AVDTP socket?

Hi

Is AVDTP complete enough to pass data around? When I try to create a socket

socket(PF_BLUETOOTH, SOCK_STREAM, BTPROTO_AVDTP)

perror gave me "Protocol not supported." Is this the right way to get an AVDTP socket?

I'm using debian-testing w/libbluetooth 2.10-1, kernel 2.6.9 (unpatched)

Brad


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Hi Brad,

there is no AVDTP in the kernel. The only thing I did, was to reserve a protocol number for it when it was clear to me that this code should run inside the kernel.

However what we really need first is an open source implementation of the SBC codec. I am not an audio expert and so I am a little bit lost here.

Regards

Marcel

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Marcel,

What I wanted to do was start by using the reference SBC codec or just one of the reference streams (cleverly hidden in the *zipfile* A2DP_TS.pdf at bluetooth.org) and then experiment with sending it to the headset.

I'm looking at the diagram on p. 17 of the A2DP spec... it looks like the AVDTP layer takes the encoded stream, breaks it up and puts the media packet headers on it. Is that right?

I am a little confused about this though... does this show the A2DP sink as using the rfcomm protocol?

$ sdptool search --bdaddr 00:08:F4:30:05:BB 0x1101
Class 0x1101
Searching for 0x1101 on 00:08:F4:30:05:BB ...
Service Name: Audio Sink
Service RecHandle: 0x10005
Service Class ID List:
  "Serial Port" (0x1101)
Protocol Descriptor List:
  "L2CAP" (0x0100)
  "RFCOMM" (0x0003)
    Channel: 1

Brad

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Hi Brad,

I looked a little bit into ALSA and actually what we really need is the support in the ALSA library for understanding SBC. So the kernel don't has to worry about it.

In general yes, but the AVDTP is crap. It needs at least two L2CAP channels on PSM 25.

I saw that too and actually I don't know what is it good for. Maybe you wanna try to connect it to a TTY and then copy the SBC files into it.

Regards

Marcel

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Marcel

good, so it's an alsa project. :)

in the perfect world, you should be able to hand an mp3 over to a userspace library that only transcodes into SBC if the headset does not advertise its own mp3 codec. same with aac. avoiding the transcode would be good for reducing latency and preserving batteries on a portable player. it may be academic if none of the headsets implement optional codecs.

it may be ugly, but it's all we've got, eh?

I copied a few of the sbc streams. They resulted in a whole lot of static on the headset. I think we need to tell the headset we're using avdtp and get that media packet header in there.

I don't know where to start with the avdtp protocol handling. Can you give me a pointer?

Brad

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Hi Brad,

it is not really an ALSA project, but from my understanding it is the best way to go. We must extend the ALSA kernel part to allow a PCM with the type SBC and then expect from the library to do the rest for us.

And yes, if there are headphones with MP3 or any other codec in their endpoint list, then we should use a native one. Actually as far as I got through the ALSA code we simply set what we support and then the rest is done automatically by the library.

This was not what I meant. You must open the signal channel first and then open the media channel. This is the basic mode.

Don't ask me. I don't know what this RFCOMM channel is for.

I think you should start with a simple test implementation in user space that fulfils your purpose. I haven't understand the AVDTP enough to move any part of it into the kernel at the moment.

Regards

Marcel

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Marcel

[avdtp]

The avdtp document is almost 150 pages long and very terse! It looks to me like I start with sdp to find the avdtp psm, open it with raw l2cap, then do stream endpoint discovery and get capabilities, set stream config, open the l2cap stream psm and send the stream. I am planning to write an app that sends one of the reference sbc streams.

