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-rw-r--r--research/sample-data/mp3_psa_streamer.py347
-rw-r--r--research/sample-data/random-music/30_tagesschau-gong.mp3bin0 -> 20046 bytes
2 files changed, 104 insertions, 243 deletions
diff --git a/research/sample-data/mp3_psa_streamer.py b/research/sample-data/mp3_psa_streamer.py
index 36d2974..f79d674 100644
--- a/research/sample-data/mp3_psa_streamer.py
+++ b/research/sample-data/mp3_psa_streamer.py
@@ -8,46 +8,36 @@ import argparse
import random
import tempfile
import subprocess
-from tqdm import tqdm # progress bar sexxy
-
-# need that for lame transcoding
-try:
- from mutagen.mp3 import MP3 # type: ignore
- MUTAGEN_AVAILABLE = True
-except ImportError:
- MUTAGEN_AVAILABLE = False
+from tqdm import tqdm # For progress bar
# Default configuration values
-DEFAULT_MULTICAST_ADDR = "172.16.20.109"
+DEFAULT_MULTICAST_ADDR = "239.192.55.1"
DEFAULT_PORT = 1681
-DEFAULT_ZONE_INFO = "01 1000 0000 0000 0000 0000 0000" # Zone 1
-DEFAULT_SEND_DUPLICATES = True # Do it like Bodet SIGMA does and just send duplicates "for redundancy" (UDP for emergency systems who the fuck though that was a good idea)
-DEFAULT_CHUNK_SIZE = 1000 # Need to use 1000 due to not understanding protocol fully yet
-DEFAULT_TTL = 3 # We want it going to Multicast and then to the client, nomore otherwise bodeter speaker gets confused
-DEFAULT_HEADER = "4d454c" # MEL goofy ah header
-DEFAULT_COMMAND = "0703" # Probably the stream command
-DEFAULT_STREAM_ID = "110c" # Write none without qoutes to use random stream ID
+DEFAULT_ZONE_INFO = "100000000000000000000000" # Zone 1
+DEFAULT_SEND_DUPLICATES = True
+DEFAULT_CHUNK_SIZE = 900
+DEFAULT_TTL = 2
+DEFAULT_HEADER = "4d454c" # MEL header
+DEFAULT_COMMAND = "070301" # Stream command
-# Prem prem maximum it supports
+# Audio quality presets
HIGH_QUALITY = {
- "bitrate": "256k", # 256k max
- "sample_rate": 48000, # 48000 max
- "channels": 1, # mono because uhm one speaker only
- "sample_format": "s32", # max s32
- "description": "Highest Quality (256kbps, 48kHz)"
+ "bitrate": "64k",
+ "sample_rate": 48000,
+ "channels": 1, # mono
+ "sample_format": "s32", # 16-bit samples (default for MP3)
+ "description": "Ass but no Durchfall"
+ # Note: Original software appears to use LAME 3.99.5
}
-# Bodeter Shitsy Sigma software defaults
LOW_QUALITY = {
- "bitrate": "64k", # Bodet Defaults
- "sample_rate": 32000, # Bodet MP3 file default
- "channels": 1,
- "sample_format": "s16", # bodet MP3 file default
- "description": "Normal Quality (64kbps, 32kHz)"
+ "bitrate": "64k",
+ "sample_rate": 32000,
+ "channels": 1, # mono
+ "sample_format": "s16", # 16-bit samples (default for MP3)
+ "description": "Ass with durchfalls"
}
-DEFAULT_TIMING_FACTOR = 1 # Used back when chat gpt suggested i should adjust the timing... but it was really just that I have to use 1000 byte chunks ...
-
def compute_psa_checksum(data: bytes) -> bytes:
"""Compute PSA checksum for packet data."""
var_e = 0x0000 # Starting seed value
@@ -55,7 +45,7 @@ def compute_psa_checksum(data: bytes) -> bytes:
var_e ^= (data[i] + i) & 0xFFFF
return var_e.to_bytes(2, 'big') # 2-byte checksum, big-endian
-def create_mp3_packet(sequence_num, zone_info, stream_id, mp3_chunk, show_packet_length=False):
+def create_mp3_packet(sequence_num, zone_info, stream_id, mp3_chunk):
"""Create a PSA packet containing MP3 data."""
