Start using ArrayView in AudioFrame, update PushResampler

Start introducing ArrayView to AudioFrame and code that flows down
from there.  In this first step:
* Add `data_view()` that returns a read-only ArrayView for the
  audio buffer. When AudioFrame is not initialized however, data_view()
  will return a nullptr whereas the current data() method never returns
  nullptr.
* Add `mutable_data()` that requires two arguments for properly setting
  the samples per channel and number of channels that's required for
  accurately reserving the returned mutable ArrayView.
  A notable behavior change is that if the requested number of channels
  is larger than supported or the calculated buffer size is too large,
  the function will trigger a check.
* Add TODOs for following work.

Bug: chromium:335805780
Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42202}

api/audio/BUILD.gn

@@ -34,6 +34,7 @@ rtc_library("audio_frame_api") {
   ]
 
   deps = [
+    "..:array_view",
     "..:rtp_packet_info",
     "../../rtc_base:checks",
     "../../rtc_base:logging",

api/audio/audio_frame.cc

@@ -22,6 +22,20 @@ AudioFrame::AudioFrame() {
   static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
 }
 
+AudioFrame::AudioFrame(int sample_rate_hz,
+                       size_t num_channels,
+                       ChannelLayout layout /*= CHANNEL_LAYOUT_UNSUPPORTED*/)
+    : samples_per_channel_(SampleRateToDefaultChannelSize(sample_rate_hz)),
+      sample_rate_hz_(sample_rate_hz),
+      num_channels_(num_channels),
+      channel_layout_(layout == CHANNEL_LAYOUT_UNSUPPORTED
+                          ? GuessChannelLayout(num_channels)
+                          : layout) {
+  RTC_DCHECK_LE(num_channels_, kMaxConcurrentChannels);
+  RTC_DCHECK_GT(sample_rate_hz_, 0);
+  RTC_DCHECK_GT(samples_per_channel_, 0u);
+}
+
 void AudioFrame::Reset() {
   ResetWithoutMuting();
   muted_ = true;
@@ -51,6 +65,7 @@ void AudioFrame::UpdateFrame(uint32_t timestamp,
                              SpeechType speech_type,
                              VADActivity vad_activity,
                              size_t num_channels) {
+  RTC_CHECK_LE(num_channels, kMaxConcurrentChannels);
   timestamp_ = timestamp;
   samples_per_channel_ = samples_per_channel;
   sample_rate_hz_ = sample_rate_hz;
@@ -110,12 +125,26 @@ int64_t AudioFrame::ElapsedProfileTimeMs() const {
 }
 
 const int16_t* AudioFrame::data() const {
-  return muted_ ? empty_data() : data_;
+  return muted_ ? zeroed_data().begin() : data_;
+}
+
+rtc::ArrayView<const int16_t> AudioFrame::data_view() const {
+  const auto samples = samples_per_channel_ * num_channels_;
+  // If you get a nullptr from `data_view()`, it's likely because the
+  // samples_per_channel_ and/or num_channels_ haven't been properly set.
+  // Since `data_view()` returns an rtc::ArrayView<>, we inherit the behavior
+  // in ArrayView when the view size is 0 that ArrayView<>::data() will always
+  // return nullptr. So, even when an AudioFrame is muted and we want to
+  // return `zeroed_data()`, if samples_per_channel_ or  num_channels_ is 0,
+  // the view will point to nullptr.
+  return muted_ ? zeroed_data().subview(0, samples)
+                : rtc::ArrayView<const int16_t>(&data_[0], samples);
 }
 
-// TODO(henrik.lundin) Can we skip zeroing the buffer?
-// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
 int16_t* AudioFrame::mutable_data() {
+  // TODO: bugs.webrtc.org/5647 - Can we skip zeroing the buffer?
+  // Consider instead if we should rather zero the buffer when `muted_` is set
+  // to `true`.
   if (muted_) {
     memset(data_, 0, kMaxDataSizeBytes);
     muted_ = false;
@@ -123,6 +152,29 @@ int16_t* AudioFrame::mutable_data() {
   return data_;
 }
 
