diff --git a/DeepDrftPublic.Client/Clients/TrackMediaClient.cs b/DeepDrftPublic.Client/Clients/TrackMediaClient.cs
index d3bef1b..f38d1d7 100644
--- a/DeepDrftPublic.Client/Clients/TrackMediaClient.cs
+++ b/DeepDrftPublic.Client/Clients/TrackMediaClient.cs
@@ -11,12 +11,20 @@ public class TrackMediaResponse : IDisposable
 {
     public Stream Stream { get; }
     public long ContentLength { get; }
+
+    /// <summary>
+    /// The response media type (e.g. "audio/wav", "audio/mpeg"). Drives format-decoder
+    /// selection on the JS side. Falls back to "audio/wav" when the server omits the header.
+    /// </summary>
+    public string ContentType { get; }
+
     private readonly HttpResponseMessage _response;
 
-    public TrackMediaResponse(Stream stream, long contentLength, HttpResponseMessage response)
+    public TrackMediaResponse(Stream stream, long contentLength, string contentType, HttpResponseMessage response)
     {
         Stream = stream;
         ContentLength = contentLength;
+        ContentType = contentType;
         _response = response;
     }
 
@@ -61,11 +69,14 @@ public class TrackMediaClient
             response.EnsureSuccessStatusCode();
 
             var contentLength = response.Content.Headers.ContentLength ?? 0;
+            // Default to WAV when the server omits the header — the only format shipping
+            // today — so the JS factory always receives a usable media type.
+            var contentType = response.Content.Headers.ContentType?.MediaType ?? "audio/wav";
             var stream = await response.Content.ReadAsStreamAsync(cancellationToken);
 
             // TrackMediaResponse takes ownership of both stream and response;
             // do NOT dispose response here — the caller disposes via TrackMediaResponse.Dispose().
-            return ApiResult<TrackMediaResponse>.CreatePassResult(new TrackMediaResponse(stream, contentLength, response));
+            return ApiResult<TrackMediaResponse>.CreatePassResult(new TrackMediaResponse(stream, contentLength, contentType, response));
         }
         catch (Exception e)
         {
diff --git a/DeepDrftPublic.Client/Services/AudioInteropService.cs b/DeepDrftPublic.Client/Services/AudioInteropService.cs
index f27fe3f..f7e1f1f 100644
--- a/DeepDrftPublic.Client/Services/AudioInteropService.cs
+++ b/DeepDrftPublic.Client/Services/AudioInteropService.cs
@@ -65,9 +65,9 @@ public class AudioInteropService : IAsyncDisposable
     }
 
     // Streaming methods
-    public async Task<AudioOperationResult> InitializeStreaming(string playerId, long totalStreamLength)
+    public async Task<AudioOperationResult> InitializeStreaming(string playerId, long totalStreamLength, string contentType)
     {
-        return await InvokeJsAsync<AudioOperationResult>("DeepDrftAudio.initializeStreaming", playerId, totalStreamLength);
+        return await InvokeJsAsync<AudioOperationResult>("DeepDrftAudio.initializeStreaming", playerId, totalStreamLength, contentType);
     }
 
     public async Task<StreamingResult> ProcessStreamingChunk(string playerId, byte[] audioChunk)
diff --git a/DeepDrftPublic.Client/Services/StreamingAudioPlayerService.cs b/DeepDrftPublic.Client/Services/StreamingAudioPlayerService.cs
index 7568df8..774d005 100644
--- a/DeepDrftPublic.Client/Services/StreamingAudioPlayerService.cs
+++ b/DeepDrftPublic.Client/Services/StreamingAudioPlayerService.cs
@@ -143,8 +143,9 @@ public class StreamingAudioPlayerService : AudioPlayerService, IStreamingPlayerS
 
             using var audio = mediaResult.Value;
 
-            // Initialize streaming mode with content length
-            var streamingResult = await _audioInterop.InitializeStreaming(PlayerId, audio.ContentLength);
+            // Initialize streaming mode with content length and media type (drives
+            // JS format-decoder selection).
+            var streamingResult = await _audioInterop.InitializeStreaming(PlayerId, audio.ContentLength, audio.ContentType);
             if (!streamingResult.Success)
             {
                 var technicalError = $"Failed to initialize streaming: {streamingResult.Error}";
diff --git a/DeepDrftPublic/Interop/audio/AudioPlayer.ts b/DeepDrftPublic/Interop/audio/AudioPlayer.ts
index 17f2617..92bdbf7 100644
--- a/DeepDrftPublic/Interop/audio/AudioPlayer.ts
+++ b/DeepDrftPublic/Interop/audio/AudioPlayer.ts
@@ -10,6 +10,8 @@
 import { AudioContextManager } from './AudioContextManager.js';
 import { StreamDecoder } from './StreamDecoder.js';
 import { PlaybackScheduler } from './PlaybackScheduler.js';
+import { IFormatDecoder } from './IFormatDecoder.js';
+import { WavFormatDecoder } from './WavFormatDecoder.js';
 
 export interface AudioResult {
     success: boolean;
@@ -89,7 +91,7 @@ export class AudioPlayer {
 