# Part 1: Header (static "MEL")
header = bytes.fromhex("4d454c")
@@ -64,7 +54,7 @@ def create_mp3_packet(sequence_num, zone_info, stream_id, mp3_chunk, show_packet
# Start marker (0100)
start_marker = bytes.fromhex("0100")
- # Sequence number (1 byte, Little Endian with FF padding)
+ # Sequence number (2 bytes, Little Endian with FF padding)
seq = sequence_num.to_bytes(1, 'little') + b'\xff'
# Command (static 070301)
@@ -73,19 +63,19 @@ def create_mp3_packet(sequence_num, zone_info, stream_id, mp3_chunk, show_packet
# Zone info
zones = bytes.fromhex(zone_info)
- # Stream ID (2 bytes)
- stream_id_bytes = stream_id.to_bytes(2, 'little')
+ # Stream ID (4 bytes)
+ stream_id_bytes = stream_id.to_bytes(4, 'little')
# Constant data (appears in all original packets)
constant_data = bytes.fromhex("05080503e8")
-
# Build the payload (everything that comes after start_marker)
payload = seq + command + zones + stream_id_bytes + constant_data + mp3_chunk
- # Calculate the length for the header - IMPORTANT: This matches what the PSA software does
- # Length is the total packet length MINUS the header (4d454c) and the length field itself (2 bytes)
- length_value = len(start_marker) + len(payload) + 7 # +2 for checksum
+ # Calculate the length for the header
+ # Length is everything AFTER the length field: start_marker + payload
+ # But NOT including checksum (which is calculated last)
+ length_value = len(start_marker + payload) + 7
# Insert the length as a 2-byte value (big endian)
length_bytes = length_value.to_bytes(2, 'big')
@@ -96,11 +86,9 @@ def create_mp3_packet(sequence_num, zone_info, stream_id, mp3_chunk, show_packet
# Calculate and append checksum
checksum = compute_psa_checksum(packet_data)
- # Create final packet
+ # Debug the packet length
final_packet = packet_data + checksum
-
- if show_packet_length:
- print(f"Packet length: {len(final_packet)} bytes, Length field: {length_value} (0x{length_value:04x})")
+ print(f"Packet length: {len(final_packet)} bytes, Length field: {length_value} (0x{length_value:04x})")
return final_packet
@@ -110,7 +98,7 @@ def send_multicast_packet(sock, packet, multicast_addr, port, send_duplicates=Tr
# Send the packet
sock.sendto(packet, (multicast_addr, port))
- # Send a duplicate (common in multicast to improve "reliability")
+ # Send a duplicate (common in multicast to improve reliability)
if send_duplicates:
sock.sendto(packet, (multicast_addr, port))
@@ -128,86 +116,54 @@ def setup_multicast_socket(ttl=DEFAULT_TTL):
def check_dependencies():
"""Check if required dependencies are installed."""
try:
- subprocess.run(['lame', '--version'], stdout=subprocess.DEVNULL, stderr=subprocess.DEVNULL)
+ subprocess.run(['ffmpeg', '-version'], stdout=subprocess.DEVNULL, stderr=subprocess.DEVNULL)
except FileNotFoundError:
- print("Error: LAME encoder is required but not found.")
- print("Please install LAME and make sure it's available in your PATH.")
+ print("Error: ffmpeg is required but not found.")
+ print("Please install ffmpeg and make sure it's available in your PATH.")
sys.exit(1)
-
- if not MUTAGEN_AVAILABLE:
- print("Warning: mutagen library not found. Limited audio file analysis available.")
- print("Install with: pip install mutagen")
def get_audio_info(mp3_file):
- """Get audio file information using mutagen if available."""
+ """Get audio file information using ffprobe."""