What should the sdp search look like? The more promising sdptool options I found didn't work flawlessly... I'm assuming 0x110d is A2DP and 0x19 is AVDTP (http://www.bluetooth.org/assigned-numbers.htm redirects to http://www.bluetoothsig.org/assigned-numbers/ which won't resolve). What is the uint16 under AVDTP trying to tell us?

sdptool search --bdaddr 00:08:F4:30:05:BB 0x110d
Class 0x110D
Searching for 0x110d on 00:08:F4:30:05:BB ...
Service Name: Audio Sink
Service RecHandle: 0x10002
Service Class ID List:
"" (0x110b)
Protocol Descriptor List:
"L2CAP" (0x0100)

    PSM: 25
  "" (0x0019)
    uint16: 0x0
Profile Descriptor List:
  "" (0x110d)
    Version: 0x0100

Brad

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Hi Brad,

no need to search for the PSM via a SDP request. It is always PSM 25, so you can simply skip this like we do for RFCOMM. Open it with a raw L2CAP socket is wrong, because this gives you only access to the L2CAP signal channel. Simply open it with SOCK_SEQPACKET and send the AVDTP commands on it. Before you start streaming you must open another L2CAP channel on PSM 25. There are no different PSMs like for HID or HCRP. It took me some time to understand this and actually at that time I never realized that this is even possible, but we already support it ;)

The CVS version of sdptool should be able to decode everything A2DP and AVDTP related. I think the uint16 for AVDTP shouldn't be there. It is a bug as far as I know. However you don't need this, because it is always PSM 25 and my sender device simply only looks if a A2DP sink is present or not.

Regards

Marcel

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Marcel,

have you coded up something basic already? could I have a look at it as a starting point? (this is all very new)

thanks
Brad

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Hi Bard,

I haven't written any code yet. I only used sdptool, l2test and hcidump.

Regards

Marcel

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Marcel

My a2play.c (committed to cvs btw) opens the l2cap socket and exchanges a couple of messages with the bt420n before I have reached something I don't know how to interpret:

send avdtp_discover
receive avdtp_discover accept, seid=1
send avdt_get_capabilities for seid=1
receive avdt_get_capabilities accept

The accept looks like this byte-for-byte:

02 -> response accept
02 -> AVDT_GET_CAPABILITIES
01 -> category = media transport
00 -> length of service capabilities = 0
07 -> 7 'elements' follow
06 -> elements[0]
00 ->
00 ->
ff ->
ff ->
02 ->
fa -> elements[6]

The avdt spec drops off sharply around page 86 when talking about what comes after the service capabilities length. Any ideas of what to make of the elements?

Brad

btw, I did have code in there to be totally sure I had drained off all the avdtp_discovery accept packets. Currently the code assumes it will only get one seid. The code gets ugly when I try to make it more generic. A nonblocking read would help but I haven't figured out how to get that.

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Hi Brad,

this should be the codec specific part. Look at A2DP for more details.

Regards

Marcel

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Marcel

It doesn't totally seem to match up. The last 4 bytes seem to be the capabilities of SBC (p.20 of a2dp)

ff -> supports all sampling frequencies, channel modes ff -> supports all block lengths, subbands, allocation methods 02 -> minimum bitpool val
fa -> max bitpool val

I can't find where the first 06, 00, 00 fit. Also, does the seid=1 in the avdtp_discover accept correspond to only supporting SBC? I wasn't expecting to know the supported codecs until after the first avdt_get_capabilities.

Does anyone have a cached copy of assigned-numbers.htm? It's extremely frustrating to have all these pointers to a web page (http://www.bluetooth.org/assigned-numbers.htm) that won't resolve.

Brad

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Hi Brad,

I looked it up and the GET_CAPABILITIES response you get back contains two block. The first is the "media transport" with 0 bytes of additional information and the second is the "media codec" part with 6 bytes of codec specific information.

	0x01 = media transport
	0x07 = media codec

The first 0x00 represents the "media type" and the second the "media codec type". And this means "audio" and "SBC". And the rest is SBC specific as you already recognized.

Regards

Marcel

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Marcel

Argh... I'm stuck again. the avdtp_set_configuration command goes out but reading the response just blocks. I'm not sure what is wrong... I send:

00 -> header = request
03 -> AVDTP_SET_CONFIGURATION
04 -> acp_seid = 1
04 -> int_seid = 1 (how should I choose the int_seid?) 01 -> service category = media transport
00 -> length of service capabilities = 0
07 -> capability type = media codec
06 -> bytes remaining? not sure if this is right. 00 -> media type = audio media type
00 -> media codec type = SBC
22 -> 44.1khz stereo
15 -> 16-byte blocks, 8 subbands, "loudness" allocation 02 -> min bitpool
fa -> max bitpool

a2play.c in cvs contains the code I'm trying...