     // ==================== Streaming ====================
 
-    initializeStreaming(totalStreamLength: number): AudioResult {
+    initializeStreaming(totalStreamLength: number, contentType: string): AudioResult {
         try {
             // Full cleanup before starting new stream
             this.stopProgressTracking();
@@ -97,15 +99,26 @@ export class AudioPlayer {
             this.streamDecoder.reset();
             this.resetState();
 
-            // Initialize new stream
+            // Initialize new stream with the format decoder selected from Content-Type.
             this.isStreamingMode = true;
-            this.streamDecoder.initialize(totalStreamLength);
+            const formatDecoder = AudioPlayer.createFormatDecoder(contentType);
+            this.streamDecoder.initialize(totalStreamLength, formatDecoder);
             return { success: true };
         } catch (error) {
             return { success: false, error: (error as Error).message };
         }
     }
 
+    /**
+     * Select a format decoder from the response Content-Type. MP3 and FLAC decoders
+     * arrive in Wave 2; until then every format falls back to WAV. When Wave 2 lands,
+     * add the MP3/FLAC branches here, e.g.:
+     *   if (contentType.includes('audio/mpeg')) return new Mp3FormatDecoder();
+     */
+    private static createFormatDecoder(_contentType: string): IFormatDecoder {
+        return new WavFormatDecoder();
+    }
+
     /**
      * Signal to the decoder that the C# streaming loop has finished sending bytes.
      * This sets streamComplete=true and flushes any remaining decoded tail segments.
@@ -321,22 +334,16 @@ export class AudioPlayer {
      */
     private seekBeyondBuffer(position: number): AudioResult {
         try {
-            const audioOffset = this.streamDecoder.calculateByteOffset(position);
-            // 0 is a valid offset (seek to start of audio data). Only a negative result
-            // indicates calculation failure — typically a missing/unparsed WAV header.
-            if (audioOffset < 0) {
+            // The header must be parsed for byte-offset math; without it we cannot
+            // build a valid Range request.
+            if (!this.streamDecoder.getFormatInfo()) {
                 return { success: false, error: 'Cannot calculate byte offset' };
             }
 
-            // The Range request is file-absolute: byte position from the start of the
-            // file on disk, header included. calculateByteOffset returns an audio-data-
-            // relative offset, so add headerSize to land on the right byte. (The old
-            // ?offset= contract was audio-relative; the server added the header itself.)
-            const header = this.streamDecoder.getWavHeader();
-            if (!header) {
-                return { success: false, error: 'Cannot calculate byte offset' };
-            }
-            const fileOffset = header.headerSize + audioOffset;
+            // calculateByteOffset returns a file-absolute offset (byte position from the
+            // start of the file on disk, header included) — exactly what the Range request
+            // needs. The format decoder owns the header-offset addition and frame alignment.
+            const fileOffset = this.streamDecoder.calculateByteOffset(position);
 
