Use std::array<> consistently for reusable audio buffers.

This is a minor change for places where we use
AudioFrame::kMaxDataSizeSamples sized intermediary buffers. The change
uses `std::array<>` instead of C style arrays which allows for use
of utility templates that incorporate type based buffer size checking.
Also adding `ClearSamples()` method, which complements CopySamples.

Bug: chromium:335805780
Change-Id: I813feb32937e020ceb9ca4b00632dc90907c93fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351681
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42400}

api/audio/audio_frame.cc

@@ -78,9 +78,9 @@ void AudioFrame::UpdateFrame(uint32_t timestamp,
   }
 
   const size_t length = samples_per_channel * num_channels;
-  RTC_CHECK_LE(length, kMaxDataSizeSamples);
+  RTC_CHECK_LE(length, data_.size());
   if (data != nullptr) {
-    memcpy(data_, data, sizeof(int16_t) * length);
+    memcpy(data_.data(), data, sizeof(int16_t) * length);
     muted_ = false;
   } else {
     muted_ = true;
@@ -98,7 +98,7 @@ void AudioFrame::CopyFrom(const AudioFrame& src) {
     // copying over new values. If we don't, msan might complain in some tests.
     // Consider locking down construction, avoiding the default constructor and
     // prefering construction that initializes all state.
-    memset(data_, 0, kMaxDataSizeBytes);
+    ClearSamples(data_);
   }
 
   timestamp_ = src.timestamp_;
@@ -115,7 +115,7 @@ void AudioFrame::CopyFrom(const AudioFrame& src) {
   absolute_capture_timestamp_ms_ = src.absolute_capture_timestamp_ms();
 
   auto data = src.data_view();
-  RTC_CHECK_LE(data.size(), kMaxDataSizeSamples);
+  RTC_CHECK_LE(data.size(), data_.size());
   if (!muted_ && !data.empty()) {
     memcpy(&data_[0], &data[0], sizeof(int16_t) * data.size());
   }
@@ -134,7 +134,7 @@ int64_t AudioFrame::ElapsedProfileTimeMs() const {
 }
 
 const int16_t* AudioFrame::data() const {
-  return muted_ ? zeroed_data().begin() : data_;
+  return muted_ ? zeroed_data().begin() : data_.data();
 }
 
 InterleavedView<const int16_t> AudioFrame::data_view() const {
@@ -155,16 +155,16 @@ int16_t* AudioFrame::mutable_data() {
   // Consider instead if we should rather zero the buffer when `muted_` is set
   // to `true`.
   if (muted_) {
-    memset(data_, 0, kMaxDataSizeBytes);
+    ClearSamples(data_);
     muted_ = false;
   }
-  return data_;
+  return &data_[0];
 }
 
 InterleavedView<int16_t> AudioFrame::mutable_data(size_t samples_per_channel,
                                                   size_t num_channels) {
   const size_t total_samples = samples_per_channel * num_channels;
-  RTC_CHECK_LE(total_samples, kMaxDataSizeSamples);
+  RTC_CHECK_LE(total_samples, data_.size());
   RTC_CHECK_LE(num_channels, kMaxConcurrentChannels);
   // Sanity check for valid argument values during development.
   // If `samples_per_channel` is < `num_channels` but larger than 0,
@@ -178,7 +178,7 @@ InterleavedView<int16_t> AudioFrame::mutable_data(size_t samples_per_channel,
   // Consider instead if we should rather zero the whole buffer when `muted_` is
   // set to `true`.
   if (muted_) {
-    memset(data_, 0, total_samples * sizeof(int16_t));
+    ClearSamples(data_, total_samples);
     muted_ = false;
   }
   samples_per_channel_ = samples_per_channel;
@@ -206,7 +206,7 @@ void AudioFrame::SetLayoutAndNumChannels(ChannelLayout layout,
     RTC_DCHECK_EQ(expected_num_channels, num_channels_);
   }
 #endif
-  RTC_CHECK_LE(samples_per_channel_ * num_channels_, kMaxDataSizeSamples);
+  RTC_CHECK_LE(samples_per_channel_ * num_channels_, data_.size());
 }
 
