diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index f9e2895b31..73b2beb568 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -49,7 +49,7 @@ rtc_library("adaptive_digital_gain_controller") {
     ":common",
     ":gain_applier",
     "..:apm_logging",
-    "..:audio_frame_view",
+    "../../../api/audio:audio_frame_api",
     "../../../api/audio:audio_processing",
     "../../../common_audio",
     "../../../rtc_base:checks",
@@ -174,7 +174,7 @@ rtc_library("gain_applier") {
   deps = [
     ":common",
     "..:audio_frame_view",
-    "../../../api:array_view",
+    "../../../api/audio:audio_frame_api",
     "../../../rtc_base:safe_minmax",
   ]
 }
@@ -232,8 +232,7 @@ rtc_library("noise_level_estimator") {
   deps = [
     ":biquad_filter",
     "..:apm_logging",
-    "..:audio_frame_view",
-    "../../../api:array_view",
+    "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
     "../../../system_wrappers",
   ]
@@ -266,8 +265,7 @@ rtc_library("vad_wrapper") {
   deps = [
     ":common",
     ":cpu_features",
-    "..:audio_frame_view",
-    "../../../api:array_view",
+    "../../../api/audio:audio_frame_api",
     "../../../common_audio",
     "../../../rtc_base:checks",
     "rnn_vad",
@@ -335,7 +333,7 @@ rtc_library("gain_applier_unittest") {
   deps = [
     ":gain_applier",
     ":test_utils",
-    "..:audio_frame_view",
+    "../../../api/audio:audio_frame_api",
     "../../../rtc_base:gunit_helpers",
     "../../../test:test_support",
   ]
@@ -435,9 +433,8 @@ rtc_library("noise_estimator_unittests") {
     ":noise_level_estimator",
     ":test_utils",
     "..:apm_logging",
-    "..:audio_frame_view",
-    "../../../api:array_view",
     "../../../api:function_view",
+    "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
     "../../../rtc_base:gunit_helpers",
   ]
@@ -449,7 +446,7 @@ rtc_library("vad_wrapper_unittests") {
   deps = [
     ":common",
     ":vad_wrapper",
-    "..:audio_frame_view",
+    "../../../api/audio:audio_frame_api",
     "../../../rtc_base:checks",
     "../../../rtc_base:gunit_helpers",
     "../../../rtc_base:safe_compare",
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc b/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
index e8edab602c..5f924cbbcf 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
@@ -124,7 +124,7 @@ AdaptiveDigitalGainController::AdaptiveDigitalGainController(
 }
 
 void AdaptiveDigitalGainController::Process(const FrameInfo& info,
-                                            AudioFrameView<float> frame) {
+                                            DeinterleavedView<float> frame) {
   RTC_DCHECK_GE(info.speech_level_dbfs, -150.0f);
   RTC_DCHECK_GE(frame.num_channels(), 1);
   RTC_DCHECK(
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_controller.h b/modules/audio_processing/agc2/adaptive_digital_gain_controller.h
index 9ae74a2dc8..d464dc6b2c 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_controller.h
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_controller.h
@@ -14,8 +14,8 @@
 #include <vector>
 
 #include "api/audio/audio_processing.h"
+#include "api/audio/audio_view.h"
 #include "modules/audio_processing/agc2/gain_applier.h"
-#include "modules/audio_processing/include/audio_frame_view.h"
 
 namespace webrtc {
 
@@ -46,7 +46,7 @@ class AdaptiveDigitalGainController {
 
   // Analyzes `info`, updates the digital gain and applies it to a 10 ms
   // `frame`. Supports any sample rate supported by APM.
-  void Process(const FrameInfo& info, AudioFrameView<float> frame);
+  void Process(const FrameInfo& info, DeinterleavedView<float> frame);
 
  private:
   ApmDataDumper* const apm_data_dumper_;
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc b/modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc
index 88fb792f2d..39e175d403 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc
@@ -83,7 +83,7 @@ TEST(GainController2AdaptiveDigitalGainControllerTest,
   // Make one call with reasonable audio level values and settings.
   VectorFloatFrame fake_audio(kStereo, kFrameLen10ms48kHz, 10000.0f);
   helper.gain_applier->Process(GetFrameInfoToNotAdapt(kDefaultConfig),
-                               fake_audio.float_frame_view());
+                               fake_audio.view());
 }
 