             // Signal that C# needs to request a new stream from this file-absolute offset
             return {
diff --git a/DeepDrftPublic/Interop/audio/IFormatDecoder.ts b/DeepDrftPublic/Interop/audio/IFormatDecoder.ts
new file mode 100644
index 0000000..23c1135
--- /dev/null
+++ b/DeepDrftPublic/Interop/audio/IFormatDecoder.ts
@@ -0,0 +1,100 @@
+/**
+ * FormatInfo: parsed header data needed to stream and seek an audio file.
+ * Populated by IFormatDecoder.tryParseHeader; used by StreamDecoder throughout playback.
+ */
+export interface FormatInfo {
+    /** Samples per second (e.g. 44100). */
+    sampleRate: number;
+    /** Number of audio channels. */
+    channels: number;
+    /**
+     * Nominal bit depth — 16 for MP3 (conventional), 16/24/32 for WAV/FLAC.
+     * Used for display; decoders handle the actual sample format internally.
+     */
+    bitsPerSample: number;
+    /**
+     * Average bytes per second. Used for CBR byte-offset estimation.
+     * For WAV: exact (sampleRate * blockAlign).
+     * For MP3 CBR: bitrate_kbps * 125.
+     * For FLAC: approximate (fileSize / duration).
+     */
+    byteRate: number;
+    /**
+     * For WAV: PCM frame size in bytes (channels * bitsPerSample / 8).
+     * For MP3: frame size in bytes (constant for CBR, 0 for VBR with TOC).
+     * For FLAC: 0 (frame sizes vary; use sync scan instead).
+     * Used by getAlignedSegmentSize to round to clean frame boundaries.
+     */
+    blockAlign: number;
+    /** Total duration in seconds, from the header (null if unavailable). */
+    totalDuration: number | null;
+    /** Byte offset where audio frames begin in the original file. */
+    audioDataOffset: number;
+    /**
+     * Format-specific accelerator for seek-beyond-buffer byte calculation.
+     * WAV: null (uses byteRate/blockAlign directly).
+     * MP3 VBR: Xing/VBRI TOC (100-entry Uint8Array, values are file-percentage * 255).
+     * FLAC: SeekTable (array of {sampleNumber: number, streamOffset: number} — stream_offset
+     *        is bytes from the start of audio frames, i.e. after all metadata blocks).
+     */
+    seekData?: Mp3VbrSeekData | FlacSeekData | null;
+}
+
+export interface Mp3VbrSeekData {
+    kind: 'mp3-vbr';
+    toc: Uint8Array;           // 100 entries; toc[i] = file-byte-fraction at i% of duration
+    totalBytes: number;        // total audio bytes (from Xing header)
+}
+
+export interface FlacSeekData {
+    kind: 'flac-seektable';
+    points: Array<{ sampleNumber: number; streamOffset: number }>;
+    streamInfoBytes: Uint8Array; // raw STREAMINFO metadata block for segment wrapping
+    metadataBlocksSize: number;  // total bytes of all metadata blocks (for stream_offset relative math)
+}
+
+/**
+ * IFormatDecoder: per-format strategy for header parsing, segment boundary detection,
+ * segment wrapping, and seek offset calculation.
+ *
+ * Implementations: WavFormatDecoder (Wave 1), Mp3FormatDecoder (Wave 2), FlacFormatDecoder (Wave 2).
+ */
+export interface IFormatDecoder {
+    /**
+     * Attempt to parse the header from accumulated bytes. Returns null if more bytes are needed.
+     * Called with growing chunks array until it succeeds or exceeds MAX_HEADER_SEARCH_BYTES.
+     */
+    tryParseHeader(chunks: Uint8Array[], totalSize: number): FormatInfo | null;
+
+    /**
+     * Return the largest decodable byte count ≤ requestedSize that ends on a clean frame/block
+     * boundary in the audio data (post-header). Returns 0 if not enough data yet.
+     * @param info - the parsed FormatInfo
+     * @param availableBytes - bytes available starting at the current processedBytes position
+     * @param requestedSize - maximum desired segment size
+     * @param streamComplete - true when the stream has ended (allows draining the tail)
+     */
+    getAlignedSegmentSize(
+        info: FormatInfo,
+        availableBytes: number,
+        requestedSize: number,
+        streamComplete: boolean
+    ): number;
+
+    /**
+     * Wrap raw audio bytes in the minimal decodable container for decodeAudioData.
+     * WAV: prepend a 44-byte standard PCM header.
+     * MP3: pass through unchanged (raw frames are self-contained).
+     * FLAC: prepend fLaC marker + STREAMINFO metadata block.
+     */
+    wrapSegment(info: FormatInfo, rawBytes: Uint8Array): Uint8Array;
+
+    /**
+     * Calculate the file-absolute byte offset for a seek-beyond-buffer Range request.
+     * The returned value includes the header offset (result ≥ info.audioDataOffset).
+     * WAV: exact PCM frame alignment.
+     * MP3 CBR: frame-aligned estimate; MP3 VBR: TOC interpolation.
+     * FLAC: SEEKTABLE lookup when available.
+     */
+    calculateByteOffset(info: FormatInfo, positionSeconds: number): number;
+}
diff --git a/DeepDrftPublic/Interop/audio/StreamDecoder.ts b/DeepDrftPublic/Interop/audio/StreamDecoder.ts
index 73a6c3a..5288487 100644
--- a/DeepDrftPublic/Interop/audio/StreamDecoder.ts
+++ b/DeepDrftPublic/Interop/audio/StreamDecoder.ts
@@ -1,11 +1,15 @@
 /**
- * StreamDecoder - Handles WAV stream parsing and AudioBuffer decoding.
+ * StreamDecoder - Handles audio stream parsing and AudioBuffer decoding.
  *
- * Single Responsibility: Convert raw WAV stream data into decoded AudioBuffers.
+ * Single Responsibility: Convert a raw audio stream into decoded AudioBuffers.
+ * Format-specific work (header parsing, segment alignment, segment wrapping, seek
+ * byte math) is delegated to an IFormatDecoder supplied at initialize time; this
+ * class owns only the format-agnostic concerns: chunk accumulation, header search
+ * bounding, stream-complete detection, decode timeout/retry, and range continuation.
  */
 
