From e3c53f9736edd1ff7f4a707d9370d4c308827e67 Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 01:24:42 +0000
Subject: [PATCH 01/14] Initial plan
From 017f31cfe27fe5ac3e5acd79cbe4296c060810ce Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 01:33:40 +0000
Subject: [PATCH 02/14] Fix AAC-ELD decoder: switch from ADTS to LATM wrapping,
fix decoder race condition
ADTS format cannot represent AAC-ELD (its 2-bit profile field only supports
AAC-Main/LC/SSR). Using LATM/LOAS wrapping which supports all AAC profiles
including AAC-ELD (AOT 39).
Also added thread-safety lock on decoder initialization to prevent race
condition between concurrent OnRawCSocketAsync and OnRawDSocketAsync handlers
that was causing duplicate FFmpeg processes to be spawned.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
.../Implementations/FFmpegAacEldDecoder.cs | 199 +++++++++++++++---
AirPlay/Listeners/AudioListener.cs | 140 ++++++------
2 files changed, 240 insertions(+), 99 deletions(-)
diff --git a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
index 72bc7f7..f74a849 100644
--- a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
+++ b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
@@ -5,14 +5,15 @@
* for AAC-ELD because the pre-built binary doesn't include ER format support.
* SharpJaad.AAC also doesn't support AAC-ELD.
*
- * This decoder pipes raw AAC-ELD frames (wrapped in ADTS) through FFmpeg for
- * decoding, which supports AAC-ELD natively. This approach works with any
- * FFmpeg version the user has installed.
+ * This decoder pipes raw AAC-ELD frames wrapped in LOAS/LATM through FFmpeg
+ * for decoding. ADTS format does NOT support AAC-ELD (its 2-bit profile field
+ * cannot represent AOT 39). LATM supports all AAC profiles including AAC-ELD.
*
* Reference: UxPlay uses GStreamer's avdec_aac (which wraps FFmpeg) for the same purpose.
*/
using System;
+using System.Collections.Generic;
using System.Diagnostics;
using System.IO;
using System.Threading;
@@ -34,10 +35,8 @@ public class FFmpegAacEldDecoder : IDecoder, IDisposable
private int _frameLength;
private bool _disposed;
private bool _initialized;
- private readonly byte[] _adtsHeader = new byte[7];
- private int _adtsProfile;
- private int _adtsFreqIdx;
- private int _adtsChanCfg;
+ private bool _firstFrame = true;
+ private int _freqIdx;
public AudioFormat Type => AudioFormat.AAC_ELD;
@@ -51,22 +50,17 @@ public int Config(int sampleRate, int channels, int bitDepth, int frameLength)
_sampleRate = sampleRate;
_frameLength = frameLength;
_pcmOutputSize = frameLength * channels * (bitDepth / 8);
-
- // ADTS profile=2 (AAC-LC) since ADTS doesn't support AAC-ELD profile encoding.
- // FFmpeg's parser will still decode the content correctly based on the actual bitstream.
- _adtsProfile = 2;
- _adtsFreqIdx = GetSampleRateIndex(sampleRate);
- _adtsChanCfg = channels;
+ _freqIdx = GetSampleRateIndex(sampleRate);
try
{
// Start FFmpeg process:
- // Input: AAC frames wrapped in ADTS headers via stdin pipe
+ // Input: AAC-ELD frames wrapped in LOAS/LATM via stdin pipe
// Output: raw PCM S16LE via stdout pipe
var psi = new ProcessStartInfo
{
FileName = "ffmpeg",
- Arguments = $"-hide_banner -loglevel error -f aac -i pipe:0 -f s16le -acodec pcm_s16le -ar {sampleRate} -ac {channels} pipe:1",
+ Arguments = $"-hide_banner -loglevel error -f latm -i pipe:0 -f s16le -acodec pcm_s16le -ar {sampleRate} -ac {channels} pipe:1",
UseShellExecute = false,
RedirectStandardInput = true,
RedirectStandardOutput = true,
@@ -90,7 +84,7 @@ public int Config(int sampleRate, int channels, int bitDepth, int frameLength)
stderrThread.Start();
_initialized = true;
- Console.WriteLine($"FFmpeg AAC-ELD decoder started: {sampleRate}Hz, {channels}ch, {bitDepth}bit, frameLength={frameLength}");
+ Console.WriteLine($"FFmpeg AAC-ELD decoder started (LATM): {sampleRate}Hz, {channels}ch, {bitDepth}bit, frameLength={frameLength}");
return 0;
}
catch (Exception ex)
@@ -113,13 +107,11 @@ public int DecodeFrame(byte[] input, ref byte[] output, int length)
try
{
- // Wrap the raw AAC frame in an ADTS header for FFmpeg to parse
- int frameLen = input.Length + 7; // ADTS header is 7 bytes
- BuildAdtsHeader(_adtsHeader, frameLen, _adtsProfile, _adtsFreqIdx, _adtsChanCfg);
+ // Build LOAS/LATM frame wrapping the raw AAC-ELD data
+ byte[] loasFrame = BuildLoasFrame(input, _firstFrame);
+ _firstFrame = false;
- // Write ADTS header + raw AAC data to FFmpeg stdin
- _ffmpegInput.Write(_adtsHeader, 0, 7);
- _ffmpegInput.Write(input, 0, input.Length);
+ _ffmpegInput.Write(loasFrame, 0, loasFrame.Length);
_ffmpegInput.Flush();
// Read decoded PCM from FFmpeg stdout
@@ -154,17 +146,124 @@ public int DecodeFrame(byte[] input, ref byte[] output, int length)
}
///
- /// Build an ADTS header for wrapping raw AAC frames.
+ /// Build a LOAS/LATM frame containing the raw AAC-ELD data.
+ /// LOAS (Low Overhead Audio Stream) sync layer wraps an AudioMuxElement.
+ /// The first frame includes the full StreamMuxConfig; subsequent frames
+ /// reuse it (useSameStreamMux=1).
+ ///
+ private byte[] BuildLoasFrame(byte[] aacFrame, bool includeConfig)
+ {
+ // Build the AudioMuxElement as a bitstream
+ var bits = new BitWriter();
+
+ if (includeConfig)
+ {
+ // useSameStreamMux = 0 (include StreamMuxConfig)
+ bits.WriteBits(0, 1);
+ WriteStreamMuxConfig(bits);
+ }
+ else
+ {
+ // useSameStreamMux = 1 (reuse previous config)
+ bits.WriteBits(1, 1);
+ }
+
+ // PayloadLengthInfo for allStreamsSameTimeFraming=1, progSIndx=0, laySIndx=0
+ // Length is encoded as: N bytes of 0xFF followed by a final byte < 0xFF
+ int remaining = aacFrame.Length;
+ while (remaining >= 255)
+ {
+ bits.WriteBits(255, 8);
+ remaining -= 255;
+ }
+ bits.WriteBits(remaining, 8);
+
+ // PayloadMux: the raw AAC-ELD frame data
+ for (int i = 0; i < aacFrame.Length; i++)
+ {
+ bits.WriteBits(aacFrame[i], 8);
+ }
+
+ // otherDataPresent = 0 already implied by StreamMuxConfig setting
+
+ byte[] audioMuxElement = bits.ToByteArray();
+
+ // LOAS sync layer: 0x56E0 | (length & 0x1FFF)
+ int audioMuxLength = audioMuxElement.Length;
+ byte[] loasFrame = new byte[3 + audioMuxLength];
+ loasFrame[0] = 0x56;
+ loasFrame[1] = (byte)(0xE0 | ((audioMuxLength >> 8) & 0x1F));
+ loasFrame[2] = (byte)(audioMuxLength & 0xFF);
+ Array.Copy(audioMuxElement, 0, loasFrame, 3, audioMuxLength);
+
+ return loasFrame;
+ }
+
+ ///
+ /// Write StreamMuxConfig for AAC-ELD into the bitstream.
+ /// ISO 14496-3 Table 1.42
+ ///
+ private void WriteStreamMuxConfig(BitWriter bits)
+ {
+ // audioMuxVersion = 0
+ bits.WriteBits(0, 1);
+ // allStreamsSameTimeFraming = 1
+ bits.WriteBits(1, 1);
+ // numSubFrames = 0 (1 subframe)
+ bits.WriteBits(0, 6);
+ // numProgram = 0 (1 program)
+ bits.WriteBits(0, 4);
+ // numLayer = 0 (1 layer)
+ bits.WriteBits(0, 3);
+
+ // AudioSpecificConfig for AAC-ELD (ISO 14496-3 Table 1.15)
+ WriteAudioSpecificConfig(bits);
+
+ // frameLengthType = 0 (variable frame length)
+ bits.WriteBits(0, 3);
+ // latmBufferFullness = 0xFF (variable bitrate)
+ bits.WriteBits(0xFF, 8);
+
+ // otherDataPresent = 0
+ bits.WriteBits(0, 1);
+ // crcCheckPresent = 0
+ bits.WriteBits(0, 1);
+ }
+
+ ///
+ /// Write AudioSpecificConfig for AAC-ELD.
+ /// ISO 14496-3 Table 1.15
///
- private static void BuildAdtsHeader(byte[] header, int packetLen, int profile, int freqIdx, int chanCfg)
+ private void WriteAudioSpecificConfig(BitWriter bits)
{
- header[0] = 0xFF;
- header[1] = 0xF1; // MPEG-4, Layer 0, no CRC
- header[2] = (byte)(((profile - 1) << 6) | (freqIdx << 2) | (chanCfg >> 2));
- header[3] = (byte)(((chanCfg & 3) << 6) | (packetLen >> 11));
- header[4] = (byte)((packetLen >> 3) & 0xFF);
- header[5] = (byte)(((packetLen & 7) << 5) | 0x1F);
- header[6] = 0xFC;
+ // audioObjectType = 39 (AAC-ELD)
+ // Since 39 >= 31, write 5 bits of 31 + 6 bits of (39-32) = 7
+ bits.WriteBits(31, 5);
+ bits.WriteBits(7, 6);
+
+ // samplingFrequencyIndex
+ bits.WriteBits(_freqIdx, 4);
+ // If freqIdx == 0xF, write 24-bit samplingFrequency (not needed for standard rates)
+
+ // channelConfiguration
+ bits.WriteBits(_channels, 4);
+
+ // SBR/PS extension: not present for AAC-ELD
+ // (AAC-ELD specific config follows)
+
+ // ELDSpecificConfig (ISO 14496-3 Table 4.180)
+ // frameLengthFlag: 0 = 512 samples (480 after windowing), 1 = 480 samples
+ bits.WriteBits(_frameLength == 480 ? 1 : 0, 1);
+ // aacSectionDataResilienceFlag = 0
+ bits.WriteBits(0, 1);
+ // aacScalefactorDataResilienceFlag = 0
+ bits.WriteBits(0, 1);
+ // aacSpectralDataResilienceFlag = 0
+ bits.WriteBits(0, 1);
+
+ // ldSbrPresentFlag = 0 (no LD-SBR)
+ bits.WriteBits(0, 1);
+ // No further extension data
}
private static int GetSampleRateIndex(int sampleRate)
@@ -202,5 +301,43 @@ public void Dispose()
catch { /* ignore cleanup errors */ }
}
}
+
+ ///
+ /// Helper class for writing individual bits into a byte array.
+ /// Used to construct LOAS/LATM bitstream fields.
+ ///
+ private class BitWriter
+ {
+ private readonly List _bytes = new List();
+ private int _currentByte = 0;
+ private int _bitsInCurrentByte = 0;
+
+ public void WriteBits(int value, int numBits)
+ {
+ for (int i = numBits - 1; i >= 0; i--)
+ {
+ _currentByte = (_currentByte << 1) | ((value >> i) & 1);
+ _bitsInCurrentByte++;
+
+ if (_bitsInCurrentByte == 8)
+ {
+ _bytes.Add((byte)_currentByte);
+ _currentByte = 0;
+ _bitsInCurrentByte = 0;
+ }
+ }
+ }
+
+ public byte[] ToByteArray()
+ {
+ if (_bitsInCurrentByte > 0)
+ {
+ // Pad remaining bits with zeros
+ _currentByte <<= (8 - _bitsInCurrentByte);
+ _bytes.Add((byte)_currentByte);
+ }
+ return _bytes.ToArray();
+ }
+ }
}
}
diff --git a/AirPlay/Listeners/AudioListener.cs b/AirPlay/Listeners/AudioListener.cs
index e906a08..8fbbaf8 100644
--- a/AirPlay/Listeners/AudioListener.cs
+++ b/AirPlay/Listeners/AudioListener.cs
@@ -27,6 +27,7 @@ public class AudioListener : BaseUdpListener
private readonly OmgHax _omgHax = new OmgHax();
private IDecoder _decoder;
+ private readonly object _decoderLock = new object();
private ulong _sync_time;
private ulong _sync_timestamp;
private ushort _controlSequenceNumber = 0;
@@ -494,94 +495,97 @@ private int RaopRtpResendCallback(Socket cSocket, ushort control_seqnum, ushort
private void InitializeDecoder (Session session)
{
- if (_decoder != null) return;
-
- var audioFormat = session.AudioFormat;
- var spf = session.AudioSamplesPerFrame;
-
- if (audioFormat == AudioFormat.ALAC)
+ lock (_decoderLock)
{
- // RTP info: 96 AppleLossless, 96 352 0 16 40 10 14 2 255 0 0 44100
- // (ALAC -> PCM)
+ if (_decoder != null) return;
- var frameLength = spf > 0 ? spf : 352;
- var numChannels = 2;
- var bitDepth = 16;
- var sampleRate = 44100;
+ var audioFormat = session.AudioFormat;
+ var spf = session.AudioSamplesPerFrame;
- _decoder = new ALACDecoder();
- _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
- }
- else if (audioFormat == AudioFormat.AAC)
- {
- // RTP info: 96 mpeg4-generic/44100/2, 96 mode=AAC-main; constantDuration=1024
- // (AAC-MAIN -> PCM)
+ if (audioFormat == AudioFormat.ALAC)
+ {
+ // RTP info: 96 AppleLossless, 96 352 0 16 40 10 14 2 255 0 0 44100
+ // (ALAC -> PCM)
- var frameLength = spf > 0 ? spf : 1024;
- var numChannels = 2;
- var bitDepth = 16;
- var sampleRate = 44100;
+ var frameLength = spf > 0 ? spf : 352;
+ var numChannels = 2;
+ var bitDepth = 16;
+ var sampleRate = 44100;
- _decoder = new AACDecoder(TransportType.TT_MP4_RAW, AudioObjectType.AOT_AAC_MAIN, 1);
- _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
- }
- else if(audioFormat == AudioFormat.AAC_ELD)
- {
- // RTP info: 96 mpeg4-generic/44100/2, 96 mode=AAC-eld; constantDuration=480
- // (AAC-ELD -> PCM) using FFmpeg subprocess decoder
+ _decoder = new ALACDecoder();
+ _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ }
+ else if (audioFormat == AudioFormat.AAC)
+ {
+ // RTP info: 96 mpeg4-generic/44100/2, 96 mode=AAC-main; constantDuration=1024
+ // (AAC-MAIN -> PCM)
- var frameLength = spf > 0 ? spf : 480;
- var numChannels = 2;
- var bitDepth = 16;
- var sampleRate = 44100;
+ var frameLength = spf > 0 ? spf : 1024;
+ var numChannels = 2;
+ var bitDepth = 16;
+ var sampleRate = 44100;
- try
+ _decoder = new AACDecoder(TransportType.TT_MP4_RAW, AudioObjectType.AOT_AAC_MAIN, 1);
+ _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ }
+ else if(audioFormat == AudioFormat.AAC_ELD)
{
- var aacEldDecoder = new Decoders.Implementations.FFmpegAacEldDecoder();
- var ret = aacEldDecoder.Config(sampleRate, numChannels, bitDepth, frameLength);
- if (ret == 0)
+ // RTP info: 96 mpeg4-generic/44100/2, 96 mode=AAC-eld; constantDuration=480
+ // (AAC-ELD -> PCM) using FFmpeg subprocess decoder with LATM wrapping
+
+ var frameLength = spf > 0 ? spf : 480;
+ var numChannels = 2;
+ var bitDepth = 16;
+ var sampleRate = 44100;
+
+ try
{
- _decoder = aacEldDecoder;
+ var aacEldDecoder = new Decoders.Implementations.FFmpegAacEldDecoder();
+ var ret = aacEldDecoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ if (ret == 0)
+ {
+ _decoder = aacEldDecoder;
+ }
+ else
+ {
+ Console.WriteLine($"FFmpeg AAC-ELD decoder config failed (error {ret}), falling back to SharpJaad AAC-LC");
+ aacEldDecoder.Dispose();
+ _decoder = new AACDecoder(TransportType.TT_MP4_RAW, AudioObjectType.AOT_AAC_LC, 1);
+ _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ }
}
- else
+ catch (Exception ex)
{
- Console.WriteLine($"FFmpeg AAC-ELD decoder config failed (error {ret}), falling back to SharpJaad AAC-LC");
- aacEldDecoder.Dispose();
+ Console.WriteLine($"FFmpeg AAC-ELD decoder unavailable ({ex.Message}), falling back to SharpJaad AAC-LC");
_decoder = new AACDecoder(TransportType.TT_MP4_RAW, AudioObjectType.AOT_AAC_LC, 1);
_decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
}
}
- catch (Exception ex)
+ else if (audioFormat == AudioFormat.PCM)
{
- Console.WriteLine($"FFmpeg AAC-ELD decoder unavailable ({ex.Message}), falling back to SharpJaad AAC-LC");
- _decoder = new AACDecoder(TransportType.TT_MP4_RAW, AudioObjectType.AOT_AAC_LC, 1);
- _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
- }
- }
- else if (audioFormat == AudioFormat.PCM)
- {
- // Raw PCM audio - no decoding needed
- _decoder = new PCMDecoder();
- }
- else
- {
- // Determine format from compression type if audioFormat is unknown
- if (session.AudioCompressionType == 1)
- {
- // ct=1 = ALAC
- var frameLength = spf > 0 ? spf : 352;
- _decoder = new ALACDecoder();
- _decoder.Config(44100, 2, 16, frameLength);
- }
- else if (session.AudioCompressionType == 0)
- {
- // ct=0 = PCM
+ // Raw PCM audio - no decoding needed
_decoder = new PCMDecoder();
}
else
{
- // Default fallback
- _decoder = new PCMDecoder();
+ // Determine format from compression type if audioFormat is unknown
+ if (session.AudioCompressionType == 1)
+ {
+ // ct=1 = ALAC
+ var frameLength = spf > 0 ? spf : 352;
+ _decoder = new ALACDecoder();
+ _decoder.Config(44100, 2, 16, frameLength);
+ }
+ else if (session.AudioCompressionType == 0)
+ {
+ // ct=0 = PCM
+ _decoder = new PCMDecoder();
+ }
+ else
+ {
+ // Default fallback
+ _decoder = new PCMDecoder();
+ }
}
}
}
From e37794672696c86bb90de0ec69ca2b273406a043 Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 01:34:43 +0000
Subject: [PATCH 03/14] Address code review: fix BitWriter thread safety and
_firstFrame race condition
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
.../Decoders/Implementations/FFmpegAacEldDecoder.cs | 13 +++++++------
1 file changed, 7 insertions(+), 6 deletions(-)
diff --git a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
index f74a849..3a76fe7 100644
--- a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
+++ b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
@@ -35,7 +35,7 @@ public class FFmpegAacEldDecoder : IDecoder, IDisposable
private int _frameLength;
private bool _disposed;
private bool _initialized;
- private bool _firstFrame = true;
+ private int _firstFrame = 1; // 1 = true, 0 = false (for thread-safe Interlocked access)
private int _freqIdx;
public AudioFormat Type => AudioFormat.AAC_ELD;
@@ -108,8 +108,8 @@ public int DecodeFrame(byte[] input, ref byte[] output, int length)
try
{
// Build LOAS/LATM frame wrapping the raw AAC-ELD data
- byte[] loasFrame = BuildLoasFrame(input, _firstFrame);
- _firstFrame = false;
+ bool includeConfig = Interlocked.Exchange(ref _firstFrame, 0) == 1;
+ byte[] loasFrame = BuildLoasFrame(input, includeConfig);
_ffmpegInput.Write(loasFrame, 0, loasFrame.Length);
_ffmpegInput.Flush();
@@ -330,13 +330,14 @@ public void WriteBits(int value, int numBits)
public byte[] ToByteArray()
{
+ var result = new List(_bytes);
if (_bitsInCurrentByte > 0)
{
// Pad remaining bits with zeros
- _currentByte <<= (8 - _bitsInCurrentByte);
- _bytes.Add((byte)_currentByte);
+ int padded = _currentByte << (8 - _bitsInCurrentByte);
+ result.Add((byte)padded);
}
- return _bytes.ToArray();
+ return result.ToArray();
}
}
}
From 0983bd0a8ac654c8be4d28698944bf99db47aa39 Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 01:48:45 +0000
Subject: [PATCH 04/14] Fix FFmpeg AAC-ELD decoder: revert from LATM to ADTS
wrapping
The previous change used `-f latm` as FFmpeg input format, but FFmpeg has
no `latm` demuxer (only a muxer). This caused FFmpeg to exit immediately,
and every subsequent DecodeFrame call returned -1 (HasExited check),
producing the "Decoding error. Decoder: AAC_ELD Code: -1" error spam.
