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_internal.h

@@ -22,6 +22,16 @@
 #define NR_PART 12  // Number of partitions in filter.
 #define PREF_BAND_SIZE 24
 
+// Number of partitions for the extended filter mode. The first one is an enum
+// to be used in array declarations, as it represents the maximum filter length.
+enum { kExtendedNumPartitions = 32 };
+static const int kNormalNumPartitions = 12;
+
+// Extended filter adaptation parameters.
+// TODO(ajm): No narrowband tuning yet.
+static const float kExtendedMu = 0.4f;
+static const float kExtendedErrorThreshold = 1.0e-6f;
+
 typedef struct PowerLevel {
   float sfrsum;
   int sfrcounter;
@@ -56,11 +66,12 @@ struct AecCore {
   float dInitMinPow[PART_LEN1];
   float *noisePow;
 
-  float xfBuf[2][NR_PART * PART_LEN1];  // farend fft buffer
+  float xfBuf[2][kExtendedNumPartitions * PART_LEN1];  // farend fft buffer
-  float wfBuf[2][NR_PART * PART_LEN1];  // filter fft
+  float wfBuf[2][kExtendedNumPartitions * PART_LEN1];  // filter fft
   complex_t sde[PART_LEN1];  // cross-psd of nearend and error
   complex_t sxd[PART_LEN1];  // cross-psd of farend and nearend
-  complex_t xfwBuf[NR_PART * PART_LEN1];  // farend windowed fft buffer
+  // Farend windowed fft buffer.
+  complex_t xfwBuf[kExtendedNumPartitions * PART_LEN1];
 
   float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1];  // far, near, error psd
   float hNs[PART_LEN1];
@@ -85,8 +96,8 @@ struct AecCore {
   int sampFreq;
   uint32_t seed;
 
-  float mu;  // stepsize
+  float normal_mu;  // stepsize
-  float errThresh;  // error threshold
+  float normal_error_threshold;  // error threshold
 
   int noiseEstCtr;
 
@@ -112,6 +123,11 @@ struct AecCore {
   void* delay_estimator_farend;
   void* delay_estimator;
 
+  // 1 = extended filter mode enabled, 0 = disabled.
+  int extended_filter_enabled;
+  // Runtime selection of number of filter partitions.
+  int num_partitions;
+
 #ifdef WEBRTC_AEC_DEBUG_DUMP
   RingBuffer* far_time_buf;
   FILE *farFile;

media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_sse2.c

@@ -34,13 +34,14 @@ __inline static float MulIm(float aRe, float aIm, float bRe, float bIm)
 static void FilterFarSSE2(AecCore* aec, float yf[2][PART_LEN1])
 {
   int i;
-  for (i = 0; i < NR_PART; i++) {
+  const int num_partitions = aec->num_partitions;
+  for (i = 0; i < num_partitions; i++) {
     int j;
     int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
+    if (i + aec->xfBufBlockPos >= num_partitions) {
-      xPos -= NR_PART*(PART_LEN1);
+      xPos -= num_partitions*(PART_LEN1);
     }
 
     // vectorized code (four at once)
@@ -75,8 +76,11 @@ static void FilterFarSSE2(AecCore* aec, float yf[2][PART_LEN1])
 static void ScaleErrorSignalSSE2(AecCore* aec, float ef[2][PART_LEN1])
 {
   const __m128 k1e_10f = _mm_set1_ps(1e-10f);
-  const __m128 kThresh = _mm_set1_ps(aec->errThresh);
+  const __m128 kMu = aec->extended_filter_enabled ?
-  const __m128 kMu = _mm_set1_ps(aec->mu);
+      _mm_set1_ps(kExtendedMu) : _mm_set1_ps(aec->normal_mu);
+  const __m128 kThresh = aec->extended_filter_enabled ?
+      _mm_set1_ps(kExtendedErrorThreshold) :
+      _mm_set1_ps(aec->normal_error_threshold);
 
   int i;
   // vectorized code (four at once)
@@ -110,32 +114,39 @@ static void ScaleErrorSignalSSE2(AecCore* aec, float ef[2][PART_LEN1])
     _mm_storeu_ps(&ef[1][i], ef_im);
   }
   // scalar code for the remaining items.
+  {
+    const float mu = aec->extended_filter_enabled ?
+        kExtendedMu : aec->normal_mu;
+    const float error_threshold = aec->extended_filter_enabled ?
+        kExtendedErrorThreshold : aec->normal_error_threshold;
     for (; i < (PART_LEN1); i++) {
-    float absEf;
+    float abs_ef;
       ef[0][i] /= (aec->xPow[i] + 1e-10f);
       ef[1][i] /= (aec->xPow[i] + 1e-10f);
-    absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
+      abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
 
-    if (absEf > aec->errThresh) {
+      if (abs_ef > error_threshold) {
-      absEf = aec->errThresh / (absEf + 1e-10f);
+        abs_ef = error_threshold / (abs_ef + 1e-10f);
-      ef[0][i] *= absEf;
+        ef[0][i] *= abs_ef;
-      ef[1][i] *= absEf;
+        ef[1][i] *= abs_ef;
       }
 
       // Stepsize factor
-    ef[0][i] *= aec->mu;
+      ef[0][i] *= mu;
-    ef[1][i] *= aec->mu;
+      ef[1][i] *= mu;
+    }
   }
 }
 
 static void FilterAdaptationSSE2(AecCore* aec, float *fft, float ef[2][PART_LEN1]) {
   int i, j;
-  for (i = 0; i < NR_PART; i++) {
+  const int num_partitions = aec->num_partitions;
+  for (i = 0; i < num_partitions; i++) {
     int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
     int pos = i * PART_LEN1;
     // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
+    if (i + aec->xfBufBlockPos >= num_partitions) {
-      xPos -= NR_PART * PART_LEN1;
+      xPos -= num_partitions * PART_LEN1;
     }
 
     // Process the whole array...
@@ -413,3 +424,4 @@ void WebRtcAec_InitAec_SSE2(void) {
   WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
   WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
 }
+

media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.c

@@ -27,6 +27,61 @@
 #include "webrtc/modules/audio_processing/utility/ring_buffer.h"
 #include "webrtc/typedefs.h"
 
+// Measured delays [ms]
+// Device                Chrome  GTP
+// MacBook Air           10
+// MacBook Retina        10      100
+// MacPro                30?
+//
+// Win7 Desktop          70      80?
+// Win7 T430s            110
+// Win8 T420s            70
+//
+// Daisy                 50
+// Pixel (w/ preproc?)           240
+// Pixel (w/o preproc?)  110     110
+
+// The extended filter mode gives us the flexibility to ignore the system's
+// reported delays. We do this for platforms which we believe provide results
+// which are incompatible with the AEC's expectations. Based on measurements
+// (some provided above) we set a conservative (i.e. lower than measured)
+// fixed delay.
+//
+// WEBRTC_UNTRUSTED_DELAY will only have an impact when |extended_filter_mode|
+// is enabled. See the note along with |DelayCorrection| in
+// echo_cancellation_impl.h for more details on the mode.
+//
+// Justification:
+// Chromium/Mac: Here, the true latency is so low (~10-20 ms), that it plays
+// havoc with the AEC's buffering. To avoid this, we set a fixed delay of 20 ms
+// and then compensate by rewinding by 10 ms (in wideband) through
+// kDelayDiffOffsetSamples. This trick does not seem to work for larger rewind
+// values, but fortunately this is sufficient.
+//
+// Chromium/Linux(ChromeOS): The values we get on this platform don't correspond
+// well to reality. The variance doesn't match the AEC's buffer changes, and the
+// bulk values tend to be too low. However, the range across different hardware
+// appears to be too large to choose a single value.
+//
+// GTP/Linux(ChromeOS): TBD, but for the moment we will trust the values.
+#if defined(WEBRTC_CHROMIUM_BUILD) && defined(WEBRTC_MAC)
+#define WEBRTC_UNTRUSTED_DELAY
+#endif
+
+#if defined(WEBRTC_MAC)
+static const int kFixedDelayMs = 20;
+static const int kDelayDiffOffsetSamples = -160;
+#elif defined(WEBRTC_WIN)
+static const int kFixedDelayMs = 50;
+static const int kDelayDiffOffsetSamples = 0;
+#else
+// Essentially ChromeOS.
+static const int kFixedDelayMs = 50;
+static const int kDelayDiffOffsetSamples = 0;
+#endif
+static const int kMinTrustedDelayMs = 20;
+static const int kMaxTrustedDelayMs = 500;
+
 // Maximum length of resampled signal. Must be an integer multiple of frames
 // (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
 // The factor of 2 handles wb, and the + 1 is as a safety margin
@@ -43,7 +98,14 @@ int webrtc_aec_instance_count = 0;
 
 // Estimates delay to set the position of the far-end buffer read pointer
 // (controlled by knownDelay)
-static int EstBufDelay(aecpc_t *aecInst);
+static void EstBufDelayNormal(aecpc_t *aecInst);
+static void EstBufDelayExtended(aecpc_t *aecInst);
+static int ProcessNormal(aecpc_t* self, const int16_t* near,
+    const int16_t* near_high, int16_t* out, int16_t* out_high,
+    int16_t num_samples, int16_t reported_delay_ms, int32_t skew);
+static void ProcessExtended(aecpc_t* self, const int16_t* near,
+    const int16_t* near_high, int16_t* out, int16_t* out_high,
+    int16_t num_samples, int16_t reported_delay_ms, int32_t skew);
 
 int32_t WebRtcAec_Create(void **aecInst)
 {
@@ -135,10 +197,6 @@ int32_t WebRtcAec_Init(void *aecInst, int32_t sampFreq, int32_t scSampFreq)
     aecpc_t *aecpc = aecInst;
     AecConfig aecConfig;
 
-    if (aecpc == NULL) {
-        return -1;
-    }
-
     if (sampFreq != 8000 && sampFreq != 16000  && sampFreq != 32000) {
         aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
         return -1;
@@ -177,31 +235,31 @@ int32_t WebRtcAec_Init(void *aecInst, int32_t sampFreq, int32_t scSampFreq)
         aecpc->splitSampFreq = sampFreq;
     }
 
-    aecpc->skewFrCtr = 0;
-    aecpc->activity = 0;
-
     aecpc->delayCtr = 0;
+    aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
+    // Sampling frequency multiplier (SWB is processed as 160 frame size).
+    aecpc->rate_factor = aecpc->splitSampFreq / 8000;
 
     aecpc->sum = 0;
     aecpc->counter = 0;
     aecpc->checkBuffSize = 1;
     aecpc->firstVal = 0;
 
-    aecpc->ECstartup = 1;
+    aecpc->startup_phase = 1;
     aecpc->bufSizeStart = 0;
     aecpc->checkBufSizeCtr = 0;
-    aecpc->filtDelay = 0;
+    aecpc->msInSndCardBuf = 0;
+    aecpc->filtDelay = -1;  // -1 indicates an initialized state.
     aecpc->timeForDelayChange = 0;
     aecpc->knownDelay = 0;
     aecpc->lastDelayDiff = 0;
 
-    aecpc->skew = 0;
+    aecpc->skewFrCtr = 0;
     aecpc->resample = kAecFalse;
     aecpc->highSkewCtr = 0;
-    aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
+    aecpc->skew = 0;
 
-    // Sampling frequency multiplier (SWB is processed as 160 frame size).
+    aecpc->farend_started = 0;
-    aecpc->rate_factor = aecpc->splitSampFreq / 8000;
 
     // Default settings.
     aecConfig.nlpMode = kAecNlpModerate;
@@ -239,10 +297,6 @@ int32_t WebRtcAec_BufferFarend(void *aecInst, const int16_t *farend,
     float skew;
     int i = 0;
 
-    if (aecpc == NULL) {
-        return -1;
-    }
-
     if (farend == NULL) {
         aecpc->lastError = AEC_NULL_POINTER_ERROR;
         return -1;
@@ -268,6 +322,7 @@ int32_t WebRtcAec_BufferFarend(void *aecInst, const int16_t *farend,
         farend_ptr = (const int16_t*) newFarend;
     }
 
+    aecpc->farend_started = 1;
     WebRtcAec_SetSystemDelay(aecpc->aec, WebRtcAec_system_delay(aecpc->aec) +
                              newNrOfSamples);
 
@@ -311,17 +366,6 @@ int32_t WebRtcAec_Process(void *aecInst, const int16_t *nearend,
 {
     aecpc_t *aecpc = aecInst;
     int32_t retVal = 0;
-    short i;
-    short nBlocks10ms;
-    short nFrames;
-    // Limit resampling to doubling/halving of signal
-    const float minSkewEst = -0.5f;
-    const float maxSkewEst = 1.0f;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
     if (nearend == NULL) {
         aecpc->lastError = AEC_NULL_POINTER_ERROR;
         return -1;
@@ -354,143 +398,20 @@ int32_t WebRtcAec_Process(void *aecInst, const int16_t *nearend,
         aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
         retVal = -1;
     }
-    else if (msInSndCardBuf > 500) {
+    else if (msInSndCardBuf > kMaxTrustedDelayMs) {
-        msInSndCardBuf = 500;
+        // The clamping is now done in ProcessExtended/Normal().
         aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
         retVal = -1;
     }
-    // TODO(andrew): we need to investigate if this +10 is really wanted.
-    msInSndCardBuf += 10;
-    aecpc->msInSndCardBuf = msInSndCardBuf;
 