-import { WavHeader, WavUtils } from '../wavutils.js';
 import { AudioContextManager } from './AudioContextManager.js';
+import { FormatInfo, IFormatDecoder } from './IFormatDecoder.js';
 
 export interface DecodedChunkResult {
     buffer: AudioBuffer;
@@ -43,13 +47,14 @@ export class DecodeError extends Error {
 }
 
 export class StreamDecoder {
-    // Upper bound on pre-header accumulation. 256 KB is far beyond any sane WAV
-    // header (including extended LIST/INFO/JUNK chunks). If we have accumulated
-    // this many bytes without finding a valid header the stream is corrupt.
+    // Upper bound on pre-header accumulation. 256 KB is far beyond any sane audio
+    // header (WAV with extended LIST/INFO/JUNK chunks, FLAC metadata blocks, etc.).
+    // If we have accumulated this many bytes without a valid header the stream is corrupt.
     private static readonly MAX_HEADER_SEARCH_BYTES = 256 * 1024;
 
     private contextManager: AudioContextManager;
-    private wavHeader: WavHeader | null = null;
+    private formatDecoder: IFormatDecoder | null = null;
+    private formatInfo: FormatInfo | null = null;
     private rawChunks: Uint8Array[] = [];
     // totalRawBytes and processedBytes are JS number (IEEE 754 double), which can
     // represent integers exactly up to 2^53 bytes (~8 PB). WAV files are bounded
@@ -61,17 +66,17 @@ export class StreamDecoder {
     private headerError: string | null = null;
 
     // Range-continuation state. After a seek-beyond-buffer the server responds 206
-    // with raw PCM from a file-absolute offset (no WAV header). We retain the header
+    // with raw audio from a file-absolute offset (no header). We retain the FormatInfo
     // parsed from the initial stream and treat the whole body as audio data. The
     // stream-complete check then counts raw bytes against the 206 Content-Length
-    // (remainingByteLength) rather than the full-file totalStreamLength + headerSize.
+    // (remainingByteLength) rather than the full-file totalStreamLength + audioDataOffset.
     private isContinuation: boolean = false;
     private remainingByteLength: number = 0;
 
-    // Pre-header accumulator. WAV headers can span multiple network chunks
-    // (small first segment, extended LIST/INFO/JUNK chunks before 'data', etc.),
-    // so we buffer raw bytes here until parseHeader succeeds rather than assuming
-    // the whole header lives in the first chunk.
+    // Pre-header accumulator. Audio headers can span multiple network chunks
+    // (small first segment, extended WAV LIST/INFO/JUNK chunks before 'data',
+    // FLAC metadata blocks, etc.), so we buffer raw bytes here until the format
+    // decoder parses a header rather than assuming it lives in the first chunk.
     private headerBytesReceived: number = 0;
     private headerSearchChunks: Uint8Array[] = [];
 
@@ -80,10 +85,12 @@ export class StreamDecoder {
     }
 
     /**
-     * Initialize for a new stream
+     * Initialize for a new stream. The format decoder owns all format-specific
+     * parsing/wrapping/seek math for this stream's lifetime.
      */
-    initialize(totalStreamLength: number): void {
-        this.wavHeader = null;
+    initialize(totalStreamLength: number, formatDecoder: IFormatDecoder): void {
+        this.formatDecoder = formatDecoder;
+        this.formatInfo = null;
         this.rawChunks = [];
         this.totalRawBytes = 0;
         this.processedBytes = 0;
@@ -105,12 +112,12 @@ export class StreamDecoder {
      * audio data to decode immediately.
      */
     async processChunk(chunk: Uint8Array): Promise<DecodedChunkResult[]> {
-        // If the header search already failed (corrupt/non-WAV stream), stop processing.
+        // If the header search already failed (corrupt/unrecognised stream), stop processing.
         if (this.headerError) {
             throw new Error(this.headerError);
         }
 