Revert to ADTS wrapping with `-f aac` which keeps FFmpeg alive. While
ADTS headers can only encode AAC-LC profile (2-bit field), FFmpeg's AAC
decoder handles the actual bitstream content. The ADTS header serves as
a framing/sync mechanism.
Also improved stderr handling to filter per-frame warnings and only log
critical errors with case-insensitive matching.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
.../Implementations/FFmpegAacEldDecoder.cs | 214 ++++--------------
1 file changed, 43 insertions(+), 171 deletions(-)
diff --git a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
index 3a76fe7..c2f419a 100644
--- a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
+++ b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
@@ -5,15 +5,15 @@
* for AAC-ELD because the pre-built binary doesn't include ER format support.
* SharpJaad.AAC also doesn't support AAC-ELD.
*
- * This decoder pipes raw AAC-ELD frames wrapped in LOAS/LATM through FFmpeg
- * for decoding. ADTS format does NOT support AAC-ELD (its 2-bit profile field
- * cannot represent AOT 39). LATM supports all AAC profiles including AAC-ELD.
+ * This decoder pipes raw AAC-ELD frames wrapped in ADTS headers through FFmpeg
+ * for decoding. While ADTS headers can only encode AAC-LC profile (2-bit field),
+ * FFmpeg's AAC decoder handles the actual bitstream content regardless of the
+ * ADTS profile indicator. The ADTS header serves as a framing/sync mechanism.
*
* Reference: UxPlay uses GStreamer's avdec_aac (which wraps FFmpeg) for the same purpose.
*/
using System;
-using System.Collections.Generic;
using System.Diagnostics;
using System.IO;
using System.Threading;
@@ -35,8 +35,8 @@ public class FFmpegAacEldDecoder : IDecoder, IDisposable
private int _frameLength;
private bool _disposed;
private bool _initialized;
- private int _firstFrame = 1; // 1 = true, 0 = false (for thread-safe Interlocked access)
- private int _freqIdx;
+ private readonly byte[] _adtsHeader = new byte[7];
+ private int _adtsFreqIdx;
public AudioFormat Type => AudioFormat.AAC_ELD;
@@ -50,17 +50,17 @@ public int Config(int sampleRate, int channels, int bitDepth, int frameLength)
_sampleRate = sampleRate;
_frameLength = frameLength;
_pcmOutputSize = frameLength * channels * (bitDepth / 8);
- _freqIdx = GetSampleRateIndex(sampleRate);
+ _adtsFreqIdx = GetSampleRateIndex(sampleRate);
try
{
// Start FFmpeg process:
- // Input: AAC-ELD frames wrapped in LOAS/LATM via stdin pipe
+ // Input: AAC frames wrapped in ADTS headers via stdin pipe
// Output: raw PCM S16LE via stdout pipe
var psi = new ProcessStartInfo
{
FileName = "ffmpeg",
- Arguments = $"-hide_banner -loglevel error -f latm -i pipe:0 -f s16le -acodec pcm_s16le -ar {sampleRate} -ac {channels} pipe:1",
+ Arguments = $"-hide_banner -loglevel error -f aac -i pipe:0 -f s16le -acodec pcm_s16le -ar {sampleRate} -ac {channels} pipe:1",
UseShellExecute = false,
RedirectStandardInput = true,
RedirectStandardOutput = true,
@@ -77,14 +77,26 @@ public int Config(int sampleRate, int channels, int bitDepth, int frameLength)
// Drain stderr in background to prevent pipe deadlock
var stderrThread = new Thread(() =>
{
- try { _ffmpegProcess.StandardError.ReadToEnd(); }
+ try
+ {
+ using var reader = _ffmpegProcess.StandardError;
+ string line;
+ while ((line = reader.ReadLine()) != null)
+ {
+ // Only log critical errors, not per-frame decode warnings
+ if (line.Contains("error", StringComparison.OrdinalIgnoreCase))
+ {
+ Console.WriteLine($"FFmpeg: {line}");
+ }
+ }
+ }
catch { /* ignore */ }
});
stderrThread.IsBackground = true;
stderrThread.Start();
_initialized = true;
- Console.WriteLine($"FFmpeg AAC-ELD decoder started (LATM): {sampleRate}Hz, {channels}ch, {bitDepth}bit, frameLength={frameLength}");
+ Console.WriteLine($"FFmpeg AAC-ELD decoder started: {sampleRate}Hz, {channels}ch, {bitDepth}bit, frameLength={frameLength}");
return 0;
}
catch (Exception ex)
@@ -107,11 +119,16 @@ public int DecodeFrame(byte[] input, ref byte[] output, int length)
try
{
- // Build LOAS/LATM frame wrapping the raw AAC-ELD data
- bool includeConfig = Interlocked.Exchange(ref _firstFrame, 0) == 1;
- byte[] loasFrame = BuildLoasFrame(input, includeConfig);
-
- _ffmpegInput.Write(loasFrame, 0, loasFrame.Length);
+ // Wrap the raw AAC frame in an ADTS header for FFmpeg to parse.
+ // ADTS profile is set to AAC-LC (profile=2) since ADTS only has a 2-bit
+ // profile field and cannot represent AAC-ELD. FFmpeg's decoder handles
+ // the actual bitstream content regardless of the ADTS profile indicator.
+ int frameLen = input.Length + 7; // ADTS header is 7 bytes
+ BuildAdtsHeader(_adtsHeader, frameLen, 2, _adtsFreqIdx, _channels);
+
+ // Write ADTS header + raw AAC data to FFmpeg stdin
+ _ffmpegInput.Write(_adtsHeader, 0, 7);
+ _ffmpegInput.Write(input, 0, input.Length);
_ffmpegInput.Flush();
// Read decoded PCM from FFmpeg stdout
@@ -146,124 +163,18 @@ public int DecodeFrame(byte[] input, ref byte[] output, int length)
}
///
- /// Build a LOAS/LATM frame containing the raw AAC-ELD data.
- /// LOAS (Low Overhead Audio Stream) sync layer wraps an AudioMuxElement.
- /// The first frame includes the full StreamMuxConfig; subsequent frames
- /// reuse it (useSameStreamMux=1).
- ///
- private byte[] BuildLoasFrame(byte[] aacFrame, bool includeConfig)
- {
- // Build the AudioMuxElement as a bitstream
- var bits = new BitWriter();
-
- if (includeConfig)
- {
- // useSameStreamMux = 0 (include StreamMuxConfig)
- bits.WriteBits(0, 1);
- WriteStreamMuxConfig(bits);
- }
- else
- {
- // useSameStreamMux = 1 (reuse previous config)
- bits.WriteBits(1, 1);
- }
-
- // PayloadLengthInfo for allStreamsSameTimeFraming=1, progSIndx=0, laySIndx=0
- // Length is encoded as: N bytes of 0xFF followed by a final byte < 0xFF
- int remaining = aacFrame.Length;
- while (remaining >= 255)
- {
- bits.WriteBits(255, 8);
- remaining -= 255;
- }
- bits.WriteBits(remaining, 8);
-
- // PayloadMux: the raw AAC-ELD frame data
- for (int i = 0; i < aacFrame.Length; i++)
- {
- bits.WriteBits(aacFrame[i], 8);
- }
-
- // otherDataPresent = 0 already implied by StreamMuxConfig setting
-
- byte[] audioMuxElement = bits.ToByteArray();
-
- // LOAS sync layer: 0x56E0 | (length & 0x1FFF)
- int audioMuxLength = audioMuxElement.Length;
- byte[] loasFrame = new byte[3 + audioMuxLength];
- loasFrame[0] = 0x56;
- loasFrame[1] = (byte)(0xE0 | ((audioMuxLength >> 8) & 0x1F));
- loasFrame[2] = (byte)(audioMuxLength & 0xFF);
- Array.Copy(audioMuxElement, 0, loasFrame, 3, audioMuxLength);
-
- return loasFrame;
- }
-
- ///
- /// Write StreamMuxConfig for AAC-ELD into the bitstream.
- /// ISO 14496-3 Table 1.42
- ///
- private void WriteStreamMuxConfig(BitWriter bits)
- {
- // audioMuxVersion = 0
- bits.WriteBits(0, 1);
- // allStreamsSameTimeFraming = 1
- bits.WriteBits(1, 1);
- // numSubFrames = 0 (1 subframe)
- bits.WriteBits(0, 6);
- // numProgram = 0 (1 program)
- bits.WriteBits(0, 4);
- // numLayer = 0 (1 layer)
- bits.WriteBits(0, 3);
-
- // AudioSpecificConfig for AAC-ELD (ISO 14496-3 Table 1.15)
- WriteAudioSpecificConfig(bits);
-
- // frameLengthType = 0 (variable frame length)
- bits.WriteBits(0, 3);
- // latmBufferFullness = 0xFF (variable bitrate)
- bits.WriteBits(0xFF, 8);
-
- // otherDataPresent = 0
- bits.WriteBits(0, 1);
- // crcCheckPresent = 0
- bits.WriteBits(0, 1);
- }
-
- ///
- /// Write AudioSpecificConfig for AAC-ELD.
- /// ISO 14496-3 Table 1.15
+ /// Build a 7-byte ADTS header for wrapping raw AAC frames.
+ /// Used as a framing/sync mechanism for FFmpeg's AAC demuxer.
///
- private void WriteAudioSpecificConfig(BitWriter bits)
+ private static void BuildAdtsHeader(byte[] header, int packetLen, int profile, int freqIdx, int chanCfg)
{
- // audioObjectType = 39 (AAC-ELD)
- // Since 39 >= 31, write 5 bits of 31 + 6 bits of (39-32) = 7
- bits.WriteBits(31, 5);
- bits.WriteBits(7, 6);
-
- // samplingFrequencyIndex
- bits.WriteBits(_freqIdx, 4);
- // If freqIdx == 0xF, write 24-bit samplingFrequency (not needed for standard rates)
-
- // channelConfiguration
- bits.WriteBits(_channels, 4);
-
- // SBR/PS extension: not present for AAC-ELD
- // (AAC-ELD specific config follows)
-
- // ELDSpecificConfig (ISO 14496-3 Table 4.180)
- // frameLengthFlag: 0 = 512 samples (480 after windowing), 1 = 480 samples
- bits.WriteBits(_frameLength == 480 ? 1 : 0, 1);
- // aacSectionDataResilienceFlag = 0
- bits.WriteBits(0, 1);
- // aacScalefactorDataResilienceFlag = 0
- bits.WriteBits(0, 1);
- // aacSpectralDataResilienceFlag = 0
- bits.WriteBits(0, 1);
-
- // ldSbrPresentFlag = 0 (no LD-SBR)
- bits.WriteBits(0, 1);
- // No further extension data
+ header[0] = 0xFF;
+ header[1] = 0xF1; // MPEG-4, Layer 0, no CRC
+ header[2] = (byte)(((profile - 1) << 6) | (freqIdx << 2) | (chanCfg >> 2));
+ header[3] = (byte)(((chanCfg & 3) << 6) | (packetLen >> 11));
+ header[4] = (byte)((packetLen >> 3) & 0xFF);
+ header[5] = (byte)(((packetLen & 7) << 5) | 0x1F);
+ header[6] = 0xFC;
}
private static int GetSampleRateIndex(int sampleRate)
@@ -301,44 +212,5 @@ public void Dispose()
catch { /* ignore cleanup errors */ }
}
}
-
- ///
- /// Helper class for writing individual bits into a byte array.
- /// Used to construct LOAS/LATM bitstream fields.
- ///
- private class BitWriter
- {
- private readonly List _bytes = new List();
- private int _currentByte = 0;
- private int _bitsInCurrentByte = 0;
-
- public void WriteBits(int value, int numBits)
- {
- for (int i = numBits - 1; i >= 0; i--)
- {
- _currentByte = (_currentByte << 1) | ((value >> i) & 1);
- _bitsInCurrentByte++;
-
- if (_bitsInCurrentByte == 8)
- {
- _bytes.Add((byte)_currentByte);
- _currentByte = 0;
- _bitsInCurrentByte = 0;
- }
- }
- }
-
- public byte[] ToByteArray()
- {
- var result = new List(_bytes);
- if (_bitsInCurrentByte > 0)
- {
- // Pad remaining bits with zeros
- int padded = _currentByte << (8 - _bitsInCurrentByte);
- result.Add((byte)padded);
- }
- return result.ToArray();
- }
- }
}
}
From b6f299c16454655f066a8b0947063f7be2096f53 Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 09:47:18 +0000
Subject: [PATCH 05/14] Fix audio threading: separate cipher per handler, add
buffer locking, add exception handling
The AES-CBC cipher is stateful (tracks IV internally). Sharing one instance
between the concurrent control and data socket handlers caused cipher state
corruption, leading to decryption failures and crashes.
