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, 24 insertions, 19 deletions

diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index 81790a159b..ff64267e8c 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -26,7 +26,7 @@ const size_t kSamplesPer48kHzChannel = 480;
 int KeyboardChannelIndex(const StreamConfig& stream_config) {
   if (!stream_config.has_keyboard()) {
     assert(false);
-    return -1;
+    return 0;
   }
 
   return stream_config.num_channels();
@@ -44,9 +44,9 @@ size_t NumBandsFromSamplesPerChannel(size_t num_frames) {
 }  // namespace
 
 AudioBuffer::AudioBuffer(size_t input_num_frames,
-                         int num_input_channels,
+                         size_t num_input_channels,
                          size_t process_num_frames,
-                         int num_process_channels,
+                         size_t num_process_channels,
                          size_t output_num_frames)
   : input_num_frames_(input_num_frames),
     num_input_channels_(num_input_channels),
@@ -74,7 +74,7 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
                                                    num_proc_channels_));
 
     if (input_num_frames_ != proc_num_frames_) {
-      for (int i = 0; i < num_proc_channels_; ++i) {
+      for (size_t i = 0; i < num_proc_channels_; ++i) {
         input_resamplers_.push_back(
             new PushSincResampler(input_num_frames_,
                                   proc_num_frames_));
@@ -82,7 +82,7 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
     }
 
     if (output_num_frames_ != proc_num_frames_) {
-      for (int i = 0; i < num_proc_channels_; ++i) {
+      for (size_t i = 0; i < num_proc_channels_; ++i) {
         output_resamplers_.push_back(
             new PushSincResampler(proc_num_frames_,
                                   output_num_frames_));
@@ -130,7 +130,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
 
   // Resample.
   if (input_num_frames_ != proc_num_frames_) {
-    for (int i = 0; i < num_proc_channels_; ++i) {
+    for (size_t i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_[i]->Resample(data_ptr[i],
                                      input_num_frames_,
                                      process_buffer_->channels()[i],
@@ -140,7 +140,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
   }
 
   // Convert to the S16 range.
-  for (int i = 0; i < num_proc_channels_; ++i) {
+  for (size_t i = 0; i < num_proc_channels_; ++i) {
     FloatToFloatS16(data_ptr[i],
                     proc_num_frames_,
                     data_->fbuf()->channels()[i]);
@@ -150,7 +150,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
 void AudioBuffer::CopyTo(const StreamConfig& stream_config,
                          float* const* data) {
   assert(stream_config.num_frames() == output_num_frames_);
-  assert(stream_config.num_channels() == num_channels_);
+  assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1);
 
   // Convert to the float range.
   float* const* data_ptr = data;
@@ -158,7 +158,7 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
     // Convert to an intermediate buffer for subsequent resampling.
     data_ptr = process_buffer_->channels();
   }
-  for (int i = 0; i < num_channels_; ++i) {
+  for (size_t i = 0; i < num_channels_; ++i) {
     FloatS16ToFloat(data_->fbuf()->channels()[i],
                     proc_num_frames_,
                     data_ptr[i]);
@@ -166,13 +166,18 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
 
   // Resample.
   if (output_num_frames_ != proc_num_frames_) {
-    for (int i = 0; i < num_channels_; ++i) {
+    for (size_t i = 0; i < num_channels_; ++i) {
       output_resamplers_[i]->Resample(data_ptr[i],
                                       proc_num_frames_,
                                       data[i],
                                       output_num_frames_);
     }
   }
+
+  // Upmix.
+  for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
+    memcpy(data[i], data[0], output_num_frames_ * sizeof(**data));
+  }
 }
 
 void AudioBuffer::InitForNewData() {
@@ -192,13 +197,13 @@ int16_t* const* AudioBuffer::channels() {
   return data_->ibuf()->channels();
 }
 
-const int16_t* const* AudioBuffer::split_bands_const(int channel) const {
+const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
   return split_data_.get() ?
          split_data_->ibuf_const()->bands(channel) :
          data_->ibuf_const()->bands(channel);
 }
 
-int16_t* const* AudioBuffer::split_bands(int channel) {
+int16_t* const* AudioBuffer::split_bands(size_t channel) {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ?
          split_data_->ibuf()->bands(channel) :
@@ -249,13 +254,13 @@ float* const* AudioBuffer::channels_f() {
   return data_->fbuf()->channels();
 }
 
-const float* const* AudioBuffer::split_bands_const_f(int channel) const {
+const float* const* AudioBuffer::split_bands_const_f(size_t channel) const {
   return split_data_.get() ?
          split_data_->fbuf_const()->bands(channel) :
          data_->fbuf_const()->bands(channel);
 }
 
-float* const* AudioBuffer::split_bands_f(int channel) {
+float* const* AudioBuffer::split_bands_f(size_t channel) {
   mixed_low_pass_valid_ = false;
   return split_data_.get() ?
          split_data_->fbuf()->bands(channel) :
@@ -336,11 +341,11 @@ AudioFrame::VADActivity AudioBuffer::activity() const {
   return activity_;
 }
 
-int AudioBuffer::num_channels() const {
+size_t AudioBuffer::num_channels() const {
   return num_channels_;
 }
 
-void AudioBuffer::set_num_channels(int num_channels) {
+void AudioBuffer::set_num_channels(size_t num_channels) {
   num_channels_ = num_channels;
 }
 
@@ -393,7 +398,7 @@ void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
 
   // Resample.
   if (input_num_frames_ != proc_num_frames_) {
-    for (int i = 0; i < num_proc_channels_; ++i) {
+    for (size_t i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
                                      input_num_frames_,
                                      data_->fbuf()->channels()[i],
@@ -418,7 +423,7 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) {
       output_buffer_.reset(
           new IFChannelBuffer(output_num_frames_, num_channels_));
     }
-    for (int i = 0; i < num_channels_; ++i) {
+    for (size_t i = 0; i < num_channels_; ++i) {
       output_resamplers_[i]->Resample(
           data_->fbuf()->channels()[i], proc_num_frames_,
           output_buffer_->fbuf()->channels()[i], output_num_frames_);
@@ -443,7 +448,7 @@ void AudioBuffer::CopyLowPassToReference() {
         new ChannelBuffer<int16_t>(num_split_frames_,
                                    num_proc_channels_));
   }
-  for (int i = 0; i < num_proc_channels_; i++) {
+  for (size_t i = 0; i < num_proc_channels_; i++) {
     memcpy(low_pass_reference_channels_->channels()[i],
            split_bands_const(i)[kBand0To8kHz],
            low_pass_reference_channels_->num_frames_per_band() *