 // Checks that the maximum allowed gain is applied.
@@ -103,7 +103,7 @@ TEST(GainController2AdaptiveDigitalGainControllerTest, MaxGainApplied) {
   float applied_gain;
   for (int i = 0; i < kNumFramesToAdapt; ++i) {
     VectorFloatFrame fake_audio(kMono, kFrameLen10ms8kHz, 1.0f);
-    helper.gain_applier->Process(info, fake_audio.float_frame_view());
+    helper.gain_applier->Process(info, fake_audio.view());
     applied_gain = fake_audio.float_frame_view().channel(0)[0];
   }
   const float applied_gain_db = 20.0f * std::log10f(applied_gain);
@@ -129,8 +129,8 @@ TEST(GainController2AdaptiveDigitalGainControllerTest, GainDoesNotChangeFast) {
     AdaptiveDigitalGainController::FrameInfo info =
         GetFrameInfoToNotAdapt(kDefaultConfig);
     info.speech_level_dbfs = initial_level_dbfs;
-    helper.gain_applier->Process(info, fake_audio.float_frame_view());
-    float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
+    helper.gain_applier->Process(info, fake_audio.view());
+    float current_gain_linear = fake_audio.view()[0][0];
     EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
               max_change_per_frame_linear);
     last_gain_linear = current_gain_linear;
@@ -143,8 +143,8 @@ TEST(GainController2AdaptiveDigitalGainControllerTest, GainDoesNotChangeFast) {
     AdaptiveDigitalGainController::FrameInfo info =
         GetFrameInfoToNotAdapt(kDefaultConfig);
     info.speech_level_dbfs = 0.f;
-    helper.gain_applier->Process(info, fake_audio.float_frame_view());
-    float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
+    helper.gain_applier->Process(info, fake_audio.view());
+    float current_gain_linear = fake_audio.view()[0][0];
     EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
               max_change_per_frame_linear);
     last_gain_linear = current_gain_linear;
@@ -160,10 +160,10 @@ TEST(GainController2AdaptiveDigitalGainControllerTest, GainIsRampedInAFrame) {
   AdaptiveDigitalGainController::FrameInfo info =
       GetFrameInfoToNotAdapt(kDefaultConfig);
   info.speech_level_dbfs = initial_level_dbfs;
-  helper.gain_applier->Process(info, fake_audio.float_frame_view());
+  helper.gain_applier->Process(info, fake_audio.view());
   float maximal_difference = 0.0f;
   float current_value = 1.0f * DbToRatio(kDefaultConfig.initial_gain_db);
-  for (const auto& x : fake_audio.float_frame_view().channel(0)) {
+  for (const auto& x : fake_audio.view()[0]) {
     const float difference = std::abs(x - current_value);
     maximal_difference = std::max(maximal_difference, difference);
     current_value = x;
@@ -195,13 +195,13 @@ TEST(GainController2AdaptiveDigitalGainControllerTest, NoiseLimitsGain) {
         GetFrameInfoToNotAdapt(kDefaultConfig);
     info.speech_level_dbfs = initial_level_dbfs;
     info.noise_rms_dbfs = kWithNoiseDbfs;
-    helper.gain_applier->Process(info, fake_audio.float_frame_view());
+    auto fake_view = fake_audio.view();
+    helper.gain_applier->Process(info, fake_view);
 
     // Wait so that the adaptive gain applier has time to lower the gain.
     if (i > num_initial_frames) {
       const float maximal_ratio =
-          *std::max_element(fake_audio.float_frame_view().channel(0).begin(),
-                            fake_audio.float_frame_view().channel(0).end());
+          *std::max_element(fake_view[0].begin(), fake_view[0].end());
 