-        if (!this.wavHeader) {
+        if (!this.formatInfo) {
             await this.tryParseHeader(chunk);
             // Check again: tryParseHeader may have just set headerError.
             if (this.headerError) {
@@ -135,16 +142,17 @@ export class StreamDecoder {
     }
 
     /**
-     * Accumulate bytes into the header-search buffer and retry parseHeader.
-     * Once a header is recognised, anything past headerSize becomes audio data.
+     * Accumulate bytes into the header-search buffer and retry the format decoder's
+     * header parse. Once a header is recognised, anything past audioDataOffset
+     * becomes audio data.
      */
     private async tryParseHeader(chunk: Uint8Array): Promise<void> {
         this.headerSearchChunks.push(chunk);
         this.headerBytesReceived += chunk.length;
 
-        // Guard against unbounded accumulation from a corrupt or non-WAV stream.
+        // Guard against unbounded accumulation from a corrupt or unrecognised stream.
         if (this.headerBytesReceived > StreamDecoder.MAX_HEADER_SEARCH_BYTES) {
-            this.headerError = `WAV header not found after ${this.headerBytesReceived} bytes — stream may be corrupt or not a WAV file`;
+            this.headerError = `Audio header not found after ${this.headerBytesReceived} bytes — stream may be corrupt or an unsupported format`;
             console.error(this.headerError);
             // Drop the search buffer so subsequent chunks are not accumulated either.
             this.headerSearchChunks = [];
@@ -152,8 +160,8 @@ export class StreamDecoder {
             return;
         }
 
-        const header = WavUtils.parseHeader(this.headerSearchChunks, this.headerBytesReceived);
-        if (!header) {
+        const info = this.formatDecoder!.tryParseHeader(this.headerSearchChunks, this.headerBytesReceived);
+        if (!info) {
             // Not enough bytes yet — wait for the next chunk. If the stream ends
             // without ever producing a valid header, the final processChunk will
             // mark streamComplete and the player will report no audio decoded;
@@ -161,22 +169,22 @@ export class StreamDecoder {
             return;
         }
 
-        this.wavHeader = header;
+        this.formatInfo = info;
 
         // Recreate AudioContext with correct sample rate if needed
-        if (this.contextManager.sampleRate !== header.sampleRate) {
-            await this.contextManager.recreateWithSampleRate(header.sampleRate);
+        if (this.contextManager.sampleRate !== info.sampleRate) {
+            await this.contextManager.recreateWithSampleRate(info.sampleRate);
         }
 
         // Concatenate all header-search chunks and push the audio-data tail
-        // (everything past headerSize) into the raw audio buffer.
+        // (everything past audioDataOffset) into the raw audio buffer.
         const concatenated = new Uint8Array(this.headerBytesReceived);
         let offset = 0;
         for (const c of this.headerSearchChunks) {
             concatenated.set(c, offset);
             offset += c.length;
         }
-        const audioData = concatenated.subarray(header.headerSize);
+        const audioData = concatenated.subarray(info.audioDataOffset);
         if (audioData.length > 0) {
             this.addRawData(audioData);
         }
@@ -204,8 +212,8 @@ export class StreamDecoder {
         }
 
         if (this.totalStreamLength <= 0) return;
-        const totalReceived = this.wavHeader
-            ? this.totalRawBytes + this.wavHeader.headerSize
+        const totalReceived = this.formatInfo
+            ? this.totalRawBytes + this.formatInfo.audioDataOffset
             : this.headerBytesReceived;
         if (totalReceived >= this.totalStreamLength) {
             this.streamComplete = true;
@@ -232,16 +240,16 @@ export class StreamDecoder {
      * silently consume the failed segment.
      */
     private async tryDecodeNextSegment(): Promise<DecodedChunkResult | null> {
-        if (!this.wavHeader) return null;
+        if (!this.formatInfo) return null;
 
         const segmentSize = 64 * 1024; // 64KB segments
         const availableBytes = this.totalRawBytes - this.processedBytes;
         // Passing streamComplete lets the aligner relax the min-frame guard
         // for the final tail; otherwise residual <512-byte tails get dropped.
-        const alignedSize = WavUtils.getSampleAlignedChunkSize(
-            this.wavHeader,
-            segmentSize,
+        const alignedSize = this.formatDecoder!.getAlignedSegmentSize(
+            this.formatInfo,
             availableBytes,
+            segmentSize,
             this.streamComplete
         );
 