+rtc::ArrayView<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(num_channels, kMaxConcurrentChannels);
+  // Sanity check for valid argument values during development.
+  // If `samples_per_channel` is <= kMaxConcurrentChannels but larger than 0,
+  // then chances are the order of arguments is incorrect.
+  RTC_DCHECK((samples_per_channel == 0 && num_channels == 0) ||
+             samples_per_channel > kMaxConcurrentChannels);
+
+  // TODO: bugs.webrtc.org/5647 - Can we skip zeroing the buffer?
+  // 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));
+    muted_ = false;
+  }
+  samples_per_channel_ = samples_per_channel;
+  num_channels_ = num_channels;
+  return rtc::ArrayView<int16_t>(&data_[0], total_samples);
+}
+
 void AudioFrame::Mute() {
   muted_ = true;
 }
@@ -146,10 +198,20 @@ void AudioFrame::SetLayoutAndNumChannels(ChannelLayout layout,
   RTC_CHECK_LE(samples_per_channel_ * num_channels_, kMaxDataSizeSamples);
 }
 
+void AudioFrame::SetSampleRateAndChannelSize(int sample_rate) {
+  sample_rate_hz_ = sample_rate;
+  // We could call `AudioProcessing::GetFrameSize()` here, but that requires
+  // adding a dependency on the ":audio_processing" build target, which can
+  // complicate the dependency tree. Some refactoring is probably in order to
+  // get some consistency around this since there are many places across the
+  // code that assume this default buffer size.
+  samples_per_channel_ = SampleRateToDefaultChannelSize(sample_rate_hz_);
+}
+
 // static
-const int16_t* AudioFrame::empty_data() {
+rtc::ArrayView<const int16_t> AudioFrame::zeroed_data() {
   static int16_t* null_data = new int16_t[kMaxDataSizeSamples]();
-  return &null_data[0];
+  return rtc::ArrayView<const int16_t>(null_data, kMaxDataSizeSamples);
 }
 
 }  // namespace webrtc

api/audio/audio_frame.h

@@ -14,11 +14,30 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include "api/array_view.h"
 #include "api/audio/channel_layout.h"
 #include "api/rtp_packet_infos.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 
+// Default webrtc buffer size in milliseconds.
+constexpr size_t kDefaultAudioBufferLengthMs = 10u;
+
+// Default total number of audio buffers per second based on the default length.
+constexpr size_t kDefaultAudioBuffersPerSec =
+    1000u / kDefaultAudioBufferLengthMs;
+
+// Returns the number of samples a buffer needs to hold for ~10ms of a single
+// audio channel at a given sample rate.
+// See also `AudioProcessing::GetFrameSize()`.
+inline size_t SampleRateToDefaultChannelSize(size_t sample_rate) {
+  // Basic sanity check. 192kHz is the highest supported input sample rate.
+  RTC_DCHECK_LE(sample_rate, 192000);
+  return sample_rate / kDefaultAudioBuffersPerSec;
+}
+/////////////////////////////////////////////////////////////////////
+
 /* This class holds up to 120 ms of super-wideband (32 kHz) stereo audio. It
  * allows for adding and subtracting frames while keeping track of the resulting
  * states.
@@ -57,6 +76,15 @@ class AudioFrame {
 
   AudioFrame();
 
+  // Construct an audio frame with frame length properties and channel
+  // information. `samples_per_channel()` will be initialized to a 10ms buffer
+  // size and if `layout` is not specified (default value of
+  // CHANNEL_LAYOUT_UNSUPPORTED is set), then the channel layout is derived
+  // (guessed) from `num_channels`.
+  AudioFrame(int sample_rate_hz,
+             size_t num_channels,
+             ChannelLayout layout = CHANNEL_LAYOUT_UNSUPPORTED);
+
   AudioFrame(const AudioFrame&) = delete;
   AudioFrame& operator=(const AudioFrame&) = delete;
 