       EXPECT_NEAR(maximal_ratio, 1.0f, 0.001f);
     }
@@ -217,7 +217,7 @@ TEST(GainController2AdaptiveDigitalGainControllerTest,
   AdaptiveDigitalGainController::FrameInfo info =
       GetFrameInfoToNotAdapt(kDefaultConfig);
   info.speech_level_dbfs = 5.0f;
-  helper.gain_applier->Process(info, fake_audio.float_frame_view());
+  helper.gain_applier->Process(info, fake_audio.view());
 }
 
 TEST(GainController2AdaptiveDigitalGainControllerTest, AudioLevelLimitsGain) {
@@ -239,13 +239,13 @@ TEST(GainController2AdaptiveDigitalGainControllerTest, AudioLevelLimitsGain) {
     info.speech_level_dbfs = initial_level_dbfs;
     info.limiter_envelope_dbfs = 1.0f;
     info.speech_level_reliable = false;
-    helper.gain_applier->Process(info, fake_audio.float_frame_view());
+    auto fake_view = fake_audio.view();
+    helper.gain_applier->Process(info, fake_view);
 
     // Wait so that the adaptive gain applier has time to lower the gain.
     if (i > num_initial_frames) {
       const float maximal_ratio =
-          *std::max_element(fake_audio.float_frame_view().channel(0).begin(),
-                            fake_audio.float_frame_view().channel(0).end());
+          *std::max_element(fake_view[0].begin(), fake_view[0].end());
 
       EXPECT_NEAR(maximal_ratio, 1.0f, 0.001f);
     }
@@ -271,8 +271,8 @@ TEST_P(AdaptiveDigitalGainControllerParametrizedTest,
   for (int i = 0; i < adjacent_speech_frames_threshold(); ++i) {
     SCOPED_TRACE(i);
     VectorFloatFrame audio(kMono, kFrameLen10ms48kHz, 1.0f);
-    helper.gain_applier->Process(info, audio.float_frame_view());
-    const float gain = audio.float_frame_view().channel(0)[0];
+    helper.gain_applier->Process(info, audio.view());
+    const float gain = audio.view()[0][0];
     if (i > 0) {
       EXPECT_EQ(prev_gain, gain);  // No gain increase applied.
     }
@@ -293,16 +293,16 @@ TEST_P(AdaptiveDigitalGainControllerParametrizedTest,
   for (int i = 0; i < adjacent_speech_frames_threshold(); ++i) {
     SCOPED_TRACE(i);
     VectorFloatFrame audio(kMono, kFrameLen10ms48kHz, 1.0f);
-    helper.gain_applier->Process(info, audio.float_frame_view());
-    prev_gain = audio.float_frame_view().channel(0)[0];
+    helper.gain_applier->Process(info, audio.view());
+    prev_gain = audio.view()[0][0];
   }
 
   // Process one more speech frame.
   VectorFloatFrame audio(kMono, kFrameLen10ms48kHz, 1.0f);
-  helper.gain_applier->Process(info, audio.float_frame_view());
+  helper.gain_applier->Process(info, audio.view());
 
   // An increased gain has been applied.
-  EXPECT_GT(audio.float_frame_view().channel(0)[0], prev_gain);
+  EXPECT_GT(audio.view()[0][0], prev_gain);
 }
 
 INSTANTIATE_TEST_SUITE_P(
diff --git a/modules/audio_processing/agc2/gain_applier.cc b/modules/audio_processing/agc2/gain_applier.cc
index f9e276d3a8..f833ad1fbe 100644
--- a/modules/audio_processing/agc2/gain_applier.cc
+++ b/modules/audio_processing/agc2/gain_applier.cc
@@ -10,7 +10,7 @@
 
 #include "modules/audio_processing/agc2/gain_applier.h"
 
-#include "api/array_view.h"
+#include "api/audio/audio_view.h"
 #include "modules/audio_processing/agc2/agc2_common.h"
 #include "rtc_base/numerics/safe_minmax.h"
 
@@ -24,9 +24,9 @@ bool GainCloseToOne(float gain_factor) {
          gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
 }
 