-    if (aecpc->skewMode == kAecTrue) {
+    // This returns the value of aec->extended_filter_enabled.
-        if (aecpc->skewFrCtr < 25) {
+    if (WebRtcAec_delay_correction_enabled(aecpc->aec)) {
-            aecpc->skewFrCtr++;
+      ProcessExtended(aecpc, nearend, nearendH, out, outH, nrOfSamples,
-        }
+                      msInSndCardBuf, skew);
-        else {
-            retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
-            if (retVal == -1) {
-                aecpc->skew = 0;
-                aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
-            }
-
-            aecpc->skew /= aecpc->sampFactor*nrOfSamples;
-
-            if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
-                aecpc->resample = kAecFalse;
-            }
-            else {
-                aecpc->resample = kAecTrue;
-            }
-
-            if (aecpc->skew < minSkewEst) {
-                aecpc->skew = minSkewEst;
-            }
-            else if (aecpc->skew > maxSkewEst) {
-                aecpc->skew = maxSkewEst;
-            }
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-            (void)fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
-#endif
-        }
-    }
-
-    nFrames = nrOfSamples / FRAME_LEN;
-    nBlocks10ms = nFrames / aecpc->rate_factor;
-
-    if (aecpc->ECstartup) {
-        if (nearend != out) {
-            // Only needed if they don't already point to the same place.
-            memcpy(out, nearend, sizeof(short) * nrOfSamples);
-        }
-
-        // The AEC is in the start up mode
-        // AEC is disabled until the system delay is OK
-
-        // Mechanism to ensure that the system delay is reasonably stable.
-        if (aecpc->checkBuffSize) {
-            aecpc->checkBufSizeCtr++;
-            // Before we fill up the far-end buffer we require the system delay
-            // to be stable (+/-8 ms) compared to the first value. This
-            // comparison is made during the following 6 consecutive 10 ms
-            // blocks. If it seems to be stable then we start to fill up the
-            // far-end buffer.
-            if (aecpc->counter == 0) {
-                aecpc->firstVal = aecpc->msInSndCardBuf;
-                aecpc->sum = 0;
-            }
-
-            if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
-                WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
-                aecpc->sum += aecpc->msInSndCardBuf;
-                aecpc->counter++;
-            }
-            else {
-                aecpc->counter = 0;
-            }
-
-            if (aecpc->counter * nBlocks10ms >= 6) {
-                // The far-end buffer size is determined in partitions of
-                // PART_LEN samples. Use 75% of the average value of the system
-                // delay as buffer size to start with.
-                aecpc->bufSizeStart = WEBRTC_SPL_MIN((3 * aecpc->sum *
-                  aecpc->rate_factor * 8) / (4 * aecpc->counter * PART_LEN),
-                  kMaxBufSizeStart);
-                // Buffer size has now been determined.
-                aecpc->checkBuffSize = 0;
-            }
-
-            if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
-                // For really bad systems, don't disable the echo canceller for
-                // more than 0.5 sec.
-                aecpc->bufSizeStart = WEBRTC_SPL_MIN((aecpc->msInSndCardBuf *
-                    aecpc->rate_factor * 3) / 40, kMaxBufSizeStart);
-                aecpc->checkBuffSize = 0;
-            }
-        }
-
-        // If |checkBuffSize| changed in the if-statement above.
-        if (!aecpc->checkBuffSize) {
-            // The system delay is now reasonably stable (or has been unstable
-            // for too long). When the far-end buffer is filled with
-            // approximately the same amount of data as reported by the system
-            // we end the startup phase.
-            int overhead_elements =
-                WebRtcAec_system_delay(aecpc->aec) / PART_LEN -
-                aecpc->bufSizeStart;
-            if (overhead_elements == 0) {
-                // Enable the AEC
-                aecpc->ECstartup = 0;
-            } else if (overhead_elements > 0) {
-                // TODO(bjornv): Do we need a check on how much we actually
-                // moved the read pointer? It should always be possible to move
-                // the pointer |overhead_elements| since we have only added data
-                // to the buffer and no delay compensation nor AEC processing
-                // has been done.
-                WebRtcAec_MoveFarReadPtr(aecpc->aec, overhead_elements);
-
-                // Enable the AEC
-                aecpc->ECstartup = 0;
-            }
-        }
     } else {
-        // AEC is enabled.
+      if (ProcessNormal(aecpc, nearend, nearendH, out, outH, nrOfSamples,
-
+                        msInSndCardBuf, skew) != 0) {
-        EstBufDelay(aecpc);
+        retVal = -1;
-
-        // Note that 1 frame is supported for NB and 2 frames for WB.
-        for (i = 0; i < nFrames; i++) {
-            // Call the AEC.
-            WebRtcAec_ProcessFrame(aecpc->aec,
-                                   &nearend[FRAME_LEN * i],
-                                   &nearendH[FRAME_LEN * i],
-                                   aecpc->knownDelay,
-                                   &out[FRAME_LEN * i],
-                                   &outH[FRAME_LEN * i]);
-            // TODO(bjornv): Re-structure such that we don't have to pass
-            // |aecpc->knownDelay| as input. Change name to something like
-            // |system_buffer_diff|.
       }
     }
 
@@ -509,11 +430,6 @@ int32_t WebRtcAec_Process(void *aecInst, const int16_t *nearend,
 
 int WebRtcAec_set_config(void* handle, AecConfig config) {
   aecpc_t* self = (aecpc_t*)handle;
-
-  if (handle == NULL ) {
-    return -1;
-  }
-
   if (self->initFlag != initCheck) {
     self->lastError = AEC_UNINITIALIZED_ERROR;
     return -1;
@@ -548,10 +464,6 @@ int WebRtcAec_set_config(void* handle, AecConfig config) {
 
 int WebRtcAec_get_echo_status(void* handle, int* status) {
   aecpc_t* self = (aecpc_t*)handle;
-
-  if (handle == NULL ) {
-    return -1;
-  }
   if (status == NULL ) {
     self->lastError = AEC_NULL_POINTER_ERROR;
     return -1;
@@ -665,10 +577,6 @@ int WebRtcAec_GetMetrics(void* handle, AecMetrics* metrics) {
 
 int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std) {
   aecpc_t* self = handle;
-
-  if (handle == NULL) {
-    return -1;
-  }
   if (median == NULL) {
     self->lastError = AEC_NULL_POINTER_ERROR;
     return -1;
@@ -693,11 +601,6 @@ int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std) {
 int32_t WebRtcAec_get_error_code(void *aecInst)
 {
     aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
     return aecpc->lastError;
 }
 
@@ -708,7 +611,225 @@ AecCore* WebRtcAec_aec_core(void* handle) {
   return ((aecpc_t*) handle)->aec;
 }
 
-static int EstBufDelay(aecpc_t* aecpc) {
+static int ProcessNormal(aecpc_t *aecpc, const int16_t *nearend,
+                         const int16_t *nearendH, int16_t *out, int16_t *outH,
+                         int16_t nrOfSamples, int16_t msInSndCardBuf,
+                         int32_t skew) {
+  int retVal = 0;
+  short i;
+  short nBlocks10ms;
+  short nFrames;
+  // Limit resampling to doubling/halving of signal
+  const float minSkewEst = -0.5f;
+  const float maxSkewEst = 1.0f;
+
+  msInSndCardBuf = msInSndCardBuf > kMaxTrustedDelayMs ?
+      kMaxTrustedDelayMs : msInSndCardBuf;
+  // TODO(andrew): we need to investigate if this +10 is really wanted.
+  msInSndCardBuf += 10;
+  aecpc->msInSndCardBuf = msInSndCardBuf;
+
+  if (aecpc->skewMode == kAecTrue) {
+    if (aecpc->skewFrCtr < 25) {
+      aecpc->skewFrCtr++;
+    }
+    else {
+      retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
+      if (retVal == -1) {
+        aecpc->skew = 0;
+        aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
+      }
+
+      aecpc->skew /= aecpc->sampFactor*nrOfSamples;
+
+      if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
+        aecpc->resample = kAecFalse;
+      }
+      else {
+        aecpc->resample = kAecTrue;
+      }
+
+      if (aecpc->skew < minSkewEst) {
+        aecpc->skew = minSkewEst;
+      }
+      else if (aecpc->skew > maxSkewEst) {
+        aecpc->skew = maxSkewEst;
+      }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+      (void)fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
+#endif
+    }
+  }
+
+  nFrames = nrOfSamples / FRAME_LEN;
+  nBlocks10ms = nFrames / aecpc->rate_factor;
+
+  if (aecpc->startup_phase) {
+    // Only needed if they don't already point to the same place.
+    if (nearend != out) {
+      memcpy(out, nearend, sizeof(short) * nrOfSamples);
+    }
+    if (nearendH != outH) {
+      memcpy(outH, nearendH, sizeof(short) * nrOfSamples);
+    }
+
+    // The AEC is in the start up mode
+    // AEC is disabled until the system delay is OK
+
+    // Mechanism to ensure that the system delay is reasonably stable.
+    if (aecpc->checkBuffSize) {
+      aecpc->checkBufSizeCtr++;
+      // Before we fill up the far-end buffer we require the system delay
+      // to be stable (+/-8 ms) compared to the first value. This
+      // comparison is made during the following 6 consecutive 10 ms
+      // blocks. If it seems to be stable then we start to fill up the
+      // far-end buffer.
+      if (aecpc->counter == 0) {
+        aecpc->firstVal = aecpc->msInSndCardBuf;
+        aecpc->sum = 0;
+      }
+
+      if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
+        WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
+        aecpc->sum += aecpc->msInSndCardBuf;
+        aecpc->counter++;
+      }
+      else {
+        aecpc->counter = 0;
+      }
+
+      if (aecpc->counter * nBlocks10ms >= 6) {
+        // The far-end buffer size is determined in partitions of
+        // PART_LEN samples. Use 75% of the average value of the system
+        // delay as buffer size to start with.
+        aecpc->bufSizeStart = WEBRTC_SPL_MIN((3 * aecpc->sum *
+            aecpc->rate_factor * 8) / (4 * aecpc->counter * PART_LEN),
+            kMaxBufSizeStart);
+        // Buffer size has now been determined.
+        aecpc->checkBuffSize = 0;
+      }
+
+      if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
+        // For really bad systems, don't disable the echo canceller for
+        // more than 0.5 sec.
+        aecpc->bufSizeStart = WEBRTC_SPL_MIN((aecpc->msInSndCardBuf *
+            aecpc->rate_factor * 3) / 40, kMaxBufSizeStart);
+        aecpc->checkBuffSize = 0;
+      }
+    }
+
+    // If |checkBuffSize| changed in the if-statement above.
+    if (!aecpc->checkBuffSize) {
+      // The system delay is now reasonably stable (or has been unstable
+      // for too long). When the far-end buffer is filled with
+      // approximately the same amount of data as reported by the system
+      // we end the startup phase.
+      int overhead_elements =
+          WebRtcAec_system_delay(aecpc->aec) / PART_LEN - aecpc->bufSizeStart;
+      if (overhead_elements == 0) {
+        // Enable the AEC
+        aecpc->startup_phase = 0;
+      } else if (overhead_elements > 0) {
+        // TODO(bjornv): Do we need a check on how much we actually
+        // moved the read pointer? It should always be possible to move
+        // the pointer |overhead_elements| since we have only added data
+        // to the buffer and no delay compensation nor AEC processing
+        // has been done.
+        WebRtcAec_MoveFarReadPtr(aecpc->aec, overhead_elements);
+
+        // Enable the AEC
+        aecpc->startup_phase = 0;
+      }
+    }
+  } else {
+    // AEC is enabled.
+    EstBufDelayNormal(aecpc);
+
+    // Note that 1 frame is supported for NB and 2 frames for WB.
+    for (i = 0; i < nFrames; i++) {
+      // Call the AEC.
+      WebRtcAec_ProcessFrame(aecpc->aec,
+                             &nearend[FRAME_LEN * i],
+                             &nearendH[FRAME_LEN * i],
+                             aecpc->knownDelay,
+                             &out[FRAME_LEN * i],
+                             &outH[FRAME_LEN * i]);
+      // TODO(bjornv): Re-structure such that we don't have to pass
+      // |aecpc->knownDelay| as input. Change name to something like
+      // |system_buffer_diff|.
+    }
+  }
+
+  return retVal;
+}
+
+static void ProcessExtended(aecpc_t* self, const int16_t* near,
+    const int16_t* near_high, int16_t* out, int16_t* out_high,
+    int16_t num_samples, int16_t reported_delay_ms, int32_t skew) {
+  int i;
+  const int num_frames = num_samples / FRAME_LEN;
+#if defined(WEBRTC_UNTRUSTED_DELAY)
+  const int delay_diff_offset = kDelayDiffOffsetSamples;
+  reported_delay_ms = kFixedDelayMs;
+#else
+  // This is the usual mode where we trust the reported system delay values.
+  const int delay_diff_offset = 0;
+  // Due to the longer filter, we no longer add 10 ms to the reported delay
+  // to reduce chance of non-causality. Instead we apply a minimum here to avoid
+  // issues with the read pointer jumping around needlessly.
+  reported_delay_ms = reported_delay_ms < kMinTrustedDelayMs ?
+      kMinTrustedDelayMs : reported_delay_ms;
+  // If the reported delay appears to be bogus, we attempt to recover by using
+  // the measured fixed delay values. We use >= here because higher layers
+  // may already clamp to this maximum value, and we would otherwise not
+  // detect it here.
+  reported_delay_ms = reported_delay_ms >= kMaxTrustedDelayMs ?
+      kFixedDelayMs : reported_delay_ms;
+#endif
+  self->msInSndCardBuf = reported_delay_ms;
+
+  if (!self->farend_started) {
+    // Only needed if they don't already point to the same place.
+    if (near != out) {
+      memcpy(out, near, sizeof(short) * num_samples);
+    }
+    if (near_high != out_high) {
+      memcpy(out_high, near_high, sizeof(short) * num_samples);
+    }
+    return;
+  }
+  if (self->startup_phase) {
+    // In the extended mode, there isn't a startup "phase", just a special
+    // action on the first frame. In the trusted delay case, we'll take the
+    // current reported delay, unless it's less then our conservative
+    // measurement.
+    int startup_size_ms = reported_delay_ms < kFixedDelayMs ?
+        kFixedDelayMs : reported_delay_ms;
+    int overhead_elements = (WebRtcAec_system_delay(self->aec) -
+        startup_size_ms / 2 * self->rate_factor * 8) / PART_LEN;
+    WebRtcAec_MoveFarReadPtr(self->aec, overhead_elements);
+    self->startup_phase = 0;
+  }
+
+  EstBufDelayExtended(self);
+
+  {
+    // |delay_diff_offset| gives us the option to manually rewind the delay on
+    // very low delay platforms which can't be expressed purely through
+    // |reported_delay_ms|.
+    const int adjusted_known_delay =
+        WEBRTC_SPL_MAX(0, self->knownDelay + delay_diff_offset);
+
+    for (i = 0; i < num_frames; ++i) {
+      WebRtcAec_ProcessFrame(self->aec, &near[FRAME_LEN * i],
+          &near_high[FRAME_LEN * i], adjusted_known_delay,
+          &out[FRAME_LEN * i], &out_high[FRAME_LEN * i]);
+    }
+  }
+}
+
+static void EstBufDelayNormal(aecpc_t* aecpc) {
   int nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->rate_factor;
   int current_delay = nSampSndCard - WebRtcAec_system_delay(aecpc->aec);
   int delay_difference = 0;
@@ -732,6 +853,9 @@ static int EstBufDelay(aecpc_t* aecpc) {
     current_delay += WebRtcAec_MoveFarReadPtr(aecpc->aec, 1) * PART_LEN;
   }
 
+  // We use -1 to signal an initialized state in the "extended" implementation;
+  // compensate for that.
+  aecpc->filtDelay = aecpc->filtDelay < 0 ? 0 : aecpc->filtDelay;
   aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short) (0.8 * aecpc->filtDelay +
       0.2 * current_delay));
 
@@ -756,6 +880,58 @@ static int EstBufDelay(aecpc_t* aecpc) {
   if (aecpc->timeForDelayChange > 25) {
     aecpc->knownDelay = WEBRTC_SPL_MAX((int) aecpc->filtDelay - 160, 0);
   }
-
+}
-  return 0;
+
+static void EstBufDelayExtended(aecpc_t* self) {
+  int reported_delay = self->msInSndCardBuf * sampMsNb * self->rate_factor;
+  int current_delay = reported_delay - WebRtcAec_system_delay(self->aec);
+  int delay_difference = 0;
+
+  // Before we proceed with the delay estimate filtering we:
+  // 1) Compensate for the frame that will be read.
+  // 2) Compensate for drift resampling.
+  // 3) Compensate for non-causality if needed, since the estimated delay can't
+  //    be negative.
+
+  // 1) Compensating for the frame(s) that will be read/processed.
+  current_delay += FRAME_LEN * self->rate_factor;
+
+  // 2) Account for resampling frame delay.
+  if (self->skewMode == kAecTrue && self->resample == kAecTrue) {
+    current_delay -= kResamplingDelay;
+  }
+
+  // 3) Compensate for non-causality, if needed, by flushing two blocks.
+  if (current_delay < PART_LEN) {
+    current_delay += WebRtcAec_MoveFarReadPtr(self->aec, 2) * PART_LEN;
+  }
+
+  if (self->filtDelay == -1) {
+    self->filtDelay = WEBRTC_SPL_MAX(0, 0.5 * current_delay);
+  } else {
+    self->filtDelay = WEBRTC_SPL_MAX(0, (short) (0.95 * self->filtDelay +
+        0.05 * current_delay));
+  }
+
+  delay_difference = self->filtDelay - self->knownDelay;
+  if (delay_difference > 384) {
+    if (self->lastDelayDiff < 128) {
+      self->timeForDelayChange = 0;
+    } else {
+      self->timeForDelayChange++;
+    }
+  } else if (delay_difference < 128 && self->knownDelay > 0) {
+    if (self->lastDelayDiff > 384) {
+      self->timeForDelayChange = 0;
+    } else {
+      self->timeForDelayChange++;
+    }
+  } else {
+    self->timeForDelayChange = 0;
+  }
+  self->lastDelayDiff = delay_difference;
+
+  if (self->timeForDelayChange > 25) {
+    self->knownDelay = WEBRTC_SPL_MAX((int) self->filtDelay - 256, 0);
+  }
 }

media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h

@@ -20,8 +20,6 @@ typedef struct {
   int splitSampFreq;
   int scSampFreq;
   float sampFactor;  // scSampRate / sampFreq
-  short autoOnOff;
-  short activity;
   short skewMode;
   int bufSizeStart;
   int knownDelay;
@@ -39,7 +37,7 @@ typedef struct {
   short msInSndCardBuf;
   short filtDelay;  // Filtered delay estimate.
   int timeForDelayChange;
-  int ECstartup;
+  int startup_phase;
   int checkBuffSize;
   short lastDelayDiff;
 