@@ -68,6 +96,7 @@ class AudioFrame {
   // ResetWithoutMuting() to skip this wasteful zeroing.
   void ResetWithoutMuting();
 
+  // TODO: b/335805780 - Accept ArrayView.
   void UpdateFrame(uint32_t timestamp,
                    const int16_t* data,
                    size_t samples_per_channel,
@@ -90,11 +119,29 @@ class AudioFrame {
   int64_t ElapsedProfileTimeMs() const;
 
   // data() returns a zeroed static buffer if the frame is muted.
-  // mutable_frame() always returns a non-static buffer; the first call to
+  // TODO: b/335805780 - Return ArrayView.
-  // mutable_frame() zeros the non-static buffer and marks the frame unmuted.
   const int16_t* data() const;
+
+  // Returns a read-only view of all the valid samples held by the AudioFrame.
+  // Note that for a muted AudioFrame, the size of the returned view will be
+  // 0u and the contained data will be nullptr.
+  rtc::ArrayView<const int16_t> data_view() const;
+
+  // mutable_frame() always returns a non-static buffer; the first call to
+  // mutable_frame() zeros the buffer and marks the frame as unmuted.
+  // TODO: b/335805780 - Return ArrayView based on the current values for
+  // samples per channel and num channels.
   int16_t* mutable_data();
 
+  // Grants write access to the audio buffer. The size of the returned writable
+  // view is determined by the `samples_per_channel` and `num_channels`
+  // dimensions which the function checks for correctness and stores in the
+  // internal member variables; `samples_per_channel()` and `num_channels()`
+  // respectively.
+  // If the state is currently muted, the returned view will be zeroed out.
+  rtc::ArrayView<int16_t> mutable_data(size_t samples_per_channel,
+                                       size_t num_channels);
+
   // Prefer to mute frames using AudioFrameOperations::Mute.
   void Mute();
   // Frame is muted by default.
@@ -119,6 +166,10 @@ class AudioFrame {
     return absolute_capture_timestamp_ms_;
   }
 
+  // Sets the sample_rate_hz and samples_per_channel properties based on a
+  // given sample rate and calculates a default 10ms samples_per_channel value.
+  void SetSampleRateAndChannelSize(int sample_rate);
+
   // RTP timestamp of the first sample in the AudioFrame.
   uint32_t timestamp_ = 0;
   // Time since the first frame in milliseconds.
@@ -157,9 +208,9 @@ class AudioFrame {
 
  private:
   // A permanently zeroed out buffer to represent muted frames. This is a
-  // header-only class, so the only way to avoid creating a separate empty
+  // header-only class, so the only way to avoid creating a separate zeroed
   // buffer per translation unit is to wrap a static in an inline function.
-  static const int16_t* empty_data();
+  static rtc::ArrayView<const int16_t> zeroed_data();
 
   int16_t data_[kMaxDataSizeSamples];
   bool muted_ = true;

api/audio/test/audio_frame_unittest.cc

@@ -19,10 +19,27 @@ namespace webrtc {
 
 namespace {
 
+bool AllSamplesAre(int16_t sample, rtc::ArrayView<const int16_t> samples) {
+  for (const auto s : samples) {
+    if (s != sample) {
+      return false;
+    }
+  }
+  return true;
+}
+
 bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
-  const int16_t* frame_data = frame.data();
+  return AllSamplesAre(sample, frame.data_view());
-  for (size_t i = 0; i < frame.max_16bit_samples(); i++) {
+}
-    if (frame_data[i] != sample) {
+
+// Checks the values of samples in the AudioFrame buffer, regardless of whether
+// they're valid or not, and disregard the `muted()` state of the frame.
+// I.e. use `max_16bit_samples()` instead of the audio properties
+// `num_samples * samples_per_channel`.
+bool AllBufferSamplesAre(int16_t sample, const AudioFrame& frame) {
+  const auto* data = frame.data_view().data();
+  for (size_t i = 0; i < frame.max_16bit_samples(); ++i) {
+    if (data[i] != sample) {
       return false;
     }
   }
@@ -38,29 +55,46 @@ constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
 