Fix: Each handler now creates its own cipher instance. RaopBuffer access is
synchronized with a lock. Try-catch blocks prevent socket errors from
silently killing the handler tasks.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
AirPlay/Listeners/AudioListener.cs | 235 +++++++++++++++++------------
1 file changed, 140 insertions(+), 95 deletions(-)
diff --git a/AirPlay/Listeners/AudioListener.cs b/AirPlay/Listeners/AudioListener.cs
index 8fbbaf8..8a02d16 100644
--- a/AirPlay/Listeners/AudioListener.cs
+++ b/AirPlay/Listeners/AudioListener.cs
@@ -23,11 +23,11 @@ public class AudioListener : BaseUdpListener
private readonly IRtspReceiver _receiver;
private readonly string _sessionId;
- private IBufferedCipher _aesCbcDecrypt;
private readonly OmgHax _omgHax = new OmgHax();
private IDecoder _decoder;
private readonly object _decoderLock = new object();
+ private readonly object _bufferLock = new object();
private ulong _sync_time;
private ulong _sync_timestamp;
private ushort _controlSequenceNumber = 0;
@@ -43,7 +43,6 @@ public AudioListener(IRtspReceiver receiver, string sessionId, ushort cport, ush
_dConfig = dConfig ?? throw new ArgumentNullException(nameof(dConfig));
_raopBuffer = RaopBufferInit();
- _aesCbcDecrypt = CipherUtilities.GetCipher("AES/CBC/NoPadding");
}
public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken cancellationToken)
@@ -52,6 +51,9 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
_cSocket = cSocket;
+ // Each handler gets its own cipher instance (cipher is stateful, not thread-safe)
+ var aesCbcDecrypt = CipherUtilities.GetCipher("AES/CBC/NoPadding");
+
// Get session by active-remove header value
var session = await SessionManager.Current.GetSessionAsync(_sessionId);
@@ -72,70 +74,90 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
do
{
- var cret = cSocket.Receive(packet, 0, RAOP_PACKET_LENGTH, SocketFlags.None, out SocketError error);
- if(error != SocketError.Success)
- {
- continue;
- }
-
- var mem = new MemoryStream(packet);
- using (var reader = new BinaryReader(mem))
+ try
{
- mem.Position = 1;
- int type_c = reader.ReadByte() & ~0x80;
- if (type_c == 0x56)
+ var cret = cSocket.Receive(packet, 0, RAOP_PACKET_LENGTH, SocketFlags.None, out SocketError error);
+ if(error != SocketError.Success)
{
- InitAesCbcCipher(session.DecryptedAesKey, session.EcdhShared, session.AesIv);
-
- mem.Position = 4;
- var data = reader.ReadBytes(cret - 4);
-
- var ret = RaopBufferQueue(_raopBuffer, data, (ushort)data.Length, session);
+ continue;
+ }
- // Dequeue and play audio received on control socket (used during screen mirroring)
- byte[] audiobuf;
- int audiobuflen = 0;
- uint timestamp = 0;
- while ((audiobuf = RaopBufferDequeue(_raopBuffer, ref audiobuflen, ref timestamp, true)) != null)
+ var mem = new MemoryStream(packet);
+ using (var reader = new BinaryReader(mem))
+ {
+ mem.Position = 1;
+ int type_c = reader.ReadByte() & ~0x80;
+ if (type_c == 0x56)
{
- if (audiobuf.Length == 0 || audiobuflen <= 0)
- continue;
-
- var pcmData = new PcmData();
- pcmData.Length = audiobuflen;
- pcmData.Data = audiobuf;
- pcmData.Pts = (ulong)(timestamp - _sync_timestamp) * 1000000UL / 44100 + _sync_time;
-
- _receiver.OnPCMData(pcmData);
+ InitAesCbcCipher(aesCbcDecrypt, session.DecryptedAesKey, session.EcdhShared, session.AesIv);
+
+ mem.Position = 4;
+ var data = reader.ReadBytes(cret - 4);
+
+ int ret;
+ lock (_bufferLock)
+ {
+ ret = RaopBufferQueue(_raopBuffer, data, (ushort)data.Length, session, aesCbcDecrypt);
+ }
+
+ // Dequeue and play audio received on control socket (used during screen mirroring)
+ byte[] audiobuf;
+ int audiobuflen = 0;
+ uint timestamp = 0;
+ lock (_bufferLock)
+ {
+ while ((audiobuf = RaopBufferDequeue(_raopBuffer, ref audiobuflen, ref timestamp, true)) != null)
+ {
+ if (audiobuf.Length == 0 || audiobuflen <= 0)
+ continue;
+
+ var pcmData = new PcmData();
+ pcmData.Length = audiobuflen;
+ pcmData.Data = audiobuf;
+ pcmData.Pts = (ulong)(timestamp - _sync_timestamp) * 1000000UL / 44100 + _sync_time;
+
+ _receiver.OnPCMData(pcmData);
+ }
+ }
+ }
+ else if (type_c == 0x54)
+ {
+ /**
+ * packetlen = 20
+ * bytes description
+ 8 RTP header without SSRC
+ 8 current NTP time
+ 4 RTP timestamp for the next audio packet
+ */
+
+ mem.Position = 8;
+ uint ntp_seconds = (uint)reader.ReadInt32();
+ uint ntp_fraction = (uint)reader.ReadInt32();
+ ulong ntp_time = ((ulong)ntp_seconds * 1000000UL) + (((ulong)ntp_fraction * 1000000UL) >> 32);
+ uint rtp_timestamp = (uint)((packet[4] << 24) | (packet[5] << 16) | (packet[6] << 8) | packet[7]);
+ uint next_timestamp = (uint)((packet[16] << 24) | (packet[17] << 16) | (packet[18] << 8) | packet[19]);
+
+ _sync_time = ntp_time - OFFSET_1900_TO_1970 * 1000000UL;
+ _sync_timestamp = rtp_timestamp;
}
}
- else if (type_c == 0x54)
- {
- /**
- * packetlen = 20
- * bytes description
- 8 RTP header without SSRC
- 8 current NTP time
- 4 RTP timestamp for the next audio packet
- */
-
- mem.Position = 8;
- uint ntp_seconds = (uint)reader.ReadInt32();
- uint ntp_fraction = (uint)reader.ReadInt32();
- ulong ntp_time = ((ulong)ntp_seconds * 1000000UL) + (((ulong)ntp_fraction * 1000000UL) >> 32);
- uint rtp_timestamp = (uint)((packet[4] << 24) | (packet[5] << 16) | (packet[6] << 8) | packet[7]);
- uint next_timestamp = (uint)((packet[16] << 24) | (packet[17] << 16) | (packet[18] << 8) | packet[19]);
-
- _sync_time = ntp_time - OFFSET_1900_TO_1970 * 1000000UL;
- _sync_timestamp = rtp_timestamp;
- }
- else
- {
- Console.WriteLine("Unknown packet");
- }
- }
- Array.Fill(packet, 0);
+ Array.Fill(packet, 0);
+ }
+ catch (ObjectDisposedException)
+ {
+ // Socket was closed (StopAsync called)
+ break;
+ }
+ catch (SocketException)
+ {
+ // Socket error (e.g., ICMP port unreachable on Windows)
+ break;
+ }
+ catch (Exception ex)
+ {
+ Console.WriteLine($"Audio control socket error: {ex.Message}");
+ }
} while (!cancellationToken.IsCancellationRequested);
Console.WriteLine("Closing audio control socket..");
@@ -145,6 +167,9 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
{
Console.WriteLine("Initializing recevie audio data from socket..");
+ // Each handler gets its own cipher instance (cipher is stateful, not thread-safe)
+ var aesCbcDecrypt = CipherUtilities.GetCipher("AES/CBC/NoPadding");
+
// Get current session
var session = await SessionManager.Current.GetSessionAsync(_sessionId);
@@ -165,53 +190,73 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
do
{
- var dret = dSocket.Receive(packet, 0, RAOP_PACKET_LENGTH, SocketFlags.None, out SocketError error);
- if (error != SocketError.Success)
+ try
{
- continue;
- }
+ var dret = dSocket.Receive(packet, 0, RAOP_PACKET_LENGTH, SocketFlags.None, out SocketError error);
+ if (error != SocketError.Success)
+ {
+ continue;
+ }
- // RTP payload type
- int type_d = packet[1] & ~0x80;
+ // RTP payload type
+ int type_d = packet[1] & ~0x80;
- if (packet.Length >= 12)
- {
- InitAesCbcCipher(session.DecryptedAesKey, session.EcdhShared, session.AesIv);
+ if (packet.Length >= 12)
+ {
+ InitAesCbcCipher(aesCbcDecrypt, session.DecryptedAesKey, session.EcdhShared, session.AesIv);
- bool no_resend = false;
- int buf_ret;
- byte[] audiobuf;
- int audiobuflen = 0;
- uint timestamp = 0;
+ bool no_resend = false;
+ int buf_ret;
+ byte[] audiobuf;
+ int audiobuflen = 0;
+ uint timestamp = 0;
- buf_ret = RaopBufferQueue(_raopBuffer, packet, (ushort)dret, session);
+ lock (_bufferLock)
+ {
+ buf_ret = RaopBufferQueue(_raopBuffer, packet, (ushort)dret, session, aesCbcDecrypt);
+ }
- //if(_raopBuffer.LastSeqNum - _raopBuffer.FirstSeqNum > (RAOP_BUFFER_LENGTH / 8))
- //{
// Dequeue all available frames from buffer
- while ((audiobuf = RaopBufferDequeue(_raopBuffer, ref audiobuflen, ref timestamp, no_resend)) != null)
+ lock (_bufferLock)
{
- if (audiobuf.Length == 0 || audiobuflen <= 0)
- continue;
+ while ((audiobuf = RaopBufferDequeue(_raopBuffer, ref audiobuflen, ref timestamp, no_resend)) != null)
+ {
+ if (audiobuf.Length == 0 || audiobuflen <= 0)
+ continue;
- var pcmData = new PcmData();
- pcmData.Length = audiobuflen;
- pcmData.Data = audiobuf;
+ var pcmData = new PcmData();
+ pcmData.Length = audiobuflen;
+ pcmData.Data = audiobuf;
- pcmData.Pts = (ulong)(timestamp - _sync_timestamp) * 1000000UL / 44100 + _sync_time;
+ pcmData.Pts = (ulong)(timestamp - _sync_timestamp) * 1000000UL / 44100 + _sync_time;
- _receiver.OnPCMData(pcmData);
+ _receiver.OnPCMData(pcmData);
+ }
}
- //}
- /* Handle possible resend requests */
- if (!no_resend)
- {
- RaopBufferHandleResends(_raopBuffer, _cSocket, _controlSequenceNumber);
+ /* Handle possible resend requests */
+ if (!no_resend)
+ {
+ RaopBufferHandleResends(_raopBuffer, _cSocket, _controlSequenceNumber);
+ }
}
- }
- Array.Clear(packet, 0, packet.Length);
+ Array.Clear(packet, 0, packet.Length);
+ }
+ catch (ObjectDisposedException)
+ {
+ // Socket was closed (StopAsync called)
+ break;
+ }
+ catch (SocketException)
+ {
+ // Socket error (e.g., ICMP port unreachable on Windows)
+ break;
+ }
+ catch (Exception ex)
+ {
+ Console.WriteLine($"Audio data socket error: {ex.Message}");
+ }
} while (!cancellationToken.IsCancellationRequested);
Console.WriteLine("Closing audio data socket..");
@@ -224,7 +269,7 @@ public Task FlushAsync(int nextSequence)
return Task.CompletedTask;
}
- private void InitAesCbcCipher(byte[] aesKey, byte[] ecdhShared, byte[] aesIv)
+ private void InitAesCbcCipher(IBufferedCipher aesCbcDecrypt, byte[] aesKey, byte[] ecdhShared, byte[] aesIv)
{
byte[] hash = Utilities.Hash(aesKey, ecdhShared);
byte[] eaesKey = Utilities.CopyOfRange(hash, 0, 16);
@@ -232,7 +277,7 @@ private void InitAesCbcCipher(byte[] aesKey, byte[] ecdhShared, byte[] aesIv)
var keyParameter = ParameterUtilities.CreateKeyParameter("AES", eaesKey);
var cipherParameters = new ParametersWithIV(keyParameter, aesIv, 0, aesIv.Length);
- _aesCbcDecrypt.Init(false, cipherParameters);
+ aesCbcDecrypt.Init(false, cipherParameters);
}
private RaopBuffer RaopBufferInit()
@@ -259,7 +304,7 @@ private RaopBuffer RaopBufferInit()
return raop_buffer;
}
- public int RaopBufferQueue(RaopBuffer raop_buffer, byte[] data, ushort datalen, Session session)
+ public int RaopBufferQueue(RaopBuffer raop_buffer, byte[] data, ushort datalen, Session session, IBufferedCipher aesCbcDecrypt)
{
int encryptedlen;
RaopBufferEntry entry;
@@ -310,7 +355,7 @@ public int RaopBufferQueue(RaopBuffer raop_buffer, byte[] data, ushort datalen,
if (encryptedlen > 0)
{
- _aesCbcDecrypt.ProcessBytes(data, 12, encryptedlen, data, 12);
+ aesCbcDecrypt.ProcessBytes(data, 12, encryptedlen, data, 12);
Array.Copy(data, 12, raw, 0, encryptedlen);
}
From 383b580db947c5ab955a3854e0f79e919161e5da Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 09:49:51 +0000
Subject: [PATCH 06/14] Move OnPCMData calls outside buffer lock to avoid
blocking
Dequeue PCM data into a batch while holding the lock, then deliver
outside the lock. This prevents the receiver's processing time from
blocking the other socket handler.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
AirPlay/Listeners/AudioListener.cs | 18 ++++++++++++++++--
1 file changed, 16 insertions(+), 2 deletions(-)
diff --git a/AirPlay/Listeners/AudioListener.cs b/AirPlay/Listeners/AudioListener.cs
index 8a02d16..45824e0 100644
--- a/AirPlay/Listeners/AudioListener.cs
+++ b/AirPlay/Listeners/AudioListener.cs
@@ -101,6 +101,7 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
}
// Dequeue and play audio received on control socket (used during screen mirroring)
+ var pcmBatch = new System.Collections.Generic.List();
byte[] audiobuf;
int audiobuflen = 0;
uint timestamp = 0;
@@ -116,9 +117,15 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
pcmData.Data = audiobuf;
pcmData.Pts = (ulong)(timestamp - _sync_timestamp) * 1000000UL / 44100 + _sync_time;
- _receiver.OnPCMData(pcmData);
+ pcmBatch.Add(pcmData);
}
}
+
+ // Deliver PCM outside the lock to avoid blocking the data handler
+ foreach (var pcm in pcmBatch)
+ {
+ _receiver.OnPCMData(pcm);
+ }
}
else if (type_c == 0x54)
{
@@ -217,6 +224,7 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
}
// Dequeue all available frames from buffer
+ var pcmBatch = new System.Collections.Generic.List();
lock (_bufferLock)
{
while ((audiobuf = RaopBufferDequeue(_raopBuffer, ref audiobuflen, ref timestamp, no_resend)) != null)
@@ -230,10 +238,16 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
pcmData.Pts = (ulong)(timestamp - _sync_timestamp) * 1000000UL / 44100 + _sync_time;
- _receiver.OnPCMData(pcmData);
+ pcmBatch.Add(pcmData);
}
}
+ // Deliver PCM outside the lock to avoid blocking the control handler
+ foreach (var pcm in pcmBatch)
+ {
+ _receiver.OnPCMData(pcm);
+ }
+
/* Handle possible resend requests */
if (!no_resend)
{
From 7c9b8d91af5da00221ea60f51b38186a37bedda3 Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 10:02:49 +0000
Subject: [PATCH 07/14] Add comprehensive debug logging throughout audio
pipeline
All log lines prefixed with [DEBUG-*] tags for easy filtering:
- [DEBUG-C] Control socket (port 7002): packet types, counts, exit reason
- [DEBUG-D] Data socket (port 7003): packet counts, queue/dequeue stats
- [DEBUG-FFMPEG] FFmpeg decoder: frame I/O, process health, ALL stderr
- [DEBUG-FFMPEG-STDERR] Raw FFmpeg stderr output
- [DEBUG-PCM] PCM data delivery with silence detection
- [DEBUG-AUDIO] AudioOutputService sample delivery and queue state
- [DEBUG-SETUP] AudioListener creation/stop lifecycle
- [DEBUG-TEARDOWN] Session teardown with stream type
- [DEBUG-UDP] Socket start/stop with port numbers
This will show exactly where audio data flow stops.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
AirPlay/AirPlayService.cs | 13 ++
.../Implementations/FFmpegAacEldDecoder.cs | 30 +++-
AirPlay/Listeners/AirTunesListener.cs | 9 +-
AirPlay/Listeners/AudioListener.cs | 130 +++++++++++++++---
AirPlay/Listeners/Bases/BaseUdpListener.cs | 16 ++-
AirPlay/Services/AudioOutputService.cs | 21 ++-
6 files changed, 184 insertions(+), 35 deletions(-)
diff --git a/AirPlay/AirPlayService.cs b/AirPlay/AirPlayService.cs
index dea74d3..0d4781e 100644
--- a/AirPlay/AirPlayService.cs
+++ b/AirPlay/AirPlayService.cs
@@ -147,8 +147,21 @@ public async Task StartAsync(CancellationToken cancellationToken)
};
_audiobuf = new List();
+ int pcmReceivedCount = 0;
_airPlayReceiver.OnPCMDataReceived += (s, e) =>
{
+ pcmReceivedCount++;
+ if (pcmReceivedCount <= 5 || pcmReceivedCount % 500 == 0)
+ {
+ bool allZeros = true;
+ int checkLen = Math.Min(e.Length, 100);
+ for (int i = 0; i < checkLen; i++)
+ {
+ if (e.Data[i] != 0) { allZeros = false; break; }
+ }
+ Console.WriteLine($"[DEBUG-PCM] OnPCMDataReceived #{pcmReceivedCount}: len={e.Length}, allZeros={allZeros}");
+ }
+
// Play audio through speakers
lock (_audioOutputLock)
{
diff --git a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
index c2f419a..4d5caae 100644
--- a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
+++ b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
@@ -37,6 +37,8 @@ public class FFmpegAacEldDecoder : IDecoder, IDisposable
private bool _initialized;
private readonly byte[] _adtsHeader = new byte[7];
private int _adtsFreqIdx;
+ private int _decodeCallCount;
+ private int _totalFramesDecoded;
public AudioFormat Type => AudioFormat.AAC_ELD;
@@ -83,11 +85,8 @@ public int Config(int sampleRate, int channels, int bitDepth, int frameLength)
string line;
while ((line = reader.ReadLine()) != null)
{
- // Only log critical errors, not per-frame decode warnings
- if (line.Contains("error", StringComparison.OrdinalIgnoreCase))
- {
- Console.WriteLine($"FFmpeg: {line}");
- }
+ // Log all FFmpeg stderr output for debugging
+ Console.WriteLine($"[DEBUG-FFMPEG-STDERR] {line}");
}
}
catch { /* ignore */ }
@@ -115,10 +114,17 @@ public int GetOutputStreamLength()
public int DecodeFrame(byte[] input, ref byte[] output, int length)
{
if (!_initialized || _ffmpegProcess == null || _ffmpegProcess.HasExited)
+ {
+ _decodeCallCount++;
+ if (_decodeCallCount <= 3 || _decodeCallCount % 500 == 0)
+ Console.WriteLine($"[DEBUG-FFMPEG] DecodeFrame called but process not ready: initialized={_initialized}, process={_ffmpegProcess != null}, hasExited={_ffmpegProcess?.HasExited}");
return -1;
+ }
try
{
+ _decodeCallCount++;
+
// Wrap the raw AAC frame in an ADTS header for FFmpeg to parse.
