diff --git a/src/host/asio/stream.rs b/src/host/asio/stream.rs
index e64459f57..f99bc6ff0 100644
--- a/src/host/asio/stream.rs
+++ b/src/host/asio/stream.rs
@@ -993,6 +993,7 @@ fn check_config(
         channels,
         sample_rate,
         buffer_size,
+        ..
     } = config;
 
     // Validate buffer size if `Fixed` is specified. This is necessary because ASIO's
diff --git a/src/host/wasapi/device.rs b/src/host/wasapi/device.rs
index 563dd9249..53d636147 100644
--- a/src/host/wasapi/device.rs
+++ b/src/host/wasapi/device.rs
@@ -1,10 +1,10 @@
 use crate::{
     error::ResultExt,
     host::{com::ComString, ErrorCallbackArc},
-    BufferSize, Data, DeviceDescription, DeviceDescriptionBuilder, DeviceDirection, DeviceId,
-    DeviceType, Error, ErrorKind, FrameCount, InputCallbackInfo, InterfaceType, OutputCallbackInfo,
-    SampleFormat, SampleRate, StreamConfig, SupportedBufferSize, SupportedStreamConfig,
-    SupportedStreamConfigRange, COMMON_SAMPLE_RATES,
+    AudioProcessing, BufferSize, Data, DeviceDescription, DeviceDescriptionBuilder,
+    DeviceDirection, DeviceId, DeviceType, Error, ErrorKind, FrameCount, InputCallbackInfo,
+    InterfaceType, OutputCallbackInfo, SampleFormat, SampleRate, StreamConfig, SupportedBufferSize,
+    SupportedStreamConfig, SupportedStreamConfigRange, COMMON_SAMPLE_RATES,
 };
 
 impl From<Audio::EDataFlow> for DeviceDirection {
@@ -665,6 +665,7 @@ impl Device {
                             channels: format.channels,
                             sample_rate,
                             buffer_size: BufferSize::Default,
+                            audio_processing: AudioProcessing::Default,
                         },
                         sample_format,
                     ) {
@@ -783,6 +784,18 @@ impl Device {
                 .build_audioclient(activation_timeout)
                 .context("failed to build audio client")?;
 
+            if config.audio_processing == AudioProcessing::PreferRaw {
+                if let Ok(audio_client2) = audio_client.cast::<Audio::IAudioClient2>() {
+                    // Disable audio processing if `PreferRaw` is set and the interface supports it.
+                    let props = Audio::AudioClientProperties {
+                        cbSize: std::mem::size_of::<Audio::AudioClientProperties>() as u32,
+                        Options: Audio::AUDCLNT_STREAMOPTIONS_RAW,
+                        ..Default::default()
+                    };
+                    audio_client2.SetClientProperties(&props).ok();
+                }
+            }
+
             // Note: Buffer size validation is not needed here - `IAudioClient::Initialize`
             // will return `AUDCLNT_E_BUFFER_SIZE_ERROR` if the buffer size is not supported.
             let buffer_duration = buffer_size_to_duration(&config.buffer_size, config.sample_rate);
@@ -898,6 +911,18 @@ impl Device {
                 .build_audioclient(activation_timeout)
                 .context("failed to build audio client")?;
 
+            if config.audio_processing == AudioProcessing::PreferRaw {
+                if let Ok(audio_client2) = audio_client.cast::<Audio::IAudioClient2>() {
+                    // Disable audio processing if `PreferRaw` is set and the interface supports it.
+                    let props = Audio::AudioClientProperties {
+                        cbSize: std::mem::size_of::<Audio::AudioClientProperties>() as u32,
+                        Options: Audio::AUDCLNT_STREAMOPTIONS_RAW,
+                        ..Default::default()
+                    };
+                    audio_client2.SetClientProperties(&props).ok();
+                }
+            }
+
             // Note: Buffer size validation is not needed here - `IAudioClient::Initialize`
             // will return `AUDCLNT_E_BUFFER_SIZE_ERROR` if the buffer size is not supported.
             let buffer_duration = buffer_size_to_duration(&config.buffer_size, config.sample_rate);
diff --git a/src/lib.rs b/src/lib.rs
index 37d0adbd8..f9e36ef9c 100644
--- a/src/lib.rs
+++ b/src/lib.rs
@@ -344,6 +344,22 @@ pub enum BufferSize {
     Fixed(FrameCount),
 }
 
+/// Specifies whether audio signal processing should be applied to the stream.
+///
+/// On Windows, the audio stack applies default signal processing (e.g. echo cancellation,
+/// noise suppression, automatic gain control) to audio streams. Using the [`AudioProcessing::PreferRaw`]
+/// option allows applications to request that such processing be bypassed.
+///
+/// If raw streams are not supported (e.g. on older Windows versions), the default behavior
+/// will be used. This option has no effect on platforms that do not apply system audio processing
+/// or do not provide a way to bypass it.
+#[derive(Clone, Copy, Debug, Default, PartialEq, Eq)]
+pub enum AudioProcessing {
+    #[default]
+    Default,
+    PreferRaw,
+}
+
 #[cfg(all(target_arch = "wasm32", feature = "wasm-bindgen"))]
 impl wasm_bindgen::describe::WasmDescribe for BufferSize {
     fn describe() {
@@ -387,6 +403,11 @@ pub struct StreamConfig {
     pub channels: ChannelCount,
     pub sample_rate: SampleRate,
     pub buffer_size: BufferSize,
+    #[cfg_attr(
+        all(target_arch = "wasm32", feature = "wasm-bindgen"),
+        wasm_bindgen(skip)
+    )]
+    pub audio_processing: AudioProcessing,
 }
 
 /// Describes the minimum and maximum supported buffer size for the device
@@ -498,6 +519,7 @@ impl SupportedStreamConfig {
             channels: self.channels,
             sample_rate: self.sample_rate,
             buffer_size: BufferSize::Default,
+            audio_processing: AudioProcessing::Default,
         }
     }
 }