 void AudioFrame::SetSampleRateAndChannelSize(int sample_rate) {

api/audio/audio_frame.h

@@ -14,6 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <array>
+
 #include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "api/audio/channel_layout.h"
@@ -146,7 +148,7 @@ class AudioFrame {
   // Frame is muted by default.
   bool muted() const;
 
-  size_t max_16bit_samples() const { return kMaxDataSizeSamples; }
+  size_t max_16bit_samples() const { return data_.size(); }
   size_t samples_per_channel() const { return samples_per_channel_; }
   size_t num_channels() const { return num_channels_; }
 
@@ -211,7 +213,7 @@ class AudioFrame {
   // buffer per translation unit is to wrap a static in an inline function.
   static rtc::ArrayView<const int16_t> zeroed_data();
 
-  int16_t data_[kMaxDataSizeSamples];
+  std::array<int16_t, kMaxDataSizeSamples> data_;
   bool muted_ = true;
   ChannelLayout channel_layout_ = CHANNEL_LAYOUT_NONE;
 

api/audio/audio_view.h

@@ -248,6 +248,22 @@ void CopySamples(D& destination, const S& source) {
          source.size() * sizeof(typename S::value_type));
 }
 
+// Sets all the samples in a view to 0. This template function is a simple
+// wrapper around `memset()` but adds the benefit of automatically calculating
+// the byte size from the number of samples and sample type.
+template <typename T>
+void ClearSamples(T& view) {
+  memset(&view[0], 0, view.size() * sizeof(typename T::value_type));
+}
+
+// Same as `ClearSamples()` above but allows for clearing only the first
+// `sample_count` number of samples.
+template <typename T>
+void ClearSamples(T& view, size_t sample_count) {
+  RTC_DCHECK_LE(sample_count, view.size());
+  memset(&view[0], 0, sample_count * sizeof(typename T::value_type));
+}
+
 }  // namespace webrtc
 
 #endif  // API_AUDIO_AUDIO_VIEW_H_

api/audio/test/audio_view_unittest.cc

@@ -10,6 +10,8 @@
 
 #include "api/audio/audio_view.h"
 
+#include <array>
+
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -155,4 +157,34 @@ TEST(AudioViewTest, CopySamples) {
     ASSERT_EQ(dest_arr[i], source_arr[i]) << "i == " << i;
   }
 }
+
+TEST(AudioViewTest, ClearSamples) {
+  std::array<int16_t, 100u> samples = {};
+  FillBuffer(rtc::ArrayView<int16_t>(samples));
+  ASSERT_NE(samples[0], 0);
+  ClearSamples(samples);
+  for (const auto s : samples) {
+    ASSERT_EQ(s, 0);
+  }
+
+  std::array<float, 100u> samples_f = {};
+  FillBuffer(rtc::ArrayView<float>(samples_f));
+  ASSERT_NE(samples_f[0], 0.0);
+  ClearSamples(samples_f);
+  for (const auto s : samples_f) {
+    ASSERT_EQ(s, 0.0);
+  }
+
+  // Clear only half of the buffer
+  FillBuffer(rtc::ArrayView<int16_t>(samples));
+  const auto half_way = samples.size() / 2;
+  ClearSamples(samples, half_way);
+  for (size_t i = 0u; i < samples.size(); ++i) {
+    if (i < half_way) {
+      ASSERT_EQ(samples[i], 0);
+    } else {
+      ASSERT_NE(samples[i], 0);
+    }
+  }
+}
 }  // namespace webrtc

audio/remix_resample.cc

@@ -10,6 +10,8 @@
 
 #include "audio/remix_resample.h"
 
