diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index 2556bcda31..2746e61580 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -20,7 +20,8 @@ namespace {
 enum {
   kSamplesPer8kHzChannel = 80,
   kSamplesPer16kHzChannel = 160,
-  kSamplesPer32kHzChannel = 320
+  kSamplesPer32kHzChannel = 320,
+  kSamplesPer48kHzChannel = 480
 };
 
 bool HasKeyboardChannel(AudioProcessing::ChannelLayout layout) {
@@ -171,13 +172,18 @@ AudioBuffer::AudioBuffer(int input_samples_per_channel,
     }
   }
 
-  if (proc_samples_per_channel_ == kSamplesPer32kHzChannel) {
+  if (proc_samples_per_channel_ == kSamplesPer32kHzChannel ||
+      proc_samples_per_channel_ == kSamplesPer48kHzChannel) {
     samples_per_split_channel_ = kSamplesPer16kHzChannel;
     split_channels_low_.reset(new IFChannelBuffer(samples_per_split_channel_,
                                                   num_proc_channels_));
     split_channels_high_.reset(new IFChannelBuffer(samples_per_split_channel_,
                                                    num_proc_channels_));
     splitting_filter_.reset(new SplittingFilter(num_proc_channels_));
+    if (proc_samples_per_channel_ == kSamplesPer48kHzChannel) {
+      split_channels_super_high_.reset(
+          new IFChannelBuffer(samples_per_split_channel_, num_proc_channels_));
+    }
   }
 }
 
@@ -391,6 +397,18 @@ float* const* AudioBuffer::high_pass_split_channels_f() {
       : NULL;
 }
 
+const float* const* AudioBuffer::super_high_pass_split_channels_f() const {
+  return split_channels_super_high_.get()
+      ? split_channels_super_high_->fbuf_const()->channels()
+      : NULL;
+}
+
+float* const* AudioBuffer::super_high_pass_split_channels_f() {
+  return split_channels_super_high_.get()
+      ? split_channels_super_high_->fbuf()->channels()
+      : NULL;
+}
+
 const int16_t* AudioBuffer::mixed_low_pass_data() {
   // Currently only mixing stereo to mono is supported.
   assert(num_proc_channels_ == 1 || num_proc_channels_ == 2);
@@ -513,15 +531,29 @@ void AudioBuffer::CopyLowPassToReference() {
 }
 
 void AudioBuffer::SplitIntoFrequencyBands() {
-  splitting_filter_->TwoBandsAnalysis(
-      channels(), samples_per_channel(), num_proc_channels_,
-      low_pass_split_channels(), high_pass_split_channels());
+  if (samples_per_channel() == kSamplesPer32kHzChannel) {
+    splitting_filter_->TwoBandsAnalysis(
+        channels(), samples_per_channel(), num_proc_channels_,
+        low_pass_split_channels(), high_pass_split_channels());
+  } else if (samples_per_channel() == kSamplesPer48kHzChannel) {
+    splitting_filter_->ThreeBandsAnalysis(
+        channels_f(), samples_per_channel(), num_proc_channels_,
+        low_pass_split_channels_f(), high_pass_split_channels_f(),
+        super_high_pass_split_channels_f());
+  }
 }
 
 void AudioBuffer::MergeFrequencyBands() {
-  splitting_filter_->TwoBandsSynthesis(
-      low_pass_split_channels(), high_pass_split_channels(),
-      samples_per_split_channel(), num_proc_channels_, channels());
+  if (samples_per_channel() == kSamplesPer32kHzChannel) {
+    splitting_filter_->TwoBandsSynthesis(
+        low_pass_split_channels(), high_pass_split_channels(),
+        samples_per_split_channel(), num_proc_channels_, channels());
+  } else if (samples_per_channel() == kSamplesPer48kHzChannel) {
+    splitting_filter_->ThreeBandsSynthesis(
+        low_pass_split_channels_f(), high_pass_split_channels_f(),
+        super_high_pass_split_channels_f(), samples_per_split_channel(),
+        num_proc_channels_, channels_f());
+  }
 }
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index 7faa8a18c4..471fe7bc31 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -78,6 +78,8 @@ class AudioBuffer {
   const float* const* low_pass_split_channels_f() const;
   float* const* high_pass_split_channels_f();
   const float* const* high_pass_split_channels_f() const;
+  float* const* super_high_pass_split_channels_f();
+  const float* const* super_high_pass_split_channels_f() const;
 
   const float* keyboard_data() const;
 