@@ -250,10 +258,10 @@ export class StreamDecoder {
         const segmentOffset = this.processedBytes;
 
         const rawSegment = this.extractAlignedData(alignedSize);
-        const wavFile = this.createWavFile(rawSegment);
+        const decodableSegment = this.formatDecoder!.wrapSegment(this.formatInfo, rawSegment);
 
         try {
-            const buffer = await this.decodeWithRetry(wavFile, segmentOffset, alignedSize);
+            const buffer = await this.decodeWithRetry(decodableSegment, segmentOffset, alignedSize);
             // Advance only after a successful decode so a thrown timeout/decode
             // failure does not silently drop the segment.
             this.processedBytes += alignedSize;
@@ -346,30 +354,19 @@ export class StreamDecoder {
         return extracted;
     }
 
-    /**
-     * Create a complete WAV file from raw audio data
-     */
-    private createWavFile(rawData: Uint8Array): Uint8Array {
-        const header = WavUtils.createHeader(this.wavHeader!, rawData.length);
-        const wavFile = new Uint8Array(header.length + rawData.length);
-        wavFile.set(header, 0);
-        wavFile.set(rawData, header.length);
-        return wavFile;
-    }
-
     /**
      * Decode with timeout to prevent hanging. Throws DecodeTimeoutError if the
      * deadline expires so callers can distinguish timeout from corrupt-data
      * failures (decodeAudioData throws DOMException for the latter).
      */
-    private async decodeWithTimeout(wavData: Uint8Array, timeoutMs: number = 5000): Promise<AudioBuffer> {
-        const buffer = new ArrayBuffer(wavData.length);
-        new Uint8Array(buffer).set(wavData);
+    private async decodeWithTimeout(audioData: Uint8Array, timeoutMs: number = 5000): Promise<AudioBuffer> {
+        const buffer = new ArrayBuffer(audioData.length);
+        new Uint8Array(buffer).set(audioData);
 
         const decodePromise = this.contextManager.decodeAudioData(buffer);
         let timer: ReturnType<typeof setTimeout> | null = null;
         const timeoutPromise = new Promise<never>((_, reject) => {
-            timer = setTimeout(() => reject(new DecodeTimeoutError(-1, wavData.length)), timeoutMs);
+            timer = setTimeout(() => reject(new DecodeTimeoutError(-1, audioData.length)), timeoutMs);
         });
 
         try {
@@ -380,23 +377,28 @@ export class StreamDecoder {
     }
 
     /**
-     * Get calculated duration from WAV header
+     * Get calculated duration from the parsed format header.
+     *
+     * Prefer the decoder's header-derived totalDuration (for WAV this is the exact
+     * dataSize/byteRate). When the header omits a usable size (e.g. a WAV data chunk
+     * size of 0 in a streamed/unknown-length file), fall back to deriving it from the
+     * total stream length minus the header — identical to the original WAV behavior.
      */
     getEstimatedDuration(): number | null {
-        if (!this.wavHeader || this.wavHeader.byteRate <= 0) return null;
-
-        const audioDataSize = this.wavHeader.dataSize > 0
-            ? this.wavHeader.dataSize
-            : (this.totalStreamLength - this.wavHeader.headerSize);
-
-        return audioDataSize / this.wavHeader.byteRate;
+        if (!this.formatInfo) return null;
+        if (this.formatInfo.totalDuration && this.formatInfo.totalDuration > 0) {
+            return this.formatInfo.totalDuration;
+        }
+        if (this.formatInfo.byteRate <= 0) return null;
+        const audioDataSize = this.totalStreamLength - this.formatInfo.audioDataOffset;
+        return audioDataSize / this.formatInfo.byteRate;
     }
 
     /**
-     * Check if WAV header has been parsed
+     * Check if the format header has been parsed
      */
     get headerParsed(): boolean {
-        return this.wavHeader !== null;
+        return this.formatInfo !== null;
     }
 
     /**
@@ -407,22 +409,21 @@ export class StreamDecoder {
     }
 
     /**
-     * Get the WAV header info for byte offset calculation
+     * Get the parsed format info (sample rate, channels, audio-data offset, …).
+     * Used by the player for seek byte-offset math and header-dependent decisions.
      */
-    getWavHeader(): WavHeader | null {
-        return this.wavHeader;
+    getFormatInfo(): FormatInfo | null {
+        return this.formatInfo;
     }
 
