Replace slave -> helper, master -> reference

A slight simplification of the NetEq code is also included.

The subtrees below common_audio, modules/audio_coding and
modules/audio_processing were scanned while making this CL.

Bug: webrtc:11680
Change-Id: I33bb1c75b2e3d1c6793fd1c5741ca59f4b6e8455
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178361
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31578}

common_audio/resampler/include/resampler.h

@@ -90,8 +90,8 @@ class Resampler {
   size_t num_channels_;
 
   // Extra instance for stereo
-  Resampler* slave_left_;
+  Resampler* helper_left_;
-  Resampler* slave_right_;
+  Resampler* helper_right_;
 };
 
 }  // namespace webrtc

common_audio/resampler/resampler.cc

@@ -37,8 +37,8 @@ Resampler::Resampler()
       my_out_frequency_khz_(0),
       my_mode_(kResamplerMode1To1),
       num_channels_(0),
-      slave_left_(nullptr),
+      helper_left_(nullptr),
-      slave_right_(nullptr) {}
+      helper_right_(nullptr) {}
 
 Resampler::Resampler(int inFreq, int outFreq, size_t num_channels)
     : Resampler() {
@@ -61,11 +61,11 @@ Resampler::~Resampler() {
   if (out_buffer_) {
     free(out_buffer_);
   }
-  if (slave_left_) {
+  if (helper_left_) {
-    delete slave_left_;
+    delete helper_left_;
   }
-  if (slave_right_) {
+  if (helper_right_) {
-    delete slave_right_;
+    delete helper_right_;
   }
 }
 
@@ -120,13 +120,13 @@ int Resampler::Reset(int inFreq, int outFreq, size_t num_channels) {
     free(out_buffer_);
     out_buffer_ = nullptr;
   }
-  if (slave_left_) {
+  if (helper_left_) {
-    delete slave_left_;
+    delete helper_left_;
-    slave_left_ = nullptr;
+    helper_left_ = nullptr;
   }
-  if (slave_right_) {
+  if (helper_right_) {
-    delete slave_right_;
+    delete helper_right_;
-    slave_right_ = nullptr;
+    helper_right_ = nullptr;
   }
 
   in_buffer_size_ = 0;
@@ -140,8 +140,8 @@ int Resampler::Reset(int inFreq, int outFreq, size_t num_channels) {
 
   if (num_channels_ == 2) {
     // Create two mono resamplers.
-    slave_left_ = new Resampler(inFreq, outFreq, 1);
+    helper_left_ = new Resampler(inFreq, outFreq, 1);
-    slave_right_ = new Resampler(inFreq, outFreq, 1);
+    helper_right_ = new Resampler(inFreq, outFreq, 1);
   }
 
   // Now create the states we need.
@@ -401,7 +401,7 @@ int Resampler::Push(const int16_t* samplesIn,
                     size_t maxLen,
                     size_t& outLen) {
   if (num_channels_ == 2) {
-    // Split up the signal and call the slave object for each channel
+    // Split up the signal and call the helper object for each channel
     int16_t* left =
         static_cast<int16_t*>(malloc(lengthIn * sizeof(int16_t) / 2));
     int16_t* right =
@@ -422,9 +422,9 @@ int Resampler::Push(const int16_t* samplesIn,
     size_t actualOutLen_left = 0;
     size_t actualOutLen_right = 0;
     // Do resampling for right channel
-    res |= slave_left_->Push(left, lengthIn, out_left, maxLen / 2,
+    res |= helper_left_->Push(left, lengthIn, out_left, maxLen / 2,
                               actualOutLen_left);
-    res |= slave_right_->Push(right, lengthIn, out_right, maxLen / 2,
+    res |= helper_right_->Push(right, lengthIn, out_right, maxLen / 2,
                                actualOutLen_right);
     if (res || (actualOutLen_left != actualOutLen_right)) {
       free(left);

modules/audio_coding/neteq/merge.cc

@@ -80,7 +80,7 @@ size_t Merge::Process(int16_t* input,
 
     if (channel == 0) {
       // Downsample, correlate, and find strongest correlation period for the
-      // master (i.e., first) channel only.
+      // reference (i.e., first) channel only.
       // Downsample to 4kHz sample rate.
       Downsample(input_channel.get(), input_length_per_channel,
                  expanded_channel.get(), expanded_length);

modules/audio_coding/neteq/time_stretch.cc

@@ -43,7 +43,7 @@ TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
     signal_len = input_len / num_channels_;
     signal_array.reset(new int16_t[signal_len]);
     signal = signal_array.get();
-    size_t j = master_channel_;
+    size_t j = kRefChannel;
     for (size_t i = 0; i < signal_len; ++i) {
       signal_array[i] = input[j];
       j += num_channels_;
@@ -187,7 +187,7 @@ bool TimeStretch::SpeechDetection(int32_t vec1_energy,
       (static_cast<int64_t>(vec1_energy) + vec2_energy) / 16);
   int32_t right_side;
   if (background_noise_.initialized()) {
-    right_side = background_noise_.Energy(master_channel_);
+    right_side = background_noise_.Energy(kRefChannel);
   } else {
     // If noise parameters have not been estimated, use a fixed threshold.
     right_side = 75000;

modules/audio_coding/neteq/time_stretch.h

@@ -40,13 +40,11 @@ class TimeStretch {
       : sample_rate_hz_(sample_rate_hz),
         fs_mult_(sample_rate_hz / 8000),
         num_channels_(num_channels),
-        master_channel_(0),  // First channel is master.
         background_noise_(background_noise),
         max_input_value_(0) {
     assert(sample_rate_hz_ == 8000 || sample_rate_hz_ == 16000 ||
            sample_rate_hz_ == 32000 || sample_rate_hz_ == 48000);
     assert(num_channels_ > 0);
-    assert(master_channel_ < num_channels_);
     memset(auto_correlation_, 0, sizeof(auto_correlation_));
   }
 
@@ -86,11 +84,11 @@ class TimeStretch {
   static const size_t kMaxLag = 60;
   static const size_t kDownsampledLen = kCorrelationLen + kMaxLag;
   static const int kCorrelationThreshold = 14746;  // 0.9 in Q14.
+  static constexpr size_t kRefChannel = 0;  // First channel is reference.
 
   const int sample_rate_hz_;
   const int fs_mult_;  // Sample rate multiplier = sample_rate_hz_ / 8000.
   const size_t num_channels_;
-  const size_t master_channel_;
   const BackgroundNoise& background_noise_;
   int16_t max_input_value_;
   int16_t downsampled_input_[kDownsampledLen];