Brad

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Hi Brad,

I think you really need to fill in the transaction number in the header byte. So it should be 0x00, 0x10, 0x20, 0x30 etc.

The bytes remaining value is right.

Aiptek devices I see that they use 0x22 0x46 for the SBC setting. And this is 44.1kHz, stereo, blocklen 64, 8 subbands, SNR if I decoded it correct.

Regards

Marcel

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Marcel

You noticed I have code to open and send a stream. When I send the reference sbc streams byte-for-byte, I can hear a signal mixed with a lot of noise. But then I don't know what the sbc parameters were for any of the reference streams. Maybe it would just work if I knew that.

The reference decoder does not report the sbc parameters and I'm only assuming that they have broken up the data and wrapped it in media packet headers and sbc headers for a conservative mtu. The reference encoder/decoder is undocumented and I didn't figure out how to set parameters in the encoder. I don't want to spend any more time trying to figure out the reference stuff.

I guess the next step for me is to put together some kind of basic encoder and try to send over something simple like a square wave.

Brad

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Hi Brad,

the reference encoder/decoder has some command line options. You will find them in "A2DP Test v10.pdf". The file format of the *.sbc is also somewhere described. However I couldn't find it right now, but every file contains a number of frames and all of them with a SBC header:

struct sbc_frame_header {

        uint8_t syncword:8;             /* Sync word */
        uint8_t subbands:1;             /* Subbands */
        uint8_t allocation_method:1;    /* Allocation method */
        uint8_t channel_mode:2;         /* Channel mode */
        uint8_t blocks:2;               /* Blocks */
        uint8_t sampling_frequency:2;   /* Sampling frequency */
        uint8_t bitpool:8;              /* Bitpool */
        uint8_t crc_check:8;            /* CRC check */

} __attribute__ ((packed));

Regards

Marcel

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Marcel,

I tried reading this from the header but I get nonsensical results like "3" for frequency. Only one bit is supposed to be specified in fields like frequency, so the headset (rightly) rejects the set_config.

i committed it to cvs

brad

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Hi Brad,

I think you have to convert it first. I don't know why the SBC example files have this header. From my sniffs of the AVDTP media channel I haven't seen this header data anywhere. So it is somekind of SBC file format. Try this conversion:

static char *sf2str(uint8_t sf)

{
        switch (sf) {
        case 0:
                return "16 kHz";
        case 1:
                return "32 kHz";
        case 2:
                return "44.1 kHz";
        case 3:
                return "48 kHz";
        }
}

static char *cm2str(uint8_t cm)

{
        switch (cm) {
        case 0:
                return "Mono";
        case 1:
                return "Dual channel";
        case 2:
                return "Stereo";
        case 3:
                return "Joint stereo";
        }
}

Regards

Marcel

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Moin,

I'm currently working (more or less; more less than more) on SBC and trying to do a decoder first because I figure that would be easier (because I know what the result is supposed to look like). I have problems synthesizing the PCM audio samples but can parse all of the example SBC streams just fine.

I'll attach what I have now. The sbc-test.c currently dumps subband samples on stdout for testing, so just redirect that to /dev/null. The decoded frame headers are written to stderr.

Henryk Plötz
Grüße aus Berlin

Henryk,

Great! Marcel and I don't have a lot of confidence in taking on sbc.

Can I put what you've given us in a module in our bluetooth-alsa sourceforge cvs and make you a developer?

I think the structs you have should probably be marked as __attribute__ ((packed)) so they don't get realigned by gcc. I'm not sure if there's a portable alternative.

Brad

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Hi Brad,

please do this for now. Later on I think we have to move some stuff back into the BlueZ CVS.

And guys, please change your coding style ;)

Regards

Marcel

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Moin,

On Thu, 18 Nov 2004 08:26:02 -0700, Brad Midgley <[email protected]> wr= ote:

Sure. My sourceforge username is henryk.

Ok, will do that.