+#include <array>
+
 #include "api/audio/audio_frame.h"
 #include "audio/utility/audio_frame_operations.h"
 #include "common_audio/resampler/include/push_resampler.h"
@@ -40,7 +42,7 @@ void RemixAndResample(const int16_t* src_data,
                       AudioFrame* dst_frame) {
   const int16_t* audio_ptr = src_data;
   size_t audio_ptr_num_channels = num_channels;
-  int16_t downmixed_audio[AudioFrame::kMaxDataSizeSamples];
+  std::array<int16_t, AudioFrame::kMaxDataSizeSamples> downmixed_audio;
 
   // Downmix before resampling.
   if (num_channels > dst_frame->num_channels_) {
@@ -54,7 +56,7 @@ void RemixAndResample(const int16_t* src_data,
                                        num_channels),
         InterleavedView<int16_t>(&downmixed_audio[0], samples_per_channel,
                                  dst_frame->num_channels_));
-    audio_ptr = downmixed_audio;
+    audio_ptr = downmixed_audio.data();
     audio_ptr_num_channels = dst_frame->num_channels_;
   }
 

modules/audio_coding/acm2/acm_receiver.cc

@@ -56,15 +56,13 @@ AcmReceiver::Config::Config(const Config&) = default;
 AcmReceiver::Config::~Config() = default;
 
 AcmReceiver::AcmReceiver(const Config& config)
-    : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
+    : neteq_(CreateNetEq(config.neteq_factory,
-      neteq_(CreateNetEq(config.neteq_factory,
                          config.neteq_config,
                          &config.clock,
                          config.decoder_factory)),
       clock_(config.clock),
       resampled_last_output_frame_(true) {
-  memset(last_audio_buffer_.get(), 0,
+  ClearSamples(last_audio_buffer_);
-         sizeof(int16_t) * AudioFrame::kMaxDataSizeSamples);
 }
 
 AcmReceiver::~AcmReceiver() = default;
@@ -170,7 +168,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
     // Prime the resampler with the last frame.
     int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
     int samples_per_channel_int = resampler_.Resample10Msec(
-        last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+        last_audio_buffer_.data(), current_sample_rate_hz, desired_freq_hz,
         audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
         temp_output);
     if (samples_per_channel_int < 0) {
@@ -206,7 +204,8 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
   }
 
   // Store current audio in `last_audio_buffer_` for next time.
-  memcpy(last_audio_buffer_.get(), audio_frame->data(),
+  // TODO: b/335805780 - Use CopySamples().
+  memcpy(last_audio_buffer_.data(), audio_frame->data(),
          sizeof(int16_t) * audio_frame->samples_per_channel_ *
              audio_frame->num_channels_);
 

modules/audio_coding/acm2/acm_receiver.h

@@ -13,6 +13,7 @@
 
 #include <stdint.h>
 
+#include <array>
 #include <map>
 #include <memory>
 #include <string>
@@ -21,6 +22,7 @@
 
 #include "absl/types/optional.h"
 #include "api/array_view.h"
+#include "api/audio/audio_frame.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
@@ -233,11 +235,12 @@ class AcmReceiver {
   mutable Mutex mutex_;
   absl::optional<DecoderInfo> last_decoder_ RTC_GUARDED_BY(mutex_);
   ACMResampler resampler_ RTC_GUARDED_BY(mutex_);
-  std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(mutex_);
   CallStatistics call_stats_ RTC_GUARDED_BY(mutex_);
   const std::unique_ptr<NetEq> neteq_;  // NetEq is thread-safe; no lock needed.
   Clock& clock_;
   bool resampled_last_output_frame_ RTC_GUARDED_BY(mutex_);
+  std::array<int16_t, AudioFrame::kMaxDataSizeSamples> last_audio_buffer_
+      RTC_GUARDED_BY(mutex_);
 };
 
 }  // namespace acm2