-void ClipSignal(AudioFrameView<float> signal) {
-  for (int k = 0; k < signal.num_channels(); ++k) {
-    rtc::ArrayView<float> channel_view = signal.channel(k);
+void ClipSignal(DeinterleavedView<float> signal) {
+  for (size_t k = 0; k < signal.num_channels(); ++k) {
+    MonoView<float> channel_view = signal[k];
     for (auto& sample : channel_view) {
       sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
     }
@@ -36,7 +36,7 @@ void ClipSignal(AudioFrameView<float> signal) {
 void ApplyGainWithRamping(float last_gain_linear,
                           float gain_at_end_of_frame_linear,
                           float inverse_samples_per_channel,
-                          AudioFrameView<float> float_frame) {
+                          DeinterleavedView<float> float_frame) {
   // Do not modify the signal.
   if (last_gain_linear == gain_at_end_of_frame_linear &&
       GainCloseToOne(gain_at_end_of_frame_linear)) {
@@ -45,8 +45,8 @@ void ApplyGainWithRamping(float last_gain_linear,
 
   // Gain is constant and different from 1.
   if (last_gain_linear == gain_at_end_of_frame_linear) {
-    for (int k = 0; k < float_frame.num_channels(); ++k) {
-      rtc::ArrayView<float> channel_view = float_frame.channel(k);
+    for (size_t k = 0; k < float_frame.num_channels(); ++k) {
+      MonoView<float> channel_view = float_frame[k];
       for (auto& sample : channel_view) {
         sample *= gain_at_end_of_frame_linear;
       }
@@ -57,12 +57,12 @@ void ApplyGainWithRamping(float last_gain_linear,
   // The gain changes. We have to change slowly to avoid discontinuities.
   const float increment = (gain_at_end_of_frame_linear - last_gain_linear) *
                           inverse_samples_per_channel;
-  float gain = last_gain_linear;
-  for (int i = 0; i < float_frame.samples_per_channel(); ++i) {
-    for (int ch = 0; ch < float_frame.num_channels(); ++ch) {
-      float_frame.channel(ch)[i] *= gain;
+  for (size_t ch = 0; ch < float_frame.num_channels(); ++ch) {
+    float gain = last_gain_linear;
+    for (float& sample : float_frame[ch]) {
+      sample *= gain;
+      gain += increment;
     }
-    gain += increment;
   }
 }
 
@@ -73,7 +73,7 @@ GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
       last_gain_factor_(initial_gain_factor),
       current_gain_factor_(initial_gain_factor) {}
 
-void GainApplier::ApplyGain(AudioFrameView<float> signal) {
+void GainApplier::ApplyGain(DeinterleavedView<float> signal) {
   if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) {
     Initialize(signal.samples_per_channel());
   }
diff --git a/modules/audio_processing/agc2/gain_applier.h b/modules/audio_processing/agc2/gain_applier.h
index ba8a4a4cd2..82ae82eeef 100644
--- a/modules/audio_processing/agc2/gain_applier.h
+++ b/modules/audio_processing/agc2/gain_applier.h
@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 
+#include "api/audio/audio_view.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
 
 namespace webrtc {
@@ -20,10 +21,15 @@ class GainApplier {
  public:
   GainApplier(bool hard_clip_samples, float initial_gain_factor);
 
-  void ApplyGain(AudioFrameView<float> signal);
+  void ApplyGain(DeinterleavedView<float> signal);
   void SetGainFactor(float gain_factor);
   float GetGainFactor() const { return current_gain_factor_; }
 
+  [[deprecated("Use DeinterleavedView<> version")]] void ApplyGain(
+      AudioFrameView<float> signal) {
+    ApplyGain(signal.view());
+  }
+
  private:
   void Initialize(int samples_per_channel);
 
diff --git a/modules/audio_processing/agc2/gain_applier_unittest.cc b/modules/audio_processing/agc2/gain_applier_unittest.cc
index 3296345e62..7548faa61b 100644
--- a/modules/audio_processing/agc2/gain_applier_unittest.cc
+++ b/modules/audio_processing/agc2/gain_applier_unittest.cc
@@ -15,6 +15,7 @@
 #include <algorithm>
 #include <limits>
 
+#include "api/audio/audio_view.h"
 #include "modules/audio_processing/agc2/vector_float_frame.h"
 #include "rtc_base/gunit.h"
 