--=20
Henryk Pl=F6tz
Gr=FC=DFe aus Berlin

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Henryk,

I put you in as a developer. Your code appears in the btsco cvs module in the subdirectory "sbc" for now. The makefiles aren't linked together at all and I'm not sure I'd make that effort. I get the feeling your stuff ultimately belongs in the alsa lib, but if it needs to be used outside alsa then that's not a perfect fit.

Thanks!

Brad

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Hi Henryk,

I just tested it and the result so far is very great. Keep on working on it. What format do you think is the easiest to encode/decode?

And btw your file creation should be this command:

	fh = open("bla_.au", O_WRONLY|O_CREAT|O_TRUNC, 0644);

Regards

Marcel

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Marcel

I cleaned up the bitfields for the avdtp packets and committed it...

Do you think this looks right for covering the header on avdtp page 45?

// remember htobs() for converting from host to bluetooth byte ordering // when putting data in this struct
typedef struct {
uint8_t v:2;
uint8_t p:1;
uint8_t x:1;
uint8_t cc:4;
uint8_t m:1;
uint8_t pt:7;
uint16_t sequence_number;
uint32_t timestamp;
uint32_t ssrc;
uint32_t csrc[0];
} __attribute__ ((packed)) media_packet_header;

If that looks like the right direction, I need to work on delineating the sbc frames inside the sbc file so i can put on the media packet headers and the media payload headers. Is it straightforward?

Brad

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Hi Brad,

this looks ok to me. From the trace of my Aiptek set I saw that these header is filled with the following values:

	0x80 0x00 0x00 0x01 0x00 0x00 0x00 0x00 0x00 0x00 0x00 0x00

So they only use the sequence number and nothing more. The next two bytes are always 0xff 0xff and then the payload differs. After that I see the 0x9c sync byte from the SBC file.

So what I think is that you can simply put the header and the other two bytes into a L2CAP packet and then add as much data from the SBC file as the MTU allows.

If the SBC frame header is always repeated I don't understand why we need to set a specific SBC configuration with AVDTP.

Regards

Marcel

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Marcel

I still have to break the stream into chunks and put the payload header on them (a2dp p.24).

I'm almost there but now I need to figure out how to find sbc frame boundaries in the file. Henryk? :)

Brad

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Hi Brad,

but the boundary don't cares. I see that in the trace. You can break the SBC file where you want and add the media header.

Regards

Marcel

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Hi Brad,

actually like with the attached patch. Most of the example *.sbc files are working. Some of them are not working and this is maybe because the frame header information don't matches the SBC configuration. Or maybe hcidump is decoding this wrong.

Regards

Marcel

Index: a2play.c
===================================================================
RCS file: /cvsroot/bluetooth-alsa/btsco/a2play.c,v retrieving revision 1.10

diff -u -r1.10 a2play.c
@@ -285,7 +285,7 @@
   return psm;
 }
 
-int do_connect(char *svr, int psm, int linger, int reliable)
+int do_connect(char *svr, int psm, int linger, int reliable, uint16_t *mtu)
 {
   struct sockaddr_l2 rem_addr, loc_addr;
   struct l2cap_options opts;

@@ -359,6 +359,9 @@
   fprintf(stderr, "Connected [imtu %d, omtu %d, flush_to %d]\n",
          opts.imtu, opts.omtu, opts.flush_to);
   
+  if (mtu)
+    *mtu = opts.omtu;
+
   return s;
 }
 

@@ -482,6 +485,7 @@
   bdaddr_t local;
   bdaddr_t bdaddr;
   int psm_cmd, psm_stream;
+  uint16_t mtu, seq_num;
 
   char *addrstr;
   char *filename;

@@ -538,7 +542,7 @@
   sigaction(SIGCHLD, &sa, NULL);
   sigaction(SIGPIPE, &sa, NULL);
   
-  s = do_connect(addrstr, psm_cmd, 0, 0);
+  s = do_connect(addrstr, psm_cmd, 0, 0, NULL);
   if(s < 0) {
     fprintf(stderr, "cannot open psm_cmd = %d\n", psm_cmd);
     exit(-1);