// ADTS profile is set to AAC-LC (profile=2) since ADTS only has a 2-bit
// profile field and cannot represent AAC-ELD. FFmpeg's decoder handles
@@ -131,6 +137,9 @@ public int DecodeFrame(byte[] input, ref byte[] output, int length)
_ffmpegInput.Write(input, 0, input.Length);
_ffmpegInput.Flush();
+ if (_decodeCallCount <= 3)
+ Console.WriteLine($"[DEBUG-FFMPEG] Wrote frame #{_decodeCallCount}: inputLen={input.Length}, adtsFrameLen={frameLen}, expecting {_pcmOutputSize} bytes PCM output");
+
// Read decoded PCM from FFmpeg stdout
int bytesToRead = _pcmOutputSize;
int totalRead = 0;
@@ -148,16 +157,25 @@ public int DecodeFrame(byte[] input, ref byte[] output, int length)
totalRead += read;
}
+ _totalFramesDecoded++;
+
+ if (_decodeCallCount <= 5 || _decodeCallCount % 500 == 0)
+ Console.WriteLine($"[DEBUG-FFMPEG] Frame #{_decodeCallCount}: read {totalRead}/{bytesToRead} bytes (attemptsLeft={maxAttempts}), totalDecoded={_totalFramesDecoded}");
+
if (totalRead < bytesToRead)
{
// Partial read - zero fill the rest
Array.Clear(output, totalRead, bytesToRead - totalRead);
+ if (_decodeCallCount <= 10)
+ Console.WriteLine($"[DEBUG-FFMPEG] WARNING: Partial read! Only {totalRead} of {bytesToRead} bytes, zero-filled remainder");
}
return 0;
}
- catch (Exception)
+ catch (Exception ex)
{
+ if (_decodeCallCount <= 5 || _decodeCallCount % 500 == 0)
+ Console.WriteLine($"[DEBUG-FFMPEG] DecodeFrame exception #{_decodeCallCount}: {ex.GetType().Name}: {ex.Message}");
return -1;
}
}
diff --git a/AirPlay/Listeners/AirTunesListener.cs b/AirPlay/Listeners/AirTunesListener.cs
index 669ba1d..82f463b 100644
--- a/AirPlay/Listeners/AirTunesListener.cs
+++ b/AirPlay/Listeners/AirTunesListener.cs
@@ -454,14 +454,17 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
// (ports 7002/7003 must be released first)
if (session.AudioControlListener != null)
{
+ Console.WriteLine("[DEBUG-SETUP] Stopping existing AudioListener before creating new one...");
try { await session.AudioControlListener.StopAsync(); }
- catch (Exception ex) { Console.WriteLine($"Error stopping old audio listener: {ex.Message}"); }
+ catch (Exception ex) { Console.WriteLine($"[DEBUG-SETUP] Error stopping old audio listener: {ex.Message}"); }
session.AudioControlListener = null;
}
+ Console.WriteLine($"[DEBUG-SETUP] Creating new AudioListener: format={session.AudioFormat}, ct={session.AudioCompressionType}, spf={session.AudioSamplesPerFrame}, mirroring={session.MirroringSession}");
// Start 'AudioListener' (handle PCM/AAC/ALAC data received from iOS/macOS
var control = new AudioListener(_receiver, session.SessionId, 7002, 7003, _dumpConfig);
await control.StartAsync(cancellationToken).ConfigureAwait(false);
+ Console.WriteLine("[DEBUG-SETUP] AudioListener started successfully");
session.AudioControlListener = control;
}
@@ -564,6 +567,7 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
}
if (request.Type == RequestType.TEARDOWN)
{
+ Console.WriteLine("[DEBUG-TEARDOWN] TEARDOWN request received");
var plistReader = new BinaryPlistReader();
using (var mem = new MemoryStream(request.Body))
{
@@ -574,10 +578,12 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
// Always one foreach request
var stream = (Dictionary)((object[])plist["streams"]).Last();
var type = (short)stream["type"];
+ Console.WriteLine($"[DEBUG-TEARDOWN] Stream type: {type}");
// If screen Mirroring
if (type == 110)
{
+ Console.WriteLine("[DEBUG-TEARDOWN] Stopping mirroring session");
// Stop mirroring session
if (session.MirroringListener != null)
{
@@ -592,6 +598,7 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
// If audio session
if (type == 96)
{
+ Console.WriteLine("[DEBUG-TEARDOWN] Stopping audio session");
// Stop audio session
if (session.AudioControlListener != null)
{
diff --git a/AirPlay/Listeners/AudioListener.cs b/AirPlay/Listeners/AudioListener.cs
index 45824e0..3148534 100644
--- a/AirPlay/Listeners/AudioListener.cs
+++ b/AirPlay/Listeners/AudioListener.cs
@@ -47,7 +47,7 @@ public AudioListener(IRtspReceiver receiver, string sessionId, ushort cport, ush
public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken cancellationToken)
{
- Console.WriteLine("Initializing recevie audio control from socket..");
+ Console.WriteLine("[DEBUG-C] OnRawCSocketAsync started");
_cSocket = cSocket;
@@ -56,6 +56,7 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
// Get session by active-remove header value
var session = await SessionManager.Current.GetSessionAsync(_sessionId);
+ Console.WriteLine($"[DEBUG-C] Session loaded: AesKey={session.AesKey != null}, AesIv={session.AesIv != null}, EcdhShared={session.EcdhShared != null}, KeyMsg={session.KeyMsg != null}, AudioFormat={session.AudioFormat}");
// If we have not decripted session AesKey
if (session.DecryptedAesKey == null)
@@ -63,14 +64,27 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
byte[] decryptedAesKey = new byte[16];
_omgHax.DecryptAesKey(session.KeyMsg, session.AesKey, decryptedAesKey);
session.DecryptedAesKey = decryptedAesKey;
+ Console.WriteLine("[DEBUG-C] AES key decrypted");
}
// Initialize decoder (needed for type 0x56 audio packets during mirroring)
InitializeDecoder(session);
+ Console.WriteLine($"[DEBUG-C] Decoder initialized: type={_decoder?.Type}, outputLen={_decoder?.GetOutputStreamLength()}");
await SessionManager.Current.CreateOrUpdateSessionAsync(_sessionId, session);
var packet = new byte[RAOP_PACKET_LENGTH];
+ int cPacketCount = 0;
+ int c56Count = 0;
+ int c54Count = 0;
+ int cOtherCount = 0;
+ int cQueuedCount = 0;
+ int cDequeuedCount = 0;
+ int cPcmDelivered = 0;
+ int cSocketErrors = 0;
+ string exitReason = "loop-end";
+
+ Console.WriteLine("[DEBUG-C] Entering receive loop...");
do
{
@@ -79,9 +93,16 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
var cret = cSocket.Receive(packet, 0, RAOP_PACKET_LENGTH, SocketFlags.None, out SocketError error);
if(error != SocketError.Success)
{
+ cSocketErrors++;
+ if (cSocketErrors <= 5)
+ Console.WriteLine($"[DEBUG-C] Socket.Receive error: {error}");
continue;
}
+ cPacketCount++;
+ if (cPacketCount == 1)
+ Console.WriteLine($"[DEBUG-C] First packet received! size={cret}");
+
var mem = new MemoryStream(packet);
using (var reader = new BinaryReader(mem))
{
@@ -89,17 +110,26 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
int type_c = reader.ReadByte() & ~0x80;
if (type_c == 0x56)
{
+ c56Count++;
InitAesCbcCipher(aesCbcDecrypt, session.DecryptedAesKey, session.EcdhShared, session.AesIv);
mem.Position = 4;
var data = reader.ReadBytes(cret - 4);
+ if (c56Count <= 3)
+ Console.WriteLine($"[DEBUG-C] 0x56 packet #{c56Count}: dataLen={data.Length}, seqNum={(ushort)((data[2] << 8) | data[3])}");
+
int ret;
lock (_bufferLock)
{
ret = RaopBufferQueue(_raopBuffer, data, (ushort)data.Length, session, aesCbcDecrypt);
}
+ if (c56Count <= 3)
+ Console.WriteLine($"[DEBUG-C] RaopBufferQueue returned: {ret}");
+
+ if (ret > 0) cQueuedCount++;
+
// Dequeue and play audio received on control socket (used during screen mirroring)
var pcmBatch = new System.Collections.Generic.List();
byte[] audiobuf;
@@ -112,6 +142,7 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
if (audiobuf.Length == 0 || audiobuflen <= 0)
continue;
+ cDequeuedCount++;
var pcmData = new PcmData();
pcmData.Length = audiobuflen;
pcmData.Data = audiobuf;
@@ -121,22 +152,19 @@ public override async Task OnRawCSocketAsync(Socket cSocket, CancellationToken c
}
}
+ if (c56Count <= 3 && pcmBatch.Count > 0)
+ Console.WriteLine($"[DEBUG-C] Dequeued {pcmBatch.Count} PCM frames, first len={pcmBatch[0].Length}");
+
// Deliver PCM outside the lock to avoid blocking the data handler
foreach (var pcm in pcmBatch)
{
_receiver.OnPCMData(pcm);
+ cPcmDelivered++;
}
}
else if (type_c == 0x54)
{
- /**
- * packetlen = 20
- * bytes description
- 8 RTP header without SSRC
- 8 current NTP time
- 4 RTP timestamp for the next audio packet
- */
-
+ c54Count++;
mem.Position = 8;
uint ntp_seconds = (uint)reader.ReadInt32();
uint ntp_fraction = (uint)reader.ReadInt32();
@@ -146,39 +174,60 @@ 8 current NTP time
_sync_time = ntp_time - OFFSET_1900_TO_1970 * 1000000UL;
_sync_timestamp = rtp_timestamp;
+
+ if (c54Count <= 3)
+ Console.WriteLine($"[DEBUG-C] 0x54 sync packet #{c54Count}: rtp_ts={rtp_timestamp}");
+ }
+ else
+ {
+ cOtherCount++;
+ if (cOtherCount <= 5)
+ Console.WriteLine($"[DEBUG-C] Unknown packet type: 0x{type_c:X2}, size={cret}");
}
}
+ // Log summary periodically
+ if (cPacketCount % 500 == 0)
+ {
+ Console.WriteLine($"[DEBUG-C] Stats: packets={cPacketCount}, 0x56={c56Count}, 0x54={c54Count}, other={cOtherCount}, queued={cQueuedCount}, dequeued={cDequeuedCount}, pcmDelivered={cPcmDelivered}, socketErrors={cSocketErrors}");
+ }
+
Array.Fill(packet, 0);
}
catch (ObjectDisposedException)
{
- // Socket was closed (StopAsync called)
+ exitReason = "ObjectDisposedException (socket closed)";
break;
}
- catch (SocketException)
+ catch (SocketException ex)
{
- // Socket error (e.g., ICMP port unreachable on Windows)
+ exitReason = $"SocketException: {ex.SocketErrorCode} - {ex.Message}";
break;
}
catch (Exception ex)
{
- Console.WriteLine($"Audio control socket error: {ex.Message}");
+ Console.WriteLine($"[DEBUG-C] Exception in receive loop: {ex.GetType().Name}: {ex.Message}");
+ Console.WriteLine($"[DEBUG-C] Stack trace: {ex.StackTrace}");
}
} while (!cancellationToken.IsCancellationRequested);
- Console.WriteLine("Closing audio control socket..");
+ if (cancellationToken.IsCancellationRequested)
+ exitReason = "CancellationToken requested";
+
+ Console.WriteLine($"[DEBUG-C] Closing audio control socket. Reason: {exitReason}");
+ Console.WriteLine($"[DEBUG-C] Final stats: packets={cPacketCount}, 0x56={c56Count}, 0x54={c54Count}, other={cOtherCount}, queued={cQueuedCount}, dequeued={cDequeuedCount}, pcmDelivered={cPcmDelivered}, socketErrors={cSocketErrors}");
}
public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken cancellationToken)
{
- Console.WriteLine("Initializing recevie audio data from socket..");
+ Console.WriteLine("[DEBUG-D] OnRawDSocketAsync started");
// Each handler gets its own cipher instance (cipher is stateful, not thread-safe)
var aesCbcDecrypt = CipherUtilities.GetCipher("AES/CBC/NoPadding");
// Get current session
var session = await SessionManager.Current.GetSessionAsync(_sessionId);
+ Console.WriteLine($"[DEBUG-D] Session loaded: AesKey={session.AesKey != null}, AesIv={session.AesIv != null}, EcdhShared={session.EcdhShared != null}, KeyMsg={session.KeyMsg != null}, AudioFormat={session.AudioFormat}");
// If we have not decripted session AesKey
if (session.DecryptedAesKey == null)
@@ -186,14 +235,24 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
byte[] decryptedAesKey = new byte[16];
_omgHax.DecryptAesKey(session.KeyMsg, session.AesKey, decryptedAesKey);
session.DecryptedAesKey = decryptedAesKey;
+ Console.WriteLine("[DEBUG-D] AES key decrypted");
}
// Initialize decoder
InitializeDecoder(session);
+ Console.WriteLine($"[DEBUG-D] Decoder initialized: type={_decoder?.Type}, outputLen={_decoder?.GetOutputStreamLength()}");
await SessionManager.Current.CreateOrUpdateSessionAsync(_sessionId, session);
var packet = new byte[RAOP_PACKET_LENGTH];
+ int dPacketCount = 0;
+ int dQueuedCount = 0;
+ int dDequeuedCount = 0;
+ int dPcmDelivered = 0;
+ int dSocketErrors = 0;
+ string exitReason = "loop-end";
+
+ Console.WriteLine("[DEBUG-D] Entering receive loop...");
do
{
@@ -202,12 +261,22 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
var dret = dSocket.Receive(packet, 0, RAOP_PACKET_LENGTH, SocketFlags.None, out SocketError error);
if (error != SocketError.Success)
{
+ dSocketErrors++;
+ if (dSocketErrors <= 5)
+ Console.WriteLine($"[DEBUG-D] Socket.Receive error: {error}");
continue;
}
+ dPacketCount++;
+ if (dPacketCount == 1)
+ Console.WriteLine($"[DEBUG-D] First packet received! size={dret}");
+
// RTP payload type
int type_d = packet[1] & ~0x80;
+ if (dPacketCount <= 3)
+ Console.WriteLine($"[DEBUG-D] Packet #{dPacketCount}: type=0x{type_d:X2}, size={dret}");
+
if (packet.Length >= 12)
{
InitAesCbcCipher(aesCbcDecrypt, session.DecryptedAesKey, session.EcdhShared, session.AesIv);
@@ -223,6 +292,11 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
buf_ret = RaopBufferQueue(_raopBuffer, packet, (ushort)dret, session, aesCbcDecrypt);
}
+ if (dPacketCount <= 3)
+ Console.WriteLine($"[DEBUG-D] RaopBufferQueue returned: {buf_ret}");
+
+ if (buf_ret > 0) dQueuedCount++;
+
// Dequeue all available frames from buffer
var pcmBatch = new System.Collections.Generic.List();
lock (_bufferLock)
@@ -232,6 +306,7 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
if (audiobuf.Length == 0 || audiobuflen <= 0)
continue;
+ dDequeuedCount++;
var pcmData = new PcmData();
pcmData.Length = audiobuflen;
pcmData.Data = audiobuf;
@@ -242,10 +317,14 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
}
}
+ if (dPacketCount <= 3 && pcmBatch.Count > 0)
+ Console.WriteLine($"[DEBUG-D] Dequeued {pcmBatch.Count} PCM frames, first len={pcmBatch[0].Length}");
+
// Deliver PCM outside the lock to avoid blocking the control handler
foreach (var pcm in pcmBatch)
{
_receiver.OnPCMData(pcm);
+ dPcmDelivered++;
}
/* Handle possible resend requests */
@@ -255,25 +334,36 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
}
}
+ // Log summary periodically
+ if (dPacketCount % 500 == 0)
+ {
+ Console.WriteLine($"[DEBUG-D] Stats: packets={dPacketCount}, queued={dQueuedCount}, dequeued={dDequeuedCount}, pcmDelivered={dPcmDelivered}, socketErrors={dSocketErrors}");
+ }
+
Array.Clear(packet, 0, packet.Length);
}
catch (ObjectDisposedException)
{
- // Socket was closed (StopAsync called)
+ exitReason = "ObjectDisposedException (socket closed)";
break;
}
- catch (SocketException)
+ catch (SocketException ex)
{
- // Socket error (e.g., ICMP port unreachable on Windows)
+ exitReason = $"SocketException: {ex.SocketErrorCode} - {ex.Message}";
break;
}
catch (Exception ex)
{
- Console.WriteLine($"Audio data socket error: {ex.Message}");
+ Console.WriteLine($"[DEBUG-D] Exception in receive loop: {ex.GetType().Name}: {ex.Message}");
+ Console.WriteLine($"[DEBUG-D] Stack trace: {ex.StackTrace}");
}
} while (!cancellationToken.IsCancellationRequested);
- Console.WriteLine("Closing audio data socket..");
+ if (cancellationToken.IsCancellationRequested)
+ exitReason = "CancellationToken requested";
+
+ Console.WriteLine($"[DEBUG-D] Closing audio data socket. Reason: {exitReason}");
+ Console.WriteLine($"[DEBUG-D] Final stats: packets={dPacketCount}, queued={dQueuedCount}, dequeued={dDequeuedCount}, pcmDelivered={dPcmDelivered}, socketErrors={dSocketErrors}");
}
public Task FlushAsync(int nextSequence)
diff --git a/AirPlay/Listeners/Bases/BaseUdpListener.cs b/AirPlay/Listeners/Bases/BaseUdpListener.cs
index 049d9bb..a16f3df 100644
--- a/AirPlay/Listeners/Bases/BaseUdpListener.cs
+++ b/AirPlay/Listeners/Bases/BaseUdpListener.cs
@@ -1,4 +1,5 @@
-using System.Net;
+using System;
+using System.Net;
using System.Net.Sockets;
using System.Threading;
using System.Threading.Tasks;
@@ -6,15 +7,19 @@
namespace AirPlay.Listeners
{
public class BaseUdpListener : BaseListener
- {
- public const int CloseTimeout = 1000;
-
+ {
+ public const int CloseTimeout = 1000;
+
private readonly Socket _cSocket;
private readonly Socket _dSocket;
+ private readonly ushort _cPort;
+ private readonly ushort _dPort;
private readonly CancellationTokenSource _cancellationTokenSource;
public BaseUdpListener(ushort cPort, ushort dPort)
{
+ _cPort = cPort;
+ _dPort = dPort;
_cSocket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, System.Net.Sockets.ProtocolType.Udp);
_dSocket = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, System.Net.Sockets.ProtocolType.Udp);
@@ -28,6 +33,7 @@ public override Task StartAsync(CancellationToken cancellationToken)
{
var source = CancellationTokenSource.CreateLinkedTokenSource(cancellationToken, _cancellationTokenSource.Token);
+ Console.WriteLine($"[DEBUG-UDP] StartAsync: cSocket on port {_cPort}, dSocket on port {_dPort}");
Task.Run(() => OnRawCSocketAsync(_cSocket, source.Token), source.Token);
Task.Run(() => OnRawDSocketAsync(_dSocket, source.Token), source.Token);
@@ -36,10 +42,12 @@ public override Task StartAsync(CancellationToken cancellationToken)
public override Task StopAsync()
{
+ Console.WriteLine("[DEBUG-UDP] StopAsync called - cancelling tokens and closing sockets");
_cancellationTokenSource.Cancel();
_cSocket.Close(CloseTimeout);
_dSocket.Close(CloseTimeout);
+ Console.WriteLine("[DEBUG-UDP] StopAsync completed");
return Task.CompletedTask;
}
diff --git a/AirPlay/Services/AudioOutputService.cs b/AirPlay/Services/AudioOutputService.cs
index af16922..b08cd9b 100644
--- a/AirPlay/Services/AudioOutputService.cs
+++ b/AirPlay/Services/AudioOutputService.cs
@@ -182,10 +182,23 @@ public void AddSamples(byte[] buffer, int offset, int count)
_streamProvider.AddSamples(buffer, offset, count);
_sampleCount++;
+ // Log first few samples for debugging
+ if (_sampleCount <= 5 || _sampleCount % 500 == 0)
+ {
+ // Check if data is all zeros (silence)
+ bool allZeros = true;
+ int checkLen = Math.Min(count, 100);
+ for (int i = offset; i < offset + checkLen; i++)
+ {
+ if (buffer[i] != 0) { allZeros = false; break; }
+ }
+ Console.WriteLine($"[DEBUG-AUDIO] AddSamples #{_sampleCount}: count={count}, allZeros={allZeros}, queue={_streamProvider.QueuedBuffers}, initialized={_initialized}, playing={_isPlaying}");
+ }
+
// Initialize DirectSound AFTER prebuffering
if (!_initialized && _sampleCount >= PREBUFFER_PACKETS)
{
- Console.WriteLine($"Prebuffered {_streamProvider.QueuedBuffers} packets, calling Init()...");
+ Console.WriteLine($"[DEBUG-AUDIO] Prebuffered {_streamProvider.QueuedBuffers} packets, calling Init()...");
_waveOut.Init(_streamProvider);
_initialized = true;
}
@@ -195,7 +208,7 @@ public void AddSamples(byte[] buffer, int offset, int count)
{
_waveOut.Play();
_isPlaying = true;
- Console.WriteLine($"Audio playback started ({_streamProvider.QueuedBuffers} packets buffered)");
+ Console.WriteLine($"[DEBUG-AUDIO] Audio playback started ({_streamProvider.QueuedBuffers} packets buffered)");
