make latency setting more consistent

README.md

@@ -144,8 +144,7 @@ sudo apt update
 sudo apt remove -y libportaudio2 portaudio19-dev libportaudiocpp0
 echo "y" | rpi-update stable
 
-# TODO: needed  ?
-sudo apt install pipewire wireplumber pipewire-audio-client-libraries rtkit
+sudo apt install -y pipewire wireplumber pipewire-audio-client-libraries rtkit
 mkdir -p ~/.config/pipewire/pipewire.conf.d
 cp src/service/pipewire/99-lowlatency.conf ~/.config/pipewire/pipewire.conf.d/
 

src/auracast/multicast.py

@@ -131,9 +131,9 @@ class ModSoundDeviceAudioInput(audio_io.ThreadedAudioInput):
 
         # Read tunables
         try:
-            blocksize = int(os.environ.get('AURACAST_SD_BLOCKSIZE', '64'))
+            blocksize = int(os.environ.get('AURACAST_SD_BLOCKSIZE', '128'))
         except Exception:
-            blocksize = 64
+            blocksize = 128
         raw_latency = os.environ.get('AURACAST_SD_LATENCY', 'low')
         if raw_latency in ('low', 'high', 'default'):
             latency: typing.Union[str, float] = raw_latency
@@ -143,6 +143,21 @@ class ModSoundDeviceAudioInput(audio_io.ThreadedAudioInput):
             except Exception:
                 latency = 'low'
 
+        # Log the chosen parameters and device info
+        try:
+            dev_info = sounddevice.query_devices(self._device, 'input') if self._device is not None else sounddevice.query_devices(kind='input')
+        except Exception:
+            dev_info = {}
+        logging.info(
+            "SoundDevice RawInputStream: rate=%s ch=%s blocksize=%s latency=%s device=%s hostapi=%s",
+            self._pcm_format.sample_rate,
+            self._pcm_format.channels,
+            blocksize,
+            latency,
+            dev_info.get('name', self._device),
+            dev_info.get('hostapi', 'unknown'),
+        )
+
         # Create the raw input stream with tighter buffering
         self._stream = sounddevice.RawInputStream(
             samplerate=self._pcm_format.sample_rate,
@@ -167,7 +182,7 @@ class ModSoundDeviceAudioInput(audio_io.ThreadedAudioInput):
             return b''
         pcm_buffer, overflowed = self._stream.read(frame_size)
         if overflowed:
-            logging.warning("input overflow")
+            logging.warning("SoundDevice input overflow: frame_size=%s", frame_size)
 
         # Duplicate mono to stereo for downstream expectations
         if self._pcm_format.channels == 1:
@@ -252,7 +267,7 @@ async def init_broadcast(
                     # Broadcast Audio Immediate Rendering flag (type 0x09), zero-length value
                     le_audio.Metadata.Entry(tag = le_audio.Metadata.Tag.BROADCAST_AUDIO_IMMEDIATE_RENDERING_FLAG, data=b"")
                 ]
-                if global_config.immediate_rendering
+                if global_config.immediate_rendering #TODO: verify this
                 else []
             )
         )

src/auracast/multicast_script.py

@@ -13,15 +13,12 @@ if __name__ == "__main__":
     )
     os.chdir(os.path.dirname(__file__))
     
-    # Hint PipeWire to use a very small processing quantum for this client
-    # (we will refine after SRATE is known). Defaults target ~0.33–0.67 ms.
-    os.environ.setdefault("PIPEWIRE_LATENCY", "16/48000")
-    # Lower PulseAudio-side buffering (PortAudio backend typically goes through PA)
+    # Minimal setting to request PipeWire quantum ~128 at 48 kHz via PortAudio
+    os.environ.setdefault("AURACAST_SD_BLOCKSIZE", "128")
+    # Also hint PortAudio/Pulse and PipeWire about small buffers
+    os.environ.setdefault("AURACAST_SD_LATENCY", "0.0027")  # ~128/48000 s
     os.environ.setdefault("PULSE_LATENCY_MSEC", "1")
-    # Request smaller PortAudio blocks via sounddevice shim
-    os.environ.setdefault("AURACAST_SD_BLOCKSIZE", "32")
-    # Accepts 'low'/'high'/'default' or seconds (float). Our shim parses number strings to float.
-    os.environ.setdefault("AURACAST_SD_LATENCY", "0.0015")
+    os.environ.setdefault("PIPEWIRE_LATENCY", "128/48000")
     logging.info("USB pw inputs:")
     usb_inputs = list_usb_pw_inputs()
     logging.info("AEs67 pw inputs:")
@@ -40,15 +37,11 @@ if __name__ == "__main__":
     LC3_SRATE = 24000
     OCTETS_PER_FRAME=60
 
-    # After capture rate is known, constrain client buffers precisely relative to
-    # the active capture rate (so the client quantum matches the device rate).
+    # Ensure PipeWire latency hint reflects the actual capture rate
     try:
-        os.environ["PIPEWIRE_LATENCY"] = f"16/{CAPTURE_SRATE}"
+        os.environ["PIPEWIRE_LATENCY"] = f"128/{CAPTURE_SRATE}"
     except Exception:
         pass
-    os.environ["PULSE_LATENCY_MSEC"] = os.environ.get("PULSE_LATENCY_MSEC", "1")
-    os.environ["AURACAST_SD_BLOCKSIZE"] = os.environ.get("AURACAST_SD_BLOCKSIZE", "32")
-    os.environ["AURACAST_SD_LATENCY"] = os.environ.get("AURACAST_SD_LATENCY", "0.0015")
 
     config = auracast_config.AuracastConfigGroup(
         bigs = [
@@ -62,6 +55,7 @@ if __name__ == "__main__":
             #auracast_config.AuracastBigConfigEng(),
         ],
         immediate_rendering=True,
+        presentation_delay_us=40000,
         qos_config=auracast_config.AuracastQosHigh(),
         auracast_sampling_rate_hz = LC3_SRATE,
         octets_per_frame = OCTETS_PER_FRAME, # 32kbps@16kHz