@@ -572,7 +576,8 @@
 
   // open the stream l2cap
 
-  streamfd = do_connect(addrstr, psm_stream, 0, 0);
+  mtu = 48;
+  streamfd = do_connect(addrstr, psm_stream, 0, 0, &mtu);
   if(streamfd < 0) {
     fprintf(stderr, "cannot open psm_stream = %d\n", psm_stream);
     exit(-1);

@@ -607,16 +612,38 @@
 
   char buf[BUFS];
 
+  if (mtu > BUFS)
+    mtu = BUFS;
+
   // TODO remove
-  terminate = 1;
+  terminate = 0;
+
+  seq_num = 1;
+  memset(buf, 0, 12);
+  buf[0] = 0x80;
+  buf[2] = seq_num >> 8;
+  buf[3] = seq_num & 0xff;
+  buf[12] = 0x0ff;
+  buf[13] = 0x0ff;
+  memcpy(buf + 14, &sbc_info, sizeof(sbc_info));
+  size = read(fd, buf + 14 + sizeof(sbc_info), mtu - sizeof(sbc_info) - 14);
+  write(streamfd, buf, size + 14 + sizeof(sbc_info));
+  seq_num++;
 
   while(!terminate) {
-    size = read(fd, &buf, BUFS);
+    memset(buf, 0, 12);
+    buf[0] = 0x80;
+    buf[2] = seq_num >> 8;
+    buf[3] = seq_num & 0xff;
+    buf[12] = 0xff;
+    buf[13] = 0xff;
+    size = read(fd, buf + 14, mtu - 14);
+    seq_num++;
     if(size <= 0) {
       terminate = 1;
     } else {
       // framing, fragmenting, a2dp headers: avdtp p.45 and a2dp spec p.23
-      write(streamfd, buf, size);
+      write(streamfd, buf, size + 14);
     }
   }
 

Hi Brad,

and I figured out what the extra RFCOMM channel 1 on my Aiptek headphone does. You can simply stream the SBC files over it. The attached program will do this for you, but actually it only works on streams with two channels. I don't know what to change to support all test SBC files.

Regards

Marcel

/*
*
* Stream SBC files over RFCOMM
*
* Copyright (C) 2004 Marcel Holtmann <[email protected]> *
*

 *  This program is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License version 2 as
 *  published by the Free Software Foundation;
 *
 *  THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
 *  OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 *  FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OF THIRD PARTY RIGHTS.
 *  IN NO EVENT SHALL THE COPYRIGHT HOLDER(S) AND AUTHOR(S) BE LIABLE FOR ANY
 *  CLAIM, OR ANY SPECIAL INDIRECT OR CONSEQUENTIAL DAMAGES, OR ANY DAMAGES 
 *  WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN 
 *  ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF 
 *  OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
 *
 *  ALL LIABILITY, INCLUDING LIABILITY FOR INFRINGEMENT OF ANY PATENTS, 
 *  COPYRIGHTS, TRADEMARKS OR OTHER RIGHTS, RELATING TO USE OF THIS 
 *  SOFTWARE IS DISCLAIMED.

*
*/

#include <stdio.h>
#include <errno.h>
#include <fcntl.h>
#include <sys/socket.h>

#include <bluetooth/bluetooth.h>
#include <bluetooth/rfcomm.h>

static int rfcomm_connect(bdaddr_t *src, bdaddr_t *dst, uint8_t channel)

{
	struct sockaddr_rc addr;
	int sk;

	sk = socket(PF_BLUETOOTH, SOCK_STREAM, BTPROTO_RFCOMM);
	if (sk < 0)
		return -1;

	memset(&addr, 0, sizeof(addr));
	addr.rc_family = AF_BLUETOOTH;
	bacpy(&addr.rc_bdaddr, src);
	if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
		close(sk);
		return -1;
	}

	memset(&addr, 0, sizeof(addr));
	addr.rc_family = AF_BLUETOOTH;
	bacpy(&addr.rc_bdaddr, dst);
	addr.rc_channel = channel;
	if (connect(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
		close(sk);
		return -1;
	}

	return sk;
}

static int stream_sbc(fd, sk)

