Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 7da30673d742075b52c63bb96b78f2c35cc93991

This commit was generated by merge_from_chromium.py.

Change-Id: I6ae37fb65fc9bcff451322ab8e1f52b667dc5d10

7 files changed, 24 insertions, 35 deletions

diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index 36740e87..9091c42a 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -90,21 +90,16 @@ source_set("common_audio") {
 
   deps = [ "../system_wrappers" ]
 
-  # TODO(ajm): Enable when GN support for openmax_dl is added.
-  # See: crbug.com/419206
-  # Not needed immediately, since nothing built by GN depends on these bits.
-  # TODO(ajm): Workaround until openmax_dl has non-Android ARM support.
-  # See: crbug.com/415393
-  #if (cpu_arch != "arm" or (cpu_arch == "arm" and is_android)) {
-  #  'sources' += [
-  #    'lapped_transform.cc',
-  #    'lapped_transform.h',
-  #    'real_fourier.cc',
-  #    'real_fourier.h',
-  #  ]
-  #
-  #  deps += [ "//third_party/openmax_dl/dl" ]
-  #}
+  if (rtc_use_openmax_dl) {
+    sources += [
+      "lapped_transform.cc",
+      "lapped_transform.h",
+      "real_fourier.cc",
+      "real_fourier.h",
+    ]
+
+    deps += [ "//third_party/openmax_dl/dl" ]
+  }
 
   if (cpu_arch == "arm") {
     sources += [
diff --git a/common_audio/common_audio.gyp b/common_audio/common_audio.gyp
index 4581f588..9378f729 100644
--- a/common_audio/common_audio.gyp
+++ b/common_audio/common_audio.gyp
@@ -102,9 +102,7 @@
         'window_generator.h',
       ],
       'conditions': [
-        # TODO(ajm): Workaround until openmax_dl has non-Android ARM support.
-        # See: crbug.com/415393
-        ['target_arch!="arm" or (target_arch=="arm" and OS=="android")', {
+        ['rtc_use_openmax_dl==1', {
           'sources': [
             'lapped_transform.cc',
             'lapped_transform.h',
@@ -247,9 +245,7 @@
             'window_generator_unittest.cc',
           ],
           'conditions': [
-            # TODO(ajm): Workaround until openmax_dl has non-Android ARM
-            # support. See: crbug.com/415393
-            ['target_arch!="arm" or (target_arch=="arm" and OS=="android")', {
+            ['rtc_use_openmax_dl==1', {
               'sources': [
                 'lapped_transform_unittest.cc',
                 'real_fourier_unittest.cc',
diff --git a/common_audio/signal_processing/include/signal_processing_library.h b/common_audio/signal_processing/include/signal_processing_library.h
index ba306ed3..ad95386f 100644
--- a/common_audio/signal_processing/include/signal_processing_library.h
+++ b/common_audio/signal_processing/include/signal_processing_library.h
@@ -89,8 +89,6 @@
 
 // Shifting with negative numbers not allowed
 // We cannot do casting here due to signed/unsigned problem
-#define WEBRTC_SPL_RSHIFT_W16(x, c)     ((x) >> (c))
-#define WEBRTC_SPL_LSHIFT_W16(x, c)     ((x) << (c))
 #define WEBRTC_SPL_RSHIFT_W32(x, c)     ((x) >> (c))
 #define WEBRTC_SPL_LSHIFT_W32(x, c)     ((x) << (c))
 
diff --git a/common_audio/signal_processing/refl_coef_to_lpc.c b/common_audio/signal_processing/refl_coef_to_lpc.c
index 3d81778c..17055c9c 100644
--- a/common_audio/signal_processing/refl_coef_to_lpc.c
+++ b/common_audio/signal_processing/refl_coef_to_lpc.c
@@ -27,7 +27,7 @@ void WebRtcSpl_ReflCoefToLpc(const int16_t *k, int use_order, int16_t *a)
     kptr = k;
     *a = 4096; // i.e., (Word16_MAX >> 3)+1.
     *any = *a;
-    a[1] = WEBRTC_SPL_RSHIFT_W16((*k), 3);
+    a[1] = *k >> 3;
 
     for (m = 1; m < use_order; m++)
     {
@@ -38,7 +38,7 @@ void WebRtcSpl_ReflCoefToLpc(const int16_t *k, int use_order, int16_t *a)
         anyptr = any;
         anyptr++;
 
-        any[m + 1] = WEBRTC_SPL_RSHIFT_W16((*kptr), 3);
+        any[m + 1] = *kptr >> 3;
         for (i = 0; i < m; i++)
         {
             *anyptr = (*aptr)
diff --git a/common_audio/signal_processing/signal_processing_unittest.cc b/common_audio/signal_processing/signal_processing_unittest.cc
index d5cc5f07..09518a9d 100644
--- a/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/common_audio/signal_processing/signal_processing_unittest.cc
@@ -66,8 +66,6 @@ TEST_F(SplTest, MacroTest) {
 
     // Shifting with negative numbers not allowed
     // We cannot do casting here due to signed/unsigned problem
-    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W16(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W16(a, 1));
     EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1));
     EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
 
diff --git a/common_audio/signal_processing/spl_sqrt.c b/common_audio/signal_processing/spl_sqrt.c
index d4f808ca..fff73c03 100644
--- a/common_audio/signal_processing/spl_sqrt.c
+++ b/common_audio/signal_processing/spl_sqrt.c
@@ -17,6 +17,8 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
+#include <assert.h>
+
 int32_t WebRtcSpl_SqrtLocal(int32_t in);
 
 int32_t WebRtcSpl_SqrtLocal(int32_t in)
@@ -154,15 +156,15 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
 
     x_norm = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); // x_norm = AH
 
-    nshift = WEBRTC_SPL_RSHIFT_W16(sh, 1); // nshift = sh>>1
-    nshift = -nshift; // Negate the power for later de-normalization
+    nshift = (sh / 2);
+    assert(nshift >= 0);
 
     A = (int32_t)WEBRTC_SPL_LSHIFT_W32((int32_t)x_norm, 16);
     A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
     A = WebRtcSpl_SqrtLocal(A); // A = sqrt(A)
 
-    if ((-2 * nshift) == sh)
-    { // Even shift value case
+    if (2 * nshift == sh) {
+        // Even shift value case
 
         t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); // t16 = AH
 
@@ -178,7 +180,7 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
     }
 
     A = A & ((int32_t)0x0000ffff);
-    A = (int32_t)WEBRTC_SPL_SHIFT_W32(A, nshift); // De-normalize the result
+    A >>= nshift;  // De-normalize the result.
 
     return A;
 }
diff --git a/common_audio/vad/vad_core.c b/common_audio/vad/vad_core.c
index 98da6eaf..6ebe65d8 100644
--- a/common_audio/vad/vad_core.c
+++ b/common_audio/vad/vad_core.c
@@ -639,10 +639,10 @@ int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
     // Downsample signal 32->16->8 before doing VAD
     WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]),
                            frame_length);
-    len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
+    len = frame_length / 2;
 
     WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
-    len = WEBRTC_SPL_RSHIFT_W16(len, 1);
+    len /= 2;
 
     // Do VAD on an 8 kHz signal
     vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
@@ -660,7 +660,7 @@ int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
     WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states,
                            frame_length);
 
-    len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
+    len = frame_length / 2;
     vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
 
     return vad;