Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/time_stretch.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/time_stretch.h"
 
 #include <algorithm>  // min, max
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
 
 namespace webrtc {
 
 TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
                                               size_t input_len,
                                               bool fast_mode,
                                               AudioMultiVector* output,
-                                              int16_t* length_change_samples) {
+                                              size_t* length_change_samples) {
   // Pre-calculate common multiplication with |fs_mult_|.
-  int fs_mult_120 = fs_mult_ * 120;  // Corresponds to 15 ms.
+  size_t fs_mult_120 = fs_mult_ * 120;  // Corresponds to 15 ms.

[hlundin-webrtc, 2015/08/10 11:30:02]

int

 
   const int16_t* signal;
   rtc::scoped_ptr<int16_t[]> signal_array;
   size_t signal_len;
   if (num_channels_ == 1) {
     signal = input;
     signal_len = input_len;
   } else {
     // We want |signal| to be only the first channel of |input|, which is
     // interleaved. Thus, we take the first sample, skip forward |num_channels|
     // samples, and continue like that.
     signal_len = input_len / num_channels_;
     signal_array.reset(new int16_t[signal_len]);
     signal = signal_array.get();
     size_t j = master_channel_;
     for (size_t i = 0; i < signal_len; ++i) {
       signal_array[i] = input[j];
       j += num_channels_;
     }
   }
 
   // Find maximum absolute value of input signal.
-  max_input_value_ = WebRtcSpl_MaxAbsValueW16(signal,
-                                              static_cast<int>(signal_len));
+  max_input_value_ = WebRtcSpl_MaxAbsValueW16(signal, signal_len);
 
   // Downsample to 4 kHz sample rate and calculate auto-correlation.
   DspHelper::DownsampleTo4kHz(signal, signal_len, kDownsampledLen,
                               sample_rate_hz_, true /* compensate delay*/,
                               downsampled_input_);
   AutoCorrelation();
 
   // Find the strongest correlation peak.
-  static const int kNumPeaks = 1;
-  int peak_index;
+  static const size_t kNumPeaks = 1;
+  size_t peak_index;
   int16_t peak_value;
   DspHelper::PeakDetection(auto_correlation_, kCorrelationLen, kNumPeaks,
                            fs_mult_, &peak_index, &peak_value);
   // Assert that |peak_index| stays within boundaries.
-  assert(peak_index >= 0);
   assert(peak_index <= (2 * kCorrelationLen - 1) * fs_mult_);
 
   // Compensate peak_index for displaced starting position. The displacement
   // happens in AutoCorrelation(). Here, |kMinLag| is in the down-sampled 4 kHz
   // domain, while the |peak_index| is in the original sample rate; hence, the
   // multiplication by fs_mult_ * 2.
   peak_index += kMinLag * fs_mult_ * 2;
   // Assert that |peak_index| stays within boundaries.
   assert(peak_index >= 20 * fs_mult_);
   assert(peak_index <= 20 * fs_mult_ + (2 * kCorrelationLen - 1) * fs_mult_);
 
   // Calculate scaling to ensure that |peak_index| samples can be square-summed
   // without overflowing.
   int scaling = 31 - WebRtcSpl_NormW32(max_input_value_ * max_input_value_) -
-      WebRtcSpl_NormW32(peak_index);
+      WebRtcSpl_NormW32(static_cast<int32_t>(peak_index));

[hlundin-webrtc, 2015/08/10 11:30:02]

rtc::checked_cast

[Peter Kasting, 2015/08/17 22:49:47]

This shouldn't be necessary, as the asserts above guarantee that peak_index <=
119 * fs_mult_, which should fit in an int32_t.

[hlundin-webrtc, 2015/08/18 07:19:18]

Acknowledged.

   scaling = std::max(0, scaling);
 
   // |vec1| starts at 15 ms minus one pitch period.
   const int16_t* vec1 = &signal[fs_mult_120 - peak_index];
   // |vec2| start at 15 ms.
   const int16_t* vec2 = &signal[fs_mult_120];
   // Calculate energies for |vec1| and |vec2|, assuming they both contain
   // |peak_index| samples.
   int32_t vec1_energy =
       WebRtcSpl_DotProductWithScale(vec1, vec1, peak_index, scaling);
@@ skipping 77 matching lines @@
                              kCorrelationLen, kMaxLag - kMinLag, scaling, -1);
 
   // Normalize correlation to 14 bits and write to |auto_correlation_|.
   int32_t max_corr = WebRtcSpl_MaxAbsValueW32(auto_corr, kCorrelationLen);
   scaling = std::max(0, 17 - WebRtcSpl_NormW32(max_corr));
   WebRtcSpl_VectorBitShiftW32ToW16(auto_correlation_, kCorrelationLen,
                                    auto_corr, scaling);
 }
 
 bool TimeStretch::SpeechDetection(int32_t vec1_energy, int32_t vec2_energy,
-                                  int peak_index, int scaling) const {
+                                  size_t peak_index, int scaling) const {
   // Check if the signal seems to be active speech or not (simple VAD).
   // If (vec1_energy + vec2_energy) / (2 * peak_index) <=
   // 8 * background_noise_energy, then we say that the signal contains no
   // active speech.
   // Rewrite the inequality as:
   // (vec1_energy + vec2_energy) / 16 <= peak_index * background_noise_energy.
   // The two sides of the inequality will be denoted |left_side| and
   // |right_side|.
   int32_t left_side = (vec1_energy + vec2_energy) / 16;
   int32_t right_side;
   if (background_noise_.initialized()) {
     right_side = background_noise_.Energy(master_channel_);
   } else {
     // If noise parameters have not been estimated, use a fixed threshold.
     right_side = 75000;
   }
   int right_scale = 16 - WebRtcSpl_NormW32(right_side);
   right_scale = std::max(0, right_scale);
   left_side = left_side >> right_scale;
-  right_side = peak_index * (right_side >> right_scale);
+  right_side = static_cast<int32_t>(peak_index) * (right_side >> right_scale);

[hlundin-webrtc, 2015/08/10 11:30:02]

rtc::checked_cast

[Peter Kasting, 2015/08/17 22:49:47]

While today this shouldn't be necessary as |peak_index| is guaranteed to fit in
an int32_t by the caller (see reply above), I suppose in the future someone
could change that.  I could change the function to take an int32_t instead, but
that seems more confusing, when the rest of the file uses a size_t for this
concept.  So, done.

[hlundin-webrtc, 2015/08/18 07:19:18]

Thanks. Future changes is what worries me sometimes.

 
   // Scale |left_side| properly before comparing with |right_side|.
   // (|scaling| is the scale factor before energy calculation, thus the scale
   // factor for the energy is 2 * scaling.)
   if (WebRtcSpl_NormW32(left_side) < 2 * scaling) {
     // Cannot scale only |left_side|, must scale |right_side| too.
     int temp_scale = WebRtcSpl_NormW32(left_side);
     left_side = left_side << temp_scale;
     right_side = right_side >> (2 * scaling - temp_scale);
   } else {
     left_side = left_side << 2 * scaling;
   }
   return left_side > right_side;
 }
 
 }  // namespace webrtc