@@ -122,6 +124,7 @@ class AudioBuffer {
   scoped_ptr<IFChannelBuffer> channels_;
   scoped_ptr<IFChannelBuffer> split_channels_low_;
   scoped_ptr<IFChannelBuffer> split_channels_high_;
+  scoped_ptr<IFChannelBuffer> split_channels_super_high_;
   scoped_ptr<SplittingFilter> splitting_filter_;
   scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
   scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 61a6f00f92..1b16c7c609 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -262,7 +262,8 @@ int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz,
   // demonstrated to work well for AEC in most practical scenarios.
   rev_proc_format_.set(rev_proc_rate, 1);
 
-  if (fwd_proc_format_.rate() == kSampleRate32kHz) {
+  if (fwd_proc_format_.rate() == kSampleRate32kHz ||
+      fwd_proc_format_.rate() == kSampleRate48kHz) {
     split_rate_ = kSampleRate16kHz;
   } else {
     split_rate_ = fwd_proc_format_.rate();
@@ -404,7 +405,8 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
   // Must be a native rate.
   if (frame->sample_rate_hz_ != kSampleRate8kHz &&
       frame->sample_rate_hz_ != kSampleRate16kHz &&
-      frame->sample_rate_hz_ != kSampleRate32kHz) {
+      frame->sample_rate_hz_ != kSampleRate32kHz &&
+      frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
   }
   if (echo_control_mobile_->is_enabled() &&
@@ -540,7 +542,8 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
   // Must be a native rate.
   if (frame->sample_rate_hz_ != kSampleRate8kHz &&
       frame->sample_rate_hz_ != kSampleRate16kHz &&
-      frame->sample_rate_hz_ != kSampleRate32kHz) {
+      frame->sample_rate_hz_ != kSampleRate32kHz &&
+      frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
   }
   // This interface does not tolerate different forward and reverse rates.
@@ -775,14 +778,16 @@ bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
 }
 
 bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const {
-  return (is_data_processed && fwd_proc_format_.rate() == kSampleRate32kHz);
+  return (is_data_processed && (fwd_proc_format_.rate() == kSampleRate32kHz ||
+          fwd_proc_format_.rate() == kSampleRate48kHz));
 }
 
 bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
   if (!is_data_processed && !voice_detection_->is_enabled()) {
     // Only level_estimator_ is enabled.
     return false;
-  } else if (fwd_proc_format_.rate() == kSampleRate32kHz) {
+  } else if (fwd_proc_format_.rate() == kSampleRate32kHz ||
+             fwd_proc_format_.rate() == kSampleRate48kHz) {
     // Something besides level_estimator_ is enabled, and we have super-wb.
     return true;
   }
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 10d2b8b8fc..1cca0edf57 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -380,7 +380,8 @@ class AudioProcessing {
   enum NativeRate {
     kSampleRate8kHz = 8000,
     kSampleRate16kHz = 16000,
-    kSampleRate32kHz = 32000
+    kSampleRate32kHz = 32000,
+    kSampleRate48kHz = 48000
   };
 
   static const int kChunkSizeMs = 10;
diff --git a/webrtc/modules/audio_processing/splitting_filter.cc b/webrtc/modules/audio_processing/splitting_filter.cc
index d2b5cc072e..8765dc987f 100644
--- a/webrtc/modules/audio_processing/splitting_filter.cc
+++ b/webrtc/modules/audio_processing/splitting_filter.cc
@@ -46,4 +46,20 @@ void SplittingFilter::TwoBandsSynthesis(const int16_t* const* low_band,
   }
 }
 
+void SplittingFilter::ThreeBandsAnalysis(const float* const* in_data,
+                                         int in_data_length,
+                                         int channels,
+                                         float* const* low_band,
+                                         float* const* high_band,
+                                         float* const* super_high_band) {
+}
+
+void SplittingFilter::ThreeBandsSynthesis(const float* const* low_band,
+                                          const float* const* high_band,
+                                          const float* const* super_high_band,
+                                          int band_length,
+                                          int channels,
+                                          float* const* out_data) {
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_processing/splitting_filter.h b/webrtc/modules/audio_processing/splitting_filter.h
index d316d9519a..1e9269ef0d 100644
--- a/webrtc/modules/audio_processing/splitting_filter.h
+++ b/webrtc/modules/audio_processing/splitting_filter.h
@@ -47,6 +47,18 @@ class SplittingFilter {
                          int band_length,
                          int channels,
                          int16_t* const* out_data);
+  void ThreeBandsAnalysis(const float* const* in_data,
+                          int in_data_length,
+                          int channels,
+                          float* const* low_band,
+                          float* const* high_band,
+                          float* const* super_high_band);
+  void ThreeBandsSynthesis(const float* const* low_band,
+                           const float* const* high_band,
+                           const float* const* super_high_band,
+                           int band_length,
+                           int channels,
+                           float* const* out_data);
 
  private:
   int channels_;