Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 17 insertions, 16 deletions

diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
index f1709ae653..9ebfabc286 100644
--- a/webrtc/common_audio/audio_converter.cc
+++ b/webrtc/common_audio/audio_converter.cc
@@ -11,6 +11,7 @@
 #include "webrtc/common_audio/audio_converter.h"
 
 #include <cstring>
+#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
@@ -24,7 +25,7 @@ namespace webrtc {
 
 class CopyConverter : public AudioConverter {
  public:
-  CopyConverter(int src_channels, size_t src_frames, int dst_channels,
+  CopyConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
                 size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
   ~CopyConverter() override {};
@@ -33,7 +34,7 @@ class CopyConverter : public AudioConverter {
                size_t dst_capacity) override {
     CheckSizes(src_size, dst_capacity);
     if (src != dst) {
-      for (int i = 0; i < src_channels(); ++i)
+      for (size_t i = 0; i < src_channels(); ++i)
         std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
     }
   }
@@ -41,7 +42,7 @@ class CopyConverter : public AudioConverter {
 
 class UpmixConverter : public AudioConverter {
  public:
-  UpmixConverter(int src_channels, size_t src_frames, int dst_channels,
+  UpmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
                  size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
   ~UpmixConverter() override {};
@@ -51,7 +52,7 @@ class UpmixConverter : public AudioConverter {
     CheckSizes(src_size, dst_capacity);
     for (size_t i = 0; i < dst_frames(); ++i) {
       const float value = src[0][i];
-      for (int j = 0; j < dst_channels(); ++j)
+      for (size_t j = 0; j < dst_channels(); ++j)
         dst[j][i] = value;
     }
   }
@@ -59,7 +60,7 @@ class UpmixConverter : public AudioConverter {
 
 class DownmixConverter : public AudioConverter {
  public:
-  DownmixConverter(int src_channels, size_t src_frames, int dst_channels,
+  DownmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
                    size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
   }
@@ -71,7 +72,7 @@ class DownmixConverter : public AudioConverter {
     float* dst_mono = dst[0];
     for (size_t i = 0; i < src_frames(); ++i) {
       float sum = 0;
-      for (int j = 0; j < src_channels(); ++j)
+      for (size_t j = 0; j < src_channels(); ++j)
         sum += src[j][i];
       dst_mono[i] = sum / src_channels();
     }
@@ -80,11 +81,11 @@ class DownmixConverter : public AudioConverter {
 
 class ResampleConverter : public AudioConverter {
  public:
-  ResampleConverter(int src_channels, size_t src_frames, int dst_channels,
+  ResampleConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
                     size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
     resamplers_.reserve(src_channels);
-    for (int i = 0; i < src_channels; ++i)
+    for (size_t i = 0; i < src_channels; ++i)
       resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
   }
   ~ResampleConverter() override {};
@@ -105,7 +106,7 @@ class ResampleConverter : public AudioConverter {
 class CompositionConverter : public AudioConverter {
  public:
   CompositionConverter(ScopedVector<AudioConverter> converters)
-      : converters_(converters.Pass()) {
+      : converters_(std::move(converters)) {
     RTC_CHECK_GE(converters_.size(), 2u);
     // We need an intermediate buffer after every converter.
     for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
@@ -135,9 +136,9 @@ class CompositionConverter : public AudioConverter {
   ScopedVector<ChannelBuffer<float>> buffers_;
 };
 
-rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
+rtc::scoped_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
                                                        size_t src_frames,
-                                                       int dst_channels,
+                                                       size_t dst_channels,
                                                        size_t dst_frames) {
   rtc::scoped_ptr<AudioConverter> sp;
   if (src_channels > dst_channels) {
@@ -147,7 +148,7 @@ rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
                                                 dst_channels, src_frames));
       converters.push_back(new ResampleConverter(dst_channels, src_frames,
                                                  dst_channels, dst_frames));
-      sp.reset(new CompositionConverter(converters.Pass()));
+      sp.reset(new CompositionConverter(std::move(converters)));
     } else {
       sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
                                     dst_frames));
@@ -159,7 +160,7 @@ rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
                                                  src_channels, dst_frames));
       converters.push_back(new UpmixConverter(src_channels, dst_frames,
                                               dst_channels, dst_frames));
-      sp.reset(new CompositionConverter(converters.Pass()));
+      sp.reset(new CompositionConverter(std::move(converters)));
     } else {
       sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
                                   dst_frames));
@@ -172,7 +173,7 @@ rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
                                dst_frames));
   }
 
-  return sp.Pass();
+  return sp;
 }
 
 // For CompositionConverter.
@@ -182,8 +183,8 @@ AudioConverter::AudioConverter()
       dst_channels_(0),
       dst_frames_(0) {}
 
-AudioConverter::AudioConverter(int src_channels, size_t src_frames,
-                               int dst_channels, size_t dst_frames)
+AudioConverter::AudioConverter(size_t src_channels, size_t src_frames,
+                               size_t dst_channels, size_t dst_frames)
     : src_channels_(src_channels),
       src_frames_(src_frames),
       dst_channels_(dst_channels),