@@ -62,6 +60,8 @@ typedef struct {
 
   int lastError;
 
+  int farend_started;
+
   AecCore* aec;
 } aecpc_t;
 

media/webrtc/trunk/webrtc/modules/audio_processing/aec/system_delay_unittest.cc

@@ -128,7 +128,7 @@ void SystemDelayTest::RunStableStartup() {
   for (; process_time_ms < kStableConvergenceMs; process_time_ms += 10) {
     RenderAndCapture(kDeviceBufMs);
     buffer_size += samples_per_frame_;
-    if (self_->ECstartup == 0) {
+    if (self_->startup_phase == 0) {
       // We have left the startup phase.
       break;
     }
@@ -222,7 +222,7 @@ TEST_F(SystemDelayTest, CorrectDelayAfterUnstableStartup) {
       RenderAndCapture(reported_delay_ms);
       buffer_size += samples_per_frame_;
       buffer_offset_ms = -buffer_offset_ms;
-      if (self_->ECstartup == 0) {
+      if (self_->startup_phase == 0) {
         // We have left the startup phase.
         break;
       }
@@ -268,7 +268,7 @@ TEST_F(SystemDelayTest, CorrectDelayAfterStableBufferBuildUp) {
     for (; process_time_ms <= kMaxConvergenceMs; process_time_ms += 10) {
       RenderAndCapture(kDeviceBufMs);
       buffer_size += samples_per_frame_;
-      if (self_->ECstartup == 0) {
+      if (self_->startup_phase == 0) {
         // We have left the startup phase.
         break;
       }

media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing.gypi

@@ -155,6 +155,7 @@
             'aec/aec_rdft_sse2.c',
           ],
           'cflags': ['-msse2',],
+          'cflags_mozilla': [ '-msse2', ],
           'xcode_settings': {
             'OTHER_CFLAGS': ['-msse2',],
           },
@@ -178,11 +179,14 @@
             'dependencies': [
               'audio_processing_offsets',
             ],
-            'sources': [
+	    #
+	    # We disable the ASM source, because our gyp->Makefile translator
+	    # does not support the build steps to get the asm offsets.
+            'sources!': [
               'aecm/aecm_core_neon.S',
               'ns/nsx_core_neon.S',
             ],
-            'sources!': [
+            'sources': [
               'aecm/aecm_core_neon.c',
               'ns/nsx_core_neon.c',
             ],

media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.cc

@@ -13,12 +13,14 @@
 #include <assert.h>
 #include <string.h>
 
+extern "C" {
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
+}
+#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
-
 namespace webrtc {
 
 typedef void Handle;
@@ -69,7 +71,8 @@ EchoCancellationImpl::EchoCancellationImpl(const AudioProcessingImpl* apm)
     stream_drift_samples_(0),
     was_stream_drift_set_(false),
     stream_has_echo_(false),
-    delay_logging_enabled_(false) {}
+    delay_logging_enabled_(false),
+    delay_correction_enabled_(true) {}
 
 EchoCancellationImpl::~EchoCancellationImpl() {}
 
@@ -338,6 +341,13 @@ int EchoCancellationImpl::Initialize() {
   return apm_->kNoError;
 }
 
+#if 0
+void EchoCancellationImpl::SetExtraOptions(const Config& config) {
+  delay_correction_enabled_ = config.Get<DelayCorrection>().enabled;
+  Configure();
+}
+#endif
+
 void* EchoCancellationImpl::CreateHandle() const {
   Handle* handle = NULL;
   if (WebRtcAec_Create(&handle) != apm_->kNoError) {
@@ -369,6 +379,8 @@ int EchoCancellationImpl::ConfigureHandle(void* handle) const {
   config.skewMode = drift_compensation_enabled_;
   config.delay_logging = delay_logging_enabled_;
 
+  WebRtcAec_enable_delay_correction(WebRtcAec_aec_core(
+      static_cast<Handle*>(handle)), delay_correction_enabled_ ? 1 : 0);
   return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
 }
 

media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl.h

@@ -14,6 +14,29 @@
 #include "webrtc/modules/audio_processing/echo_cancellation_impl_wrapper.h"
 
 namespace webrtc {
+// Use to enable the delay correction feature. This now engages an extended
+// filter mode in the AEC, along with robustness measures around the reported
+// system delays. It comes with a significant increase in AEC complexity, but is
+// much more robust to unreliable reported delays.
+//
+// Detailed changes to the algorithm:
+// - The filter length is changed from 48 to 128 ms. This comes with tuning of
+//   several parameters: i) filter adaptation stepsize and error threshold;
+//   ii) non-linear processing smoothing and overdrive.
+// - Option to ignore the reported delays on platforms which we deem
+//   sufficiently unreliable. See WEBRTC_UNTRUSTED_DELAY in echo_cancellation.c.
+// - Faster startup times by removing the excessive "startup phase" processing
+//   of reported delays.
+// - Much more conservative adjustments to the far-end read pointer. We smooth
+//   the delay difference more heavily, and back off from the difference more.
+//   Adjustments force a readaptation of the filter, so they should be avoided
+//   except when really necessary.
+struct DelayCorrection {
+  DelayCorrection() : enabled(false) {}
+  DelayCorrection(bool enabled) : enabled(enabled) {}
+
+  bool enabled;
+};
 
 class AudioProcessingImpl;
 class AudioBuffer;
@@ -34,6 +57,7 @@ class EchoCancellationImpl : public EchoCancellationImplWrapper {
 
   // ProcessingComponent implementation.
   virtual int Initialize() OVERRIDE;
+  //  virtual void SetExtraOptions(const Config& config) OVERRIDE;
 
  private:
   // EchoCancellation implementation.
@@ -70,6 +94,7 @@ class EchoCancellationImpl : public EchoCancellationImplWrapper {
   bool was_stream_drift_set_;
   bool stream_has_echo_;
   bool delay_logging_enabled_;
+  bool delay_correction_enabled_;
 };
 
 }  // namespace webrtc

media/webrtc/trunk/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc

@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2013 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 "testing/gtest/include/gtest/gtest.h"
+extern "C" {
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
+}
+#include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+TEST(EchoCancellationInternalTest, DelayCorrection) {
+  scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(0));
+  EXPECT_TRUE(ap->echo_cancellation()->aec_core() == NULL);
+
+  EXPECT_EQ(ap->kNoError, ap->echo_cancellation()->Enable(true));
+  EXPECT_TRUE(ap->echo_cancellation()->is_enabled());
+
+  AecCore* aec_core = ap->echo_cancellation()->aec_core();
+  ASSERT_TRUE(aec_core != NULL);
+  // Disabled by default.
+  EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core));
+
+  Config config;
+  config.Set<DelayCorrection>(new DelayCorrection(true));
+  ap->SetExtraOptions(config);
+  EXPECT_EQ(1, WebRtcAec_delay_correction_enabled(aec_core));
+
+  // Retains setting after initialization.
+  EXPECT_EQ(ap->kNoError, ap->Initialize());
+  EXPECT_EQ(1, WebRtcAec_delay_correction_enabled(aec_core));
+
+  config.Set<DelayCorrection>(new DelayCorrection(false));
+  ap->SetExtraOptions(config);
+  EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core));
+
+  // Retains setting after initialization.
+  EXPECT_EQ(ap->kNoError, ap->Initialize());
+  EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core));
+}
+
+}  // namespace webrtc

media/webrtc/trunk/webrtc/modules/media_file/source/media_file_utility.cc

@@ -609,13 +609,13 @@ int32_t ModuleFileUtility::ReadWavHeader(InStream& wav)
         //                 special cases?
         if(_wavFormatObj.nSamplesPerSec == 44100)
         {
-            _readSizeBytes = 440 * _wavFormatObj.nChannels *
+            _readSizeBytes = 441 * _wavFormatObj.nChannels *
                 (_wavFormatObj.nBitsPerSample / 8);
         } else if(_wavFormatObj.nSamplesPerSec == 22050) {
-            _readSizeBytes = 220 * _wavFormatObj.nChannels *
+            _readSizeBytes = 220 * _wavFormatObj.nChannels * // XXX inexact!
                 (_wavFormatObj.nBitsPerSample / 8);
         } else if(_wavFormatObj.nSamplesPerSec == 11025) {
-            _readSizeBytes = 110 * _wavFormatObj.nChannels *
+            _readSizeBytes = 110 * _wavFormatObj.nChannels * // XXX inexact!
                 (_wavFormatObj.nBitsPerSample / 8);
         } else {
             _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
@@ -677,22 +677,22 @@ int32_t ModuleFileUtility::InitWavCodec(uint32_t samplesPerSec,
         {
             strcpy(codec_info_.plname, "L16");
             _codecId = kCodecL16_16kHz;
-            codec_info_.pacsize = 110;
+            codec_info_.pacsize = 110; // XXX inexact!
-            codec_info_.plfreq = 11000;
+            codec_info_.plfreq = 11000; // XXX inexact!
         }
         else if(samplesPerSec == 22050)
         {
             strcpy(codec_info_.plname, "L16");
             _codecId = kCodecL16_16kHz;
-            codec_info_.pacsize = 220;
+            codec_info_.pacsize = 220; // XXX inexact!
-            codec_info_.plfreq = 22000;
+            codec_info_.plfreq = 22000; // XXX inexact!
         }
         else if(samplesPerSec == 44100)
         {
             strcpy(codec_info_.plname, "L16");
             _codecId = kCodecL16_16kHz;
-            codec_info_.pacsize = 440;
+            codec_info_.pacsize = 441;
-            codec_info_.plfreq = 44000;
+            codec_info_.plfreq = 44100;
         }
         else if(samplesPerSec == 48000)
         {
@@ -1125,8 +1125,6 @@ int32_t ModuleFileUtility::WriteWavHeader(
 {
 
     // Frame size in bytes for 10 ms of audio.
-    // TODO (hellner): 44.1 kHz has 440 samples frame size. Doesn't seem to
-    //                 be taken into consideration here!
     int32_t frameSize = (freq / 100) * bytesPerSample * channels;
 
     // Calculate the number of full frames that the wave file contain.

media/webrtc/trunk/webrtc/modules/modules.gyp

@@ -145,6 +145,7 @@
             'audio_coding/neteq4/mock/mock_payload_splitter.h',
             'audio_processing/aec/system_delay_unittest.cc',
             'audio_processing/aec/echo_cancellation_unittest.cc',
+            'audio_processing/echo_cancellation_impl_unittest.cc',
             'audio_processing/test/audio_processing_unittest.cc',
             'audio_processing/utility/delay_estimator_unittest.cc',
             'audio_processing/utility/ring_buffer_unittest.cc',

media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc

@@ -11,6 +11,7 @@
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 
 #include <assert.h>
+#include <cstdlib> // for abs()
 #include <string.h>
 
 #include <algorithm>

media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.cc

@@ -21,7 +21,7 @@
 #include <WinSock.h>  // timeval
 
 #include <MMSystem.h>  // timeGetTime
-#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
+#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_BSD) || (defined WEBRTC_MAC))
 #include <sys/time.h>  // gettimeofday
 #include <time.h>
 #endif
@@ -118,7 +118,7 @@ bool StringCompare(const char* str1, const char* str2,
                    const uint32_t length) {
   return (_strnicmp(str1, str2, length) == 0) ? true : false;
 }
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
 bool StringCompare(const char* str1, const char* str2,
                    const uint32_t length) {
   return (strncasecmp(str1, str2, length) == 0) ? true : false;

media/webrtc/trunk/webrtc/modules/utility/source/file_player_impl.cc

@@ -87,7 +87,7 @@ int32_t FilePlayerImpl::Frequency() const
     {
         return 32000;
     }
-    else if(_codec.plfreq == 44000)
+    else if(_codec.plfreq == 44100 || _codec.plfreq == 44000 ) // XXX just 44100?
     {
         return 32000;
     }

media/webrtc/trunk/webrtc/modules/utility/source/rtp_dump_impl.cc

@@ -19,7 +19,7 @@
 #if defined(_WIN32)
 #include <Windows.h>
 #include <mmsystem.h>
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC) || defined(WEBRTC_BSD)
 #include <string.h>
 #include <sys/time.h>
 #include <time.h>
@@ -237,7 +237,7 @@ inline uint32_t RtpDumpImpl::GetTimeInMS() const
 {
 #if defined(_WIN32)
     return timeGetTime();
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD) || defined(WEBRTC_MAC)
     struct timeval tv;
     struct timezone tz;
     unsigned long val;

media/webrtc/trunk/webrtc/modules/video_capture/android/device_info_android.cc

@@ -16,6 +16,8 @@
 #include "webrtc/system_wrappers/interface/ref_count.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
+#include "AndroidJNIWrapper.h"
+
 namespace webrtc
 {
 
@@ -52,17 +54,14 @@ DeviceInfoAndroid::~DeviceInfoAndroid() {
 }
 
 uint32_t DeviceInfoAndroid::NumberOfDevices() {
-  JNIEnv *env;
+  AutoLocalJNIFrame jniFrame;
-  jclass javaCmDevInfoClass;
+  JNIEnv* env = jniFrame.GetEnv();
-  jobject javaCmDevInfoObject;
+  if (!env)
-  bool attached = false;
-  if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
-          env,
-          javaCmDevInfoClass,
-          javaCmDevInfoObject,
-          attached) != 0)
       return 0;
 
+  jclass javaCmDevInfoClass = jniFrame.GetCmDevInfoClass();
+  jobject javaCmDevInfoObject = jniFrame.GetCmDevInfoObject();
+
   WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
                "%s GetMethodId", __FUNCTION__);
   // get the method ID for the Android Java GetDeviceUniqueName name.
@@ -76,7 +75,6 @@ uint32_t DeviceInfoAndroid::NumberOfDevices() {
                  "%s Calling Number of devices", __FUNCTION__);
     numberOfDevices = env->CallIntMethod(javaCmDevInfoObject, cid);
   }
-  VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
 
   if (numberOfDevices > 0)
     return numberOfDevices;
@@ -92,18 +90,15 @@ int32_t DeviceInfoAndroid::GetDeviceName(
     char* /*productUniqueIdUTF8*/,
     uint32_t /*productUniqueIdUTF8Length*/) {
 
-  JNIEnv *env;
-  jclass javaCmDevInfoClass;
-  jobject javaCmDevInfoObject;
   int32_t result = 0;
-  bool attached = false;
+  AutoLocalJNIFrame jniFrame;
-  if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
+  JNIEnv* env = jniFrame.GetEnv();
-          env,
+  if (!env)
-          javaCmDevInfoClass,
-          javaCmDevInfoObject,
-          attached)!= 0)
       return -1;
 