@@ -25,9 +26,9 @@ TEST(AutomaticGainController2GainApplier, InitialGainIsRespected) {
   VectorFloatFrame fake_audio(1, 1, initial_signal_level);
   GainApplier gain_applier(true, gain_factor);
 
-  gain_applier.ApplyGain(fake_audio.float_frame_view());
-  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
-              initial_signal_level * gain_factor, 0.1f);
+  auto fake_view = fake_audio.view();
+  gain_applier.ApplyGain(fake_audio.view());
+  EXPECT_NEAR(fake_view[0][0], initial_signal_level * gain_factor, 0.1f);
 }
 
 TEST(AutomaticGainController2GainApplier, ClippingIsDone) {
@@ -36,9 +37,9 @@ TEST(AutomaticGainController2GainApplier, ClippingIsDone) {
   VectorFloatFrame fake_audio(1, 1, initial_signal_level);
   GainApplier gain_applier(true, gain_factor);
 
-  gain_applier.ApplyGain(fake_audio.float_frame_view());
-  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
-              std::numeric_limits<int16_t>::max(), 0.1f);
+  gain_applier.ApplyGain(fake_audio.view());
+  EXPECT_NEAR(fake_audio.view()[0][0], std::numeric_limits<int16_t>::max(),
+              0.1f);
 }
 
 TEST(AutomaticGainController2GainApplier, ClippingIsNotDone) {
@@ -47,10 +48,10 @@ TEST(AutomaticGainController2GainApplier, ClippingIsNotDone) {
   VectorFloatFrame fake_audio(1, 1, initial_signal_level);
   GainApplier gain_applier(false, gain_factor);
 
-  gain_applier.ApplyGain(fake_audio.float_frame_view());
+  gain_applier.ApplyGain(fake_audio.view());
 
-  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
-              initial_signal_level * gain_factor, 0.1f);
+  EXPECT_NEAR(fake_audio.view()[0][0], initial_signal_level * gain_factor,
+              0.1f);
 }
 
 TEST(AutomaticGainController2GainApplier, RampingIsDone) {
@@ -64,13 +65,13 @@ TEST(AutomaticGainController2GainApplier, RampingIsDone) {
   GainApplier gain_applier(false, initial_gain_factor);
 
   gain_applier.SetGainFactor(target_gain_factor);
-  gain_applier.ApplyGain(fake_audio.float_frame_view());
+  gain_applier.ApplyGain(fake_audio.view());
 
   // The maximal gain change should be close to that in linear interpolation.
   for (size_t channel = 0; channel < num_channels; ++channel) {
     float max_signal_change = 0.f;
     float last_signal_level = initial_signal_level;
-    for (const auto sample : fake_audio.float_frame_view().channel(channel)) {
+    for (const auto sample : fake_audio.view()[channel]) {
       const float current_change = fabs(last_signal_level - sample);
       max_signal_change = std::max(max_signal_change, current_change);
       last_signal_level = sample;
@@ -84,10 +85,10 @@ TEST(AutomaticGainController2GainApplier, RampingIsDone) {
   // Next frame should have the desired level.
   VectorFloatFrame next_fake_audio_frame(num_channels, samples_per_channel,
                                          initial_signal_level);
-  gain_applier.ApplyGain(next_fake_audio_frame.float_frame_view());
+  gain_applier.ApplyGain(next_fake_audio_frame.view());
 
   // The last sample should have the new gain.
-  EXPECT_NEAR(next_fake_audio_frame.float_frame_view().channel(0)[0],
+  EXPECT_NEAR(next_fake_audio_frame.view()[0][0],
               initial_signal_level * target_gain_factor, 0.1f);
 }
 }  // namespace webrtc
diff --git a/modules/audio_processing/agc2/noise_level_estimator.cc b/modules/audio_processing/agc2/noise_level_estimator.cc
index 691513b509..c43738aad3 100644
--- a/modules/audio_processing/agc2/noise_level_estimator.cc
+++ b/modules/audio_processing/agc2/noise_level_estimator.cc
@@ -16,7 +16,7 @@
 #include <cmath>
 #include <numeric>
 
