Resampling audio output for A2DP (from PCM WAV) - audio

How do I bring stereo PCM WAV 32,000Hz with a stream of 1024 kbps (125 KB) to the headset with Bluetooth 2.1 on a CM7 smartphone with DSPManager? Is this possible? SBC is really a bad idea.
Maybe it's possible with the Headset Profile (HSP)? I don`t know about supported codecs in this profile.
#TJD: Because it compresses the compressed stream. My Epic 4G don`t have Apt-X support. My headset Gemix BH-04A yellow.

Related

ALSA Card for Respeaker 4-Mic Setup

During the installation, we are supposed to check the sound card by pressing ‘arecord -L’ to obtain a certain output like shown below,
pi#raspberrypi:~ $ arecord -L
null
Discard all samples (playback) or generate zero samples (capture)
jack
JACK Audio Connection Kit
pulse
PulseAudio Sound Server
default
playback
ac108
sysdefault:CARD=seeed4micvoicec
seeed-4mic-voicecard,
Default Audio Device
dmix:CARD=seeed4micvoicec,DEV=0
seeed-4mic-voicecard,
Direct sample mixing device
dsnoop:CARD=seeed4micvoicec,DEV=0
seeed-4mic-voicecard,
Direct sample snooping device
hw:CARD=seeed4micvoicec,DEV=0
seeed-4mic-voicecard,
Direct hardware device without any conversions
plughw:CARD=seeed4micvoicec,DEV=0
seeed-4mic-voicecard,
Hardware device with all software conversions
usbstream:CARD=seeed4micvoicec
seeed-4mic-voicecard
USB Stream Output
usbstream:CARD=ALSA
bcm2835 ALSA
USB Stream Output
However, the output that I have received is as shown below,
Screenshot of Output
It basically shows that I don’t have the ALSA soundcard, and I cant move on to the sound localization process. Please show how can I move forward, thanks!

Can Linux determine the parameters of an S/PDIF stream?

I have an external audio source that transmits audio data to my computer's sound card via S/PDIF. The sound card has an S/PDIF input. With "arecord" or "audacity" I can record over this input without any problems.
The audio source offers the data in different sample rates (32 kHz, 44.1 kHz, 48 kHz) which I cannot influence. I also can't tell from the source which sample rate the audio source has selected.
For "recording" I would now very much like to keep the sample rate and not have it converted (apparently by the sound card).
Now finally my question: Can I somehow detect with the help of Linux in which format and with which parameters the S/PDIF stream is encoded

HD Audio 44.1 SPDIF

It has been known that integrated audio codecs (on board) CAN'T natively play 44.1 sample rate via analog jacks. But whether they also can't do it via digital toss-link output? resample occurs in this case too?

RTP AAC Packet Depacketizer

I asked earlier about H264 at RTP H.264 Packet Depacketizer
My question now is about the audio packets.
I noticed via the RTP packets that audio frames like AAC, G.711, G.726 and others all have the Marker Bit set.
I think frames are independent. am I right?
My question is: Audio is small, but I know that I can have more than one frame per RTP ​​packet. Independent of how many frames I have, they are complete? Or it may be fragmented between RTP packets.
The difference between audio and video is that audio is typically encoded either in individual samples, or in certain [small] frames without reference to previous data. Additionally, amount of data is small. So audio does not typically need complicated fragmentation to be transmitted over RTP. However, for any payload type you should again refer to RFC that describes the details:
AAC - RTP Payload Format for MPEG-4 Audio/Visual Streams
G.711 - RTP Payload Format for ITU-T Recommendation G.711.1
G.726 - RTP Profile for Audio and Video Conferences with Minimal Control
Other

How to detect the audio availability in the SUSE Linux system

I got some audio pieces in flv format. Each of them is about 10 seconds long.
My question is how to detect whether the audio pieces has "sound", in other words, sometimes the audio pieces has no sound even the size of it is not 0 byte, so how to find those broken/silent audio files by some linux tool/command?
Maybe ffplay can do this? any available advice?
If you want to quickly check if the stream is video only, audio only or if it contains both audio and video, try hexdump -C filename | head. The fifth byte contains information about the contents of the file.
0x01 - video only
0x04 - audio only
0x05 - audio + video
You can also try to play the file using VLC media player. There is a menu option that enables informational messages from the media being played back.

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