{
	unsigned char buf[330];
	int err, len;

	while (1) {
		len = read(fd, buf, sizeof(buf));
		if (len <= 0)
			break;

		err = write(sk, buf, len);
		if (err < 0)
			break;
	}

	return 0;
}

int main(int argc, char *argv[])

{
	bdaddr_t bdaddr;
	int fd, sk;

	if (argc < 3) {
		fprintf(stderr, "Usage: rcplay <bdaddr> <filename>\n");
		exit(1);
	}

	str2ba(argv[1], &bdaddr);

	fd = open(argv[2], O_RDONLY);
	if (fd < 0) {
		perror("Can't open file");
		exit(1);
	}

	sk = rfcomm_connect(BDADDR_ANY, &bdaddr, 1);
	if (sk < 0) {
		perror("Can't open connection");
		close(fd);
		exit(1);
	}

	stream_sbc(fd, sk);

	close(sk);
	close(fd);

	return 0;
}

Moin,

On Fri, 19 Nov 2004 21:45:43 +0100, Marcel Holtmann <[email protected]> w= rote:

Excellent!

I know I did try this as the first thing when I discovered that channel (using rfcomm bind ..., rfcomm connect ... and cat foo.sbc > /dev/rfcomm0) but obviously didn't try hard enough. Your rcplay does work with mono files for me just fine.

decoder, just listened at it) I'd say my headphones play back the first 10 SBC test streams well, with the notable exception of number
7. So it seems that being a mono file doesn't prevent playback this way generically.

--=20
Henryk Pl=F6tz
Gr=FC=DFe aus Berlin

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Hi Henryk,

I think the problem with the other approach (I also tried it) was that the underlaying frame size is different if you connect through a TTY. I stuck it at 330 bytes which is a DH5 minus protocol overhead.

The test files 01, 03, 05 don't work for me.

Do you also use the Aiptek or the BlueTake one. The Aiptek is very old and maybe the decoder in my headphone has a bug.

Regards

Marcel

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Moin,

On Fri, 19 Nov 2004 22:33:44 +0100, Marcel Holtmann <[email protected]> w= rote:

I only have the BlueTake one.

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Marcel,

I added rcplay.c to cvs and made you a developer on bluetooth-alsa so you can tweak things if you see fit.

The bluetake sometimes works and sometimes doesn't when using a2play. The same stream will work a couple of times and then it produces loud white noise. There is probably something still not quite right.

I disabled this section in a2dp.c

memset(buf, 0, 12);
buf[0] = 0x80;
buf[2] = seq_num >> 8;
buf[3] = seq_num & 0xff;
buf[12] = 0x0ff;
buf[13] = 0x0ff;
memcpy(buf + 14, &sbc_info, sizeof(sbc_info));
size = read(fd, buf + 14 + sizeof(sbc_info), mtu - sizeof(sbc_info) -
14);
write(streamfd, buf, size + 14 + sizeof(sbc_info)); seq_num++;

but it didn't help or hurt. I couldn't see why sbc_info was being written to the stream since I thought it was just a file header. Does your headset work with a2play as it is in cvs now?

Brad

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Hi Brad,

thanks. I will do that.

Yes, perfect. I tried it with an MP3 that I convert to a wave file and used the sbc_encoder.exe to create the SBC file. It also works fine with the rcplay tool.

I think that GCT (the company that build the chips and wrote the software for both headphone types) added their own SBC media packet format. I read the A2DP specification again and as far as I understand it, we must read the SBC frame header from the first frame. Set the AVDTP configuration according to that header and then send the SBC frame with the SBC media packet header (with or without fragmenting). And then proceed with the other SBC frames in that file.

I found nothing about 0xff 0xff and even playing with content protection and multiplexing bits and bytes, nothing resulted in these two bytes.

What we need is an easy way to calculate the frame size from the frame header without decoding the frame data itself. And also scanning for the sync byte is not really an option. If we get this working, we need to check if these headphones also support real A2DP media packets.

Regards

Marcel

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Marcel,

Shouldn't this work to send the header in a separate write() and have a struct for it?