try:
- if MUTAGEN_AVAILABLE:
- # Use mutagen to analyze the file
- audio = MP3(mp3_file)
-
- # Extract relevant information
- sample_rate = audio.info.sample_rate
- bit_rate = int(audio.info.bitrate / 1000) # Convert to kbps
- channels = audio.info.channels
- codec_name = "mp3" if audio.info.layer == 3 else f"mpeg-{audio.info.layer}"
-
- return {
- 'bit_rate': bit_rate,
- 'sample_rate': sample_rate,
- 'channels': channels,
- 'codec_name': codec_name
- }
- else:
- # Fallback method: use LAME to identify file info
- cmd = ['lame', '--decode', mp3_file, '-t', '--brief', '-']
- result = subprocess.run(cmd, stdout=subprocess.DEVNULL, stderr=subprocess.PIPE, text=True)
-
- # LAME outputs limited info to stderr when using --brief
- info_text = result.stderr
-
- # Very basic parsing - this is not ideal but works as a fallback
- codec_name = "mp3" # Assume it's MP3
- sample_rate = 44100 # Default assumption
- channels = 2 # Default assumption
- bit_rate = 0 # Unknown
-
- for line in info_text.split('\n'):
- if "MPEG" in line and "Layer" in line:
- codec_name = "mp3"
- if "Hz" in line:
- parts = line.split()
- for part in parts:
- if "Hz" in part:
- try:
- sample_rate = int(part.replace("Hz", ""))
- except ValueError:
- pass
- if "stereo" in line.lower():
- channels = 2
- elif "mono" in line.lower():
- channels = 1
- if "kbps" in line:
- parts = line.split()
- for part in parts:
- if "kbps" in part:
- try:
- bit_rate = int(part.replace("kbps", ""))
- except ValueError:
- pass
-
- return {
- 'bit_rate': bit_rate,
- 'sample_rate': sample_rate,
- 'channels': channels,
- 'codec_name': codec_name
- }
+ # Get audio stream information
+ cmd = [
+ 'ffprobe', '-v', 'quiet', '-print_format', 'json',
+ '-show_streams', '-select_streams', 'a:0', mp3_file
+ ]
+ result = subprocess.run(cmd, stdout=subprocess.PIPE, stderr=subprocess.PIPE, text=True)
+
+ if result.returncode != 0:
+ print(f"Error analyzing audio file: {result.stderr}")
+ return None
+
+ import json
+ info = json.loads(result.stdout)
+
+ if 'streams' not in info or len(info['streams']) == 0:
+ print("No audio stream found in the file.")
+ return None
+
+ stream = info['streams'][0]
+
+ # Extract relevant information
+ bit_rate = int(stream.get('bit_rate', '0')) // 1000 if 'bit_rate' in stream else None
+ sample_rate = int(stream.get('sample_rate', '0'))
+ channels = int(stream.get('channels', '0'))
+ codec_name = stream.get('codec_name', 'unknown')
+
+ return {
+ 'bit_rate': bit_rate,
+ 'sample_rate': sample_rate,
+ 'channels': channels,
+ 'codec_name': codec_name
+ }
except Exception as e:
print(f"Error getting audio information: {e}")
return None
def transcode_mp3(input_file, quality_preset, include_metadata=False):
- """Transcode MP3 file using LAME encoder."""
+ """Transcode MP3 file to match required specifications."""
print(f"Transcoding audio to {quality_preset['description']}...")
# Create temporary output file
@@ -215,28 +171,25 @@ def transcode_mp3(input_file, quality_preset, include_metadata=False):
os.close(fd)
try:
- # Build LAME command
+ # Base command
cmd = [
- 'lame',
- '--quiet', # Less output
- '-m', 'm' if quality_preset['channels'] == 1 else 's', # m=mono, s=stereo
- '--resample', str(quality_preset['sample_rate'] // 1000), # Convert to kHz
- '-b', quality_preset['bitrate'].replace('k', ''), # Remove 'k' suffix
- '--cbr', # Use constant bitrate
+ 'ffmpeg',
+ '-y', # Overwrite output file
+ '-i', input_file,
+ '-codec:a', 'libmp3lame', # Force LAME encoder
+ '-ac', str(quality_preset['channels']),
+ '-ar', str(quality_preset['sample_rate']),
+ '-b:a', quality_preset['bitrate'],
+ '-sample_fmt', quality_preset['sample_format'], # Use configurable sample format
]
# Add options to minimize metadata if requested
if not include_metadata:
- cmd.append('--noreplaygain')
- # Use more compatible tag options instead of --id3v2-none
- cmd.append('--tt')
- cmd.append('') # Empty title
- cmd.append('--tc')
- cmd.append('') # Empty comment
- cmd.append('--nohist')
+ cmd.extend(['-metadata', 'title=', '-metadata', 'artist=',
+ '-metadata', 'album=', '-metadata', 'comment=',
+ '-map_metadata', '-1']) # Strip all metadata
- # Input and output files
- cmd.extend([input_file, temp_output])
+ cmd.append(temp_output)
result = subprocess.run(cmd, stdout=subprocess.DEVNULL, stderr=subprocess.PIPE, text=True)
@@ -253,16 +206,11 @@ def transcode_mp3(input_file, quality_preset, include_metadata=False):
os.remove(temp_output)
return None
-def prepare_mp3_file(mp3_file, quality, include_metadata=False, sample_format=None):
+def prepare_mp3_file(mp3_file, quality, include_metadata=False):
"""Ensure the MP3 file meets the required specifications."""
