Hi Luca,
Nice to see your progress so far. Once you have the
DAB receiver audio listening in place, I would
suggest to have an audio synchronization for continuous
playback without any buffer overflow or under-runs.
DAB+ audio super frame length is 120ms according to DAB+
standard (ETSI TS 102 563). Each audio super frame is
carried in five consecutive logical DAB frames.
Which means 120ms of audio is mapped to 5 DAB frames.
If I add a timestamp at the receiver when the first DAB frame
sample arrives, I can check the max latency when it comes to
audio renderer, I mean after buffering to adjust the variable
decoding time of compressed audio.
t_D = t_A - t_B ,
where,
t_A = time at audio out
t_B = time at input baseband sample.
t_D = maximum system delay.
The difficulty is to estimate the slow clock drift correctly
and separate it from the short-time channel/decoding jitter.
Add a delay to buffer audio at audio out, say D, which is larger
than max system delay. Whenever the audio reaches audio out, check the
delay to separate the clock drift.
drift = t_D - D
Please let me know if you need any more details.
-ben
________________________________
From: Discuss-gnuradio <discuss-gnuradio-bounces+ben.alex=outlook....@gnu.org>
on behalf of Moritz Luca Schmid <luca.moritz.sch...@gmail.com>
Sent: Friday, May 26, 2017 6:19:31 PM
To: GNURadio Discussion List
Subject: [Discuss-gnuradio] [GSoC 17] DAB: updates of the week
Hi everyone,
I just published my latest updates of my DAB project in a new blog
post<https://dabtransceiver.wordpress.com/>.
This week, I created a source block for the Fast Information Channel and
started to build a reception chain for the Main Service Channel (where the
audio data is transmitted).
Read more about it in my post.
Cheers
Luca
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