     /**
-     * Calculate byte offset from a time position (in seconds)
-     * Returns block-aligned byte offset for clean audio
+     * Calculate the file-absolute byte offset for a seek to the given time position.
+     * Delegates to the format decoder; the returned value is a byte position in the
+     * file on disk (header included), ready for a Range request.
      */
     calculateByteOffset(positionSeconds: number): number {
-        if (!this.wavHeader || this.wavHeader.byteRate <= 0) return 0;
-
-        const rawOffset = Math.floor(positionSeconds * this.wavHeader.byteRate);
-        // Align to block boundary for clean audio
-        return Math.floor(rawOffset / this.wavHeader.blockAlign) * this.wavHeader.blockAlign;
+        if (!this.formatInfo) return 0;
+        return this.formatDecoder!.calculateByteOffset(this.formatInfo, positionSeconds);
     }
 
     /**
@@ -453,10 +454,11 @@ export class StreamDecoder {
     }
 
     /**
-     * Reset decoder state
+     * Reset decoder state. The format decoder is retained — a stream's format does
+     * not change across reset; a new stream supplies a fresh decoder via initialize.
      */
     reset(): void {
-        this.wavHeader = null;
+        this.formatInfo = null;
         this.rawChunks = [];
         this.totalRawBytes = 0;
         this.processedBytes = 0;
@@ -473,10 +475,10 @@ export class StreamDecoder {
      * Reinitialize for a Range-continuation stream after seek-beyond-buffer.
      *
      * The server responds to a Range request with 206 Partial Content carrying raw
-     * PCM from a file-absolute offset — there is NO WAV header in this body. We retain
-     * the header parsed from the initial stream (its format describes every segment we
-     * synthesise via createWavFile) and feed the entire 206 body straight into the
-     * decode pipeline. The `if (!this.wavHeader)` branch in processChunk therefore goes
+     * audio from a file-absolute offset — there is NO header in this body. We retain
+     * the FormatInfo parsed from the initial stream (its format describes every segment
+     * the decoder wraps via wrapSegment) and feed the entire 206 body straight into the
+     * decode pipeline. The `if (!this.formatInfo)` branch in processChunk therefore goes
      * directly to addRawData and tryParseHeader is never re-entered.
      *
      * @param remainingByteLength the Content-Length of the 206 response — the number of
@@ -484,7 +486,7 @@ export class StreamDecoder {
      *   reached when totalRawBytes >= this value.
      */
     reinitializeForRangeContinuation(remainingByteLength: number): void {
-        // Retain this.wavHeader — the 206 body carries no header to reparse.
+        // Retain this.formatInfo and this.formatDecoder — the 206 body carries no header to reparse.
         this.rawChunks = [];
         this.totalRawBytes = 0;
         this.processedBytes = 0;
diff --git a/DeepDrftPublic/Interop/audio/WavFormatDecoder.ts b/DeepDrftPublic/Interop/audio/WavFormatDecoder.ts
new file mode 100644
index 0000000..57a89ed
--- /dev/null
+++ b/DeepDrftPublic/Interop/audio/WavFormatDecoder.ts
@@ -0,0 +1,80 @@
+/**
+ * WavFormatDecoder - WAV/PCM implementation of IFormatDecoder.
+ *
+ * All WAV-specific stream logic lives here: header parsing, frame-aligned segment
+ * sizing, segment wrapping with a synthesised 44-byte PCM header, and block-aligned
+ * seek byte-offset calculation. StreamDecoder delegates to this via IFormatDecoder
+ * and holds no WAV knowledge of its own.
+ *
+ * The low-level byte routines (parseHeader, createHeader, getSampleAlignedChunkSize)
+ * are reused from WavUtils rather than duplicated — this is the same proven logic the
+ * decoder used before the format-agnostic refactor, kept in one place.
+ */
+
+import { WavUtils } from '../wavutils.js';
+import { FormatInfo, IFormatDecoder } from './IFormatDecoder.js';
+
+export class WavFormatDecoder implements IFormatDecoder {
+    tryParseHeader(chunks: Uint8Array[], totalSize: number): FormatInfo | null {
+        const header = WavUtils.parseHeader(chunks, totalSize);
+        if (!header) return null;
+
+        return {
+            sampleRate: header.sampleRate,
+            channels: header.channels,
+            bitsPerSample: header.bitsPerSample,
+            byteRate: header.byteRate,
+            blockAlign: header.blockAlign,
+            // dataSize / byteRate is the exact PCM duration; guard against a zero
+            // byteRate (malformed fmt chunk) rather than producing Infinity/NaN.
+            totalDuration: header.byteRate > 0 ? header.dataSize / header.byteRate : null,
+            audioDataOffset: header.headerSize,
+            seekData: null
+        };
+    }
+
+    getAlignedSegmentSize(
+        info: FormatInfo,
+        availableBytes: number,
+        requestedSize: number,
+        streamComplete: boolean
+    ): number {
+        // blockAlign is the PCM frame size; reuse the original frame-alignment routine
+        // verbatim so streaming/tail-drain behavior is unchanged.
+        return WavUtils.getSampleAlignedChunkSize(
+            { blockAlign: info.blockAlign },
+            requestedSize,
+            availableBytes,
+            streamComplete
+        );
+    }
+
+    wrapSegment(info: FormatInfo, rawBytes: Uint8Array): Uint8Array {
+        const header = WavUtils.createHeader(
+            {
+                channels: info.channels,
+                sampleRate: info.sampleRate,
+                byteRate: info.byteRate,
+                blockAlign: info.blockAlign,
+                bitsPerSample: info.bitsPerSample
+            },
+            rawBytes.length
+        );
+        const wavFile = new Uint8Array(header.length + rawBytes.length);
+        wavFile.set(header, 0);
+        wavFile.set(rawBytes, header.length);
+        return wavFile;
+    }
+
+    calculateByteOffset(info: FormatInfo, positionSeconds: number): number {
+        // No header / malformed fmt chunk: cannot compute. Return the audio start so the
+        // caller still has a valid (file-absolute) Range target rather than a negative.
+        if (info.byteRate <= 0 || info.blockAlign <= 0) return info.audioDataOffset;
+
+        const rawOffset = Math.floor(positionSeconds * info.byteRate);
+        // Align to a PCM frame boundary for clean audio, then make it file-absolute by
+        // adding the header size — the Range request is a byte position in the file on disk.
+        const alignedAudioOffset = Math.floor(rawOffset / info.blockAlign) * info.blockAlign;
+        return info.audioDataOffset + alignedAudioOffset;
+    }
+}
diff --git a/DeepDrftPublic/Interop/audio/index.ts b/DeepDrftPublic/Interop/audio/index.ts
index b6dbb7b..3239f24 100644
--- a/DeepDrftPublic/Interop/audio/index.ts
+++ b/DeepDrftPublic/Interop/audio/index.ts
@@ -31,10 +31,10 @@ const DeepDrftAudio = {
         }
     },
 