# Check if quality is valid
quality_preset = HIGH_QUALITY if quality == "high" else LOW_QUALITY
- # Override sample format if specified
- if sample_format:
- quality_preset = quality_preset.copy()
- quality_preset["sample_format"] = sample_format
-
# Get audio file information
info = get_audio_info(mp3_file)
if not info:
@@ -294,73 +242,29 @@ def prepare_mp3_file(mp3_file, quality, include_metadata=False, sample_format=No
print("Audio file already meets required specifications.")
return mp3_file
-def calculate_delay_ms(chunk_size, bitrate, timing_factor=DEFAULT_TIMING_FACTOR):
- """
- Calculate the appropriate delay between chunks for real-time streaming.
-
- Args:
- chunk_size: Size of each audio chunk in bytes
- bitrate: Audio bitrate (string like '128k' or integer)
- timing_factor: Factor to adjust timing (lower = faster playback)
-
- Returns:
- Delay in milliseconds between packets
- """
- if isinstance(bitrate, str) and bitrate.endswith('k'):
- bitrate_kbps = int(bitrate[:-1])
+def calculate_delay_ms(chunk_size, quality):
+ """Calculate the appropriate delay between chunks for real-time streaming."""
+ # Calculate bytes_per_second dynamically from the bitrate
+ if quality == "high":
+ # 128kbps = 16,000 bytes/second
+ bytes_per_second = 16000
else:
- bitrate_kbps = int(bitrate)
+ # 40kbps = 5,000 bytes/second
+ bytes_per_second = 5000
- bytes_per_second = bitrate_kbps * 1000 // 8
- delay_ms = (chunk_size / bytes_per_second) * 1000 * timing_factor
+ # Calculate how long this chunk would play for in real-time
+ delay_ms = (chunk_size / bytes_per_second) * 1000
return delay_ms
-class StreamTimer:
- """Maintains proper timing for streaming audio packets."""
-
- def __init__(self, chunk_size, bitrate, timing_factor=DEFAULT_TIMING_FACTOR):
- """Initialize the stream timer with audio parameters."""
- self.chunk_size = chunk_size
- self.bitrate = bitrate
- self.timing_factor = timing_factor
- self.start_time = None
- self.packets_sent = 0
- self.delay_per_packet = calculate_delay_ms(chunk_size, bitrate, timing_factor) / 1000.0
-
- def start(self):
- """Start the stream timer."""
- self.start_time = time.time()
- self.packets_sent = 0
-
- def wait_for_next_packet(self):
- """Wait until it's time to send the next packet."""
- if self.start_time is None:
- self.start()
- return
-
- self.packets_sent += 1
-
- # Calculate when this packet should be sent
- target_time = self.start_time + (self.packets_sent * self.delay_per_packet)
-
- # Calculate how long to wait
- now = time.time()
- wait_time = target_time - now
-
- # Only wait if we're ahead of schedule
- if wait_time > 0:
- time.sleep(wait_time)
-
def stream_mp3_to_psa(mp3_file, multicast_addr, port, zone_info, send_duplicates,
- chunk_size, quality, include_metadata=False, sample_format=None,
- show_packet_length=False, timing_factor=DEFAULT_TIMING_FACTOR, fixed_stream_id=None):
+ chunk_size, quality, include_metadata=False):
"""Stream an MP3 file to PSA system."""
try:
# Check dependencies
check_dependencies()
# Prepare MP3 file (transcode if needed)
- prepared_file = prepare_mp3_file(mp3_file, quality, include_metadata, sample_format)
+ prepared_file = prepare_mp3_file(mp3_file, quality, include_metadata)
if not prepared_file:
print("Failed to prepare MP3 file.")
