Update to branch-head/58 and add ARM support.

build.sh

@@ -9,7 +9,7 @@ DEPOT_TOOLS_DIR="$THIRD_PARTY_DIR/depot_tools"
 OS=$(go env GOOS)
 ARCH=$(go env GOARCH)
 CONFIG="Release"
-COMMIT="f33698296719f956497d2dbff81b5080864a8804"  # branch-heads/52
+COMMIT="c279861207c5b15fc51069e96595782350e0ac12"  # branch-heads/58
 
 INCLUDE_DIR="$PROJECT_DIR/include"
 LIB_DIR="$PROJECT_DIR/lib"
@@ -57,6 +57,13 @@ else
 	popd
 fi
 
+if [ "$ARCH" = "arm" ]; then
+    echo "Manually fetching arm sysroot"
+    pushd $WEBRTC_SRC || exit 1
+    ./build/linux/sysroot_scripts/install-sysroot.py --arch=arm || exit 1
+    popd
+fi
+
 echo "Checking out latest tested / compatible version of webrtc ..."
 pushd $WEBRTC_SRC
 git checkout $COMMIT
@@ -67,19 +74,15 @@ pushd $WEBRTC_SRC || exit 1
 rm -rf out/$CONFIG
 popd
 
-echo "Applying webrtc patches ..."
-pushd $WEBRTC_SRC || exit 1
-for PATCH in build_at_webrtc_branch_heads_52.patch; do
-	git apply --check ${PROJECT_DIR}/webrtc_patches/${PATCH} || exit 1
-	git am < ${PROJECT_DIR}/webrtc_patches/${PATCH} || exit 1
-done
-popd
-
 echo "Building webrtc ..."
 pushd $WEBRTC_SRC
 export GYP_DEFINES="include_tests=0 include_examples=0"
-python webrtc/build/gyp_webrtc webrtc/api/api.gyp || exit 1
-ninja -C out/$CONFIG || exit 1
+if [ "$ARCH" = "arm" ]; then
+    gn gen out/$CONFIG --args='target_os="linux" target_cpu="arm" is_debug=false' || exit 1
+else
+    gn gen out/$CONFIG --args='is_debug=false' || exit 1
+fi
+ninja -C out/$CONFIG webrtc webrtc/test webrtc/pc:pc_test_utils || exit 1
 popd
 
 echo "Copying headers ..."
@@ -92,15 +95,17 @@ do
 done
 popd
 pushd $PROJECT_DIR || exit 1
-git clean -fdx "$INCLUDE_DIR"
+git clean -fd "$INCLUDE_DIR"
 popd
 
 echo "Concatenating libraries ..."
 pushd $WEBRTC_SRC/out/$CONFIG
 if [ "$OS" = "darwin" ]; then
-	ls *.a > filelist
+	find . -name '*.o' > filelist
 	libtool -static -o libwebrtc-magic.a -filelist filelist
 	strip -S -x -o libwebrtc-magic.a libwebrtc-magic.a
+elif [ "$ARCH" = "arm" ]; then
+    arm-linux-gnueabihf-ar crs libwebrtc-magic.a $(find . -name '*.o' -not -name '*.main.o')
 else
 	ar crs libwebrtc-magic.a $(find . -name '*.o' -not -name '*.main.o')
 fi

datachannel.cc

@@ -111,8 +111,24 @@ class FakeDataChannel : public DataChannelInterface {
     return state_;
   };
 
+  virtual uint32_t messages_sent() const {
+    return 1234;
+  };
+
+  virtual uint64_t bytes_sent() const {
+    return 1234;
+  };
+
+  virtual uint32_t messages_received() const {
+    return 1234;
+  };
+
+  virtual uint64_t bytes_received() const {
+    return 1234;
+  };
+
   virtual uint64_t buffered_amount() const {
-    return 0;
+    return 1234;
   };
 
   // Sends data to self.

datachannel.hpp

@@ -2,6 +2,7 @@
 #define _DATACHANNEL_H_
 
 #include <_cgo_export.h>  // Allow calling certain Go functions.
+#include <assert.h>
 
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/datachannelinterface.h"

datachannel_test.go

@@ -29,7 +29,7 @@ func TestDataStateEnums(t *testing.T) {
 		So(c.MaxRetransmits(), ShouldEqual, 0)
 		So(c.Negotiated(), ShouldBeFalse)
 		So(c.ID(), ShouldEqual, 12345)
-		So(c.BufferedAmount(), ShouldEqual, 0)
+		So(c.BufferedAmount(), ShouldEqual, 1234)
 
 		// There's not a good way to create a DataChannel without first having an
 		// available PeerConnection object with a valid session, but that's part of

include/webrtc/api/audio/audio_mixer.h

@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+#ifndef WEBRTC_API_AUDIO_AUDIO_MIXER_H_
+#define WEBRTC_API_AUDIO_AUDIO_MIXER_H_
+
+#include <memory>
+
+#include "webrtc/base/refcount.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle.
+class AudioMixer : public rtc::RefCountInterface {
+ public:
+  // A callback class that all mixer participants must inherit from/implement.
+  class Source {
+   public:
+    enum class AudioFrameInfo {
+      kNormal,  // The samples in audio_frame are valid and should be used.
+      kMuted,   // The samples in audio_frame should not be used, but
+                // should be implicitly interpreted as zero. Other
+                // fields in audio_frame may be read and should
+                // contain meaningful values.
+      kError,   // The audio_frame will not be used.
+    };
+
+    // Overwrites |audio_frame|. The data_ field is overwritten with
+    // 10 ms of new audio (either 1 or 2 interleaved channels) at
+    // |sample_rate_hz|. All fields in |audio_frame| must be updated.
+    virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
+                                                 AudioFrame* audio_frame) = 0;
+
+    // A way for a mixer implementation to distinguish participants.
+    virtual int Ssrc() const = 0;
+
+    // A way for this source to say that GetAudioFrameWithInfo called
+    // with this sample rate or higher will not cause quality loss.
+    virtual int PreferredSampleRate() const = 0;
+
+    virtual ~Source() {}
+  };
+
+  // Returns true if adding was successful. A source is never added
+  // twice. Addition and removal can happen on different threads.
+  virtual bool AddSource(Source* audio_source) = 0;
+
+  // Removal is never attempted if a source has not been successfully
+  // added to the mixer.
+  virtual void RemoveSource(Source* audio_source) = 0;
+
+  // Performs mixing by asking registered audio sources for audio. The
+  // mixed result is placed in the provided AudioFrame. This method
+  // will only be called from a single thread. The channels argument
+  // specifies the number of channels of the mix result. The mixer
+  // should mix at a rate that doesn't cause quality loss of the
+  // sources' audio. The mixing rate is one of the rates listed in
+  // AudioProcessing::NativeRate. All fields in
+  // |audio_frame_for_mixing| must be updated.
+  virtual void Mix(size_t number_of_channels,
+                   AudioFrame* audio_frame_for_mixing) = 0;
+
+ protected:
+  // Since the mixer is reference counted, the destructor may be
+  // called from any thread.
+  ~AudioMixer() override {}
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_AUDIO_AUDIO_MIXER_H_