}
// Log status periodically
@@ -203,12 +216,12 @@ public void AddSamples(byte[] buffer, int offset, int count)
{
var state = _waveOut.PlaybackState;
var queued = _streamProvider.QueuedBuffers;
- Console.WriteLine($"Audio: {_sampleCount} packets | Queue: {queued} | State: {state}");
+ Console.WriteLine($"[DEBUG-AUDIO] Audio: {_sampleCount} packets | Queue: {queued} | State: {state}");
}
}
catch (Exception ex)
{
- Console.WriteLine($"Audio error: {ex.Message}");
+ Console.WriteLine($"[DEBUG-AUDIO] Audio error in AddSamples: {ex.GetType().Name}: {ex.Message}");
}
}
}
From 679dfcd287bc37f36f3e10f57ebaa071a20318f3 Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 10:20:53 +0000
Subject: [PATCH 08/14] Add RTSP flow debug logging and SETPEERS support
Trace the full RTSP request/response flow to identify why iOS tears
down audio immediately during screen mirroring:
- [DEBUG-RTSP] logs every request type and response status
- [DEBUG-SETUP] logs all stream properties from SETUP requests
- [DEBUG-QUEUE] logs RaopBufferQueue decisions (no-data markers, old seqnums)
- [DEBUG-RTSP] logs unrecognized data that doesn't match any request pattern
Also add SETPEERS support:
- iOS sends SETPEERS requests for peer discovery
- Without a handler, these were silently dropped (no regex match, no response)
- This could cause iOS to think the connection is broken
- Added to RequestConst, RequestType enum, Request parser, and regex pattern
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
AirPlay/Listeners/AirTunesListener.cs | 17 +++++++++-
AirPlay/Listeners/AudioListener.cs | 13 ++++++++
AirPlay/Listeners/Bases/BaseTcpListener.cs | 36 ++++++++++++++--------
AirPlay/Models/Enums/RequestConst.cs | 1 +
AirPlay/Models/Enums/RequestType.cs | 3 +-
AirPlay/Models/TcpListeners/Request.cs | 4 +++
6 files changed, 60 insertions(+), 14 deletions(-)
diff --git a/AirPlay/Listeners/AirTunesListener.cs b/AirPlay/Listeners/AirTunesListener.cs
index 82f463b..c7c36d3 100644
--- a/AirPlay/Listeners/AirTunesListener.cs
+++ b/AirPlay/Listeners/AirTunesListener.cs
@@ -291,6 +291,14 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
// Always one foreach request
var stream = (Dictionary)((object[])plist["streams"])[0];
var type = (short)stream["type"];
+ Console.WriteLine($"[DEBUG-SETUP] Stream type={type}, keys=[{string.Join(", ", stream.Keys)}]");
+ foreach (var kv in stream)
+ {
+ if (kv.Value is byte[] ba)
+ Console.WriteLine($"[DEBUG-SETUP] {kv.Key} = byte[{ba.Length}]");
+ else
+ Console.WriteLine($"[DEBUG-SETUP] {kv.Key} = {kv.Value}");
+ }
// If screen Mirroring
if (type == 110)
@@ -388,6 +396,7 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
}
else
{
+ Console.WriteLine($"[DEBUG-SETUP] Initial SETUP (keys/timing): et={plist.Contains("et")}, ekey={plist.Contains("ekey")}, eiv={plist.Contains("eiv")}, isScreenMirroringSession={plist.Contains("isScreenMirroringSession")}, timingPort={plist.Contains("timingPort")}");
// Read ekey and eiv used to decode video and audio data
if (plist.Contains("et"))
{
@@ -483,6 +492,7 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
}
if (request.Type == RequestType.RECORD)
{
+ Console.WriteLine("[DEBUG-RTSP] RECORD request received");
response.Headers.Add("Audio-Latency", "0"); // 11025
// response.Headers.Add("Audio-Jack-Status", "connected; type=analog");
}
@@ -533,11 +543,16 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
}
if(request.Type == RequestType.OPTIONS)
{
- response.Headers.Add("Public", "SETUP, RECORD, PAUSE, FLUSH, TEARDOWN, OPTIONS, GET_PARAMETER, SET_PARAMETER, ANNOUNCE");
+ response.Headers.Add("Public", "SETUP, RECORD, PAUSE, FLUSH, TEARDOWN, OPTIONS, GET_PARAMETER, SET_PARAMETER, ANNOUNCE, SETPEERS");
}
if(request.Type == RequestType.ANNOUNCE)
{
+ }
+ if(request.Type == RequestType.SETPEERS)
+ {
+ Console.WriteLine("[DEBUG-RTSP] SETPEERS request received - returning 200 OK");
+ // Just acknowledge the peer list, no action needed
}
if(request.Type == RequestType.FLUSH)
{
diff --git a/AirPlay/Listeners/AudioListener.cs b/AirPlay/Listeners/AudioListener.cs
index 3148534..8aab385 100644
--- a/AirPlay/Listeners/AudioListener.cs
+++ b/AirPlay/Listeners/AudioListener.cs
@@ -408,26 +408,39 @@ private RaopBuffer RaopBufferInit()
return raop_buffer;
}
+ private int _queueCallCount = 0;
+
public int RaopBufferQueue(RaopBuffer raop_buffer, byte[] data, ushort datalen, Session session, IBufferedCipher aesCbcDecrypt)
{
int encryptedlen;
RaopBufferEntry entry;
+ _queueCallCount++;
+
/* Check packet data length is valid */
if (datalen < 12 || datalen > RAOP_PACKET_LENGTH)
{
+ if (_queueCallCount <= 5)
+ Console.WriteLine($"[DEBUG-QUEUE] #{_queueCallCount}: REJECT invalid length={datalen}");
return -1;
}
var seqnum = (ushort)((data[2] << 8) | data[3]);
if (datalen == 16 && data[12] == 0x0 && data[13] == 0x68 && data[14] == 0x34 && data[15] == 0x0)
{
+ if (_queueCallCount <= 5)
+ Console.WriteLine($"[DEBUG-QUEUE] #{_queueCallCount}: no-data marker, seqnum={seqnum}");
return 0;
}
+ if (_queueCallCount <= 10 || _queueCallCount % 500 == 0)
+ Console.WriteLine($"[DEBUG-QUEUE] #{_queueCallCount}: seqnum={seqnum}, datalen={datalen}, payloadSize={datalen - 12}, bufferEmpty={raop_buffer.IsEmpty}, firstSeq={raop_buffer.FirstSeqNum}, lastSeq={raop_buffer.LastSeqNum}");
+
// Ignore, old
if (!raop_buffer.IsEmpty && seqnum < raop_buffer.FirstSeqNum && seqnum != 0)
{
+ if (_queueCallCount <= 10)
+ Console.WriteLine($"[DEBUG-QUEUE] #{_queueCallCount}: SKIP old seqnum={seqnum} < firstSeqNum={raop_buffer.FirstSeqNum}");
return 0;
}
diff --git a/AirPlay/Listeners/Bases/BaseTcpListener.cs b/AirPlay/Listeners/Bases/BaseTcpListener.cs
index b8bf448..eaffd76 100644
--- a/AirPlay/Listeners/Bases/BaseTcpListener.cs
+++ b/AirPlay/Listeners/Bases/BaseTcpListener.cs
@@ -134,22 +134,30 @@ private async Task ReadFormattedAsync(TcpClient client, NetworkStream stream, Ca
// Because of the persistent connection we might receive more than one request at a time
// I'm using a regex to find all request by 'magic numbers' (ex. GET, POST, SETUP, ecc)
- var pattern =
- $"^{RequestConst.GET}[.]*|" +
- $"^{RequestConst.POST}[.]*|" +
- $"^{RequestConst.SETUP}[.]*|" +
- $"^{RequestConst.GET_PARAMETER}[.]*|" +
- $"^{RequestConst.RECORD}[.]*|" +
- $"^{RequestConst.SET_PARAMETER}[.]*|" +
- $"^{RequestConst.ANNOUNCE}[.]*|" +
- $"^{RequestConst.FLUSH}[.]*|" +
- $"^{RequestConst.OPTIONS}[.]*|" +
- $"^{RequestConst.PAUSE}[.]*|" +
+ var pattern =
+ $"^{RequestConst.GET}[.]*|" +
+ $"^{RequestConst.POST}[.]*|" +
+ $"^{RequestConst.SETUP}[.]*|" +
+ $"^{RequestConst.GET_PARAMETER}[.]*|" +
+ $"^{RequestConst.RECORD}[.]*|" +
+ $"^{RequestConst.SET_PARAMETER}[.]*|" +
+ $"^{RequestConst.ANNOUNCE}[.]*|" +
+ $"^{RequestConst.FLUSH}[.]*|" +
+ $"^{RequestConst.OPTIONS}[.]*|" +
+ $"^{RequestConst.PAUSE}[.]*|" +
+ $"^{RequestConst.SETPEERS}[.]*|" +
$"^{RequestConst.TEARDOWN}[.]*";
var r = new Regex(pattern, RegexOptions.Multiline);
var m = r.Matches(raw);
+ if (m.Count == 0 && raw.Length > 0)
+ {
+ // Log first 200 chars of unrecognized data for debugging
+ var preview = raw.Length > 200 ? raw.Substring(0, 200) : raw;
+ Console.WriteLine($"[DEBUG-RTSP] Unrecognized data (no regex match), len={raw.Length}, preview: {preview}");
+ }
+
// Split requests and create models
var requests = new List();
for (int i = 0; i < m.Count; i++)
@@ -172,9 +180,13 @@ private async Task ReadFormattedAsync(TcpClient client, NetworkStream stream, Ca
{
var response = request.GetBaseResponse();
+ var cseq = request.Headers.ContainsKey("CSeq") ? request.Headers["CSeq"] : "?";
+ Console.WriteLine($"[DEBUG-RTSP] >>> {request.Type} CSeq={cseq}");
+
await OnDataReceivedAsync(request, response, cancellationToken).ConfigureAwait(false);
await SendResponseAsync(stream, response);
- }
+
+ Console.WriteLine($"[DEBUG-RTSP] <<< {response.StatusCode}"); }
}
// If we have read some bytes, leave connection open and wait for next message
diff --git a/AirPlay/Models/Enums/RequestConst.cs b/AirPlay/Models/Enums/RequestConst.cs
index a2d98d5..ac40f68 100644
--- a/AirPlay/Models/Enums/RequestConst.cs
+++ b/AirPlay/Models/Enums/RequestConst.cs
@@ -13,5 +13,6 @@ public class RequestConst
public const string TEARDOWN = "54454152444F574E";
public const string OPTIONS = "4F5054494F4E53";
public const string PAUSE = "5041555345";
+ public const string SETPEERS = "534554504545525320";
}
}
diff --git a/AirPlay/Models/Enums/RequestType.cs b/AirPlay/Models/Enums/RequestType.cs
index b6865e1..788447e 100644
--- a/AirPlay/Models/Enums/RequestType.cs
+++ b/AirPlay/Models/Enums/RequestType.cs
@@ -12,6 +12,7 @@ public enum RequestType : ushort
FLUSH = 7,
TEARDOWN = 8,
OPTIONS = 9,
- PAUSE = 10
+ PAUSE = 10,
+ SETPEERS = 11
}
}
\ No newline at end of file
diff --git a/AirPlay/Models/TcpListeners/Request.cs b/AirPlay/Models/TcpListeners/Request.cs
index c53fc28..aab7006 100644
--- a/AirPlay/Models/TcpListeners/Request.cs
+++ b/AirPlay/Models/TcpListeners/Request.cs
@@ -162,6 +162,10 @@ private void Initialize ()
{
return RequestType.PAUSE;
}
+ if (hex.StartsWith(RequestConst.SETPEERS, StringComparison.OrdinalIgnoreCase))
+ {
+ return RequestType.SETPEERS;
+ }
return null;
}
From 11bb564f0c53823d1de3546a378b94c88f023970 Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 10:45:00 +0000
Subject: [PATCH 09/14] Switch AAC-ELD decoder from ADTS to LOAS framing with
correct AudioSpecificConfig
ADTS headers can only encode AAC-Main/LC/SSR profiles (2-bit field) and
cannot represent AAC-ELD (AOT 39). FFmpeg was parsing AAC-ELD frames as
AAC-LC, causing "skip_data_stream_element: Input buffer exhausted" errors.
LOAS (Low Overhead Audio Stream, ISO 14496-3) carries a full
AudioSpecificConfig that properly signals AAC-ELD. The ASC (f8e85000)
matches UxPlay's reference implementation:
- AOT 39 (ER AAC-ELD)
- 44100 Hz
- 2 channels (stereo)
- frameLengthFlag = 1
FFmpeg demuxer changed from `-f aac` to `-f loas`.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
.../Implementations/FFmpegAacEldDecoder.cs | 224 +++++++++++++++---
1 file changed, 190 insertions(+), 34 deletions(-)
diff --git a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
index 4d5caae..a5f4f0b 100644
--- a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
+++ b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
@@ -1,16 +1,17 @@
/*
- * AAC-ELD Decoder using FFmpeg as a subprocess.
+ * AAC-ELD Decoder using FFmpeg as a subprocess with LOAS/LATM framing.
*
* The fdk-aac NuGet package returns error 0x5 (AAC_DEC_UNSUPPORTED_ER_FORMAT)
* for AAC-ELD because the pre-built binary doesn't include ER format support.
* SharpJaad.AAC also doesn't support AAC-ELD.
*
- * This decoder pipes raw AAC-ELD frames wrapped in ADTS headers through FFmpeg
- * for decoding. While ADTS headers can only encode AAC-LC profile (2-bit field),
- * FFmpeg's AAC decoder handles the actual bitstream content regardless of the
- * ADTS profile indicator. The ADTS header serves as a framing/sync mechanism.
+ * This decoder wraps raw AAC-ELD frames in LOAS (Low Overhead Audio Stream)
+ * format per ISO 14496-3 and pipes them through FFmpeg's LOAS demuxer (-f loas).
+ * Unlike ADTS which only supports AAC-Main/LC/SSR profiles (2-bit field), LOAS
+ * carries a full AudioSpecificConfig that can properly signal AAC-ELD (AOT 39).
*
- * Reference: UxPlay uses GStreamer's avdec_aac (which wraps FFmpeg) for the same purpose.
+ * AudioSpecificConfig (f8e85000): AOT=39 (ER AAC-ELD), 44100Hz, 2ch
+ * Reference: UxPlay uses this same ASC with GStreamer's avdec_aac decoder.
*/
using System;
@@ -35,10 +36,13 @@ public class FFmpegAacEldDecoder : IDecoder, IDisposable
private int _frameLength;
private bool _disposed;
private bool _initialized;
- private readonly byte[] _adtsHeader = new byte[7];
- private int _adtsFreqIdx;
private int _decodeCallCount;
private int _totalFramesDecoded;
+ private bool _firstFrame = true;
+
+ // Pre-built LOAS framing components
+ private byte[] _streamMuxConfigBits; // StreamMuxConfig as bit array
+ private int _streamMuxConfigBitLen;
public AudioFormat Type => AudioFormat.AAC_ELD;
@@ -52,17 +56,19 @@ public int Config(int sampleRate, int channels, int bitDepth, int frameLength)
_sampleRate = sampleRate;
_frameLength = frameLength;
_pcmOutputSize = frameLength * channels * (bitDepth / 8);
- _adtsFreqIdx = GetSampleRateIndex(sampleRate);
+
+ // Build AudioSpecificConfig for AAC-ELD
+ BuildStreamMuxConfig(sampleRate, channels, frameLength);
try
{
// Start FFmpeg process:
- // Input: AAC frames wrapped in ADTS headers via stdin pipe
+ // Input: LOAS-wrapped AAC-ELD frames via stdin pipe
// Output: raw PCM S16LE via stdout pipe
var psi = new ProcessStartInfo
{
FileName = "ffmpeg",
- Arguments = $"-hide_banner -loglevel error -f aac -i pipe:0 -f s16le -acodec pcm_s16le -ar {sampleRate} -ac {channels} pipe:1",
+ Arguments = $"-hide_banner -loglevel error -f loas -i pipe:0 -f s16le -acodec pcm_s16le -ar {sampleRate} -ac {channels} pipe:1",
UseShellExecute = false,
RedirectStandardInput = true,
RedirectStandardOutput = true,
@@ -85,7 +91,6 @@ public int Config(int sampleRate, int channels, int bitDepth, int frameLength)
string line;
while ((line = reader.ReadLine()) != null)
{
- // Log all FFmpeg stderr output for debugging
Console.WriteLine($"[DEBUG-FFMPEG-STDERR] {line}");
}
}
@@ -125,20 +130,16 @@ public int DecodeFrame(byte[] input, ref byte[] output, int length)
{
_decodeCallCount++;
- // Wrap the raw AAC frame in an ADTS header for FFmpeg to parse.
- // ADTS profile is set to AAC-LC (profile=2) since ADTS only has a 2-bit
- // profile field and cannot represent AAC-ELD. FFmpeg's decoder handles
- // the actual bitstream content regardless of the ADTS profile indicator.
- int frameLen = input.Length + 7; // ADTS header is 7 bytes
- BuildAdtsHeader(_adtsHeader, frameLen, 2, _adtsFreqIdx, _channels);
+ // Build LOAS frame containing the raw AAC-ELD data
+ byte[] loasFrame = BuildLoasFrame(input, _firstFrame);
+ _firstFrame = false;
- // Write ADTS header + raw AAC data to FFmpeg stdin
- _ffmpegInput.Write(_adtsHeader, 0, 7);
- _ffmpegInput.Write(input, 0, input.Length);
+ // Write LOAS frame to FFmpeg stdin
+ _ffmpegInput.Write(loasFrame, 0, loasFrame.Length);
_ffmpegInput.Flush();
if (_decodeCallCount <= 3)
- Console.WriteLine($"[DEBUG-FFMPEG] Wrote frame #{_decodeCallCount}: inputLen={input.Length}, adtsFrameLen={frameLen}, expecting {_pcmOutputSize} bytes PCM output");
+ Console.WriteLine($"[DEBUG-FFMPEG] Wrote LOAS frame #{_decodeCallCount}: inputLen={input.Length}, loasLen={loasFrame.Length}, expecting {_pcmOutputSize} bytes PCM output");
// Read decoded PCM from FFmpeg stdout
int bytesToRead = _pcmOutputSize;
@@ -181,18 +182,133 @@ public int DecodeFrame(byte[] input, ref byte[] output, int length)
}
///
- /// Build a 7-byte ADTS header for wrapping raw AAC frames.
- /// Used as a framing/sync mechanism for FFmpeg's AAC demuxer.
+ /// Build a LOAS (Low Overhead Audio Stream) frame per ISO 14496-3.
+ /// First frame includes full StreamMuxConfig with AudioSpecificConfig.
+ /// Subsequent frames use useSameStreamMux=1 to reuse the config.
+ ///
+ private byte[] BuildLoasFrame(byte[] aacPayload, bool includeConfig)
+ {
+ // Build AudioMuxElement as bit stream
+ var muxBits = new BitList();
+
+ if (includeConfig)
+ {
+ // useSameStreamMux = 0 (send new StreamMuxConfig)
+ muxBits.Add(false);
+
+ // Write pre-built StreamMuxConfig bits
+ for (int i = 0; i < _streamMuxConfigBitLen; i++)
+ {
+ muxBits.Add(_streamMuxConfigBits[i] != 0);
+ }
+ }
+ else
+ {
+ // useSameStreamMux = 1 (reuse previous config)
+ muxBits.Add(true);
+ }
+
+ // PayloadLengthInfo (for frameLengthType == 0):
+ // Encode length as sequence of 255-byte chunks + remainder
+ int remaining = aacPayload.Length;
+ while (remaining >= 255)
+ {
+ WriteBits(muxBits, 255, 8);
+ remaining -= 255;
+ }
+ WriteBits(muxBits, remaining, 8);
+
+ // PayloadMux: raw AAC frame data
+ for (int i = 0; i < aacPayload.Length; i++)
+ {
+ WriteBits(muxBits, aacPayload[i], 8);
+ }
+
+ // Convert AudioMuxElement to bytes (pad to byte boundary)
+ byte[] muxBytes = BitsToBytes(muxBits);
+
+ // Build complete LOAS frame:
+ // syncword (11 bits) = 0x2B7
+ // audioMuxLengthBytes (13 bits) = length of AudioMuxElement in bytes
+ var frameBits = new BitList();
+ WriteBits(frameBits, 0x2B7, 11);
+ WriteBits(frameBits, muxBytes.Length, 13);
+
+ // Append AudioMuxElement bytes
+ for (int i = 0; i < muxBytes.Length; i++)
+ {
+ WriteBits(frameBits, muxBytes[i], 8);
+ }
+
+ return BitsToBytes(frameBits);
+ }
+
+ ///
+ /// Build the StreamMuxConfig containing AudioSpecificConfig for AAC-ELD.
+ /// Pre-computed once during Config() for reuse in every LOAS frame.
///
- private static void BuildAdtsHeader(byte[] header, int packetLen, int profile, int freqIdx, int chanCfg)
+ private void BuildStreamMuxConfig(int sampleRate, int channels, int frameLength)
{
- header[0] = 0xFF;
- header[1] = 0xF1; // MPEG-4, Layer 0, no CRC
- header[2] = (byte)(((profile - 1) << 6) | (freqIdx << 2) | (chanCfg >> 2));
- header[3] = (byte)(((chanCfg & 3) << 6) | (packetLen >> 11));
- header[4] = (byte)((packetLen >> 3) & 0xFF);
- header[5] = (byte)(((packetLen & 7) << 5) | 0x1F);
- header[6] = 0xFC;
+ var bits = new BitList();
+
+ // audioMuxVersion = 0
+ bits.Add(false);
+ // allStreamsSameTimeFraming = 1
+ bits.Add(true);
+ // numSubFrames = 0 (1 subframe)
+ WriteBits(bits, 0, 6);
+ // numProgram = 0 (1 program)
+ WriteBits(bits, 0, 4);
+ // numLayer = 0 (1 layer)
+ WriteBits(bits, 0, 3);
+
+ // AudioSpecificConfig for AAC-ELD (ISO 14496-3)
+ // audioObjectType = 39 (ER AAC-ELD): escape(11111) + 7(000111)
+ WriteBits(bits, 0x1F, 5); // escape
+ WriteBits(bits, 39 - 32, 6); // 7
+
+ // samplingFrequencyIndex
+ int freqIdx = GetSampleRateIndex(sampleRate);
+ WriteBits(bits, freqIdx, 4);
+
+ // channelConfiguration
+ WriteBits(bits, channels, 4);
+
+ // ELD specific config (ISO 14496-3 section 4.4.2.5)
+ // frameLengthFlag: 0=480, 1=512
+ // UxPlay's working ASC (f8e85000) uses frameLengthFlag=1 (512 samples)
+ // even though iOS SETUP sends spf=480. FFmpeg's decoder handles the
+ // actual frame content regardless of this flag.