 }  // namespace
 
-TEST(AudioFrameTest, FrameStartsMuted) {
+TEST(AudioFrameTest, FrameStartsZeroedAndMuted) {
   AudioFrame frame;
   EXPECT_TRUE(frame.muted());
+  EXPECT_TRUE(frame.data_view().empty());
   EXPECT_TRUE(AllSamplesAre(0, frame));
 }
 
+// TODO: b/335805780 - Delete test when `mutable_data()` returns ArrayView.
+TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroedLegacy) {
+  AudioFrame frame(kSampleRateHz, kNumChannelsMono, CHANNEL_LAYOUT_NONE);
+  frame.mutable_data();
+  EXPECT_FALSE(frame.muted());
+  EXPECT_TRUE(AllSamplesAre(0, frame));
+  EXPECT_TRUE(AllBufferSamplesAre(0, frame));
+}
+
 TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
   AudioFrame frame;
-  frame.mutable_data();
+  auto data = frame.mutable_data(kSamplesPerChannel, kNumChannelsMono);
   EXPECT_FALSE(frame.muted());
+  EXPECT_EQ(frame.data_view().size(), kSamplesPerChannel);
+  EXPECT_EQ(data.size(), kSamplesPerChannel);
   EXPECT_TRUE(AllSamplesAre(0, frame));
 }
 
 TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
   AudioFrame frame;
-  int16_t* frame_data = frame.mutable_data();
+  int16_t* frame_data =
+      frame.mutable_data(kSamplesPerChannel, kNumChannelsMono).begin();
+  EXPECT_FALSE(frame.muted());
+  // Fill the reserved buffer with non-zero data.
   for (size_t i = 0; i < frame.max_16bit_samples(); i++) {
     frame_data[i] = 17;
   }
   ASSERT_TRUE(AllSamplesAre(17, frame));
+  ASSERT_TRUE(AllBufferSamplesAre(17, frame));
   frame.Mute();
   EXPECT_TRUE(frame.muted());
   EXPECT_TRUE(AllSamplesAre(0, frame));
+  ASSERT_TRUE(AllBufferSamplesAre(0, frame));
 }
 
 TEST(AudioFrameTest, UpdateFrameMono) {
@@ -95,11 +129,17 @@ TEST(AudioFrameTest, UpdateFrameMultiChannel) {
   EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel());
   EXPECT_EQ(kNumChannelsStereo, frame.num_channels());
   EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame.channel_layout());
+  EXPECT_TRUE(frame.muted());
 
-  frame.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
+  // Initialize the frame with valid `kNumChannels5_1` data to make sure we
+  // get an unmuted frame with valid samples.
+  int16_t samples[kSamplesPerChannel * kNumChannels5_1] = {17};
+  frame.UpdateFrame(kTimestamp, samples /* data */, kSamplesPerChannel,
                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
                     kNumChannels5_1);
+  EXPECT_FALSE(frame.muted());
   EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel());
+  EXPECT_EQ(kSamplesPerChannel * kNumChannels5_1, frame.data_view().size());
   EXPECT_EQ(kNumChannels5_1, frame.num_channels());
   EXPECT_EQ(CHANNEL_LAYOUT_5_1, frame.channel_layout());
 }
@@ -121,6 +161,7 @@ TEST(AudioFrameTest, CopyFrom) {
   EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_);
   EXPECT_EQ(frame2.num_channels_, frame1.num_channels_);
 