+  jclass javaCmDevInfoClass = jniFrame.GetCmDevInfoClass();
+  jobject javaCmDevInfoObject = jniFrame.GetCmDevInfoObject();
+
   // get the method ID for the Android Java GetDeviceUniqueName name.
   jmethodID cid = env->GetMethodID(javaCmDevInfoClass, "GetDeviceUniqueName",
                                    "(I)Ljava/lang/String;");
@@ -151,8 +146,6 @@ int32_t DeviceInfoAndroid::GetDeviceName(
     result = -1;
   }
 
-  VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
-
   WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
                "%s: result %d", __FUNCTION__, (int) result);
   return result;
@@ -168,21 +161,17 @@ int32_t DeviceInfoAndroid::CreateCapabilityMap(
     delete it->second;
   _captureCapabilities.clear();
 
-  JNIEnv *env;
+  AutoLocalJNIFrame jniFrame;
-  jclass javaCmDevInfoClass;
+  JNIEnv* env = jniFrame.GetEnv();
-  jobject javaCmDevInfoObject;
+  if (!env)
-  bool attached = false;
-  if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
-          env,
-          javaCmDevInfoClass,
-          javaCmDevInfoObject,
-          attached) != 0)
       return -1;
 
+  jclass javaCmDevInfoClass = jniFrame.GetCmDevInfoClass();
+  jobject javaCmDevInfoObject = jniFrame.GetCmDevInfoObject();
+
   // Find the capability class
-  jclass javaCapClass = g_capabilityClass;
+  jclass javaCapClass = jsjni_GetGlobalClassRef(AndroidJavaCaptureCapabilityClass);
   if (javaCapClass == NULL) {
-    VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
                  "%s: SetAndroidCaptureClasses must be called first!",
                  __FUNCTION__);
@@ -195,7 +184,6 @@ int32_t DeviceInfoAndroid::CreateCapabilityMap(
       "GetCapabilityArray",
       "(Ljava/lang/String;)[Lorg/webrtc/videoengine/CaptureCapabilityAndroid;");
   if (cid == NULL) {
-    VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
                  "%s: Can't find method GetCapabilityArray.", __FUNCTION__);
     return -1;
@@ -204,7 +192,6 @@ int32_t DeviceInfoAndroid::CreateCapabilityMap(
   jstring capureIdString = env->NewStringUTF((char*) deviceUniqueIdUTF8);
 
   if (capureIdString == NULL) {
-    VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
                  "%s: Can't create string for  method GetCapabilityArray.",
                  __FUNCTION__);
@@ -214,7 +201,6 @@ int32_t DeviceInfoAndroid::CreateCapabilityMap(
   jobject javaCapabilitiesObj = env->CallObjectMethod(javaCmDevInfoObject,
                                                       cid, capureIdString);
   if (!javaCapabilitiesObj) {
-    VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
                  "%s: Failed to call java GetCapabilityArray.",
                  __FUNCTION__);
@@ -225,7 +211,6 @@ int32_t DeviceInfoAndroid::CreateCapabilityMap(
   jfieldID heigtField = env->GetFieldID(javaCapClass, "height", "I");
   jfieldID maxFpsField = env->GetFieldID(javaCapClass, "maxFPS", "I");
   if (widthField == NULL || heigtField == NULL || maxFpsField == NULL) {
-    VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
                  "%s: Failed to get field Id.", __FUNCTION__);
     return -1;
@@ -258,7 +243,8 @@ int32_t DeviceInfoAndroid::CreateCapabilityMap(
          deviceUniqueIdUTF8,
          _lastUsedDeviceNameLength + 1);
 
-  VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
+  env->DeleteGlobalRef(javaCapClass);
+
   WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
                "CreateCapabilityMap %d", _captureCapabilities.size());
 
@@ -268,22 +254,18 @@ int32_t DeviceInfoAndroid::CreateCapabilityMap(
 int32_t DeviceInfoAndroid::GetOrientation(
     const char* deviceUniqueIdUTF8,
     VideoCaptureRotation& orientation) {
-  JNIEnv *env;
+  AutoLocalJNIFrame jniFrame;
-  jclass javaCmDevInfoClass;
+  JNIEnv* env = jniFrame.GetEnv();
-  jobject javaCmDevInfoObject;
+  if (!env)
-  bool attached = false;
-  if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
-          env,
-          javaCmDevInfoClass,
-          javaCmDevInfoObject,
-          attached) != 0)
       return -1;
 
+  jclass javaCmDevInfoClass = jniFrame.GetCmDevInfoClass();
+  jobject javaCmDevInfoObject = jniFrame.GetCmDevInfoObject();
+
   // get the method ID for the Android Java GetOrientation .
   jmethodID cid = env->GetMethodID(javaCmDevInfoClass, "GetOrientation",
                                    "(Ljava/lang/String;)I");
   if (cid == NULL) {
-    VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
                  "%s: Can't find method GetOrientation.", __FUNCTION__);
     return -1;
@@ -291,7 +273,6 @@ int32_t DeviceInfoAndroid::GetOrientation(
   // Create a jstring so we can pass the deviceUniquName to the java method.
   jstring capureIdString = env->NewStringUTF((char*) deviceUniqueIdUTF8);
   if (capureIdString == NULL) {
-    VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
                  "%s: Can't create string for  method GetCapabilityArray.",
                  __FUNCTION__);
@@ -300,7 +281,6 @@ int32_t DeviceInfoAndroid::GetOrientation(
   // Call the java class and get the orientation.
   jint jorientation = env->CallIntMethod(javaCmDevInfoObject, cid,
                                          capureIdString);
-  VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
 
   int32_t retValue = 0;
   switch (jorientation) {

media/webrtc/trunk/webrtc/modules/video_capture/android/device_info_android.h

@@ -16,6 +16,9 @@
 #include "webrtc/modules/video_capture/device_info_impl.h"
 #include "webrtc/modules/video_capture/video_capture_impl.h"
 
+#define AndroidJavaCaptureDeviceInfoClass "org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid"
+#define AndroidJavaCaptureCapabilityClass "org/webrtc/videoengine/CaptureCapabilityAndroid"
+
 namespace webrtc
 {
 namespace videocapturemodule

media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java

@@ -25,14 +25,25 @@ import android.graphics.YuvImage;
 import android.hardware.Camera;
 import android.hardware.Camera.PreviewCallback;
 import android.util.Log;
+import android.view.Surface;
 import android.view.SurfaceHolder;
 import android.view.SurfaceHolder.Callback;
+import android.view.SurfaceView;
+import android.view.TextureView;
+import android.view.TextureView.SurfaceTextureListener;
+import android.view.View;
+
+import org.mozilla.gecko.GeckoApp;
+import org.mozilla.gecko.GeckoAppShell;
+import org.mozilla.gecko.GeckoAppShell.AppStateListener;
+import org.mozilla.gecko.util.ThreadUtils;
 
 public class VideoCaptureAndroid implements PreviewCallback, Callback {
 
     private final static String TAG = "WEBRTC-JC";
 
     private Camera camera;
+    private int cameraId;
     private AndroidVideoCaptureDevice currentDevice = null;
     public ReentrantLock previewBufferLock = new ReentrantLock();
     // This lock takes sync with StartCapture and SurfaceChanged
@@ -43,6 +54,9 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
     private boolean isCaptureStarted = false;
     private boolean isCaptureRunning = false;
     private boolean isSurfaceReady = false;
+    private SurfaceHolder surfaceHolder = null;
+    private SurfaceTexture surfaceTexture = null;
+    private SurfaceTexture dummySurfaceTexture = null;
 
     private final int numCaptureBuffers = 3;
     private int expectedFrameSize = 0;
@@ -51,7 +65,6 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
     // C++ callback context variable.
     private long context = 0;
     private SurfaceHolder localPreview = null;
-    private SurfaceTexture dummySurfaceTexture = null;
     // True if this class owns the preview video buffers.
     private boolean ownsBuffers = false;
 
@@ -59,25 +72,171 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
     private int mCaptureHeight = -1;
     private int mCaptureFPS = -1;
 
+    private int mCaptureRotation = 0;
+
+    private AppStateListener mAppStateListener = null;
+
+    public class MySurfaceTextureListener implements TextureView.SurfaceTextureListener {
+        public void onSurfaceTextureAvailable(SurfaceTexture surface, int width, int height) {
+            Log.d(TAG, "VideoCaptureAndroid::onSurfaceTextureAvailable");
+
+            captureLock.lock();
+            isSurfaceReady = true;
+            surfaceTexture = surface;
+
+            tryStartCapture(mCaptureWidth, mCaptureHeight, mCaptureFPS);
+            captureLock.unlock();
+        }
+
+        public void onSurfaceTextureSizeChanged(SurfaceTexture surface,
+                                                int width, int height) {
+            // Ignored, Camera does all the work for us
+            // Note that for a TextureView we start on onSurfaceTextureAvailable,
+            // for a SurfaceView we start on surfaceChanged. TextureView
+            // will not give out an onSurfaceTextureSizeChanged during creation.
+        }
+
+        public boolean onSurfaceTextureDestroyed(SurfaceTexture surface) {
+            Log.d(TAG, "VideoCaptureAndroid::onSurfaceTextureDestroyed");
+            isSurfaceReady = false;
+            DetachCamera();
+            return true;
+        }
+
+        public void onSurfaceTextureUpdated(SurfaceTexture surface) {
+            // Invoked every time there's a new Camera preview frame
+        }
+    }
     public static
     void DeleteVideoCaptureAndroid(VideoCaptureAndroid captureAndroid) {
         Log.d(TAG, "DeleteVideoCaptureAndroid");
-        if (captureAndroid.camera == null) {
+
-            return;
+        GeckoAppShell.getGeckoInterface().removeAppStateListener(captureAndroid.mAppStateListener);
-        }
 
         captureAndroid.StopCapture();
+        if (captureAndroid.camera != null) {
             captureAndroid.camera.release();
             captureAndroid.camera = null;
+        }
         captureAndroid.context = 0;
+
+        View cameraView = GeckoAppShell.getGeckoInterface().getCameraView();
+        if (cameraView instanceof SurfaceView) {
+            ((SurfaceView)cameraView).getHolder().removeCallback(captureAndroid);
+        } else if (cameraView instanceof TextureView) {
+            // No need to explicitly remove the Listener:
+            // i.e. ((SurfaceView)cameraView).setSurfaceTextureListener(null);
+        }
+        ThreadUtils.getUiHandler().post(new Runnable() {
+            @Override
+            public void run() {
+                try {
+                    GeckoAppShell.getGeckoInterface().disableCameraView();
+                } catch (Exception e) {
+                    Log.e(TAG,
+                          "VideoCaptureAndroid disableCameraView exception: " +
+                          e.getLocalizedMessage());
+                }
+           }
+        });
     }
 
     public VideoCaptureAndroid(int in_id, long in_context, Camera in_camera,
-            AndroidVideoCaptureDevice in_device) {
+                               AndroidVideoCaptureDevice in_device,
+                               int in_cameraId) {
         id = in_id;
         context = in_context;
         camera = in_camera;
+        cameraId = in_cameraId;
         currentDevice = in_device;
+        mCaptureRotation = GetRotateAmount();
+
+        try {
+            View cameraView = GeckoAppShell.getGeckoInterface().getCameraView();
+            if (cameraView instanceof SurfaceView) {
+                ((SurfaceView)cameraView).getHolder().addCallback(this);
+            } else if (cameraView instanceof TextureView) {
+                MySurfaceTextureListener listener = new MySurfaceTextureListener();
+                ((TextureView)cameraView).setSurfaceTextureListener(listener);
+            }
+            ThreadUtils.getUiHandler().post(new Runnable() {
+                @Override
+                public void run() {
+                    try {
+                        GeckoAppShell.getGeckoInterface().enableCameraView();
+                    } catch (Exception e) {
+                        Log.e(TAG,
+                              "VideoCaptureAndroid enableCameraView exception: "
+                               + e.getLocalizedMessage());
+                    }
+                }
+            });
+        } catch (Exception ex) {
+            Log.e(TAG, "VideoCaptureAndroid constructor exception: " +
+                  ex.getLocalizedMessage());
+        }
+
+        mAppStateListener = new AppStateListener() {
+            @Override
+            public void onPause() {
+                StopCapture();
+                if (camera != null) {
+                    camera.release();
+                   camera = null;
+                }
+            }
+            @Override
+            public void onResume() {
+                try {
+                    if(android.os.Build.VERSION.SDK_INT>8) {
+                        camera = Camera.open(cameraId);
+                    } else {
+                        camera = Camera.open();
+                    }
+                } catch (Exception ex) {
+                    Log.e(TAG, "Error reopening to the camera: " + ex.getMessage());
+                }
+                captureLock.lock();
+                isCaptureStarted = true;
+                tryStartCapture(mCaptureWidth, mCaptureHeight, mCaptureFPS);
+                captureLock.unlock();
+            }
+            @Override
+            public void onOrientationChanged() {
+                mCaptureRotation = GetRotateAmount();
+            }
+        };
+
+        GeckoAppShell.getGeckoInterface().addAppStateListener(mAppStateListener);
+    }
+
+    public int GetRotateAmount() {
+        int rotation = GeckoAppShell.getGeckoInterface().getActivity().getWindowManager().getDefaultDisplay().getRotation();
+        int degrees = 0;
+        switch (rotation) {
+            case Surface.ROTATION_0: degrees = 0; break;
+            case Surface.ROTATION_90: degrees = 90; break;
+            case Surface.ROTATION_180: degrees = 180; break;
+            case Surface.ROTATION_270: degrees = 270; break;
+        }
+        if(android.os.Build.VERSION.SDK_INT>8) {
+            android.hardware.Camera.CameraInfo info =
+                new android.hardware.Camera.CameraInfo();
+            android.hardware.Camera.getCameraInfo(cameraId, info);
+            int result;
+            if (info.facing == Camera.CameraInfo.CAMERA_FACING_FRONT) {
+                result = (info.orientation + degrees) % 360;
+            } else {  // back-facing
+                result = (info.orientation - degrees + 360) % 360;
+            }
+            return result;
+        } else {
+            // Assume 90deg orientation for Froyo devices.
+            // Only back-facing cameras are supported in Froyo.
+            int orientation = 90;
+            int result = (orientation - degrees + 360) % 360;
+            return result;
+        }
     }
 
     private int tryStartCapture(int width, int height, int frameRate) {
@@ -86,16 +245,34 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
             return -1;
         }
 
-        Log.d(TAG, "tryStartCapture: " + width +
+        Log.d(TAG, "tryStartCapture " + width +
-            "x" + height +", frameRate: " + frameRate +
+                " height " + height +" frame rate " + frameRate +
-            ", isCaptureRunning: " + isCaptureRunning +
+                " isCaptureRunning " + isCaptureRunning +
-            ", isSurfaceReady: " + isSurfaceReady +
+                " isSurfaceReady " + isSurfaceReady +
-            ", isCaptureStarted: " + isCaptureStarted);
+                " isCaptureStarted " + isCaptureStarted);
 
-        if (isCaptureRunning || !isCaptureStarted) {
+        if (isCaptureRunning || !isSurfaceReady || !isCaptureStarted) {
             return 0;
         }
 
+        try {
+            if (surfaceHolder != null)
+                camera.setPreviewDisplay(surfaceHolder);
+            if (surfaceTexture != null)
+                camera.setPreviewTexture(surfaceTexture);
+            if (surfaceHolder == null && surfaceTexture == null) {
+                // No local renderer.  Camera won't capture without
+                // setPreview{Texture,Display}, so we create a dummy SurfaceTexture
+                // and hand it over to Camera, but never listen for frame-ready
+                // callbacks, and never call updateTexImage on it.
+                try {
+                    dummySurfaceTexture = new SurfaceTexture(42);
+                    camera.setPreviewTexture(dummySurfaceTexture);
+                } catch (IOException e) {
+                    throw new RuntimeException(e);
+                }
+            }
+
             CaptureCapabilityAndroid currentCapability =
                     new CaptureCapabilityAndroid();
             currentCapability.width = width;
@@ -103,17 +280,13 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
             currentCapability.maxFPS = frameRate;
             PixelFormat.getPixelFormatInfo(PIXEL_FORMAT, pixelFormat);
 
