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# Configuration for ALSA and the uncompressed audio mapping module
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# General Listener configuration
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# role: Sets the process as a talker or listener. Valid values are
# talker or listener
role = listener

# initial_state: Specify whether the talker or listener should be
# running or stopped on startup.  Valid values are running or stopped.
# If not specified, the default will depend on how the talker or
# listener is launched.
#initial_state = stopped

# stream_addr: Used on the listener and should be set to the 
# mac address of the talker.
stream_addr = 84:7e:40:2b:63:f4

# stream_uid: The unique stream ID. The talker and listener must
# both have this set the same.
stream_uid = 2

# dest_addr: When SRP is being used the destination address only needs to
# be set in the talker.  If SRP is not being used the destination address
# needs to be set in both side the talker and listener.
# The destination is a multicast address, not a real MAC address, so it
# does not match the talker or listener's interface MAC.  There are 
# several pools of those addresses for use by AVTP defined in 1722.
# At this time they need to be locally administered and must be in the range
# of 91:E0:F0:00:FE:00 - 91:E0:F0:00:FE:FF.
# Typically :00 for the first stream, :01 for the second, etc.
#dest_addr = 91:e0:f0:00:fe:00

# max_transit_usec: Allows manually specifying a maximum transit time. 
# On the talker this value is added to the PTP walltime to create the AVTP Timestamp.
# On the listener this value is used to validate an expected valid timestamp range.
# Note: For the listener the map_nv_item_count value must be set large enough to 
# allow buffering at least as many AVTP packets that can be transmitted  during this 
# max transit time.
max_transit_usec = 50000

# max_stale: The number of microseconds beyond the presentation time that media queue items will be purged 
# because they are too old (past the presentation time). This is only used on listener end stations.
# Note: needing to purge old media queue items is often a sign of some other problem. For example: a delay at 
# stream startup before incoming packets are ready to be processed by the media sink. If this deficit 
# in processing or purging the old (stale) packets is not handled, syncing multiple listeners will be problematic.
#max_stale = 1000

# raw_rx_buffers: The number of raw socket receive buffers. Typically 50 - 100 are good values.
# This is only used by the listener. If not set internal defaults are used.
#raw_rx_buffers = 100

# report_seconds: How often to output stats. Defaults to 10 seconds. 0 turns off the stats. 
#report_seconds = 0

# Ethernet Interface Name. Only needed on some platforms when stack is built with no endpoint functionality
# ifname = eth0

current_sampling_rate = 48000

sampling_rates = 44100,48000,96000

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# Mapping module configuration
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# map_lib: The name of the library file (commonly a .so file) that 
#  implements the Initialize function.  Comment out the map_lib name
#  and link in the .c file to the openavb_tl executable to embed the mapper
#  directly into the executable unit. There is no need to change anything
#  else. The Initialize function will still be dynamically linked in.
map_lib = ./libopenavb_map_aaf_audio.so

# map_fn: The name of the initialize function in the mapper.
map_fn = openavbMapAVTPAudioInitialize

# map_nv_item_count: The number of media queue elements to hold.
map_nv_item_count = 32

# map_nv_tx_rate: Transmit rate.
# This must be set for the AAF audio mapping module.
map_nv_tx_rate = 4000

# map_nv_packing_factor: Multiple of how many packets of audio frames to place in a media queue item.
# If sparse timestamping mode is enabled the listener should set here one of the possible
# packing factors values to be sure that proper presentation time is put into media queue item.
# Possible values are: 1, 2, 4, 8, 16, 24, 32, 40, 48, (+ 8)...
map_nv_packing_factor = 32

# map_nv_sparse_mode: if set to 0 presentation time should be
# valid in each packet. Set to 1 to use sparse mode - presentation
# time should be valid in every 8th packet.
map_nv_sparse_mode = 0

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# Interface module configuration
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# intf_lib: The name of the library file (commonly a .so file) that 
#  implements the Initialize function.  Comment out the intf_lib name
#  and link in the .c file to the openavb_tl executable to embed the interface
#  directly into the executable unit. There is no need to change anything
#  else. The Initialize function will still be dynamically linked in.
# intf_fn: The name of the initialize function in the interface.
intf_lib = ./libopenavb_intf_alsa.so

# intf_fn: The name of the initialize function in the interface.
intf_fn = openavbIntfAlsaInitialize

# intf_nv_ignore_timestamp: If set the listener will ignore the timestamp on media queue items.
# intf_nv_ignore_timestamp = 1

# intf_nv_device_name: ALSA device name. Commonly "default" or "plug:dmix"
intf_nv_device_name = default

# intf_nv_audio_rate: Valid values that are supported by AAF are:
#  8000, 16000, 24000, 32000, 44100, 48000, 88200, 96000, 176400 and 192000
intf_nv_audio_rate = 48000

# intf_nv_audio_bit_depth: Valid values that are supported by AAF are:
#  16, 24, 32
intf_nv_audio_bit_depth = 32

# intf_nv_audio_channels
intf_nv_audio_channels = 2

# intf_nv_allow_resampling: 0 = disable software resampling. 1 = allow software resampling. Default is disable.
intf_nv_allow_resampling = 1

# intf_nv_start_threshold_periods: The number of period to wait before starting playback. The larger the value to great
# the latency. The small the number the great chance for a buffer underrun. A good range is 1 - 5.
intf_nv_start_threshold_periods = 3

# intf_nv_period_time: the number of microseconds which should be set to unify latency between different platforms.
# This influence ALSA's period_time and period_size parameters and the result value should be the nearest possible.
# Initial playback latency is equal intf_nv_start_threshold_periods * intf_nv_period_time. If not set internal defaults are used.
# intf_nv_period_time = 31250

# AAF is defined to be big-endian.
intf_nv_audio_endian = big