+  EXPECT_EQ(frame2.data_view().size(), frame1.data_view().size());
   EXPECT_EQ(frame2.muted(), frame1.muted());
   EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
 

audio/audio_transport_impl.cc

@@ -70,20 +70,21 @@ void ProcessCaptureFrame(uint32_t delay_ms,
 int Resample(const AudioFrame& frame,
              const int destination_sample_rate,
              PushResampler<int16_t>* resampler,
-             int16_t* destination) {
+             rtc::ArrayView<int16_t> destination) {
   TRACE_EVENT2("webrtc", "Resample", "frame sample rate", frame.sample_rate_hz_,
                "destination_sample_rate", destination_sample_rate);
   const int number_of_channels = static_cast<int>(frame.num_channels_);
   const int target_number_of_samples_per_channel =
       destination_sample_rate / 100;
+  RTC_CHECK_EQ(destination.size(),
+               frame.num_channels_ * target_number_of_samples_per_channel);
+
   resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
                                 number_of_channels);
 
   // TODO(yujo): make resampler take an AudioFrame, and add special case
   // handling of muted frames.
-  return resampler->Resample(
+  return resampler->Resample(frame.data_view(), destination);
-      frame.data(), frame.samples_per_channel_ * number_of_channels,
-      destination, number_of_channels * target_number_of_samples_per_channel);
 }
 }  // namespace
 
@@ -232,8 +233,10 @@ int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples,
     RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
   }
 
-  nSamplesOut = Resample(mixed_frame_, samplesPerSec, &render_resampler_,
+  nSamplesOut =
-                         static_cast<int16_t*>(audioSamples));
+      Resample(mixed_frame_, samplesPerSec, &render_resampler_,
+               rtc::ArrayView<int16_t>(static_cast<int16_t*>(audioSamples),
+                                       nSamples * nChannels));
   RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
   return 0;
 }
@@ -263,8 +266,10 @@ void AudioTransportImpl::PullRenderData(int bits_per_sample,
   *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
   *ntp_time_ms = mixed_frame_.ntp_time_ms_;
 
-  auto output_samples = Resample(mixed_frame_, sample_rate, &render_resampler_,
+  int output_samples =
-                                 static_cast<int16_t*>(audio_data));
+      Resample(mixed_frame_, sample_rate, &render_resampler_,
+               rtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_data),
+                                       number_of_channels * number_of_frames));
   RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
 }
 

audio/remix_resample.cc

@@ -14,6 +14,7 @@
 #include "audio/utility/audio_frame_operations.h"
 #include "common_audio/resampler/include/push_resampler.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 
 namespace webrtc {
 namespace voe {
@@ -67,15 +68,22 @@ void RemixAndResample(const int16_t* src_data,
   // how much to zero here; or 2) make resampler accept a hint that the input is
   // zeroed.
   const size_t src_length = samples_per_channel * audio_ptr_num_channels;
-  int out_length =
+  // Ensure the `samples_per_channel_` member is set correctly based on the
-      resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
+  // destination sample rate, number of channels and assumed 10ms buffer size.
-                          AudioFrame::kMaxDataSizeSamples);
+  // TODO(tommi): Could we rather assume that this has been done by the caller?
+  dst_frame->SetSampleRateAndChannelSize(dst_frame->sample_rate_hz_);
+
+  int out_length = resampler->Resample(
+      rtc::ArrayView<const int16_t>(audio_ptr, src_length),
+      dst_frame->mutable_data(dst_frame->samples_per_channel_,
+                              dst_frame->num_channels_));
   if (out_length == -1) {
     RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
                 << ", src_length = " << src_length
                 << ", dst_frame->mutable_data() = "
                 << dst_frame->mutable_data();
   }
+
   dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
 
   // Upmix after resampling.

audio/remix_resample.h

@@ -17,6 +17,8 @@
 namespace webrtc {
 namespace voe {
 
+// Note: The RemixAndResample methods assume 10ms buffer sizes.
+
 // Upmix or downmix and resample the audio to `dst_frame`. Expects `dst_frame`
 // to have its sample rate and channels members set to the desired values.
 // Updates the `samples_per_channel_` member accordingly.

common_audio/resampler/include/push_resampler.h

@@ -14,11 +14,14 @@
 #include <memory>
 #include <vector>
 
+#include "api/array_view.h"
+
 namespace webrtc {
 
 class PushSincResampler;
 