-#include "api/array_view.h"
+#include "api/audio/audio_view.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/checks.h"
 
@@ -25,11 +25,12 @@ namespace {
 
 constexpr int kFramesPerSecond = 100;
 
-float FrameEnergy(const AudioFrameView<const float>& audio) {
+float FrameEnergy(DeinterleavedView<const float> audio) {
   float energy = 0.0f;
-  for (int k = 0; k < audio.num_channels(); ++k) {
+  for (size_t k = 0; k < audio.num_channels(); ++k) {
+    MonoView<const float> ch = audio[k];
     float channel_energy =
-        std::accumulate(audio.channel(k).begin(), audio.channel(k).end(), 0.0f,
+        std::accumulate(ch.begin(), ch.end(), 0.0f,
                         [](float a, float b) -> float { return a + b * b; });
     energy = std::max(channel_energy, energy);
   }
@@ -81,7 +82,7 @@ class NoiseFloorEstimator : public NoiseLevelEstimator {
   NoiseFloorEstimator& operator=(const NoiseFloorEstimator&) = delete;
   ~NoiseFloorEstimator() = default;
 
-  float Analyze(const AudioFrameView<const float>& frame) override {
+  float Analyze(DeinterleavedView<const float> frame) override {
     // Detect sample rate changes.
     const int sample_rate_hz =
         static_cast<int>(frame.samples_per_channel() * kFramesPerSecond);
diff --git a/modules/audio_processing/agc2/noise_level_estimator.h b/modules/audio_processing/agc2/noise_level_estimator.h
index 9f3b957486..8df4cbc93d 100644
--- a/modules/audio_processing/agc2/noise_level_estimator.h
+++ b/modules/audio_processing/agc2/noise_level_estimator.h
@@ -13,7 +13,7 @@
 
 #include <memory>
 
-#include "modules/audio_processing/include/audio_frame_view.h"
+#include "api/audio/audio_view.h"
 
 namespace webrtc {
 class ApmDataDumper;
@@ -24,7 +24,7 @@ class NoiseLevelEstimator {
   virtual ~NoiseLevelEstimator() = default;
   // Analyzes a 10 ms `frame`, updates the noise level estimation and returns
   // the value for the latter in dBFS.
-  virtual float Analyze(const AudioFrameView<const float>& frame) = 0;
+  virtual float Analyze(DeinterleavedView<const float> frame) = 0;
 };
 
 // Creates a noise level estimator based on noise floor detection.
diff --git a/modules/audio_processing/agc2/noise_level_estimator_unittest.cc b/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
index 8168c5a229..9d42bfc0fb 100644
--- a/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
+++ b/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
@@ -15,6 +15,7 @@
 #include <functional>
 #include <limits>
 
+#include "api/audio/audio_view.h"
 #include "api/function_view.h"
 #include "modules/audio_processing/agc2/agc2_testing_common.h"
 #include "modules/audio_processing/agc2/vector_float_frame.h"
@@ -36,13 +37,13 @@ float RunEstimator(rtc::FunctionView<float()> sample_generator,
       rtc::CheckedDivExact(sample_rate_hz, kFramesPerSecond);
   VectorFloatFrame signal(1, samples_per_channel, 0.0f);
   for (int i = 0; i < kNumIterations; ++i) {
-    AudioFrameView<float> frame_view = signal.float_frame_view();
+    DeinterleavedView<float> frame_view = signal.view();
     for (int j = 0; j < samples_per_channel; ++j) {
-      frame_view.channel(0)[j] = sample_generator();
+      frame_view[0][j] = sample_generator();
     }
     estimator.Analyze(frame_view);
   }
-  return estimator.Analyze(signal.float_frame_view());
+  return estimator.Analyze(signal.view());
 }
 
 class NoiseEstimatorParametrization : public ::testing::TestWithParam<int> {
diff --git a/modules/audio_processing/agc2/vad_wrapper.cc b/modules/audio_processing/agc2/vad_wrapper.cc
index 3bafdc7c56..8de8abd315 100644
--- a/modules/audio_processing/agc2/vad_wrapper.cc
+++ b/modules/audio_processing/agc2/vad_wrapper.cc
@@ -13,7 +13,6 @@
 #include <array>
 #include <utility>
 