+
             Camera.Parameters parameters = camera.getParameters();
             parameters.setPreviewSize(currentCapability.width,
                     currentCapability.height);
             parameters.setPreviewFormat(PIXEL_FORMAT);
             parameters.setPreviewFrameRate(currentCapability.maxFPS);
-        try {
             camera.setParameters(parameters);
-        } catch (RuntimeException e) {
-            Log.e(TAG, "setParameters failed", e);
-            return -1;
-        }
 
             int bufSize = width * height * pixelFormat.bitsPerPixel / 8;
             byte[] buffer = null;
@@ -130,35 +303,19 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
             isCaptureRunning = true;
             previewBufferLock.unlock();
 
+        }
+        catch (Exception ex) {
+            Log.e(TAG, "Failed to start camera: " + ex.getMessage());
+            return -1;
+        }
+
+        isCaptureRunning = true;
         return 0;
     }
 
     public int StartCapture(int width, int height, int frameRate) {
         Log.d(TAG, "StartCapture width " + width +
                 " height " + height +" frame rate " + frameRate);
-        // Get the local preview SurfaceHolder from the static render class
-        localPreview = ViERenderer.GetLocalRenderer();
-        if (localPreview != null) {
-            if (localPreview.getSurface() != null &&
-                localPreview.getSurface().isValid()) {
-                surfaceCreated(localPreview);
-            }
-            localPreview.addCallback(this);
-        } else {
-          // No local renderer.  Camera won't capture without
-          // setPreview{Texture,Display}, so we create a dummy SurfaceTexture
-          // and hand it over to Camera, but never listen for frame-ready
-          // callbacks, and never call updateTexImage on it.
-          captureLock.lock();
-          try {
-            dummySurfaceTexture = new SurfaceTexture(42);
-            camera.setPreviewTexture(dummySurfaceTexture);
-          } catch (IOException e) {
-            throw new RuntimeException(e);
-          }
-          captureLock.unlock();
-        }
-
         captureLock.lock();
         isCaptureStarted = true;
         mCaptureWidth = width;
@@ -171,36 +328,48 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
         return res;
     }
 
-    public int StopCapture() {
+    public int DetachCamera() {
-        Log.d(TAG, "StopCapture");
         try {
             previewBufferLock.lock();
             isCaptureRunning = false;
             previewBufferLock.unlock();
-            camera.stopPreview();
+            if (camera != null) {
                 camera.setPreviewCallbackWithBuffer(null);
-        } catch (RuntimeException e) {
+                camera.stopPreview();
-            Log.e(TAG, "Failed to stop camera", e);
+            }
+        } catch (Exception ex) {
+            Log.e(TAG, "Failed to stop camera: " + ex.getMessage());
             return -1;
         }
-
-        isCaptureStarted = false;
         return 0;
     }
 
-    native void ProvideCameraFrame(byte[] data, int length, long captureObject);
+    public int StopCapture() {
+        Log.d(TAG, "StopCapture");
+        isCaptureStarted = false;
+        return DetachCamera();
+    }
+
+    native void ProvideCameraFrame(byte[] data, int length, int rotation,
+                                   long captureObject);
 
     public void onPreviewFrame(byte[] data, Camera camera) {
         previewBufferLock.lock();
 
+        String dataLengthStr = "does not exist";
+        if(data != null) {
+          dataLengthStr = Integer.toString(data.length);
+        }
+
         // The following line is for debug only
-        // Log.v(TAG, "preview frame length " + data.length +
+        Log.v(TAG, "preview frame length " + data.length +
-        //            " context" + context);
+              " context" + context);
         if (isCaptureRunning) {
             // If StartCapture has been called but not StopCapture
             // Call the C++ layer with the captured frame
-            if (data.length == expectedFrameSize) {
+            if (data != null && data.length == expectedFrameSize) {
-                ProvideCameraFrame(data, expectedFrameSize, context);
+                ProvideCameraFrame(data, expectedFrameSize, mCaptureRotation,
+                                   context);
                 if (ownsBuffers) {
                     // Give the video buffer to the camera service again.
                     camera.addCallbackBuffer(data);
@@ -210,58 +379,26 @@ public class VideoCaptureAndroid implements PreviewCallback, Callback {
         previewBufferLock.unlock();
     }
 
-    // Sets the rotation of the preview render window.
-    // Does not affect the captured video image.
-    public void SetPreviewRotation(int rotation) {
-        Log.v(TAG, "SetPreviewRotation:" + rotation);
-
-        if (camera == null) {
-            return;
-        }
-
-        int resultRotation = 0;
-        if (currentDevice.frontCameraType ==
-            VideoCaptureDeviceInfoAndroid.FrontFacingCameraType.Android23) {
-            // this is a 2.3 or later front facing camera.
-            // SetDisplayOrientation will flip the image horizontally
-            // before doing the rotation.
-            resultRotation = ( 360 - rotation ) % 360; // compensate the mirror
-        }
-        else {
-            // Back facing or 2.2 or previous front camera
-            resultRotation = rotation;
-        }
-        camera.setDisplayOrientation(resultRotation);
-    }
-
     public void surfaceChanged(SurfaceHolder holder,
                                int format, int width, int height) {
         Log.d(TAG, "VideoCaptureAndroid::surfaceChanged");
+
+        captureLock.lock();
+        isSurfaceReady = true;
+        surfaceHolder = holder;
+
+        tryStartCapture(mCaptureWidth, mCaptureHeight, mCaptureFPS);
+        captureLock.unlock();
+        return;
     }
 
     public void surfaceCreated(SurfaceHolder holder) {
         Log.d(TAG, "VideoCaptureAndroid::surfaceCreated");
-        captureLock.lock();
-        try {
-          if (camera != null) {
-              camera.setPreviewDisplay(holder);
-          }
-        } catch (IOException e) {
-            Log.e(TAG, "Failed to set preview surface!", e);
-        }
-        captureLock.unlock();
     }
 
     public void surfaceDestroyed(SurfaceHolder holder) {
         Log.d(TAG, "VideoCaptureAndroid::surfaceDestroyed");
-        captureLock.lock();
+        isSurfaceReady = false;
-        try {
+        DetachCamera();
-            if (camera != null) {
-                camera.setPreviewDisplay(null);
-            }
-        } catch (IOException e) {
-            Log.e(TAG, "Failed to clear preview surface!", e);
-        }
-        captureLock.unlock();
     }
 }

media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java

@@ -252,6 +252,7 @@ public class VideoCaptureDeviceInfoAndroid {
             Log.d(TAG, "AllocateCamera " + deviceUniqueId);
 
             Camera camera = null;
+            int cameraId = 0;
             AndroidVideoCaptureDevice deviceToUse = null;
             for (AndroidVideoCaptureDevice device: deviceList) {
                 if(device.deviceUniqueName.equals(deviceUniqueId)) {
@@ -266,10 +267,12 @@ public class VideoCaptureDeviceInfoAndroid {
                             break;
                         default:
                             // From Android 2.3 and onwards)
-                            if(android.os.Build.VERSION.SDK_INT>8)
+                            if(android.os.Build.VERSION.SDK_INT>8) {
+                                cameraId = device.index;
                                 camera = Camera.open(device.index);
-                            else
+                            } else {
-                                camera=Camera.open(); // Default camera
+                                camera = Camera.open(); // Default_ camera
+                            }
                     }
                 }
             }
@@ -279,7 +282,7 @@ public class VideoCaptureDeviceInfoAndroid {
             }
             Log.v(TAG, "AllocateCamera - creating VideoCaptureAndroid");
 
-            return new VideoCaptureAndroid(id, context, camera, deviceToUse);
+            return new VideoCaptureAndroid(id, context, camera, deviceToUse, cameraId);
         } catch (NoSuchMethodException e) {
             Log.e(TAG, "AllocateCamera Failed to open camera", e);
         } catch (ClassNotFoundException e) {

media/webrtc/trunk/webrtc/modules/video_capture/android/video_capture_android.cc

@@ -16,6 +16,9 @@
 #include "webrtc/system_wrappers/interface/ref_count.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
+#include "AndroidJNIWrapper.h"
+#include "mozilla/Assertions.h"
+
 namespace webrtc
 {
 #if defined(WEBRTC_ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
@@ -45,12 +48,19 @@ VideoCaptureModule* VideoCaptureImpl::Create(
   return implementation;
 }
 
+#ifdef DEBUG
 // Android logging, uncomment to print trace to
 // logcat instead of trace file/callback
-// #include <android/log.h>
+#include <android/log.h>
 // #undef WEBRTC_TRACE
 // #define WEBRTC_TRACE(a,b,c,...)
 // __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
+// Some functions are called before before the WebRTC logging can be brought up,
+// log those to the Android log.
+#define EARLY_WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTC-VCA", __VA_ARGS__)
+#else
+#define EARLY_WEBRTC_TRACE(a,b,c,...)
+#endif
 
 JavaVM* VideoCaptureAndroid::g_jvm = NULL;
 //VideoCaptureAndroid.java
@@ -59,7 +69,6 @@ jclass VideoCaptureAndroid::g_javaCmClass = NULL;
 jclass VideoCaptureAndroid::g_javaCmDevInfoClass = NULL;
 //static instance of VideoCaptureDeviceInfoAndroid.java
 jobject VideoCaptureAndroid::g_javaCmDevInfoObject = NULL;
-jobject VideoCaptureAndroid::g_javaContext = NULL;
 
 /*
  * Register references to Java Capture class.
@@ -67,87 +76,59 @@ jobject VideoCaptureAndroid::g_javaContext = NULL;
 int32_t VideoCaptureAndroid::SetAndroidObjects(void* javaVM,
                                                void* javaContext) {
 
+  MOZ_ASSERT(javaVM != nullptr || g_javaCmDevInfoClass != nullptr);
+  EARLY_WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+               "%s: running", __FUNCTION__);
+
   g_jvm = static_cast<JavaVM*> (javaVM);
-  g_javaContext = static_cast<jobject> (javaContext);
 
   if (javaVM) {
+    // Already done? Exit early.
+    if (g_javaCmClass != NULL
+        && g_javaCmDevInfoClass != NULL
+        && g_javaCmDevInfoObject != NULL) {
+        EARLY_WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+                     "%s: early exit", __FUNCTION__);
+        return 0;
+    }
+
     JNIEnv* env = NULL;
     if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
                    "%s: could not get Java environment", __FUNCTION__);
       return -1;
     }
     // get java capture class type (note path to class packet)
-    jclass javaCmClassLocal = env->FindClass(AndroidJavaCaptureClass);
+    g_javaCmClass = jsjni_GetGlobalClassRef(AndroidJavaCaptureClass);
-    if (!javaCmClassLocal) {
+    if (!g_javaCmClass) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
                    "%s: could not find java class", __FUNCTION__);
       return -1;
     }
-    // create a global reference to the class
-    // (to tell JNI that we are referencing it
-    // after this function has returned)
-    g_javaCmClass = static_cast<jclass>
-        (env->NewGlobalRef(javaCmClassLocal));
-    if (!g_javaCmClass) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
-                   "%s: InitVideoEngineJava(): could not create"
-                   " Java Camera class reference",
-                   __FUNCTION__);
-      return -1;
-    }
-    // Delete local class ref, we only use the global ref
-    env->DeleteLocalRef(javaCmClassLocal);
     JNINativeMethod nativeFunctions =
-        { "ProvideCameraFrame", "([BIJ)V",
+        { "ProvideCameraFrame", "([BIIJ)V",
           (void*) &VideoCaptureAndroid::ProvideCameraFrame };
     if (env->RegisterNatives(g_javaCmClass, &nativeFunctions, 1) == 0) {
-      WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
                    "%s: Registered native functions", __FUNCTION__);
     }
     else {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
                    "%s: Failed to register native functions",
                    __FUNCTION__);
       return -1;
     }
 
-    jclass capabilityClassLocal = env->FindClass(
-        "org/webrtc/videoengine/CaptureCapabilityAndroid");
-    if (!capabilityClassLocal) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
-                   "%s: could not find java class", __FUNCTION__);
-      return -1;
-    }
-    jclass capabilityClassGlobal = reinterpret_cast<jclass>(env->NewGlobalRef(
-        capabilityClassLocal));
-    DeviceInfoAndroid::SetAndroidCaptureClasses(capabilityClassGlobal);
-
     // get java capture class type (note path to class packet)
-    jclass javaCmDevInfoClassLocal = env->FindClass(
+    g_javaCmDevInfoClass = jsjni_GetGlobalClassRef(
-        "org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid");
+                 AndroidJavaCaptureDeviceInfoClass);
-    if (!javaCmDevInfoClassLocal) {
+    if (!g_javaCmDevInfoClass) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
                    "%s: could not find java class", __FUNCTION__);
       return -1;
     }
 
-    // create a global reference to the class
+    EARLY_WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
-    // (to tell JNI that we are referencing it
-    // after this function has returned)
-    g_javaCmDevInfoClass = static_cast<jclass>
-        (env->NewGlobalRef(javaCmDevInfoClassLocal));
-    if (!g_javaCmDevInfoClass) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
-                   "%s: InitVideoEngineJava(): could not create Java "
-                   "Camera Device info class reference",
-                   __FUNCTION__);
-      return -1;
-    }
-    // Delete local class ref, we only use the global ref
-    env->DeleteLocalRef(javaCmDevInfoClassLocal);
-
-    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
                  "VideoCaptureDeviceInfoAndroid get method id");
 
     // get the method ID for the Android Java CaptureClass static
@@ -158,23 +139,23 @@ int32_t VideoCaptureAndroid::SetAndroidObjects(void* javaVM,
         "(ILandroid/content/Context;)"
         "Lorg/webrtc/videoengine/VideoCaptureDeviceInfoAndroid;");
     if (cid == NULL) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
                    "%s: could not get java"
                    "VideoCaptureDeviceInfoAndroid constructor ID",
                    __FUNCTION__);
       return -1;
     }
 
-    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
+    EARLY_WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
                  "%s: construct static java device object", __FUNCTION__);
 
     // construct the object by calling the static constructor object
     jobject javaCameraDeviceInfoObjLocal =
         env->CallStaticObjectMethod(g_javaCmDevInfoClass,
                                     cid, (int) -1,
-                                    g_javaContext);
+                                    javaContext);
     if (!javaCameraDeviceInfoObjLocal) {
-      WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
                    "%s: could not create Java Capture Device info object",
                    __FUNCTION__);
       return -1;
@@ -183,7 +164,7 @@ int32_t VideoCaptureAndroid::SetAndroidObjects(void* javaVM,
     // we are referencing it after this function has returned)
     g_javaCmDevInfoObject = env->NewGlobalRef(javaCameraDeviceInfoObjLocal);
     if (!g_javaCmDevInfoObject) {
-      WEBRTC_TRACE(webrtc::kTraceError,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceError,
                    webrtc::kTraceAudioDevice,
                    -1,
                    "%s: could not create Java"
@@ -193,13 +174,16 @@ int32_t VideoCaptureAndroid::SetAndroidObjects(void* javaVM,
     }
     // Delete local object ref, we only use the global ref
     env->DeleteLocalRef(javaCameraDeviceInfoObjLocal);
+
+    EARLY_WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+                 "%s: success", __FUNCTION__);
     return 0;
   }
   else {
-    WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+    EARLY_WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
                  "%s: JVM is NULL, assuming deinit", __FUNCTION__);
     if (!g_jvm) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
                    "%s: SetAndroidObjects not called with a valid JVM.",
                    __FUNCTION__);
       return -1;
@@ -211,7 +195,7 @@ int32_t VideoCaptureAndroid::SetAndroidObjects(void* javaVM,
       // Attach this thread to JVM
       jint res = g_jvm->AttachCurrentThread(&env, NULL);
       if ((res < 0) || !env) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
+        EARLY_WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
                      -1, "%s: Could not attach thread to JVM (%d, %p)",
                      __FUNCTION__, res, env);
         return -1;
@@ -222,7 +206,7 @@ int32_t VideoCaptureAndroid::SetAndroidObjects(void* javaVM,
     env->DeleteGlobalRef(g_javaCmDevInfoClass);
     env->DeleteGlobalRef(g_javaCmClass);
     if (attached && g_jvm->DetachCurrentThread() < 0) {
-      WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
+      EARLY_WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
                    "%s: Could not detach thread from JVM", __FUNCTION__);
       return -1;
     }
@@ -232,47 +216,6 @@ int32_t VideoCaptureAndroid::SetAndroidObjects(void* javaVM,
   return 0;
 }
 