+ bits.Add(true); // frameLengthFlag = 1
+
+ // aacSectionDataResilienceFlag
+ bits.Add(false);
+ // aacScalefactorDataResilienceFlag
+ bits.Add(false);
+ // aacSpectralDataResilienceFlag
+ bits.Add(false);
+
+ // ldSbrPresentFlag
+ bits.Add(false);
+
+ // End of AudioSpecificConfig
+
+ // frameLengthType = 0 (variable length)
+ WriteBits(bits, 0, 3);
+ // latmBufferFullness = 0xFF (VBR)
+ WriteBits(bits, 0xFF, 8);
+ // otherDataPresent = 0
+ bits.Add(false);
+ // crcCheckPresent = 0
+ bits.Add(false);
+
+ // Store as bit array for reuse
+ _streamMuxConfigBitLen = bits.Count;
+ _streamMuxConfigBits = new byte[_streamMuxConfigBitLen];
+ for (int i = 0; i < _streamMuxConfigBitLen; i++)
+ {
+ _streamMuxConfigBits[i] = bits[i] ? (byte)1 : (byte)0;
+ }
}
private static int GetSampleRateIndex(int sampleRate)
@@ -206,6 +322,48 @@ private static int GetSampleRateIndex(int sampleRate)
};
}
+ #region Bit manipulation helpers
+
+ private static void WriteBits(BitList bits, int value, int numBits)
+ {
+ for (int i = numBits - 1; i >= 0; i--)
+ {
+ bits.Add(((value >> i) & 1) == 1);
+ }
+ }
+
+ private static byte[] BitsToBytes(BitList bits)
+ {
+ // Pad to byte boundary
+ int padded = ((bits.Count + 7) / 8) * 8;
+ int byteCount = padded / 8;
+ var result = new byte[byteCount];
+
+ for (int i = 0; i < byteCount; i++)
+ {
+ int val = 0;
+ for (int j = 0; j < 8; j++)
+ {
+ int bitIdx = i * 8 + j;
+ val <<= 1;
+ if (bitIdx < bits.Count && bits[bitIdx])
+ val |= 1;
+ }
+ result[i] = (byte)val;
+ }
+
+ return result;
+ }
+
+ ///
+ /// Simple growable bit list using System.Collections.Generic.List<bool>.
+ ///
+ private class BitList : System.Collections.Generic.List
+ {
+ }
+
+ #endregion
+
public void Dispose()
{
if (!_disposed)
@@ -213,12 +371,10 @@ public void Dispose()
_disposed = true;
try
{
- // Close stdin first for graceful FFmpeg shutdown
_ffmpegInput?.Close();
if (_ffmpegProcess != null && !_ffmpegProcess.HasExited)
{
- // Wait for graceful exit, then force kill if needed
if (!_ffmpegProcess.WaitForExit(PROCESS_EXIT_TIMEOUT_MS))
{
_ffmpegProcess.Kill();
From b2517b5ad6dae3b2d370b7908a5119567be5e535 Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 11:05:51 +0000
Subject: [PATCH 10/14] Add decrypted payload debug logging, fix
frameLengthFlag to match iOS spf
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Add [DEBUG-DECRYPT] logging to show first bytes of decrypted AAC-ELD
payload. Valid frames should start with 0x8c/0x8d/0x8e/0x80/0x81/0x82/0x20
(per UxPlay reference). This will help identify if AES-CBC decryption
is producing valid AAC-ELD data.
Also fix AudioSpecificConfig frameLengthFlag to match the actual frame
length from iOS SETUP (spf=480 → frameLengthFlag=0). Previously hardcoded
to 1 (512 samples) which mismatches the 480-sample frames iOS sends.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
.../Decoders/Implementations/FFmpegAacEldDecoder.cs | 8 +++-----
AirPlay/Listeners/AudioListener.cs | 12 +++++++++++-
2 files changed, 14 insertions(+), 6 deletions(-)
diff --git a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
index a5f4f0b..6427ffc 100644
--- a/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
+++ b/AirPlay/Decoders/Implementations/FFmpegAacEldDecoder.cs
@@ -275,11 +275,9 @@ private void BuildStreamMuxConfig(int sampleRate, int channels, int frameLength)
WriteBits(bits, channels, 4);
// ELD specific config (ISO 14496-3 section 4.4.2.5)
- // frameLengthFlag: 0=480, 1=512
- // UxPlay's working ASC (f8e85000) uses frameLengthFlag=1 (512 samples)
- // even though iOS SETUP sends spf=480. FFmpeg's decoder handles the
- // actual frame content regardless of this flag.
- bits.Add(true); // frameLengthFlag = 1
+ // frameLengthFlag: 0=480 samples, 1=512 samples
+ // Must match the actual frame length from iOS SETUP (spf field)
+ bits.Add(frameLength == 512); // frameLengthFlag
// aacSectionDataResilienceFlag
bits.Add(false);
diff --git a/AirPlay/Listeners/AudioListener.cs b/AirPlay/Listeners/AudioListener.cs
index 8aab385..36956a5 100644
--- a/AirPlay/Listeners/AudioListener.cs
+++ b/AirPlay/Listeners/AudioListener.cs
@@ -484,6 +484,15 @@ public int RaopBufferQueue(RaopBuffer raop_buffer, byte[] data, ushort datalen,
File.WriteAllBytes($"{fPath}raw_{seqnum}", raw);
#endif
/* RAW -> PCM */
+ if (_queueCallCount <= 5 || _queueCallCount % 500 == 0)
+ {
+ // Log first bytes of decrypted payload for verification
+ // Valid AAC-ELD frames start with: 0x8c, 0x8d, 0x8e, 0x80, 0x81, 0x82, 0x20
+ var hexDump = raw.Length >= 16
+ ? BitConverter.ToString(raw, 0, Math.Min(16, raw.Length))
+ : BitConverter.ToString(raw);
+ Console.WriteLine($"[DEBUG-DECRYPT] #{_queueCallCount}: raw[0]=0x{raw[0]:X2}, len={raw.Length}, first16={hexDump}");
+ }
var length = _decoder.GetOutputStreamLength();
var output = new byte[length];
@@ -491,7 +500,8 @@ public int RaopBufferQueue(RaopBuffer raop_buffer, byte[] data, ushort datalen,
if (res != 0)
{
output = new byte[length];
- Console.WriteLine($"Decoding error. Decoder: {_decoder.Type} Code: {res}");
+ if (_queueCallCount <= 5 || _queueCallCount % 500 == 0)
+ Console.WriteLine($"Decoding error. Decoder: {_decoder.Type} Code: {res}");
}
#if DUMP
From 948daec392b07ba0dff9599b9e54d14297d862df Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 11:37:32 +0000
Subject: [PATCH 11/14] Add native FDK-AAC P/Invoke decoder for AAC-ELD with
comprehensive debug logging
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
Replace FFmpeg subprocess + fragile ADTS/LOAS wrapping with direct native
FDK-AAC library calls via P/Invoke, matching the approach used by both
itskenny0/airplayreceiver and UxPlay reference implementations.
Key changes:
- NativeFdkAacEldDecoder: P/Invoke to libfdk-aac-2.dll with TT_MP4_RAW
transport. Raw decrypted AAC-ELD frames fed directly, no container framing.
- AudioSpecificConfig: f8e85000 (matches UxPlay/itskenny0 exactly)
- Decoder fallback chain: native FDK-AAC → FFmpeg subprocess → SharpJaad
- Debug logging: [DEBUG-FDK] for decoder, [DEBUG-DECRYPT] for payload hex,
[DEBUG-DECODE] for decode results with silence detection
- SETUP debug: shows unsigned port values for clarity
User needs libfdk-aac-2.dll in the application directory.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
.../Implementations/NativeFdkAacEldDecoder.cs | 284 ++++++++++++++++++
AirPlay/Listeners/AirTunesListener.cs | 2 +
AirPlay/Listeners/AudioListener.cs | 74 ++++-
3 files changed, 346 insertions(+), 14 deletions(-)
create mode 100644 AirPlay/Decoders/Implementations/NativeFdkAacEldDecoder.cs
diff --git a/AirPlay/Decoders/Implementations/NativeFdkAacEldDecoder.cs b/AirPlay/Decoders/Implementations/NativeFdkAacEldDecoder.cs
new file mode 100644
index 0000000..9cee74b
--- /dev/null
+++ b/AirPlay/Decoders/Implementations/NativeFdkAacEldDecoder.cs
@@ -0,0 +1,284 @@
+/*
+ * AAC-ELD Decoder using the native FDK-AAC library via P/Invoke.
+ *
+ * This decoder feeds raw (decrypted) AAC-ELD frames directly to FDK-AAC
+ * using TT_MP4_RAW transport type with AudioSpecificConfig for AAC-ELD.
+ * No ADTS/LOAS framing is needed - the raw access units are passed directly.
+ *
+ * Requires libfdk-aac-2.dll (Windows) or libfdk-aac.so.2 (Linux) in the
+ * application directory or system PATH.
+ *
+ * AudioSpecificConfig: f8e85000 (AOT=39, 44100Hz, stereo, bitDepth=16)
+ * This matches both UxPlay and itskenny0/airplayreceiver reference implementations.
+ *
+ * Reference: itskenny0/airplayreceiver AACDecoder.cs
+ */
+
+using System;
+using System.Runtime.InteropServices;
+using AirPlay.Models.Enums;
+
+namespace AirPlay.Decoders.Implementations
+{
+ public unsafe class NativeFdkAacEldDecoder : IDecoder, IDisposable
+ {
+ private const int TT_MP4_RAW = 0;
+
+ private IntPtr _decoder;
+ private int _pcmPktSize;
+ private bool _disposed;
+ private int _decodeCallCount;
+ private int _fillErrorCount;
+ private int _decodeErrorCount;
+ private int _successCount;
+
+ public AudioFormat Type => AudioFormat.AAC_ELD;
+
+ public NativeFdkAacEldDecoder()
+ {
+ }
+
+ public int Config(int sampleRate, int channels, int bitDepth, int frameLength)
+ {
+ _pcmPktSize = frameLength * channels * (bitDepth / 8);
+
+ Console.WriteLine($"[DEBUG-FDK] Config called: sampleRate={sampleRate}, channels={channels}, bitDepth={bitDepth}, frameLength={frameLength}, pcmOutputSize={_pcmPktSize}");
+
+ // Open FDK-AAC decoder with TT_MP4_RAW transport (raw access units)
+ _decoder = aacDecoder_Open(TT_MP4_RAW, 1);
+ if (_decoder == IntPtr.Zero)
+ {
+ Console.WriteLine("[DEBUG-FDK] aacDecoder_Open returned null - library loaded but Open failed");
+ return -1;
+ }
+ Console.WriteLine($"[DEBUG-FDK] aacDecoder_Open succeeded: handle=0x{_decoder:X}");
+
+ // Build AudioSpecificConfig for AAC-ELD
+ // Uses exact same format as itskenny0/airplayreceiver and UxPlay: f8e85000
+ var asc = BuildAudioSpecificConfig(39, sampleRate, channels, frameLength);
+
+ // Configure decoder with the ASC
+ int ret = ConfigRaw(asc);
+ if (ret != 0)
+ {
+ Console.WriteLine($"[DEBUG-FDK] aacDecoder_ConfigRaw FAILED: error=0x{ret:X} ({(AACDecoderError)ret})");
+ aacDecoder_Close(_decoder);
+ _decoder = IntPtr.Zero;
+ return ret;
+ }
+
+ Console.WriteLine($"[DEBUG-FDK] Decoder configured successfully: {sampleRate}Hz, {channels}ch, {bitDepth}bit, frameLength={frameLength}, pcmSize={_pcmPktSize}");
+ return 0;
+ }
+
+ public int GetOutputStreamLength()
+ {
+ return _pcmPktSize;
+ }
+
+ public int DecodeFrame(byte[] input, ref byte[] output, int pcm_pkt_size)
+ {
+ if (_decoder == IntPtr.Zero)
+ {
+ _decodeCallCount++;
+ if (_decodeCallCount <= 3)
+ Console.WriteLine("[DEBUG-FDK] DecodeFrame called but decoder is null");
+ return -1;
+ }
+
+ _decodeCallCount++;
+
+ // Log first few frames and periodically
+ bool shouldLog = _decodeCallCount <= 5 || _decodeCallCount % 200 == 0;
+
+ if (shouldLog)
+ {
+ var hexDump = input.Length >= 16
+ ? BitConverter.ToString(input, 0, Math.Min(16, input.Length))
+ : BitConverter.ToString(input);
+ Console.WriteLine($"[DEBUG-FDK] DecodeFrame #{_decodeCallCount}: inputLen={input.Length}, first16={hexDump}");
+ }
+
+ int ret;
+
+ // Fill the decoder's internal buffer with the raw AAC frame
+ ret = Fill(input);
+ if (ret != 0)
+ {
+ _fillErrorCount++;
+ if (_fillErrorCount <= 5 || _fillErrorCount % 200 == 0)
+ Console.WriteLine($"[DEBUG-FDK] aacDecoder_Fill error #{_fillErrorCount}: 0x{ret:X} ({(AACDecoderError)ret}), inputLen={input.Length}");
+ return ret;
+ }
+
+ // Decode one frame
+ ret = InternalDecodeFrame(ref output, pcm_pkt_size);
+ if (ret != 0)
+ {
+ _decodeErrorCount++;
+ if (_decodeErrorCount <= 5 || _decodeErrorCount % 200 == 0)
+ Console.WriteLine($"[DEBUG-FDK] aacDecoder_DecodeFrame error #{_decodeErrorCount}: 0x{ret:X} ({(AACDecoderError)ret}), pcmSize={pcm_pkt_size}");
+ return ret;
+ }
+
+ _successCount++;
+ if (_successCount <= 5 || _successCount % 200 == 0)
+ {
+ // Check if output is silence (all zeros)
+ bool isSilence = true;
+ for (int i = 0; i < Math.Min(output.Length, 64); i++)
+ {
+ if (output[i] != 0) { isSilence = false; break; }
+ }
+ Console.WriteLine($"[DEBUG-FDK] Decode SUCCESS #{_successCount}: outputLen={pcm_pkt_size}, isSilence={isSilence}, totalErrors(fill={_fillErrorCount},decode={_decodeErrorCount})");
+ }
+
+ return 0;
+ }
+
+ private int Fill(byte[] pBuffer)
+ {
+ uint bufferSize = (uint)pBuffer.Length;
+ uint bytesValid = (uint)pBuffer.Length;
+
+ IntPtr ptr = Marshal.AllocHGlobal(pBuffer.Length);
+ try
+ {
+ Marshal.Copy(pBuffer, 0, ptr, pBuffer.Length);
+
+ IntPtr* pBufferPtr = stackalloc IntPtr[1];
+ pBufferPtr[0] = ptr;
+
+ int ret = (int)aacDecoder_Fill(_decoder, pBufferPtr, &bufferSize, &bytesValid);
+ if (ret == 0 && _decodeCallCount <= 5)
+ Console.WriteLine($"[DEBUG-FDK] Fill OK: bufferSize={pBuffer.Length}, bytesValid after={bytesValid}");
+ return ret;
+ }
+ finally
+ {
+ Marshal.FreeHGlobal(ptr);
+ }
+ }
+
+ private int InternalDecodeFrame(ref byte[] output, int pcm_pkt_size)
+ {
+ IntPtr ptr = Marshal.AllocHGlobal(pcm_pkt_size);
+ try
+ {
+ int ret = (int)aacDecoder_DecodeFrame(_decoder, ptr, pcm_pkt_size, 0);
+ if (ret == 0)
+ {
+ Marshal.Copy(ptr, output, 0, pcm_pkt_size);
+ }
+ return ret;
+ }
+ finally
+ {
+ Marshal.FreeHGlobal(ptr);
+ }
+ }
+
+ private int ConfigRaw(byte[] config)
+ {
+ uint length = (uint)config.Length;
+
+ IntPtr ptr = Marshal.AllocHGlobal(config.Length);
+ try
+ {
+ Marshal.Copy(config, 0, ptr, config.Length);
+
+ IntPtr* confPtr = stackalloc IntPtr[1];
+ confPtr[0] = ptr;
+
+ return (int)aacDecoder_ConfigRaw(_decoder, confPtr, &length);
+ }
+ finally
+ {
+ Marshal.FreeHGlobal(ptr);
+ }
+ }
+
+ ///
+ /// Build AudioSpecificConfig for AAC-ELD per ISO 14496-3.