 // Wraps PushSincResampler to provide stereo support.
+// Note: This implementation assumes 10ms buffer sizes throughout.
 // TODO(ajm): add support for an arbitrary number of channels.
 template <typename T>
 class PushResampler {
@@ -34,7 +37,7 @@ class PushResampler {
 
   // Returns the total number of samples provided in destination (e.g. 32 kHz,
   // 2 channel audio gives 640 samples).
-  int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
+  int Resample(rtc::ArrayView<const T> src, rtc::ArrayView<T> dst);
 
  private:
   int src_sample_rate_hz_;

common_audio/resampler/push_resampler.cc

@@ -73,32 +73,31 @@ int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
 }
 
 template <typename T>
-int PushResampler<T>::Resample(const T* src,
+int PushResampler<T>::Resample(rtc::ArrayView<const T> src,
-                               size_t src_length,
+                               rtc::ArrayView<T> dst) {
-                               T* dst,
-                               size_t dst_capacity) {
   // These checks used to be factored out of this template function due to
   // Windows debug build issues with clang. http://crbug.com/615050
   const size_t src_size_10ms = (src_sample_rate_hz_ / 100) * num_channels_;
   const size_t dst_size_10ms = (dst_sample_rate_hz_ / 100) * num_channels_;
-  RTC_DCHECK_EQ(src_length, src_size_10ms);
+  RTC_DCHECK_EQ(src.size(), src_size_10ms);
-  RTC_DCHECK_GE(dst_capacity, dst_size_10ms);
+  RTC_DCHECK_GE(dst.size(), dst_size_10ms);
 
   if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
     // The old resampler provides this memcpy facility in the case of matching
     // sample rates, so reproduce it here for the sinc resampler.
-    memcpy(dst, src, src_length * sizeof(T));
+    memcpy(dst.data(), src.data(), src.size() * sizeof(T));
-    return static_cast<int>(src_length);
+    return static_cast<int>(src.size());
   }
 
-  const size_t src_length_mono = src_length / num_channels_;
+  const size_t src_length_mono = src.size() / num_channels_;
-  const size_t dst_capacity_mono = dst_capacity / num_channels_;
+  const size_t dst_capacity_mono = dst.size() / num_channels_;
 
   for (size_t ch = 0; ch < num_channels_; ++ch) {
     channel_data_array_[ch] = channel_resamplers_[ch].source.data();
   }
 
-  Deinterleave(src, src_length_mono, num_channels_, channel_data_array_.data());
+  Deinterleave(src.data(), src_length_mono, num_channels_,
+               channel_data_array_.data());
 
   size_t dst_length_mono = 0;
 
@@ -112,7 +111,8 @@ int PushResampler<T>::Resample(const T* src,
     channel_data_array_[ch] = channel_resamplers_[ch].destination.data();
   }
 
-  Interleave(channel_data_array_.data(), dst_length_mono, num_channels_, dst);
+  Interleave(channel_data_array_.data(), dst_length_mono, num_channels_,
+             dst.data());
   return static_cast<int>(dst_length_mono * num_channels_);
 }
 

modules/audio_coding/acm2/acm_resampler.cc

@@ -45,8 +45,9 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio,
     return -1;
   }
 
-  int out_length =
+  int out_length = resampler_.Resample(
-      resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
+      rtc::ArrayView<const int16_t>(in_audio, in_length),
+      rtc::ArrayView<int16_t>(out_audio, out_capacity_samples));
   if (out_length == -1) {
     RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
                       << out_audio << ", " << out_capacity_samples

modules/audio_mixer/audio_mixer_impl_unittest.cc

@@ -517,13 +517,8 @@ TEST(AudioMixerDeathTest, MultipleChannelsAndHighRate) {
   other_frame->samples_per_channel_ = kSamplesPerChannel;
   mixer->AddSource(&other_source);
 