-int32_t VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
-    JNIEnv*& env,
-    jclass& javaCmDevInfoClass,
-    jobject& javaCmDevInfoObject,
-    bool& attached) {
-  // get the JNI env for this thread
-  if (!g_jvm) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
-                 "%s: SetAndroidObjects not called with a valid JVM.",
-                 __FUNCTION__);
-    return -1;
-  }
-  attached = false;
-  if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
-    // try to attach the thread and get the env
-    // Attach this thread to JVM
-    jint res = g_jvm->AttachCurrentThread(&env, NULL);
-    if ((res < 0) || !env) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
-                   "%s: Could not attach thread to JVM (%d, %p)",
-                   __FUNCTION__, res, env);
-      return -1;
-    }
-    attached = true;
-  }
-  javaCmDevInfoClass = g_javaCmDevInfoClass;
-  javaCmDevInfoObject = g_javaCmDevInfoObject;
-  return 0;
-
-}
-
-int32_t VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(
-    bool attached) {
-  if (attached && g_jvm->DetachCurrentThread() < 0) {
-    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
-                 "%s: Could not detach thread from JVM", __FUNCTION__);
-    return -1;
-  }
-  return 0;
-}
-
 /*
  * JNI callback from Java class. Called
  * when the camera has a new frame to deliver
@@ -284,11 +227,29 @@ void JNICALL VideoCaptureAndroid::ProvideCameraFrame(JNIEnv * env,
                                                      jobject,
                                                      jbyteArray javaCameraFrame,
                                                      jint length,
+                                                     jint rotation,
                                                      jlong context) {
   VideoCaptureAndroid* captureModule =
       reinterpret_cast<VideoCaptureAndroid*>(context);
   WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture,
                -1, "%s: IncomingFrame %d", __FUNCTION__,length);
+
+  switch (rotation) {
+    case 90:
+      captureModule->SetCaptureRotation(kCameraRotate90);
+      break;
+    case 180:
+      captureModule->SetCaptureRotation(kCameraRotate180);
+      break;
+    case 270:
+      captureModule->SetCaptureRotation(kCameraRotate270);
+      break;
+    case 0:
+    default:
+      captureModule->SetCaptureRotation(kCameraRotate0);
+      break;
+  }
+
   jbyte* cameraFrame= env->GetByteArrayElements(javaCameraFrame,NULL);
   captureModule->IncomingFrame((uint8_t*) cameraFrame,
                                length,captureModule->_frameInfo,0);
@@ -338,33 +299,25 @@ int32_t VideoCaptureAndroid::Init(const int32_t id,
                  "%s: Not a valid Java VM pointer", __FUNCTION__);
     return -1;
   }
-  // get the JNI env for this thread
-  JNIEnv *env;
-  bool isAttached = false;
 
-  // get the JNI env for this thread
+  AutoLocalJNIFrame jniFrame;
-  if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
+  JNIEnv* env = jniFrame.GetEnv();
-    // try to attach the thread and get the env
+  if (!env)
-    // Attach this thread to JVM
-    jint res = g_jvm->AttachCurrentThread(&env, NULL);
-    if ((res < 0) || !env) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
-                   "%s: Could not attach thread to JVM (%d, %p)",
-                   __FUNCTION__, res, env);
       return -1;
-    }
+
-    isAttached = true;
+  jclass javaCmDevInfoClass = jniFrame.GetCmDevInfoClass();
-  }
+  jobject javaCmDevInfoObject = jniFrame.GetCmDevInfoObject();
+
+  int32_t rotation = 0;
 
   WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
                "get method id");
-
   // get the method ID for the Android Java
   // CaptureDeviceInfoClass AllocateCamera factory method.
   char signature[256];
   sprintf(signature, "(IJLjava/lang/String;)L%s;", AndroidJavaCaptureClass);
 
-  jmethodID cid = env->GetMethodID(g_javaCmDevInfoClass, "AllocateCamera",
+  jmethodID cid = env->GetMethodID(javaCmDevInfoClass, "AllocateCamera",
                                    signature);
   if (cid == NULL) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
@@ -374,7 +327,7 @@ int32_t VideoCaptureAndroid::Init(const int32_t id,
 
   jstring capureIdString = env->NewStringUTF((char*) deviceUniqueIdUTF8);
   // construct the object by calling the static constructor object
-  jobject javaCameraObjLocal = env->CallObjectMethod(g_javaCmDevInfoObject,
+  jobject javaCameraObjLocal = env->CallObjectMethod(javaCmDevInfoObject,
                                                      cid, (jint) id,
                                                      (jlong) this,
                                                      capureIdString);
@@ -394,17 +347,6 @@ int32_t VideoCaptureAndroid::Init(const int32_t id,
     return -1;
   }
 
-  // Delete local object ref, we only use the global ref
-  env->DeleteLocalRef(javaCameraObjLocal);
-
-  // Detach this thread if it was attached
-  if (isAttached) {
-    if (g_jvm->DetachCurrentThread() < 0) {
-      WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice, _id,
-                   "%s: Could not detach thread from JVM", __FUNCTION__);
-    }
-  }
-
   return 0;
 }
 
@@ -416,29 +358,15 @@ VideoCaptureAndroid::~VideoCaptureAndroid() {
                  "%s: Nothing to clean", __FUNCTION__);
   }
   else {
-    bool isAttached = false;
+    AutoLocalJNIFrame jniFrame;
-    // get the JNI env for this thread
+    JNIEnv* env = jniFrame.GetEnv();
-    JNIEnv *env;
+    if (!env)
-    if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
+        return;
-      // try to attach the thread and get the env
-      // Attach this thread to JVM
-      jint res = g_jvm->AttachCurrentThread(&env, NULL);
-      if ((res < 0) || !env) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
-                     _id,
-                     "%s: Could not attach thread to JVM (%d, %p)",
-                     __FUNCTION__, res, env);
-      }
-      else {
-        isAttached = true;
-      }
-    }
 
     // get the method ID for the Android Java CaptureClass static
     // DeleteVideoCaptureAndroid  method. Call this to release the camera so
     // another application can use it.
-    jmethodID cid = env->GetStaticMethodID(
+    jmethodID cid = env->GetStaticMethodID(g_javaCmClass,
-        g_javaCmClass,
                                            "DeleteVideoCaptureAndroid",
                                            "(Lorg/webrtc/videoengine/VideoCaptureAndroid;)V");
     if (cid != NULL) {
@@ -450,21 +378,11 @@ VideoCaptureAndroid::~VideoCaptureAndroid() {
       // Delete global object ref to the camera.
       env->DeleteGlobalRef(_javaCaptureObj);
       _javaCaptureObj = NULL;
-    }
+    } else {
-    else {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
                      "%s: Failed to find DeleteVideoCaptureAndroid id",
                      __FUNCTION__);
     }
-
-    // Detach this thread if it was attached
-    if (isAttached) {
-      if (g_jvm->DetachCurrentThread() < 0) {
-        WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice,
-                     _id, "%s: Could not detach thread from JVM",
-                     __FUNCTION__);
-      }
-    }
   }
 }
 
@@ -474,23 +392,13 @@ int32_t VideoCaptureAndroid::StartCapture(
   WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
                "%s: ", __FUNCTION__);
 
-  bool isAttached = false;
   int32_t result = 0;
-  // get the JNI env for this thread
+  int32_t rotation = 0;
-  JNIEnv *env;
+
-  if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
+  AutoLocalJNIFrame jniFrame;
-    // try to attach the thread and get the env
+  JNIEnv* env = jniFrame.GetEnv();
-    // Attach this thread to JVM
+  if (!env)
-    jint res = g_jvm->AttachCurrentThread(&env, NULL);
+      return -1;
-    if ((res < 0) || !env) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
-                   "%s: Could not attach thread to JVM (%d, %p)",
-                   __FUNCTION__, res, env);
-    }
-    else {
-      isAttached = true;
-    }
-  }
 
   if (_capInfo.GetBestMatchedCapability(_deviceUniqueId, capability,
                                         _frameInfo) < 0) {
@@ -522,13 +430,6 @@ int32_t VideoCaptureAndroid::StartCapture(
                  "%s: Failed to find StartCapture id", __FUNCTION__);
   }
 
-  // Detach this thread if it was attached
-  if (isAttached) {
-    if (g_jvm->DetachCurrentThread() < 0) {
-      WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice, _id,
-                   "%s: Could not detach thread from JVM", __FUNCTION__);
-    }
-  }
   if (result == 0) {
     _requestedCapability = capability;
     _captureStarted = true;
@@ -543,23 +444,12 @@ int32_t VideoCaptureAndroid::StopCapture() {
   WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
                "%s: ", __FUNCTION__);
 
-  bool isAttached = false;
   int32_t result = 0;
-  // get the JNI env for this thread
+
-  JNIEnv *env = NULL;
+  AutoLocalJNIFrame jniFrame;
-  if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
+  JNIEnv* env = jniFrame.GetEnv();
-    // try to attach the thread and get the env
+  if (!env)
-    // Attach this thread to JVM
+      return -1;
-    jint res = g_jvm->AttachCurrentThread(&env, NULL);
-    if ((res < 0) || !env) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
-                   "%s: Could not attach thread to JVM (%d, %p)",
-                   __FUNCTION__, res, env);
-    }
-    else {
-      isAttached = true;
-    }
-  }
 
   memset(&_requestedCapability, 0, sizeof(_requestedCapability));
   memset(&_frameInfo, 0, sizeof(_frameInfo));
@@ -577,13 +467,6 @@ int32_t VideoCaptureAndroid::StopCapture() {
                  "%s: Failed to find StopCapture id", __FUNCTION__);
   }
 
-  // Detach this thread if it was attached
-  if (isAttached) {
-    if (g_jvm->DetachCurrentThread() < 0) {
-      WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice, _id,
-                   "%s: Could not detach thread from JVM", __FUNCTION__);
-    }
-  }
   _captureStarted = false;
 
   WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
@@ -610,65 +493,7 @@ int32_t VideoCaptureAndroid::CaptureSettings(
 int32_t VideoCaptureAndroid::SetCaptureRotation(
     VideoCaptureRotation rotation) {
   CriticalSectionScoped cs(&_apiCs);
-  if (VideoCaptureImpl::SetCaptureRotation(rotation) == 0) {
+  return VideoCaptureImpl::SetCaptureRotation(rotation);
-    if (!g_jvm)
-      return -1;
-
-    // get the JNI env for this thread
-    JNIEnv *env;
-    bool isAttached = false;
-
-    // get the JNI env for this thread
-    if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK) {
-      // try to attach the thread and get the env
-      // Attach this thread to JVM
-      jint res = g_jvm->AttachCurrentThread(&env, NULL);
-      if ((res < 0) || !env) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
-                     _id,
-                     "%s: Could not attach thread to JVM (%d, %p)",
-                     __FUNCTION__, res, env);
-        return -1;
-      }
-      isAttached = true;
-    }
-
-    jmethodID cid = env->GetMethodID(g_javaCmClass, "SetPreviewRotation",
-                                     "(I)V");
-    if (cid == NULL) {
-      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
-                   "%s: could not get java SetPreviewRotation ID",
-                   __FUNCTION__);
-      return -1;
-    }
-    jint rotateFrame = 0;
-    switch (rotation) {
-      case kCameraRotate0:
-        rotateFrame = 0;
-        break;
-      case kCameraRotate90:
-        rotateFrame = 90;
-        break;
-      case kCameraRotate180:
-        rotateFrame = 180;
-        break;
-      case kCameraRotate270:
-        rotateFrame = 270;
-        break;
-    }
-    env->CallVoidMethod(_javaCaptureObj, cid, rotateFrame);
-
-    // Detach this thread if it was attached
-    if (isAttached) {
-      if (g_jvm->DetachCurrentThread() < 0) {
-        WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice,
-                     _id, "%s: Could not detach thread from JVM",
-                     __FUNCTION__);
-      }
-    }
-
-  }
-  return 0;
 }
 
 }  // namespace videocapturemodule

media/webrtc/trunk/webrtc/modules/video_capture/android/video_capture_android.h

@@ -12,6 +12,8 @@
 #define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_VIDEO_CAPTURE_ANDROID_H_
 
 #include <jni.h>
+#include <assert.h>
+#include "trace.h"
 
 #include "webrtc/modules/video_capture/android/device_info_android.h"
 #include "webrtc/modules/video_capture/video_capture_impl.h"
@@ -24,17 +26,9 @@ namespace videocapturemodule {
 class VideoCaptureAndroid : public VideoCaptureImpl {
  public:
   static int32_t SetAndroidObjects(void* javaVM, void* javaContext);
-  static int32_t AttachAndUseAndroidDeviceInfoObjects(
-      JNIEnv*& env,
-      jclass& javaCmDevInfoClass,
-      jobject& javaCmDevInfoObject,
-      bool& attached);
-  static int32_t ReleaseAndroidDeviceInfoObjects(bool attached);
-
   VideoCaptureAndroid(const int32_t id);
   virtual int32_t Init(const int32_t id, const char* deviceUniqueIdUTF8);
 
-
   virtual int32_t StartCapture(
       const VideoCaptureCapability& capability);
   virtual int32_t StopCapture();
@@ -42,12 +36,16 @@ class VideoCaptureAndroid : public VideoCaptureImpl {
   virtual int32_t CaptureSettings(VideoCaptureCapability& settings);
   virtual int32_t SetCaptureRotation(VideoCaptureRotation rotation);
 