return
@@ -368,30 +272,24 @@ def stream_mp3_to_psa(mp3_file, multicast_addr, port, zone_info, send_duplicates
using_temp_file = prepared_file != mp3_file
try:
- # Use fixed stream ID if provided, otherwise generate a random one
- if fixed_stream_id is not None:
- stream_id = fixed_stream_id
- else:
- stream_id = random.randint(0, 0xFFFF) # 16-bit (2 bytes) stream ID
+ # Generate random stream ID for this transmission
+ stream_id = random.randint(0, 0xFFFFFFFF) # Changed to 32-bit to accommodate 4 bytes
# Open the MP3 file
with open(prepared_file, 'rb') as f:
mp3_data = f.read()
# Calculate number of chunks
- chunks = chunk_mp3_data(mp3_data, chunk_size)
+ chunks = [mp3_data[i:i+chunk_size] for i in range(0, len(mp3_data), chunk_size)]
total_chunks = len(chunks)
- # Get the appropriate quality preset
- quality_preset = HIGH_QUALITY if quality == "high" else LOW_QUALITY
-
- # Calculate the appropriate delay for real-time streaming using the actual bitrate
- delay_ms = calculate_delay_ms(chunk_size, quality_preset['bitrate'])
+ # Calculate the appropriate delay for real-time streaming
+ delay_ms = calculate_delay_ms(chunk_size, quality)
print(f"Streaming {os.path.basename(mp3_file)} ({len(mp3_data)/1024:.2f} KB)")
print(f"Split into {total_chunks} packets of {chunk_size} bytes each")
print(f"Sending to multicast {multicast_addr}:{port}")
- print(f"Stream ID: {stream_id:04x}, Zone info: {zone_info}")
+ print(f"Stream ID: {stream_id:08x}, Zone info: {zone_info}")
print(f"Duplicate packets: {'Yes' if send_duplicates else 'No'}")
print(f"Quality preset: {HIGH_QUALITY['description'] if quality == 'high' else LOW_QUALITY['description']}")
print(f"Including metadata: {'Yes' if include_metadata else 'No'}")
@@ -400,17 +298,13 @@ def stream_mp3_to_psa(mp3_file, multicast_addr, port, zone_info, send_duplicates
# Setup multicast socket
sock = setup_multicast_socket()
- # Initialize the stream timer with the appropriate parameters
- timer = StreamTimer(chunk_size, quality_preset['bitrate'], timing_factor)
- timer.start()
-
# Process and send each chunk
sequence_num = 0
with tqdm(total=total_chunks, desc="Streaming progress") as pbar:
try:
for i, chunk in enumerate(chunks):
# Create packet with current sequence number
- packet = create_mp3_packet(sequence_num, zone_info, stream_id, chunk, show_packet_length)
+ packet = create_mp3_packet(sequence_num, zone_info, stream_id, chunk)
# Send the packet
success = send_multicast_packet(sock, packet, multicast_addr, port, send_duplicates)
@@ -421,9 +315,9 @@ def stream_mp3_to_psa(mp3_file, multicast_addr, port, zone_info, send_duplicates
# Update progress bar
pbar.update(1)
- # Wait for the next packet timing using our timer
+ # Delay to match real-time playback
if i < total_chunks - 1: # No need to wait after the last chunk
- timer.wait_for_next_packet()
+ time.sleep(delay_ms / 1000.0)
except Exception as e:
print(f"Error during streaming: {e}")
@@ -441,28 +335,6 @@ def stream_mp3_to_psa(mp3_file, multicast_addr, port, zone_info, send_duplicates
print(f"Error during streaming: {e}")
sys.exit(1)
-def chunk_mp3_data(data, chunk_size):
- """
- Split MP3 data into chunks of specified size.
-
- Args:
- data: The complete MP3 data as bytes
- chunk_size: Size of each chunk in bytes
-
- Returns:
- A list of data chunks
- """
- chunks = []
- # Calculate how many chunks we'll have
- num_chunks = (len(data) + chunk_size - 1) // chunk_size
-
- for i in range(num_chunks):
- start = i * chunk_size
- end = min((i + 1) * chunk_size, len(data))
- chunks.append(data[start:end])
-
- return chunks
-
def main():
# Set up command line arguments
parser = argparse.ArgumentParser(description="Stream MP3 to Bodet PSA System")
@@ -483,12 +355,6 @@ def main():
help="Include metadata in MP3 stream (default: no metadata)")
parser.add_argument("-s", "--sample-format", choices=["s16", "s24", "s32"], default=None,
help="Sample format to use for transcoding (default: s16)")
- parser.add_argument("--show-packet-length", action="store_true",
- help="Show packet length information during streaming")
- parser.add_argument("-t", "--timing-factor", type=float, default=DEFAULT_TIMING_FACTOR,
- help=f"Timing adjustment factor (lower = faster playback, default: {DEFAULT_TIMING_FACTOR})")
- parser.add_argument("--fixed-stream-id", type=int,
- help="Use a fixed stream ID instead of a random one (0-65535)")
args = parser.parse_args()
@@ -501,13 +367,8 @@ def main():
not args.no_duplicates,
args.chunk_size,
args.quality,
- args.include_metadata,
- args.sample_format,
- args.show_packet_length,
- args.timing_factor,
- args.fixed_stream_id # Pass fixed_stream_id parameter
+ args.include_metadata
)
-# Remove duplicate main() call
if __name__ == "__main__":
main()
diff --git a/research/sample-data/random-music/30_tagesschau-gong.mp3 b/research/sample-data/random-music/30_tagesschau-gong.mp3
new file mode 100644
index 0000000..417b5c6
--- /dev/null
+++ b/research/sample-data/random-music/30_tagesschau-gong.mp3
Binary files differ