+ /// Uses the exact same format as itskenny0/airplayreceiver and UxPlay:
+ /// AOT(5+6) + FreqIdx(4) + Channels(4) + BitDepth(5) + padding(8)
+ /// For AAC-ELD 44100Hz stereo 16bit: f8 e8 50 00
+ ///
+ private static byte[] BuildAudioSpecificConfig(int audioObjectType, int sampleRate, int channels, int frameLength)
+ {
+ int frequencyIndex = GetSampleRateIndex(sampleRate);
+ int bitDepth = 16;
+
+ string bin;
+ if (audioObjectType >= 31)
+ {
+ // Extended AOT encoding: 5-bit escape (11111) + 6-bit AOT-32
+ bin = Convert.ToString(31, 2).PadLeft(5, '0');
+ bin += Convert.ToString(audioObjectType - 32, 2).PadLeft(6, '0');
+ }
+ else
+ {
+ bin = Convert.ToString(audioObjectType, 2).PadLeft(5, '0');
+ }
+
+ bin += Convert.ToString(frequencyIndex, 2).PadLeft(4, '0');
+ bin += Convert.ToString(channels, 2).PadLeft(4, '0');
+ bin += Convert.ToString(bitDepth, 2).PadLeft(5, '0');
+ bin += "00000000"; // padding byte
+
+ int nBytes = bin.Length / 8;
+ byte[] bytes = new byte[nBytes];
+ for (int i = 0; i < nBytes; i++)
+ {
+ bytes[i] = Convert.ToByte(bin.Substring(8 * i, 8), 2);
+ }
+
+ Console.WriteLine($"[DEBUG-FDK] AudioSpecificConfig: {BitConverter.ToString(bytes)} (AOT={audioObjectType}, freq={sampleRate}[idx={frequencyIndex}], ch={channels}, bitDepth={bitDepth}, frameLen={frameLength})");
+ return bytes;
+ }
+
+ private static int GetSampleRateIndex(int sampleRate)
+ {
+ return sampleRate switch
+ {
+ 96000 => 0, 88200 => 1, 64000 => 2, 48000 => 3,
+ 44100 => 4, 32000 => 5, 24000 => 6, 22050 => 7,
+ 16000 => 8, 12000 => 9, 11025 => 10, 8000 => 11,
+ 7350 => 12, _ => 4,
+ };
+ }
+
+ public void Dispose()
+ {
+ if (!_disposed && _decoder != IntPtr.Zero)
+ {
+ _disposed = true;
+ Console.WriteLine($"[DEBUG-FDK] Disposing decoder: totalCalls={_decodeCallCount}, successes={_successCount}, fillErrors={_fillErrorCount}, decodeErrors={_decodeErrorCount}");
+ aacDecoder_Close(_decoder);
+ _decoder = IntPtr.Zero;
+ }
+ }
+
+ // ---- Native P/Invoke declarations ----
+ // FDK-AAC library names vary by platform:
+ // Windows: libfdk-aac-2.dll
+ // Linux: libfdk-aac.so.2
+
+ private const string FDK_AAC_LIB = "libfdk-aac-2";
+
+ [DllImport(FDK_AAC_LIB, CallingConvention = CallingConvention.Cdecl)]
+ private static extern IntPtr aacDecoder_Open(int transportFmt, uint nrOfLayers);
+
+ [DllImport(FDK_AAC_LIB, CallingConvention = CallingConvention.Cdecl)]
+ private static extern int aacDecoder_ConfigRaw(IntPtr decoder, IntPtr* conf, uint* length);
+
+ [DllImport(FDK_AAC_LIB, CallingConvention = CallingConvention.Cdecl)]
+ private static extern int aacDecoder_Fill(IntPtr decoder, IntPtr* pBuffer, uint* bufferSize, uint* pBytesValid);
+
+ [DllImport(FDK_AAC_LIB, CallingConvention = CallingConvention.Cdecl)]
+ private static extern int aacDecoder_DecodeFrame(IntPtr decoder, IntPtr output, int pcm_pkt_size, uint flags);
+
+ [DllImport(FDK_AAC_LIB, CallingConvention = CallingConvention.Cdecl)]
+ private static extern void aacDecoder_Close(IntPtr decoder);
+ }
+}
diff --git a/AirPlay/Listeners/AirTunesListener.cs b/AirPlay/Listeners/AirTunesListener.cs
index c7c36d3..b3b0418 100644
--- a/AirPlay/Listeners/AirTunesListener.cs
+++ b/AirPlay/Listeners/AirTunesListener.cs
@@ -296,6 +296,8 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
{
if (kv.Value is byte[] ba)
Console.WriteLine($"[DEBUG-SETUP] {kv.Key} = byte[{ba.Length}]");
+ else if (kv.Value is short sv)
+ Console.WriteLine($"[DEBUG-SETUP] {kv.Key} = {sv} (unsigned={(ushort)sv})");
else
Console.WriteLine($"[DEBUG-SETUP] {kv.Key} = {kv.Value}");
}
diff --git a/AirPlay/Listeners/AudioListener.cs b/AirPlay/Listeners/AudioListener.cs
index 36956a5..3a07991 100644
--- a/AirPlay/Listeners/AudioListener.cs
+++ b/AirPlay/Listeners/AudioListener.cs
@@ -484,7 +484,7 @@ public int RaopBufferQueue(RaopBuffer raop_buffer, byte[] data, ushort datalen,
File.WriteAllBytes($"{fPath}raw_{seqnum}", raw);
#endif
/* RAW -> PCM */
- if (_queueCallCount <= 5 || _queueCallCount % 500 == 0)
+ if (_queueCallCount <= 10 || _queueCallCount % 200 == 0)
{
// Log first bytes of decrypted payload for verification
// Valid AAC-ELD frames start with: 0x8c, 0x8d, 0x8e, 0x80, 0x81, 0x82, 0x20
@@ -500,8 +500,21 @@ public int RaopBufferQueue(RaopBuffer raop_buffer, byte[] data, ushort datalen,
if (res != 0)
{
output = new byte[length];
- if (_queueCallCount <= 5 || _queueCallCount % 500 == 0)
- Console.WriteLine($"Decoding error. Decoder: {_decoder.Type} Code: {res}");
+ if (_queueCallCount <= 10 || _queueCallCount % 200 == 0)
+ Console.WriteLine($"[DEBUG-DECODE] #{_queueCallCount}: ERROR decoder={_decoder.Type}, code=0x{res:X} ({res}), inputLen={raw.Length}, outputLen={length}");
+ }
+ else
+ {
+ if (_queueCallCount <= 10 || _queueCallCount % 200 == 0)
+ {
+ // Check if PCM output is silence
+ bool isSilence = true;
+ for (int i = 0; i < Math.Min(output.Length, 64); i++)
+ {
+ if (output[i] != 0) { isSilence = false; break; }
+ }
+ Console.WriteLine($"[DEBUG-DECODE] #{_queueCallCount}: OK decoder={_decoder.Type}, inputLen={raw.Length}, outputLen={length}, silence={isSilence}");
+ }
}
#if DUMP
@@ -703,34 +716,67 @@ private void InitializeDecoder (Session session)
else if(audioFormat == AudioFormat.AAC_ELD)
{
// RTP info: 96 mpeg4-generic/44100/2, 96 mode=AAC-eld; constantDuration=480
- // (AAC-ELD -> PCM) using FFmpeg subprocess decoder with LATM wrapping
+ // (AAC-ELD -> PCM)
+ // Try native FDK-AAC first (direct P/Invoke, no framing needed),
+ // then FFmpeg subprocess with LOAS wrapping as fallback.
var frameLength = spf > 0 ? spf : 480;
var numChannels = 2;
var bitDepth = 16;
var sampleRate = 44100;
+ // Try 1: Native FDK-AAC library (same approach as itskenny0/airplayreceiver)
try
{
- var aacEldDecoder = new Decoders.Implementations.FFmpegAacEldDecoder();
- var ret = aacEldDecoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ var nativeDecoder = new Decoders.Implementations.NativeFdkAacEldDecoder();
+ var ret = nativeDecoder.Config(sampleRate, numChannels, bitDepth, frameLength);
if (ret == 0)
{
- _decoder = aacEldDecoder;
+ _decoder = nativeDecoder;
+ Console.WriteLine("[DEBUG] Using native FDK-AAC decoder for AAC-ELD");
}
else
{
- Console.WriteLine($"FFmpeg AAC-ELD decoder config failed (error {ret}), falling back to SharpJaad AAC-LC");
- aacEldDecoder.Dispose();
- _decoder = new AACDecoder(TransportType.TT_MP4_RAW, AudioObjectType.AOT_AAC_LC, 1);
- _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ Console.WriteLine($"[DEBUG] Native FDK-AAC config error: 0x{ret:X}, trying FFmpeg...");
+ nativeDecoder.Dispose();
}
}
+ catch (DllNotFoundException ex)
+ {
+ Console.WriteLine($"[DEBUG] Native FDK-AAC library not found: {ex.Message}");
+ Console.WriteLine("[DEBUG] To use native decoder, place libfdk-aac-2.dll (Windows) or libfdk-aac.so.2 (Linux) in app directory");
+ }
catch (Exception ex)
{
- Console.WriteLine($"FFmpeg AAC-ELD decoder unavailable ({ex.Message}), falling back to SharpJaad AAC-LC");
- _decoder = new AACDecoder(TransportType.TT_MP4_RAW, AudioObjectType.AOT_AAC_LC, 1);
- _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ Console.WriteLine($"[DEBUG] Native FDK-AAC failed: {ex.GetType().Name}: {ex.Message}");
+ }
+
+ // Try 2: FFmpeg subprocess with LOAS wrapping
+ if (_decoder == null)
+ {
+ try
+ {
+ var ffmpegDecoder = new Decoders.Implementations.FFmpegAacEldDecoder();
+ var ret = ffmpegDecoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ if (ret == 0)
+ {
+ _decoder = ffmpegDecoder;
+ Console.WriteLine("[DEBUG] Using FFmpeg subprocess decoder for AAC-ELD");
+ }
+ else
+ {
+ Console.WriteLine($"[DEBUG] FFmpeg decoder config failed (error {ret}), falling back to SharpJaad AAC-LC");
+ ffmpegDecoder.Dispose();
+ _decoder = new AACDecoder(TransportType.TT_MP4_RAW, AudioObjectType.AOT_AAC_LC, 1);
+ _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ }
+ }
+ catch (Exception ex)
+ {
+ Console.WriteLine($"[DEBUG] FFmpeg decoder unavailable ({ex.Message}), falling back to SharpJaad AAC-LC");
+ _decoder = new AACDecoder(TransportType.TT_MP4_RAW, AudioObjectType.AOT_AAC_LC, 1);
+ _decoder.Config(sampleRate, numChannels, bitDepth, frameLength);
+ }
}
}
else if (audioFormat == AudioFormat.PCM)
From 7d80e5e995cd56620541c900a4c1c9f8b62924f1 Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 21:21:18 +0000
Subject: [PATCH 12/14] Fix audio stuttering: buffer length overflow +
no_resend for mirroring + CI deps
Three fixes for screen mirroring audio stuttering:
1. RaopBufferDequeue: Change buflen from short to int (cast via ushort).
When LastSeqNum - FirstSeqNum > 32767, short overflows negative,
causing dequeue to think buffer is empty and stop processing.
2. Set no_resend=true during mirroring. Real-time audio should not wait
for packet resends - this caused dequeue to get stuck on missing
entries while the queue grew unboundedly, adding seconds of latency.
3. FlushAsync: Add _bufferLock synchronization for thread safety.
CI workflow: Add FFmpeg (ffmpeg.exe + ffplay.exe) and libfdk-aac-2.dll
to build artifacts so users don't need to download them separately.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
.github/workflows/build.yml | 35 +++++++++++++++++++++++++++
AirPlay/Listeners/AirTunesListener.cs | 3 ++-
AirPlay/Listeners/AudioListener.cs | 22 +++++++++++------
3 files changed, 52 insertions(+), 8 deletions(-)
diff --git a/.github/workflows/build.yml b/.github/workflows/build.yml
index b7de20f..5b2fe6e 100644
--- a/.github/workflows/build.yml
+++ b/.github/workflows/build.yml
@@ -31,6 +31,41 @@ jobs:
- name: Publish
run: dotnet publish AirPlay/AirPlay.csproj --configuration Release --runtime win-x64 --self-contained true --output ./publish/win-x64
+ - name: Download FFmpeg
+ shell: pwsh
+ run: |
+ $ffmpegUrl = "https://github.com/BtbN/FFmpeg-Builds/releases/download/latest/ffmpeg-master-latest-win64-gpl.zip"
+ Invoke-WebRequest -Uri $ffmpegUrl -OutFile ffmpeg.zip
+ Expand-Archive -Path ffmpeg.zip -DestinationPath ffmpeg-tmp
+ # Copy ffmpeg.exe and ffplay.exe to publish directory
+ $binDir = Get-ChildItem -Path ffmpeg-tmp -Recurse -Directory -Filter "bin" | Select-Object -First 1
+ Copy-Item "$($binDir.FullName)\ffmpeg.exe" -Destination ./publish/win-x64/
+ Copy-Item "$($binDir.FullName)\ffplay.exe" -Destination ./publish/win-x64/
+
+ - name: Download libfdk-aac
+ shell: pwsh
+ run: |
+ # Download pre-built FDK-AAC for Windows x64
+ $fdkUrl = "https://github.com/nicholastay/libfdk-aac-build/releases/download/v2.0.3/libfdk-aac-v2.0.3-win-x64.zip"
+ Invoke-WebRequest -Uri $fdkUrl -OutFile libfdk-aac.zip
+ Expand-Archive -Path libfdk-aac.zip -DestinationPath libfdk-aac-tmp
+ # Find and copy libfdk-aac-2.dll to publish directory
+ $dllFile = Get-ChildItem -Path libfdk-aac-tmp -Recurse -Filter "libfdk-aac-2.dll" | Select-Object -First 1
+ if ($dllFile) {
+ Copy-Item $dllFile.FullName -Destination ./publish/win-x64/
+ Write-Host "Copied libfdk-aac-2.dll to publish directory"
+ } else {
+ # Try alternative name
+ $dllFile = Get-ChildItem -Path libfdk-aac-tmp -Recurse -Filter "*.dll" | Select-Object -First 1
+ if ($dllFile) {
+ Copy-Item $dllFile.FullName -Destination ./publish/win-x64/libfdk-aac-2.dll
+ Write-Host "Copied $($dllFile.Name) as libfdk-aac-2.dll to publish directory"
+ } else {
+ Write-Host "WARNING: Could not find libfdk-aac DLL in downloaded archive"
+ Get-ChildItem -Path libfdk-aac-tmp -Recurse | ForEach-Object { Write-Host $_.FullName }
+ }
+ }
+
- name: Upload build artifacts
uses: actions/upload-artifact@v4
with:
diff --git a/AirPlay/Listeners/AirTunesListener.cs b/AirPlay/Listeners/AirTunesListener.cs
index b3b0418..d55ae52 100644
--- a/AirPlay/Listeners/AirTunesListener.cs
+++ b/AirPlay/Listeners/AirTunesListener.cs
@@ -473,7 +473,8 @@ public override async Task OnDataReceivedAsync(Request request, Response respons
Console.WriteLine($"[DEBUG-SETUP] Creating new AudioListener: format={session.AudioFormat}, ct={session.AudioCompressionType}, spf={session.AudioSamplesPerFrame}, mirroring={session.MirroringSession}");
// Start 'AudioListener' (handle PCM/AAC/ALAC data received from iOS/macOS
- var control = new AudioListener(_receiver, session.SessionId, 7002, 7003, _dumpConfig);
+ bool isMirroring = session.MirroringSession.HasValue && session.MirroringSession.Value;
+ var control = new AudioListener(_receiver, session.SessionId, 7002, 7003, _dumpConfig, isMirroring);
await control.StartAsync(cancellationToken).ConfigureAwait(false);
Console.WriteLine("[DEBUG-SETUP] AudioListener started successfully");
diff --git a/AirPlay/Listeners/AudioListener.cs b/AirPlay/Listeners/AudioListener.cs
index 3a07991..3f56dd6 100644
--- a/AirPlay/Listeners/AudioListener.cs
+++ b/AirPlay/Listeners/AudioListener.cs
@@ -36,11 +36,14 @@ public class AudioListener : BaseUdpListener
private readonly DumpConfig _dConfig;
- public AudioListener(IRtspReceiver receiver, string sessionId, ushort cport, ushort dport, DumpConfig dConfig) : base(cport, dport)
+ private bool _isMirroring = false;
+
+ public AudioListener(IRtspReceiver receiver, string sessionId, ushort cport, ushort dport, DumpConfig dConfig, bool isMirroring = false) : base(cport, dport)
{
_receiver = receiver ?? throw new ArgumentNullException(nameof(receiver));
_sessionId = sessionId ?? throw new ArgumentNullException(nameof(sessionId));
_dConfig = dConfig ?? throw new ArgumentNullException(nameof(dConfig));
+ _isMirroring = isMirroring;
_raopBuffer = RaopBufferInit();
}
@@ -281,7 +284,8 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
{
InitAesCbcCipher(aesCbcDecrypt, session.DecryptedAesKey, session.EcdhShared, session.AesIv);
- bool no_resend = false;
+ // During screen mirroring, skip resend waiting (real-time audio can't wait)
+ bool no_resend = _isMirroring;
int buf_ret;
byte[] audiobuf;
int audiobuflen = 0;
@@ -327,7 +331,7 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
dPcmDelivered++;
}
- /* Handle possible resend requests */
+ /* Handle possible resend requests (not needed during mirroring) */
if (!no_resend)
{
RaopBufferHandleResends(_raopBuffer, _cSocket, _controlSequenceNumber);
@@ -368,7 +372,11 @@ public override async Task OnRawDSocketAsync(Socket dSocket, CancellationToken c
public Task FlushAsync(int nextSequence)
{
- RaopBufferFlush(_raopBuffer, nextSequence);
+ Console.WriteLine($"[DEBUG-FLUSH] FlushAsync called: nextSequence={nextSequence}, isMirroring={_isMirroring}");
+ lock (_bufferLock)
+ {
+ RaopBufferFlush(_raopBuffer, nextSequence);
+ }
_receiver.OnAudioFlush();
return Task.CompletedTask;
}
@@ -548,11 +556,11 @@ public int RaopBufferQueue(RaopBuffer raop_buffer, byte[] data, ushort datalen,
public byte[] RaopBufferDequeue(RaopBuffer raop_buffer, ref int length, ref uint pts, bool noResend)
{
- short buflen;
+ int buflen;
RaopBufferEntry entry;
- /* Calculate number of entries in the current buffer */
- buflen = (short)(raop_buffer.LastSeqNum - raop_buffer.FirstSeqNum + 1);
+ /* Calculate number of entries in the current buffer (use ushort arithmetic to handle wraparound) */
+ buflen = (ushort)(raop_buffer.LastSeqNum - raop_buffer.FirstSeqNum + 1);
/* Cannot dequeue from empty buffer */
if (raop_buffer.IsEmpty || buflen <= 0)
From 84546e35edfae5bf91f46fe1e93f22949264569d Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 21:34:39 +0000
Subject: [PATCH 13/14] Fix CI workflow, rewrite README.md, add README_zh.md
1. CI workflow: Replace non-existent nicholastay/libfdk-aac-build with
building FDK-AAC from source using MSYS2 (pre-installed on
windows-latest runners). Uses mstorsjo/fdk-aac official source
with CMake.
2. README.md: Complete rewrite reflecting current project state -
FDK-AAC native decoder for mirroring audio, FFmpeg for video,
simplified build instructions, architecture diagram.
3. README_zh.md: Chinese version of the README.
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
.github/workflows/build.yml | 44 ++---
README.md | 317 ++++++++----------------------------
README_zh.md | 98 +++++++++++
3 files changed, 189 insertions(+), 270 deletions(-)
create mode 100644 README_zh.md
diff --git a/.github/workflows/build.yml b/.github/workflows/build.yml
index 5b2fe6e..4861787 100644
--- a/.github/workflows/build.yml
+++ b/.github/workflows/build.yml
@@ -42,29 +42,29 @@ jobs:
Copy-Item "$($binDir.FullName)\ffmpeg.exe" -Destination ./publish/win-x64/
Copy-Item "$($binDir.FullName)\ffplay.exe" -Destination ./publish/win-x64/
- - name: Download libfdk-aac
- shell: pwsh
+ - name: Build libfdk-aac from source
+ shell: bash
run: |
- # Download pre-built FDK-AAC for Windows x64
- $fdkUrl = "https://github.com/nicholastay/libfdk-aac-build/releases/download/v2.0.3/libfdk-aac-v2.0.3-win-x64.zip"
- Invoke-WebRequest -Uri $fdkUrl -OutFile libfdk-aac.zip
- Expand-Archive -Path libfdk-aac.zip -DestinationPath libfdk-aac-tmp
- # Find and copy libfdk-aac-2.dll to publish directory
- $dllFile = Get-ChildItem -Path libfdk-aac-tmp -Recurse -Filter "libfdk-aac-2.dll" | Select-Object -First 1
- if ($dllFile) {
- Copy-Item $dllFile.FullName -Destination ./publish/win-x64/
- Write-Host "Copied libfdk-aac-2.dll to publish directory"
- } else {
- # Try alternative name
- $dllFile = Get-ChildItem -Path libfdk-aac-tmp -Recurse -Filter "*.dll" | Select-Object -First 1
- if ($dllFile) {
- Copy-Item $dllFile.FullName -Destination ./publish/win-x64/libfdk-aac-2.dll
- Write-Host "Copied $($dllFile.Name) as libfdk-aac-2.dll to publish directory"
- } else {
- Write-Host "WARNING: Could not find libfdk-aac DLL in downloaded archive"
- Get-ChildItem -Path libfdk-aac-tmp -Recurse | ForEach-Object { Write-Host $_.FullName }
- }
- }
+ # Build FDK-AAC from official source using MSYS2 (pre-installed on windows-latest)
+ export MSYSTEM=MINGW64
+ export PATH="/c/msys64/mingw64/bin:/c/msys64/usr/bin:$PATH"
+
+ # Install build dependencies
+ pacman -S --noconfirm --needed mingw-w64-x86_64-gcc mingw-w64-x86_64-cmake make autoconf automake libtool
+
+ # Clone and build fdk-aac
+ git clone --depth 1 https://github.com/mstorsjo/fdk-aac.git fdk-aac-src
+ cd fdk-aac-src
+
+ # Use CMake for a simpler build
+ mkdir build && cd build
+ cmake -G "MinGW Makefiles" -DCMAKE_BUILD_TYPE=Release -DBUILD_SHARED_LIBS=ON ..
+ cmake --build . --config Release
+
+ # Find and copy the DLL
+ find . -name "*.dll" -type f
+ cp -v $(find . -name "fdk-aac*.dll" -o -name "libfdk-aac*.dll" | head -1) ../../publish/win-x64/libfdk-aac-2.dll
+ echo "Successfully built and copied libfdk-aac-2.dll"
- name: Upload build artifacts
uses: actions/upload-artifact@v4
diff --git a/README.md b/README.md
index bad0543..9105115 100644
--- a/README.md
+++ b/README.md
@@ -1,277 +1,98 @@
-# AirPlay Receiver
-Open source implementation of AirPlay 2 Mirroring / Audio protocol in C# and .NET.
