Minor fixes

Author: IgorSwat

packages/react-native-executorch/common/rnexecutorch/models/speech_to_text/whisper/ASR.cpp

@@ -1,8 +1,3 @@
-#include <algorithm>
-#include <array>
-#include <numeric>
-#include <random>
-
 #include "ASR.h"
 #include "Constants.h"
 #include "Params.h"
@@ -11,6 +6,12 @@
 #include <rnexecutorch/data_processing/Numerical.h>
 #include <rnexecutorch/data_processing/gzip.h>
 
+#include <algorithm>
+#include <array>
+#include <numeric>
+#include <random>
+#include <ranges>
+
 namespace rnexecutorch::models::speech_to_text::whisper {
 
 using executorch::runtime::etensor::ScalarType;
@@ -451,7 +452,7 @@ ASR::estimateWordLevelTimestampsLinear(std::span<const uint64_t> tokens,
       puncts += w.back();
       w.pop_back();
     }
-    std::reverse(puncts.begin(), puncts.end());
+    std::ranges::reverse(puncts);
 
     // Add the core word.
     wordObjs.emplace_back(std::move(w), wStart, wEnd);

packages/react-native-executorch/common/rnexecutorch/models/speech_to_text/whisper/OnlineASR.cpp

@@ -1,14 +1,13 @@
 #include "OnlineASR.h"
+#include "Constants.h"
+#include "Params.h"
+#include "Utils.h"
 
 #include <algorithm>
 #include <iterator>
 #include <ranges>
 #include <utility>
 
-#include "Constants.h"
-#include "Params.h"
-#include "Utils.h"
-
 namespace rnexecutorch::models::speech_to_text::whisper::stream {
 
 OnlineASR::OnlineASR(const ASR *asr) : asr_(asr) {
@@ -41,8 +40,8 @@ void OnlineASR::insertAudioChunk(std::span<const float> audio) {
 }
 
 ProcessResult OnlineASR::process(const DecodingOptions &options) {
-  constexpr float kStreamSafeBufferMaxSamples =
-      params::kStreamSafeBufferDuration * constants::kSamplingRate;
+  constexpr size_t kStreamSafeBufferMaxSamples = static_cast<size_t>(
+      params::kStreamSafeBufferDuration * constants::kSamplingRate);
 
   std::vector<float> audioCopy;
 
@@ -115,8 +114,7 @@ ProcessResult OnlineASR::process(const DecodingOptions &options) {
   // in a 'good' spot - where it will remove a significant audio chunk, yet
   // won't affect most recent, unfinished speech samples.
   size_t bufferSize = audioBuffer_.size();
-  if (std::cmp_greater<size_t, size_t>(bufferSize,
-                                       kStreamSafeBufferMaxSamples)) {
+  if (bufferSize > kStreamSafeBufferMaxSamples) {
     auto newCommitted = commitAndClean(words);
 
     committed.insert(committed.end(),