-#include "api/array_view.h"
 #include "common_audio/resampler/include/push_resampler.h"
 #include "modules/audio_processing/agc2/agc2_common.h"
 #include "modules/audio_processing/agc2/rnn_vad/common.h"
@@ -36,7 +35,7 @@ class MonoVadImpl : public VoiceActivityDetectorWrapper::MonoVad {
 
   int SampleRateHz() const override { return rnn_vad::kSampleRate24kHz; }
   void Reset() override { rnn_vad_.Reset(); }
-  float Analyze(rtc::ArrayView<const float> frame) override {
+  float Analyze(MonoView<const float> frame) override {
     RTC_DCHECK_EQ(frame.size(), rnn_vad::kFrameSize10ms24kHz);
     std::array<float, rnn_vad::kFeatureVectorSize> feature_vector;
     const bool is_silence = features_extractor_.CheckSilenceComputeFeatures(
@@ -87,7 +86,8 @@ VoiceActivityDetectorWrapper::VoiceActivityDetectorWrapper(
 
 VoiceActivityDetectorWrapper::~VoiceActivityDetectorWrapper() = default;
 
-float VoiceActivityDetectorWrapper::Analyze(AudioFrameView<const float> frame) {
+float VoiceActivityDetectorWrapper::Analyze(
+    DeinterleavedView<const float> frame) {
   // Periodically reset the VAD.
   time_to_vad_reset_--;
   if (time_to_vad_reset_ <= 0) {
@@ -98,7 +98,7 @@ float VoiceActivityDetectorWrapper::Analyze(AudioFrameView<const float> frame) {
   // Resample the first channel of `frame`.
   RTC_DCHECK_EQ(frame.samples_per_channel(), frame_size_);
   MonoView<float> dst(resampled_buffer_.data(), resampled_buffer_.size());
-  resampler_.Resample(frame.channel(0), dst);
+  resampler_.Resample(frame[0], dst);
 
   return vad_->Analyze(resampled_buffer_);
 }
diff --git a/modules/audio_processing/agc2/vad_wrapper.h b/modules/audio_processing/agc2/vad_wrapper.h
index 8751dbaf75..025a48ef22 100644
--- a/modules/audio_processing/agc2/vad_wrapper.h
+++ b/modules/audio_processing/agc2/vad_wrapper.h
@@ -14,10 +14,9 @@
 #include <memory>
 #include <vector>
 
-#include "api/array_view.h"
+#include "api/audio/audio_view.h"
 #include "common_audio/resampler/include/push_resampler.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
-#include "modules/audio_processing/include/audio_frame_view.h"
 
 namespace webrtc {
 
@@ -37,7 +36,7 @@ class VoiceActivityDetectorWrapper {
     // Resets the internal state.
     virtual void Reset() = 0;
     // Analyzes an audio frame and returns the speech probability.
-    virtual float Analyze(rtc::ArrayView<const float> frame) = 0;
+    virtual float Analyze(MonoView<const float> frame) = 0;
   };
 
   // Ctor. Uses `cpu_features` to instantiate the default VAD.
@@ -63,7 +62,7 @@ class VoiceActivityDetectorWrapper {
   // Analyzes the first channel of `frame` and returns the speech probability.
   // `frame` must be a 10 ms frame with the sample rate specified in the last
   // `Initialize()` call.
-  float Analyze(AudioFrameView<const float> frame);
+  float Analyze(DeinterleavedView<const float> frame);
 
  private:
   const int vad_reset_period_frames_;
diff --git a/modules/audio_processing/agc2/vad_wrapper_unittest.cc b/modules/audio_processing/agc2/vad_wrapper_unittest.cc
index 6f66560060..9d8761d23e 100644
--- a/modules/audio_processing/agc2/vad_wrapper_unittest.cc
+++ b/modules/audio_processing/agc2/vad_wrapper_unittest.cc
@@ -16,8 +16,8 @@
 #include <utility>
 #include <vector>
 