+# AirPlay Receiver for Windows
-
-
-## Requirements
-
-- .NET 8.0 SDK or later
-- C++ build tools for compiling AAC and ALAC codecs
-
-## Generic
-
-Tested on macOS with iPhone 12 Pro iOS14.
-
-The project is fully functional, but the AAC and ALAC libraries written in C++ must be built.
-
-## Building the Project
-
-### Prerequisites
-1. Install [.NET 8.0 SDK](https://dotnet.microsoft.com/download/dotnet/8.0)
-2. Build the required codecs (see below)
-
-### Compile the Project
-```bash
-$ dotnet restore AirPlay.sln
-$ dotnet build AirPlay.sln --configuration Release
-```
-
-## How To
-
-### Build AAC Codec
-To download, build and install fdk-aac do the following:
-
-Clone the repository and cd into the folder:
-```
-$ git clone https://github.com/mstorsjo/fdk-aac.git
-$ cd fdk-aac
-```
-
-Configure the build and make the library:
-```
-$ autoreconf -fi
-$ ./configure
-$ make
-```
-
-### Build ALAC Codec
-To download, build and install alac do the following:
-
-Clone the repository and cd into the folder:
-```
-$ git clone https://github.com/mikebrady/alac.git
-$ cd alac
-```
-
-Download and paste 'GiteKat''s files in 'alac/codec' folder cloned before
-```
-$ https://github.com/GiteKat/LibALAC/tree/master/LibALAC
-```
-
-The 'mikebrady''s source code does not contains 'extern' keyword.
-We need external linkage so we use 'GiteKat''s source code files.
-
-
-
-Edit makefile.original as follow
-
-
-```
-# libalac make
-
-CFLAGS = -g -O3 -c
-LFLAGS = -Wall
-CC = g++
-
-SRCDIR = .
-OBJDIR = ./obj
-INCLUDES = .
-
-HEADERS = \
-$(SRCDIR)/EndianPortable.h \
-$(SRCDIR)/aglib.h \
-$(SRCDIR)/ALACAudioTypes.h \
-$(SRCDIR)/ALACBitUtilities.h\
-$(SRCDIR)/ALACDecoder.h \
-$(SRCDIR)/ALACEncoder.h \
-$(SRCDIR)/LibALAC.h \
-$(SRCDIR)/dplib.h \
-$(SRCDIR)/matrixlib.h
-
-SOURCES = \
-$(SRCDIR)/EndianPortable.c \
-$(SRCDIR)/ALACBitUtilities.c \
-$(SRCDIR)/ALACDecoder.cpp \
-$(SRCDIR)/ALACEncoder.cpp \
-$(SRCDIR)/LibALAC.cpp \
-$(SRCDIR)/ag_dec.c \
-$(SRCDIR)/ag_enc.c \
-$(SRCDIR)/dp_dec.c \
-$(SRCDIR)/dp_enc.c \
-$(SRCDIR)/matrix_dec.c \
-$(SRCDIR)/matrix_enc.c
-
-OBJS = \
-EndianPortable.o \
-ALACBitUtilities.o \
-ALACDecoder.o \
-ALACEncoder.o \
-LibALAC.o \
-ag_dec.o \
-ag_enc.o \
-dp_dec.o \
-dp_enc.o \
-matrix_dec.o \
-matrix_enc.o
+[中文文档](README_zh.md)
-libalac.a: $(OBJS)
- ar rcs libalac.a $(OBJS)
+Open-source AirPlay 2 receiver for Windows, supporting **screen mirroring** (with audio) and **audio streaming** from Apple devices. Built with C# and .NET 8.
-EndianPortable.o : EndianPortable.c
- $(CC) -I $(INCLUDES) $(CFLAGS) EndianPortable.c
+
-ALACBitUtilities.o : ALACBitUtilities.c
- $(CC) -I $(INCLUDES) $(CFLAGS) ALACBitUtilities.c
+## Features
-ALACDecoder.o : ALACDecoder.cpp
- $(CC) -I $(INCLUDES) $(CFLAGS) ALACDecoder.cpp
+- **Screen Mirroring** — Mirror your iPhone/iPad/Mac screen to Windows with H.264 video and AAC-ELD audio
+- **Audio Streaming** — Play music and podcasts via AirPlay (ALAC and AAC codecs)
+- **Volume Control** — Remote volume adjustment from your Apple device
+- **Auto-Discovery** — Bonjour/mDNS service advertising, your Windows PC appears as an AirPlay receiver automatically
-ALACEncoder.o : ALACEncoder.cpp
- $(CC) -I $(INCLUDES) $(CFLAGS) ALACEncoder.cpp
+## Quick Start
-LibALAC.o : LibALAC.cpp
- $(CC) -I $(INCLUDES) $(CFLAGS) LibALAC.cpp
+### Download
-ag_dec.o : ag_dec.c
- $(CC) -I $(INCLUDES) $(CFLAGS) ag_dec.c
+Download the latest build from [GitHub Actions](https://github.com/YimingZhanshen/Airplay2OnWindows/actions) artifacts. The package includes all required dependencies (FFmpeg, libfdk-aac).
-ag_enc.o : ag_enc.c
- $(CC) -I $(INCLUDES) $(CFLAGS) ag_enc.c
+### Manual Setup
-dp_dec.o : dp_dec.c
- $(CC) -I $(INCLUDES) $(CFLAGS) dp_dec.c
+#### Prerequisites
-dp_enc.o : dp_enc.c
- $(CC) -I $(INCLUDES) $(CFLAGS) dp_enc.c
+- [.NET 8.0 SDK](https://dotnet.microsoft.com/download/dotnet/8.0) or later
+- [FFmpeg](https://github.com/BtbN/FFmpeg-Builds/releases) — `ffmpeg.exe` and `ffplay.exe` in the application directory or PATH
+- [libfdk-aac](https://github.com/mstorsjo/fdk-aac) — `libfdk-aac-2.dll` in the application directory (required for screen mirroring audio)
-matrix_dec.o : matrix_dec.c
- $(CC) -I $(INCLUDES) $(CFLAGS) matrix_dec.c
+#### Build
-matrix_enc.o : matrix_enc.c
- $(CC) -I $(INCLUDES) $(CFLAGS) matrix_enc.c
-
-clean:
- -rm $(OBJS) libalac.a
+```bash
+dotnet restore AirPlay.sln
+dotnet build AirPlay.sln --configuration Release
```
-
-
-
-
-Edit makefile.am as follow
-
-
-```
-## Copyright (c) 2013 Tiancheng "Timothy" Gu
-## Modifications copyright (c) 2016 Mike Brady
-## Licensed under the Apache License, Version 2.0 (the "License");
-## you may not use this file except in compliance with the License.
-## You may obtain a copy of the License at
-##
-## http://www.apache.org/licenses/LICENSE-2.0
-##
-## Unless required by applicable law or agreed to in writing, software
-## distributed under the License is distributed on an "AS IS" BASIS,
-## WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-## See the License for the specific language governing permissions and
-## limitations under the License.
+#### Run
-lib_LTLIBRARIES = libalac.la
+1. Start the application
+2. Open a video player to receive the mirroring stream:
+ ```bash
+ ffplay -f h264 -probesize 32 -analyzeduration 0 -fflags nobuffer -flags low_delay \\.\pipe\AirPlayVideo
+ ```
+3. On your Apple device, open Control Center → Screen Mirroring → select your PC
-libalac_la_CPPFLAGS = -Wno-multichar
-libalac_la_LDFLAGS = -version-info @ALAC_VERSION@
+## Building libfdk-aac on Windows
-libalac_la_SOURCES = \
- EndianPortable.c \
- ALACBitUtilities.c \
- ALACDecoder.cpp \
- ALACEncoder.cpp \
- LibALAC.cpp \
- ag_dec.c \
- ag_enc.c \
- dp_dec.c \
- dp_enc.c \
- matrix_dec.c \
- matrix_enc.c
+The `libfdk-aac-2.dll` is required for decoding AAC-ELD audio during screen mirroring. You can build it from source:
-pkgconfigdir = $(libdir)/pkgconfig
-pkgconfig_DATA = alac.pc
+1. Install [MSYS2](https://www.msys2.org/)
+2. Open MSYS2 MinGW 64-bit terminal and run:
+ ```bash
+ pacman -S mingw-w64-x86_64-gcc mingw-w64-x86_64-cmake make
+ git clone https://github.com/mstorsjo/fdk-aac.git
+ cd fdk-aac
+ mkdir build && cd build
+ cmake -G "MinGW Makefiles" -DCMAKE_BUILD_TYPE=Release -DBUILD_SHARED_LIBS=ON ..
+ cmake --build .
+ ```
+3. Copy the resulting `libfdk-aac-2.dll` to the application directory
-# Install to include/alac
-alacincludedir = $(includedir)/alac
+## Architecture
-# Install everything
-alacinclude_HEADERS = *.h
```
-
-
-
-Configure the build and make the library:
+Apple Device ──AirPlay──► AirPlay Receiver (.NET 8)
+ │
+ ┌─────────┼─────────┐
+ ▼ ▼ ▼
+ Screen Mirror Audio Volume
+ (H.264+AAC) (ALAC) Control
+ │ │
+ ┌─────┴───┐ │
+ ▼ ▼ ▼
+ Named Pipe FDK-AAC NAudio
+ (ffplay) Decoder DirectSound
```
-$ autoreconf -fi
-$ ./configure
-$ make
-```
-
-### Linux
-On terminal type the follow to install build tools
-```
-apt-get install build-essential autoconf automake libtool
-```
-Add compiled DLL path into 'appsettings_linux.json' file.
-
-### MacOS
-On terminal type the follow to install build tools
-
-```
-brew install autoconf automake libtool
-```
-Add compiled DLL path into 'appsettings_osx.json' file.
-
-### Windows
-Use [this](http://www.gaia-gis.it/gaia-sins/mingw64_how_to.html#env) tutorial to understand how to install build tools and how to compile source code on Windows.
-You need MinGW32 or MinGW64 based on arch.
-
-Put repo folders inside msys64 home folder ('C:\msys64\home\').
-
-
-Start an mingw32.exe or mingw64.exe shell based on arch and execute commands.
-
-
-The compiled dll will be saved in 'C:\\msys64\\home\\username\\fdk-aac-master\\.libs\\'.
-Add compiled DLL path into 'appsettings_win.json' file.
-
-TIP 1: If the ALAC library gives you an error during the compilation try to insert the following arguments in the 'makefile.am' file:
-```
-libalac_la_LDFLAGS = -version-info @ALAC_VERSION@ -no-undefined -static-libgcc -static-libstdc++
-```
-TIP 2: If the error code 126 appears when loading the dll, try to import all the dlls located in the C:\msys64\bin\ folder into the bin folder of the project.
-TIP 3: If the error code 193 appears when loading the dll, it means that you are trying to load a dll with the wrong architecture, so you have to compile the dll with the other mingwXX.exe.
-
-## Wiki
-
-Here you will find an [Article](https://github.com/SteeBono/airplayreceiver/wiki/AirPlay2-Protocol) where I explain how the whole AirPlay 2 protocol works.
-
-## Disclamier
-
-All the resources in this repository are written using only opensource projects.
-The code and related resources are meant for educational purposes only.
-I do not take any responsibility for the use that will be made of it.
+- **Video**: H.264 stream written to a named pipe (`\\.\pipe\AirPlayVideo`), consumed by ffplay or any compatible player
+- **Mirroring Audio**: AAC-ELD decoded by native FDK-AAC library via P/Invoke, output through NAudio DirectSound
+- **Streaming Audio**: ALAC/AAC decoded and output through NAudio DirectSound
## Credits
-Inspired by others AirPlay open source projects.
-Big ty to OmgHax.c's author 😱.
+Based on open-source AirPlay protocol implementations. Special thanks to:
+- [SteeBono/airplayreceiver](https://github.com/SteeBono/airplayreceiver) — Original C# AirPlay receiver
+- [itskenny0/airplayreceiver](https://github.com/itskenny0/airplayreceiver) — FDK-AAC mirroring audio implementation
+- [UxPlay](https://github.com/FDH2/UxPlay) — AirPlay protocol reference
+- [mstorsjo/fdk-aac](https://github.com/mstorsjo/fdk-aac) — FDK-AAC codec library
+
+## Disclaimer
+
+All resources in this repository are written using only open-source projects.
+The code and related resources are meant for educational purposes only.
+The author does not take any responsibility for the use that will be made of it.
-## If you want support me 🔥
+## License
-If you appreciate my work, consider buying me a cup of coffee to keep me recharged ☕
-
-
-
-... or ... crypto ...
-
-BTC: 1BXhfC5U75G2H8b99wk5AedGFxtqJ6xf8q
-BCH: 1BXhfC5U75G2H8b99wk5AedGFxtqJ6xf8q
-ETH: 0x4Fc12c7C71C581aBc77945Ab9cFBA8DF9692b713 (ERC20)
+[MIT License](LICENSE)
diff --git a/README_zh.md b/README_zh.md
new file mode 100644
index 0000000..be3e216
--- /dev/null
+++ b/README_zh.md
@@ -0,0 +1,98 @@
+# AirPlay 投屏接收器（Windows）
+
+[English](README.md)
+
+基于 C# 和 .NET 8 的开源 AirPlay 2 接收器，支持 **屏幕镜像**（含音频）和 **音频推送** 功能。
+
+
+
+## 功能特性
+
+- **屏幕镜像** — 将 iPhone/iPad/Mac 的屏幕投射到 Windows 上，支持 H.264 视频和 AAC-ELD 音频
+- **音频推送** — 通过 AirPlay 播放音乐和播客（支持 ALAC 和 AAC 编解码器）
+- **音量控制** — 支持从 Apple 设备远程调节音量
+- **自动发现** — Bonjour/mDNS 服务广播，Windows 电脑自动显示为 AirPlay 接收器
+
+## 快速开始
+
+### 下载
+
+从 [GitHub Actions](https://github.com/YimingZhanshen/Airplay2OnWindows/actions) 的构建产物中下载最新版本。安装包已包含所有依赖（FFmpeg、libfdk-aac）。
+
+### 手动配置
+
+#### 前置条件
+
+- [.NET 8.0 SDK](https://dotnet.microsoft.com/download/dotnet/8.0) 或更高版本
+- [FFmpeg](https://github.com/BtbN/FFmpeg-Builds/releases) — 将 `ffmpeg.exe` 和 `ffplay.exe` 放在程序目录或 PATH 中
+- [libfdk-aac](https://github.com/mstorsjo/fdk-aac) — 将 `libfdk-aac-2.dll` 放在程序目录中（屏幕镜像音频所需）
+
+#### 编译
+
+```bash
+dotnet restore AirPlay.sln
+dotnet build AirPlay.sln --configuration Release
+```
+
+#### 运行
+
+1. 启动应用程序
+2. 打开视频播放器接收投屏画面：
+ ```bash
+ ffplay -f h264 -probesize 32 -analyzeduration 0 -fflags nobuffer -flags low_delay \\.\pipe\AirPlayVideo
+ ```
+3. 在 Apple 设备上，打开控制中心 → 屏幕镜像 → 选择你的电脑
+
+## 在 Windows 上编译 libfdk-aac
+
+屏幕镜像时需要 `libfdk-aac-2.dll` 来解码 AAC-ELD 音频。编译步骤如下：
+
+1. 安装 [MSYS2](https://www.msys2.org/)
+2. 打开 MSYS2 MinGW 64 位终端，执行：
+ ```bash
+ pacman -S mingw-w64-x86_64-gcc mingw-w64-x86_64-cmake make
+ git clone https://github.com/mstorsjo/fdk-aac.git
+ cd fdk-aac
+ mkdir build && cd build
+ cmake -G "MinGW Makefiles" -DCMAKE_BUILD_TYPE=Release -DBUILD_SHARED_LIBS=ON ..
+ cmake --build .
+ ```
+3. 将生成的 `libfdk-aac-2.dll` 复制到程序目录中
+
+## 系统架构
+
+```
+Apple 设备 ──AirPlay──► AirPlay 接收器 (.NET 8)
+ │
+ ┌─────────┼─────────┐
+ ▼ ▼ ▼
+ 屏幕镜像 音频 音量
+ (H.264+AAC) (ALAC) 控制
+ │ │
+ ┌─────┴───┐ │
+ ▼ ▼ ▼
+ 命名管道 FDK-AAC NAudio
+ (ffplay) 解码器 DirectSound
+```
+
+- **视频**：H.264 码流写入命名管道（`\\.\pipe\AirPlayVideo`），由 ffplay 或其他兼容播放器消费
+- **镜像音频**：AAC-ELD 通过原生 FDK-AAC 库（P/Invoke）解码，经 NAudio DirectSound 输出
+- **推送音频**：ALAC/AAC 解码后通过 NAudio DirectSound 输出
+
+## 致谢
+
+基于开源 AirPlay 协议实现。特别感谢：
+- [SteeBono/airplayreceiver](https://github.com/SteeBono/airplayreceiver) — 原始 C# AirPlay 接收器
+- [itskenny0/airplayreceiver](https://github.com/itskenny0/airplayreceiver) — FDK-AAC 镜像音频实现
+- [UxPlay](https://github.com/FDH2/UxPlay) — AirPlay 协议参考
+- [mstorsjo/fdk-aac](https://github.com/mstorsjo/fdk-aac) — FDK-AAC 编解码器库
+
+## 免责声明
+
+本仓库中的所有资源均使用开源项目编写。
+代码及相关资源仅供教育目的使用。
+作者不对其使用方式承担任何责任。
+
+## 许可证
+
+[MIT 许可证](LICENSE)
From b8ad2ed69fd66a71e36ec9dc72de017185e64f9a Mon Sep 17 00:00:00 2001
From: "copilot-swe-agent[bot]" <198982749+Copilot@users.noreply.github.com>
Date: Tue, 10 Feb 2026 21:35:44 +0000
Subject: [PATCH 14/14] CI: remove unused packages, add DLL build error
handling
Co-authored-by: YimingZhanshen <76594627+YimingZhanshen@users.noreply.github.com>
---
.github/workflows/build.yml | 10 ++++++++--
1 file changed, 8 insertions(+), 2 deletions(-)
diff --git a/.github/workflows/build.yml b/.github/workflows/build.yml
index 4861787..c66e837 100644
--- a/.github/workflows/build.yml
+++ b/.github/workflows/build.yml
@@ -50,7 +50,7 @@ jobs:
export PATH="/c/msys64/mingw64/bin:/c/msys64/usr/bin:$PATH"
# Install build dependencies
- pacman -S --noconfirm --needed mingw-w64-x86_64-gcc mingw-w64-x86_64-cmake make autoconf automake libtool
+ pacman -S --noconfirm --needed mingw-w64-x86_64-gcc mingw-w64-x86_64-cmake make
# Clone and build fdk-aac
git clone --depth 1 https://github.com/mstorsjo/fdk-aac.git fdk-aac-src
@@ -62,8 +62,14 @@ jobs:
cmake --build . --config Release
# Find and copy the DLL
+ echo "Built files:"
find . -name "*.dll" -type f
- cp -v $(find . -name "fdk-aac*.dll" -o -name "libfdk-aac*.dll" | head -1) ../../publish/win-x64/libfdk-aac-2.dll
+ DLL_PATH=$(find . -name "fdk-aac*.dll" -o -name "libfdk-aac*.dll" | head -1)
+ if [ -z "$DLL_PATH" ]; then
+ echo "ERROR: No FDK-AAC DLL found after build"
+ exit 1
+ fi
+ cp -v "$DLL_PATH" ../../publish/win-x64/libfdk-aac-2.dll
echo "Successfully built and copied libfdk-aac-2.dll"
- name: Upload build artifacts