+  friend class AutoLocalJNIFrame;
+
  protected:
   virtual ~VideoCaptureAndroid();
   static void JNICALL ProvideCameraFrame (JNIEnv * env,
                                           jobject,
                                           jbyteArray javaCameraFrame,
-                                          jint length, jlong context);
+                                          jint length,
+                                          jint rotation,
+                                          jlong context);
   DeviceInfoAndroid _capInfo;
   jobject _javaCaptureObj; // Java Camera object.
   VideoCaptureCapability _frameInfo;
@@ -58,7 +56,120 @@ class VideoCaptureAndroid : public VideoCaptureImpl {
   static jclass g_javaCmDevInfoClass;
   //Static java object implementing the needed device info functions;
   static jobject g_javaCmDevInfoObject;
-  static jobject g_javaContext; // Java Application context
+};
+
+// Reworked version of what is available in AndroidBridge,
+// can attach/deatch in addition to push/pop frames.
+class AutoLocalJNIFrame {
+public:
+ AutoLocalJNIFrame(int nEntries = 128)
+     : mEntries(nEntries), mHasFrameBeenPushed(false), mAttached(false)
+    {
+        mJNIEnv = InitJNIEnv();
+        Push();
+    }
+
+    JNIEnv* GetEnv() {
+        return mJNIEnv;
+    }
+
+    jclass GetCmDevInfoClass() {
+        assert(VideoCaptureAndroid::g_javaCmDevInfoClass != nullptr);
+        return VideoCaptureAndroid::g_javaCmDevInfoClass;
+    }
+
+    jobject GetCmDevInfoObject() {
+        assert(VideoCaptureAndroid::g_javaCmDevInfoObject != nullptr);
+        return VideoCaptureAndroid::g_javaCmDevInfoObject;
+    }
+
+    bool CheckForException() {
+        if (mJNIEnv->ExceptionCheck()) {
+            mJNIEnv->ExceptionDescribe();
+            mJNIEnv->ExceptionClear();
+            return true;
+        }
+
+        return false;
+    }
+
+    ~AutoLocalJNIFrame() {
+        if (!mJNIEnv)
+            return;
+
+        CheckForException();
+
+        if (mHasFrameBeenPushed)
+            mJNIEnv->PopLocalFrame(NULL);
+
+        if (mAttached) {
+            int res = VideoCaptureAndroid::g_jvm->DetachCurrentThread();
+            if (res < 0) {
+                WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+                         "%s: JVM Detach failed.", __FUNCTION__);
+            }
+        }
+    }
+
+private:
+    void Push() {
+        if (!mJNIEnv)
+            return;
+
+        // Make sure there is enough space to store a local ref to the
+        // exception.  I am not completely sure this is needed, but does
+        // not hurt.
+        jint ret = mJNIEnv->PushLocalFrame(mEntries + 1);
+        assert(ret == 0);
+        if (ret < 0)
+            CheckForException();
+        else
+            mHasFrameBeenPushed = true;
+    }
+
+    JNIEnv* InitJNIEnv()
+    {
+        JNIEnv* env = nullptr;
+
+        // Get the JNI env for this thread.
+        if (!VideoCaptureAndroid::g_jvm) {
+            WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+                         "%s: SetAndroidObjects not called with a valid JVM.",
+                         __FUNCTION__);
+            return nullptr;
+        }
+
+        jint res = VideoCaptureAndroid::g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4);
+        if (res == JNI_EDETACHED) {
+            // Try to attach this thread to the JVM and get the env.
+            res = VideoCaptureAndroid::g_jvm->AttachCurrentThread(&env, NULL);
+            if ((res < 0) || !env) {
+                // Attaching failed, error out.
+                WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
+                             "%s: Could not attach thread to JVM (%d, %p)",
+                             __FUNCTION__, res, env);
+                return nullptr;
+            }
+            mAttached = true;
+            WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+                         "%s: attach success", __FUNCTION__);
+        } else if (res == JNI_OK) {
+            // Already attached, GetEnv succeeded.
+            WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
+                         "%s: did not attach because JVM Env already present",
+                         __FUNCTION__);
+        } else {
+            // Non-recoverable error in GetEnv.
+            return nullptr;
+        }
+
+        return env;
+    }
+
+    int mEntries;
+    JNIEnv* mJNIEnv;
+    bool mHasFrameBeenPushed;
+    bool mAttached;
 };
 
 }  // namespace videocapturemodule

media/webrtc/trunk/webrtc/modules/video_capture/device_info_impl.cc

@@ -55,7 +55,7 @@ int32_t DeviceInfoImpl::NumberOfCapabilities(
     if (_lastUsedDeviceNameLength == strlen((char*) deviceUniqueIdUTF8))
     {
         // Is it the same device that is asked for again.
-#if defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
+#if defined(WEBRTC_MAC) || defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
         if(strncasecmp((char*)_lastUsedDeviceName,
                        (char*) deviceUniqueIdUTF8,
                        _lastUsedDeviceNameLength)==0)
@@ -92,7 +92,7 @@ int32_t DeviceInfoImpl::GetCapability(const char* deviceUniqueIdUTF8,
     ReadLockScoped cs(_apiLock);
 
     if ((_lastUsedDeviceNameLength != strlen((char*) deviceUniqueIdUTF8))
-#if defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
+#if defined(WEBRTC_MAC) || defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
         || (strncasecmp((char*)_lastUsedDeviceName,
                         (char*) deviceUniqueIdUTF8,
                         _lastUsedDeviceNameLength)!=0))
@@ -156,7 +156,7 @@ int32_t DeviceInfoImpl::GetBestMatchedCapability(
 
     ReadLockScoped cs(_apiLock);
     if ((_lastUsedDeviceNameLength != strlen((char*) deviceUniqueIdUTF8))
-#if defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
+#if defined(WEBRTC_MAC) || defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
         || (strncasecmp((char*)_lastUsedDeviceName,
                         (char*) deviceUniqueIdUTF8,
                         _lastUsedDeviceNameLength)!=0))

media/webrtc/trunk/webrtc/modules/video_capture/linux/device_info_linux.cc

@@ -93,9 +93,10 @@ int32_t DeviceInfoLinux::GetDeviceName(
     char device[20];
     int fd = -1;
     bool found = false;
-    for (int n = 0; n < 64; n++)
+    int device_index;
+    for (device_index = 0; device_index < 64; device_index++)
     {
-        sprintf(device, "/dev/video%d", n);
+        sprintf(device, "/dev/video%d", device_index);
         if ((fd = open(device, O_RDONLY)) != -1)
         {
             if (count == deviceNumber) {
@@ -154,6 +155,15 @@ int32_t DeviceInfoLinux::GetDeviceName(
                        "buffer passed is too small");
             return -1;
         }
+    } else {
+        // if there's no bus info to use for uniqueId, invent one - and it has to be repeatable
+        if (snprintf(deviceUniqueIdUTF8, deviceUniqueIdUTF8Length, "fake_%u", device_index) >=
+            deviceUniqueIdUTF8Length)
+        {
+            WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
+                       "buffer passed is too small");
+            return -1;
+        }
     }
 
     return 0;
@@ -165,6 +175,7 @@ int32_t DeviceInfoLinux::CreateCapabilityMap(
     int fd;
     char device[32];
     bool found = false;
+    int device_index;
 
     const int32_t deviceUniqueIdUTF8Length =
                             (int32_t) strlen((char*) deviceUniqueIdUTF8);
@@ -176,6 +187,15 @@ int32_t DeviceInfoLinux::CreateCapabilityMap(
     WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
                "CreateCapabilityMap called for device %s", deviceUniqueIdUTF8);
 
+    /* detect /dev/video [0-63] entries */
+    if (sscanf(deviceUniqueIdUTF8,"fake_%d",&device_index) == 1)
+    {
+        sprintf(device, "/dev/video%d", device_index);
+        fd = open(device, O_RDONLY);
+        if (fd != -1) {
+            found = true;
+        }
+    } else {
         /* detect /dev/video [0-63] entries */
         for (int n = 0; n < 64; ++n)
         {
@@ -198,19 +218,11 @@ int32_t DeviceInfoLinux::CreateCapabilityMap(
                         break; // fd matches with device unique id supplied
                     }
                 }
-            else //match for device name
+                // else can't be a match as the test for fake_* above would have matched it
-            {
-                if (IsDeviceNameMatches((const char*) cap.card,
-                                        (const char*) deviceUniqueIdUTF8))
-                {
-                    found = true;
-                    break;
-                }
-            }
             }
             close(fd); // close since this is not the matching device
         }
-
+    }
     if (!found)
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "no matching device found");

media/webrtc/trunk/webrtc/modules/video_capture/linux/video_capture_linux.cc

@@ -10,7 +10,6 @@
 
 #include <errno.h>
 #include <fcntl.h>
-#include <linux/videodev2.h>
 #include <stdio.h>
 #include <string.h>
 #include <sys/ioctl.h>
@@ -18,7 +17,15 @@
 #include <sys/stat.h>
 #include <unistd.h>
 
-#include <iostream>
+//v4l includes
+#if defined(__DragonFly__) || defined(__NetBSD__) || defined(__OpenBSD__)
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
+#include <linux/videodev2.h>
+#endif
+
 #include <new>
 
 #include "webrtc/modules/video_capture/linux/video_capture_linux.h"
@@ -71,6 +78,13 @@ int32_t VideoCaptureModuleV4L2::Init(const char* deviceUniqueIdUTF8)
         memcpy(_deviceUniqueId, deviceUniqueIdUTF8, len + 1);
     }
 
+    int device_index;
+    if (sscanf(deviceUniqueIdUTF8,"fake_%d", &device_index) == 1)
+    {
+      _deviceId = device_index;
+      return 0;
+    }
+
     int fd;
     char device[32];
     bool found = false;

media/webrtc/trunk/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit.mm

@@ -15,6 +15,20 @@
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
+class nsAutoreleasePool {
+public:
+    nsAutoreleasePool()
+    {
+        mLocalPool = [[NSAutoreleasePool alloc] init];
+    }
+    ~nsAutoreleasePool()
+    {
+        [mLocalPool release];
+    }
+private:
+    NSAutoreleasePool *mLocalPool;
+};
+
 namespace webrtc
 {
 
@@ -41,6 +55,7 @@ VideoCaptureMacQTKit::VideoCaptureMacQTKit(const int32_t id) :
 VideoCaptureMacQTKit::~VideoCaptureMacQTKit()
 {
 
+    nsAutoreleasePool localPool;
     WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
                  "~VideoCaptureMacQTKit() called");
     if(_captureDevice)
@@ -71,6 +86,8 @@ int32_t VideoCaptureMacQTKit::Init(
     _deviceUniqueId = new char[nameLength+1];
     memcpy(_deviceUniqueId, iDeviceUniqueIdUTF8,nameLength+1);
 
+    nsAutoreleasePool localPool;
+
     _captureDevice = [[VideoCaptureMacQTKitObjC alloc] init];
     if(NULL == _captureDevice)
     {
@@ -164,6 +181,7 @@ int32_t VideoCaptureMacQTKit::StartCapture(
     const VideoCaptureCapability& capability)
 {
 
+    nsAutoreleasePool localPool;
     _captureWidth = capability.width;
     _captureHeight = capability.height;
     _captureFrameRate = capability.maxFPS;
@@ -180,6 +198,7 @@ int32_t VideoCaptureMacQTKit::StartCapture(
 
 int32_t VideoCaptureMacQTKit::StopCapture()
 {
+    nsAutoreleasePool localPool;
     [_captureDevice stopCapture];
     _isCapturing = false;
     return 0;

media/webrtc/trunk/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info.mm

@@ -13,6 +13,20 @@
 #include "webrtc/modules/video_capture/video_capture_config.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
+class nsAutoreleasePool {
+public:
+    nsAutoreleasePool()
+    {
+        mLocalPool = [[NSAutoreleasePool alloc] init];
+    }
+    ~nsAutoreleasePool()
+    {
+        [mLocalPool release];
+    }
+private:
+    NSAutoreleasePool *mLocalPool;
+};
+
 namespace webrtc
 {
 namespace videocapturemodule
@@ -21,13 +35,14 @@ namespace videocapturemodule
 VideoCaptureMacQTKitInfo::VideoCaptureMacQTKitInfo(const int32_t id) :
     DeviceInfoImpl(id)
 {
+    nsAutoreleasePool localPool;
     _captureInfo = [[VideoCaptureMacQTKitInfoObjC alloc] init];
 }
 
 VideoCaptureMacQTKitInfo::~VideoCaptureMacQTKitInfo()
 {
+    nsAutoreleasePool localPool;
     [_captureInfo release];
-
 }
 
 int32_t VideoCaptureMacQTKitInfo::Init()
@@ -39,6 +54,7 @@ int32_t VideoCaptureMacQTKitInfo::Init()
 uint32_t VideoCaptureMacQTKitInfo::NumberOfDevices()
 {
 
+    nsAutoreleasePool localPool;
     uint32_t captureDeviceCount =
         [[_captureInfo getCaptureDeviceCount]intValue];
     return captureDeviceCount;
@@ -51,6 +67,7 @@ int32_t VideoCaptureMacQTKitInfo::GetDeviceName(
     uint32_t deviceUniqueIdUTF8Length, char* productUniqueIdUTF8,
     uint32_t productUniqueIdUTF8Length)
 {
+    nsAutoreleasePool localPool;
     int errNum = [[_captureInfo getDeviceNamesFromIndex:deviceNumber
                    DefaultName:deviceNameUTF8 WithLength:deviceNameLength
                    AndUniqueID:deviceUniqueIdUTF8
@@ -104,6 +121,7 @@ int32_t VideoCaptureMacQTKitInfo::DisplayCaptureSettingsDialogBox(
     uint32_t positionX, uint32_t positionY)
 {
 
+    nsAutoreleasePool localPool;
     return [[_captureInfo
              displayCaptureSettingsDialogBoxWithDevice:deviceUniqueIdUTF8
              AndTitle:dialogTitleUTF8

media/webrtc/trunk/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info_objc.h

@@ -25,7 +25,6 @@
 @interface VideoCaptureMacQTKitInfoObjC : NSObject{
     bool                                _OSSupportedInfo;
     NSArray*                            _captureDevicesInfo;
-    NSAutoreleasePool*                    _poolInfo;
     int                                    _captureDeviceCountInfo;
 
 }

media/webrtc/trunk/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info_objc.mm

@@ -93,8 +93,12 @@ using namespace webrtc;
         return [NSNumber numberWithInt:-1];
     }
 
-    QTCaptureDevice* tempCaptureDevice =
+    if ([_captureDevicesInfo count] <= index)
-        (QTCaptureDevice*)[_captureDevicesInfo objectAtIndex:index];
+    {
+      return [NSNumber numberWithInt:-1];
+    }
+
+    QTCaptureDevice* tempCaptureDevice = (QTCaptureDevice*)[_captureDevicesInfo objectAtIndex:index];
     if(!tempCaptureDevice)
     {
       return [NSNumber numberWithInt:-1];
@@ -137,7 +141,6 @@ using namespace webrtc;
         return [NSNumber numberWithInt:0];
     }
 
-    _poolInfo = [[NSAutoreleasePool alloc]init];
     _captureDeviceCountInfo = 0;
     [self getCaptureDevices];
 

media/webrtc/trunk/webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_objc.mm

@@ -151,7 +151,17 @@ using namespace videocapturemodule;
   if (!_capturing)
     return;
 