+#include "api/audio/audio_view.h"
 #include "modules/audio_processing/agc2/agc2_common.h"
-#include "modules/audio_processing/include/audio_frame_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/gunit.h"
 #include "rtc_base/numerics/safe_compare.h"
@@ -85,11 +85,9 @@ struct FrameWithView {
   explicit FrameWithView(int sample_rate_hz)
       : samples(rtc::CheckedDivExact(sample_rate_hz, kNumFramesPerSecond),
                 0.0f),
-        channel0(samples.data()),
-        view(&channel0, /*num_channels=*/1, samples.size()) {}
+        view(samples.data(), samples.size(), /*num_channels=*/1) {}
   std::vector<float> samples;
-  const float* const channel0;
-  const AudioFrameView<const float> view;
+  const DeinterleavedView<const float> view;
 };
 
 // Checks that the expected speech probabilities are returned.
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index f0625c5dc2..2fb021c7c3 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -1473,8 +1473,8 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
 
     absl::optional<float> voice_probability;
     if (!!submodules_.voice_activity_detector) {
-      voice_probability = submodules_.voice_activity_detector->Analyze(
-          AudioFrameView<const float>(capture_buffer->view()));
+      voice_probability =
+          submodules_.voice_activity_detector->Analyze(capture_buffer->view());
     }
 
     if (submodules_.transient_suppressor) {
diff --git a/modules/audio_processing/gain_controller2.cc b/modules/audio_processing/gain_controller2.cc
index 9b19bc8517..9bfae527d7 100644
--- a/modules/audio_processing/gain_controller2.cc
+++ b/modules/audio_processing/gain_controller2.cc
@@ -64,11 +64,11 @@ struct SpeechLevel {
 };
 
 // Computes the audio levels for the first channel in `frame`.
-AudioLevels ComputeAudioLevels(AudioFrameView<float> frame,
+AudioLevels ComputeAudioLevels(DeinterleavedView<float> frame,
                                ApmDataDumper& data_dumper) {
   float peak = 0.0f;
   float rms = 0.0f;
-  for (const auto& x : frame.channel(0)) {
+  for (const auto& x : frame[0]) {
     peak = std::max(std::fabs(x), peak);
     rms += x * x;
   }
@@ -182,8 +182,8 @@ void GainController2::Process(absl::optional<float> speech_probability,
       saturation_protector_->Reset();
   }
 
-  AudioFrameView<float> float_frame(audio->channels(), audio->num_channels(),
-                                    audio->num_frames());
+  DeinterleavedView<float> float_frame = audio->view();
+
   // Compute speech probability.
   if (vad_) {
     // When the VAD component runs, `speech_probability` should not be specified
@@ -258,7 +258,7 @@ void GainController2::Process(absl::optional<float> speech_probability,
   // computation in `limiter_`.
   fixed_gain_applier_.ApplyGain(float_frame);
 
-  limiter_.Process(float_frame.view());
+  limiter_.Process(float_frame);
 
   // Periodically log limiter stats.
   if (++calls_since_last_limiter_log_ == kLogLimiterStatsPeriodNumFrames) {
diff --git a/modules/audio_processing/gain_controller2_unittest.cc b/modules/audio_processing/gain_controller2_unittest.cc
index 71642dc38c..bccab8d060 100644
--- a/modules/audio_processing/gain_controller2_unittest.cc
+++ b/modules/audio_processing/gain_controller2_unittest.cc
@@ -16,7 +16,6 @@
 #include <numeric>
 #include <tuple>
 
-#include "api/array_view.h"
 #include "modules/audio_processing/agc2/agc2_testing_common.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_processing/test/audio_buffer_tools.h"
@@ -596,9 +595,7 @@ TEST(GainController2,
     agc2_reference.Process(absl::nullopt, /*input_volume_changed=*/false,
                            &audio_buffer_reference);
     test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
-    float speech_probability = vad.Analyze(AudioFrameView<const float>(
-        audio_buffer.channels(), audio_buffer.num_channels(),
-        audio_buffer.num_frames()));
+    float speech_probability = vad.Analyze(audio_buffer.view());
     agc2.Process(speech_probability, /*input_volume_changed=*/false,
                  &audio_buffer);
     // Check the output buffer.