-    initializeStreaming: (playerId: string, totalStreamLength: number): AudioResult => {
+    initializeStreaming: (playerId: string, totalStreamLength: number, contentType: string): AudioResult => {
         const player = audioPlayers.get(playerId);
         if (!player) return { success: false, error: 'Player not found' };
-        return player.initializeStreaming(totalStreamLength);
+        return player.initializeStreaming(totalStreamLength, contentType);
     },
 
     processStreamingChunk: async (playerId: string, chunk: Uint8Array): Promise<StreamingResult> => {
diff --git a/DeepDrftPublic/Interop/wavutils.ts b/DeepDrftPublic/Interop/wavutils.ts
index f27efcc..c8b9001 100644
--- a/DeepDrftPublic/Interop/wavutils.ts
+++ b/DeepDrftPublic/Interop/wavutils.ts
@@ -123,7 +123,9 @@ class WavUtils {
         };
     }
 
-    static createHeader(wavHeader: WavHeader, dataSize: number): Uint8Array {
+    static createHeader(
+        wavHeader: Pick<WavHeader, 'channels' | 'sampleRate' | 'byteRate' | 'blockAlign' | 'bitsPerSample'>,
+        dataSize: number): Uint8Array {
         const header = new ArrayBuffer(44);
         const view = new DataView(header);
         
@@ -195,7 +197,7 @@ class WavUtils {
         buffer[43] = (audioDataSize >> 24) & 0xFF;
     }
 
-    static getSampleAlignedChunkSize(header: WavHeader, maxChunkSize: number, availableDataSize: number, streamComplete: boolean = false): number {
+    static getSampleAlignedChunkSize(header: Pick<WavHeader, 'blockAlign'>, maxChunkSize: number, availableDataSize: number, streamComplete: boolean = false): number {
         const frameSize = header.blockAlign;
 
         // Much smaller minimum for streaming - just enough for Web Audio API.