-  [_captureSession stopRunning];
+  // This method is often called on a secondary thread.  Which means
+  // that the following can sometimes run "too early", causing crashes
+  // and/or weird errors concerning initialization.  On OS X 10.7 and
+  // 10.8, the CoreMediaIO method CMIOUninitializeGraph() is called from
+  // -[QTCaptureSession stopRunning].  If this is called too early,
+  // low-level session data gets uninitialized before low-level code
+  // is finished trying to use it.  The solution is to make stopRunning
+  // always run on the main thread.  See bug 837539.
+  [_captureSession performSelectorOnMainThread:@selector(stopRunning)
+                   withObject:nil
+                   waitUntilDone:NO];
   _capturing = NO;
 }
 

media/webrtc/trunk/webrtc/modules/video_capture/mac/video_capture_mac.mm

@@ -27,8 +27,8 @@
 #include <QuickTime/video_capture_quick_time.h>
 #include <QuickTime/video_capture_quick_time_info.h>
 #else
-#include <QTKit/video_capture_qtkit.h>
+#include "qtkit/video_capture_qtkit.h"
-#include <QTKit/video_capture_qtkit_info.h>
+#include "qtkit/video_capture_qtkit_info.h"
 #endif
 
 namespace webrtc

media/webrtc/trunk/webrtc/modules/video_capture/video_capture.gypi

@@ -16,6 +16,11 @@
         '<(webrtc_root)/common_video/common_video.gyp:common_video',
         '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
       ],
+
+      'cflags_mozilla': [
+        '$(NSPR_CFLAGS)',
+      ],
+
       'include_dirs': [
         'include',
         '../interface',
@@ -41,7 +46,7 @@
           ],
         }, {  # include_internal_video_capture == 1
           'conditions': [
-            ['OS=="linux"', {
+            ['include_v4l2_video_capture==1', {
               'include_dirs': [
                 'linux',
               ],
@@ -77,9 +82,13 @@
               },
             }],  # mac
             ['OS=="win"', {
+              'conditions': [
+                ['build_with_mozilla==0', {
                   'dependencies': [
                     '<(DEPTH)/third_party/winsdk_samples/winsdk_samples.gyp:directshow_baseclasses',
                   ],
+                }],
+              ],
               'include_dirs': [
                 'windows',
               ],
@@ -97,6 +106,10 @@
                 'windows/video_capture_factory_windows.cc',
                 'windows/video_capture_mf.cc',
                 'windows/video_capture_mf.h',
+                'windows/BasePin.cpp',
+                'windows/BaseFilter.cpp',
+                'windows/BaseInputPin.cpp',
+                'windows/MediaType.cpp',
               ],
               'link_settings': {
                 'libraries': [
@@ -162,7 +175,7 @@
             'test/video_capture_main_mac.mm',
           ],
           'conditions': [
-            ['OS=="mac" or OS=="linux"', {
+            ['OS!="win" and OS!="android"', {
               'cflags': [
                 '-Wno-write-strings',
               ],
@@ -170,11 +183,15 @@
                 '-lpthread -lm',
               ],
             }],
+            ['include_v4l2_video_capture==1', {
+              'libraries': [
+                '-lXext',
+                '-lX11',
+              ],
+            }],
             ['OS=="linux"', {
               'libraries': [
                 '-lrt',
-                '-lXext',
-                '-lX11',
               ],
             }],
             ['OS=="mac"', {

media/webrtc/trunk/webrtc/modules/video_capture/windows/device_info_ds.cc

@@ -17,7 +17,6 @@
 #include "webrtc/system_wrappers/interface/trace.h"
 
 #include <Dvdmedia.h>
-#include <Streams.h>
 
 namespace webrtc
 {
@@ -42,6 +41,23 @@ const DelayValues WindowsCaptureDelays[NoWindowsCaptureDelays] = {
   },
 };
 
+
+  void _FreeMediaType(AM_MEDIA_TYPE& mt)
+{
+    if (mt.cbFormat != 0)
+    {
+        CoTaskMemFree((PVOID)mt.pbFormat);
+        mt.cbFormat = 0;
+        mt.pbFormat = NULL;
+    }
+    if (mt.pUnk != NULL)
+    {
+        // pUnk should not be used.
+        mt.pUnk->Release();
+        mt.pUnk = NULL;
+    }
+}
+
 // static
 DeviceInfoDS* DeviceInfoDS::Create(const int32_t id)
 {
@@ -565,7 +581,7 @@ int32_t DeviceInfoDS::CreateCapabilityMap(
 
             if (hrVC == S_OK)
             {
-                LONGLONG *frameDurationList;
+                LONGLONG *frameDurationList = NULL;
                 LONGLONG maxFPS; 
                 long listSize;
                 SIZE size;
@@ -584,7 +600,9 @@ int32_t DeviceInfoDS::CreateCapabilityMap(
 
                 // On some odd cameras, you may get a 0 for duration.
                 // GetMaxOfFrameArray returns the lowest duration (highest FPS)
-                if (hrVC == S_OK && listSize > 0 &&
+                // Initialize and check the returned list for null since
+                // some broken drivers don't modify it.
+                if (hrVC == S_OK && listSize > 0 && frameDurationList &&
                     0 != (maxFPS = GetMaxOfFrameArray(frameDurationList, 
                                                       listSize)))
                 {
@@ -679,7 +697,7 @@ int32_t DeviceInfoDS::CreateCapabilityMap(
                          capability->width, capability->height,
                          capability->rawType, capability->maxFPS);
         }
-        DeleteMediaType(pmt);
+        _FreeMediaType(*pmt);
         pmt = NULL;
     }
     RELEASE_AND_CLEAR(streamConfig);

media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.cc

@@ -21,6 +21,9 @@
 DEFINE_GUID(CLSID_SINKFILTER, 0x88cdbbdc, 0xa73b, 0x4afa, 0xac, 0xbf, 0x15, 0xd5,
             0xe2, 0xce, 0x12, 0xc3);
 
+using namespace mozilla::media;
+using namespace mozilla;
+
 namespace webrtc
 {
 namespace videocapturemodule
@@ -37,10 +40,10 @@ typedef struct tagTHREADNAME_INFO
 CaptureInputPin::CaptureInputPin (int32_t moduleId,
                             IN TCHAR * szName,
                             IN CaptureSinkFilter* pFilter,
-                            IN CCritSec * pLock,
+                            IN CriticalSection * pLock,
                             OUT HRESULT * pHr,
                             IN LPCWSTR pszName)
-    : CBaseInputPin (szName, pFilter, pLock, pHr, pszName),
+    : BaseInputPin (szName, pFilter, pLock, pHr, pszName),
       _requestedCapability(),
       _resultingCapability()
 {
@@ -53,7 +56,7 @@ CaptureInputPin::~CaptureInputPin()
 }
 
 HRESULT
-CaptureInputPin::GetMediaType (IN int iPosition, OUT CMediaType * pmt)
+CaptureInputPin::GetMediaType (IN int iPosition, OUT MediaType * pmt)
 {
     // reset the thread handle
     _threadHandle = NULL;
@@ -161,7 +164,7 @@ CaptureInputPin::GetMediaType (IN int iPosition, OUT CMediaType * pmt)
 }
 
 HRESULT
-CaptureInputPin::CheckMediaType ( IN const CMediaType * pMediaType)
+CaptureInputPin::CheckMediaType ( IN const MediaType * pMediaType)
 {
     // reset the thread handle
     _threadHandle = NULL;
@@ -319,8 +322,8 @@ CaptureInputPin::Receive ( IN IMediaSample * pIMediaSample )
 {
     HRESULT hr = S_OK;
 
-    ASSERT (m_pFilter);
+    assert (mFilter);
-    ASSERT (pIMediaSample);
+    assert (pIMediaSample);
 
     // get the thread handle of the delivering thread inc its priority
     if( _threadHandle == NULL)
@@ -348,8 +351,8 @@ CaptureInputPin::Receive ( IN IMediaSample * pIMediaSample )
 
     }
 
-    reinterpret_cast <CaptureSinkFilter *>(m_pFilter)->LockReceive();
+    reinterpret_cast <CaptureSinkFilter *>(mFilter)->LockReceive();
-    hr = CBaseInputPin::Receive (pIMediaSample);
+    hr = BaseInputPin::Receive (pIMediaSample);
 
     if (SUCCEEDED (hr))
     {
@@ -358,17 +361,17 @@ CaptureInputPin::Receive ( IN IMediaSample * pIMediaSample )
         unsigned char* pBuffer = NULL;
         if(S_OK != pIMediaSample->GetPointer(&pBuffer))
         {
-            reinterpret_cast <CaptureSinkFilter *>(m_pFilter)->UnlockReceive();
+            reinterpret_cast <CaptureSinkFilter *>(mFilter)->UnlockReceive();
             return S_FALSE;
         }
 
         // NOTE: filter unlocked within Send call
-        reinterpret_cast <CaptureSinkFilter *> (m_pFilter)->ProcessCapturedFrame(
+        reinterpret_cast <CaptureSinkFilter *> (mFilter)->ProcessCapturedFrame(
                                         pBuffer,length,_resultingCapability);
     }
     else
     {
-        reinterpret_cast <CaptureSinkFilter *>(m_pFilter)->UnlockReceive();
+        reinterpret_cast <CaptureSinkFilter *>(mFilter)->UnlockReceive();
     }
 
     return hr;
@@ -389,13 +392,15 @@ CaptureSinkFilter::CaptureSinkFilter (IN TCHAR * tszName,
                               OUT HRESULT * phr,
                               VideoCaptureExternal& captureObserver,
                               int32_t moduleId)
-    : CBaseFilter(tszName,punk,& m_crtFilter,CLSID_SINKFILTER),
+    : BaseFilter(tszName, CLSID_SINKFILTER),
+      m_crtFilter("CaptureSinkFilter::m_crtFilter"),
+      m_crtRecv("CaptureSinkFilter::m_crtRecv"),
       m_pInput(NULL),
       _captureObserver(captureObserver),
       _moduleId(moduleId)
 {
     (* phr) = S_OK;
-    m_pInput = new CaptureInputPin(moduleId,NAME ("VideoCaptureInputPin"),
+    m_pInput = new CaptureInputPin(moduleId, L"VideoCaptureInputPin",
                                    this,
                                    & m_crtFilter,
                                    phr, L"VideoCapture");
@@ -418,10 +423,10 @@ int CaptureSinkFilter::GetPinCount()
     return 1;
 }
 
-CBasePin *
+BasePin *
 CaptureSinkFilter::GetPin(IN int Index)
 {
-    CBasePin * pPin;
+    BasePin * pPin;
     LockFilter ();
     if (Index == 0)
     {
@@ -438,22 +443,22 @@ CaptureSinkFilter::GetPin(IN int Index)
 STDMETHODIMP CaptureSinkFilter::Pause()
 {
     LockFilter();
-    if (m_State == State_Stopped)
+    if (mState == State_Stopped)
     {
         //  change the state, THEN activate the input pin
-        m_State = State_Paused;
+        mState = State_Paused;
         if (m_pInput && m_pInput->IsConnected())
         {
             m_pInput->Active();
         }
         if (m_pInput && !m_pInput->IsConnected())
         {
-            m_State = State_Running;
+            mState = State_Running;
         }
     }
-    else if (m_State == State_Running)
+    else if (mState == State_Running)
     {
-        m_State = State_Paused;
+        mState = State_Paused;
     }
     UnlockFilter();
     return S_OK;
@@ -465,7 +470,7 @@ STDMETHODIMP CaptureSinkFilter::Stop()
     LockFilter();
 
     //  set the state
-    m_State = State_Stopped;
+    mState = State_Stopped;
 
     //  inactivate the pins
     if (m_pInput)
@@ -479,7 +484,7 @@ STDMETHODIMP CaptureSinkFilter::Stop()
 void CaptureSinkFilter::SetFilterGraph(IGraphBuilder* graph)
 {
     LockFilter();
-    m_pGraph = graph;
+    mGraph = graph;
     UnlockFilter();
 }
 
@@ -488,7 +493,7 @@ void CaptureSinkFilter::ProcessCapturedFrame(unsigned char* pBuffer,
                                          const VideoCaptureCapability& frameInfo)
 {
     //  we have the receiver lock
-    if (m_State == State_Running)
+    if (mState == State_Running)
     {
         _captureObserver.IncomingFrame(pBuffer, length, frameInfo);
 

media/webrtc/trunk/webrtc/modules/video_capture/windows/sink_filter_ds.h

@@ -11,9 +11,10 @@
 #ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
 #define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
 
-#include <Streams.h> // Include base DS filter header files
-
 #include "webrtc/modules/video_capture/include/video_capture_defines.h"
+#include "BaseInputPin.h"
+#include "BaseFilter.h"
+#include "MediaType.h"
 
 namespace webrtc
 {
@@ -26,7 +27,7 @@ class CaptureSinkFilter;
  *	input pin for camera input
  *
  */
-class CaptureInputPin: public CBaseInputPin
+class CaptureInputPin: public mozilla::media::BaseInputPin
 {
 public:
     int32_t _moduleId;
@@ -38,18 +39,18 @@ public:
     CaptureInputPin(int32_t moduleId,
                     IN TCHAR* szName,
                     IN CaptureSinkFilter* pFilter,
-                    IN CCritSec * pLock,
+                    IN mozilla::CriticalSection * pLock,
                     OUT HRESULT * pHr,
                     IN LPCWSTR pszName);
     virtual ~CaptureInputPin();
 
-    HRESULT GetMediaType (IN int iPos, OUT CMediaType * pmt);
+    HRESULT GetMediaType (IN int iPos, OUT mozilla::media::MediaType * pmt);
-    HRESULT CheckMediaType (IN const CMediaType * pmt);
+    HRESULT CheckMediaType (IN const mozilla::media::MediaType * pmt);
     STDMETHODIMP Receive (IN IMediaSample *);
     HRESULT SetMatchingMediaType(const VideoCaptureCapability& capability);
 };
 
-class CaptureSinkFilter: public CBaseFilter
+class CaptureSinkFilter: public mozilla::media::BaseFilter
 {
 
 public:
@@ -66,34 +67,55 @@ public:
     void ProcessCapturedFrame(unsigned char* pBuffer, int32_t length,
                               const VideoCaptureCapability& frameInfo);
     //  explicit receiver lock aquisition and release
-    void LockReceive()  { m_crtRecv.Lock();}
+    void LockReceive()  { m_crtRecv.Enter();}
-    void UnlockReceive() {m_crtRecv.Unlock();}
+    void UnlockReceive() {m_crtRecv.Leave();}
+
     //  explicit filter lock aquisition and release
-    void LockFilter() {m_crtFilter.Lock();}
+    void LockFilter() {m_crtFilter.Enter();}
-    void UnlockFilter() { m_crtFilter.Unlock(); }
+    void UnlockFilter() { m_crtFilter.Leave(); }
     void SetFilterGraph(IGraphBuilder* graph); // Used if EVR
 
     //  --------------------------------------------------------------------
     //  COM interfaces
-DECLARE_IUNKNOWN    ;
+    STDMETHODIMP QueryInterface(REFIID aIId, void **aInterface)
+    {
+      return mozilla::media::BaseFilter::QueryInterface(aIId, aInterface);
+    }
+    STDMETHODIMP_(ULONG) AddRef()
+    {
+      return ::InterlockedIncrement(&mRefCnt);
+    }
+
+    STDMETHODIMP_(ULONG) Release()
+    {
+      unsigned long newRefCnt = ::InterlockedDecrement(&mRefCnt);
+
+      if (!newRefCnt) {
+        delete this;
+      }
+
+      return newRefCnt;
+    }
+
     STDMETHODIMP SetMatchingMediaType(const VideoCaptureCapability& capability);
 
     //  --------------------------------------------------------------------
     //  CBaseFilter methods
     int GetPinCount ();
-    CBasePin * GetPin ( IN int Index);
+    mozilla::media::BasePin * GetPin ( IN int Index);
     STDMETHODIMP Pause ();
     STDMETHODIMP Stop ();
     STDMETHODIMP GetClassID ( OUT CLSID * pCLSID);
     //  --------------------------------------------------------------------
     //  class factory calls this
-    static CUnknown * CreateInstance (IN LPUNKNOWN punk, OUT HRESULT * phr);
+    static IUnknown * CreateInstance (IN LPUNKNOWN punk, OUT HRESULT * phr);
 private:
-    CCritSec m_crtFilter; //  filter lock
+    mozilla::CriticalSection m_crtFilter; //  filter lock
-    CCritSec m_crtRecv;  //  receiver lock; always acquire before filter lock
+    mozilla::CriticalSection m_crtRecv;  //  receiver lock; always acquire before filter lock
     CaptureInputPin * m_pInput;
     VideoCaptureExternal& _captureObserver;
     int32_t _moduleId;
+    unsigned long mRefCnt;
 };
 }  // namespace videocapturemodule
 }  // namespace webrtc

media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/vp8.gyp

@@ -30,6 +30,12 @@
           'dependencies': [
             '<(DEPTH)/third_party/libvpx/libvpx.gyp:libvpx',
           ],
+        },{
+          'link_settings': {
+            'libraries': [
+              '$(LIBVPX_OBJ)/libvpx.a',
+            ],
+          },
         }],
       ],
       'direct_dependent_settings': {