-#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
   EXPECT_DEATH(mixer->Mix(kNumberOfChannels, &frame_for_mixing), "");
-#elif !RTC_DCHECK_IS_ON
-  mixer->Mix(kNumberOfChannels, &frame_for_mixing);
-  EXPECT_EQ(frame_for_mixing.num_channels_, kNumberOfChannels);
-  EXPECT_EQ(frame_for_mixing.sample_rate_hz_,
-            HighOutputRateCalculator::kDefaultFrequency);
 #endif
 }
 

modules/audio_mixer/frame_combiner_unittest.cc

@@ -139,8 +139,9 @@ TEST(FrameCombiner, ContainsAllRtpPacketInfos) {
   }
 }
 
-// There are DCHECKs in place to check for invalid parameters.
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST(FrameCombinerDeathTest, DebugBuildCrashesWithManyChannels) {
+// There are CHECKs in place to check for invalid parameters.
+TEST(FrameCombinerDeathTest, BuildCrashesWithManyChannels) {
   FrameCombiner combiner(true);
   for (const int rate : {8000, 18000, 34000, 48000}) {
     for (const int number_of_channels : {10, 20, 21}) {
@@ -149,7 +150,9 @@ TEST(FrameCombinerDeathTest, DebugBuildCrashesWithManyChannels) {
         continue;
       }
       const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
-      SetUpFrames(rate, number_of_channels);
+      // With an unsupported channel count, this will crash in
+      // `AudioFrame::UpdateFrame`.
+      EXPECT_DEATH(SetUpFrames(rate, number_of_channels), "");
 
       const int number_of_frames = 2;
       SCOPED_TRACE(
@@ -157,18 +160,14 @@ TEST(FrameCombinerDeathTest, DebugBuildCrashesWithManyChannels) {
       const std::vector<AudioFrame*> frames_to_combine(
           all_frames.begin(), all_frames.begin() + number_of_frames);
       AudioFrame audio_frame_for_mixing;
-#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
       EXPECT_DEATH(
           combiner.Combine(frames_to_combine, number_of_channels, rate,
                            frames_to_combine.size(), &audio_frame_for_mixing),
           "");
-#elif !RTC_DCHECK_IS_ON
-      combiner.Combine(frames_to_combine, number_of_channels, rate,
-                       frames_to_combine.size(), &audio_frame_for_mixing);
-#endif
     }
   }
 }
+#endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 TEST(FrameCombinerDeathTest, DebugBuildCrashesWithHighRate) {
   FrameCombiner combiner(true);
@@ -249,7 +248,8 @@ TEST(FrameCombiner, CombiningZeroFramesShouldProduceSilence) {
 TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
   FrameCombiner combiner(false);
   for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
-    for (const int number_of_channels : {1, 2, 4, 8, 10}) {
+    // kMaxConcurrentChannels is 8.
+    for (const int number_of_channels : {1, 2, 4, kMaxConcurrentChannels}) {
       SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1));
 
       AudioFrame audio_frame_for_mixing;

modules/audio_processing/agc2/vad_wrapper.cc

@@ -104,8 +104,7 @@ float VoiceActivityDetectorWrapper::Analyze(AudioFrameView<const float> frame) {
   }
   // Resample the first channel of `frame`.
   RTC_DCHECK_EQ(frame.samples_per_channel(), frame_size_);
-  resampler_.Resample(frame.channel(0).data(), frame_size_,
+  resampler_.Resample(frame.channel(0), resampled_buffer_);
-                      resampled_buffer_.data(), resampled_buffer_.size());
 
   return vad_->Analyze(resampled_buffer_);
 }

modules/audio_processing/audio_processing_unittest.cc

@@ -2198,7 +2198,8 @@ TEST_P(AudioProcessingTest, Formats) {
           // necessary.
           ASSERT_EQ(ref_length,
                     static_cast<size_t>(resampler.Resample(
-                        out_ptr, out_length, cmp_data.get(), ref_length)));
+                        rtc::ArrayView<const float>(out_ptr, out_length),
+                        rtc::ArrayView<float>(cmp_data.get(), ref_length))));
           out_ptr = cmp_data.get();
         }