I thought everything ended up being little-endian on the wire, but

    buf[2] = seq_num >> 8;
    buf[3] = seq_num & 0xff;

looks big endian to me (00 then 01 for 0x0001). When my code writes to sequence_number, it comes out little endian (01 then 00 for 0x0001).

Brad

while(!terminate) {

-    memset(buf, 0, 12);
-    buf[0] = 0x80;
-    buf[2] = seq_num >> 8;
-    buf[3] = seq_num & 0xff;
-    buf[12] = 0xff;
-    buf[13] = 0xff;
-    size = read(fd, buf + 14, mtu - 14);
+    memset(&packet_header, 0, sizeof(packet_header));
+    packet_header.cc = 8;
+    packet_header.sequence_number = htobs(seq_num);
+    packet_header.csrc[0] = htobl(0xffff0000);
+
+    //    dump_packet(&packet_header, sizeof(packet_header)); exit(0);
+
+    size = read(fd, buf, mtu - sizeof(packet_header));
     seq_num++;
     if(size <= 0) {
       terminate = 1;
     } else {
-      // framing, fragmenting, a2dp headers: avdtp p.45 and a2dp spec p.23
-      write(streamfd, buf, size + 14);
+      // avdtp p.45
+      write(streamfd, &packet_header, sizeof(packet_header));
+      // what about framing, fragmenting, a2dp headers (a2dp spec p.23)
+      write(streamfd, buf, size);
     }

}
close(fd);

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Hi Brad,

no, because the L2CAP is a SOCK_SEQPACKET. You must send the packet at once. Otherwise you will get bad fragmentation.

Look at the RTP RFC they mentioned in the specification. I didn't checked it. Simply used the stuff I've seen in my trace.

And btw the endianess may differ between parts of the Bluetooth protocol stack. This is the problem when you adapt other protocols.

Regards

Marcel

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Moin,

On Fri, 19 Nov 2004 23:04:06 +0100, Marcel Holtmann <[email protected]> w= rote:

I just re-discovered that in the A2DP spec. It's on page 70, section 12.9.

--=20
Henryk Pl=F6tz
Gr=FC=DFe aus Berlin

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Hi Henryk,

btw what is the format of sbc_test_01.sbc etc. Do we have a specific list of the settings in these files. I like to put it for reference on the web.

Regards

Marcel

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Moin,

On Sat, 20 Nov 2004 01:43:16 +0100, Marcel Holtmann <[email protected]> w= rote:

Yupp, that's in A2DP_TS_C1.1v10.pdf on pages 34 and following.

--=20
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Gr=FC=DFe aus Berlin

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Hi Henryk,

I've added a tool called sbcinfo to the CVS that goes through a SBC file and decodes this information. I also updated a2play and rcplay to make use of the SBC frame length.

However I was unable to get my Aiptek headphone to play some music with the original SBC media header.

And btw the AVDTP parser in the hcidump CVS repository should now be able to display the correct decoded information on the signal channel.

Regards

Marcel

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Moin,

Well, from what I understand the SBC frames can greatly vary in length and one can't know the length before unpacking the bitstream (or at least calculating the 'bits' array). That's why my code works like it
does: the sbc_unpack_frame() returns the length of the frame that it just unpacked so the calling code can know how many bytes to skip.

There should be a correlation between the bitpool value used and the length of the frame, but I didn't look into this. Even then one would have to parse the frame header to know what parameters have actually been used and calculate the length based on that.

--=20
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Gr=FC=DFe aus Berlin

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Moin,

On Fri, 19 Nov 2004 00:59:28 +0100, Marcel Holtmann <[email protected]> w= rote:

Thanks. I'm getting more and more confident in that my bitstream decoding is correct, but I can't seem to get the synthesis done. I got me some books on compression from the library and read the chapters about subband coding, but they basically said what I knew (or figured) anyways.

In what respect to?=20

Ahh, thanks. I had wondered why it kept doing funny things, but didn't investigate further.

--=20
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Gr=FC=DFe aus Berlin

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Hi Henryk,

number of subbands etc. The SBC defines a lot, but not all of them can be supported when we want to integrate it into ALSA. For the sink all are mandatory, but for the source only some of them.

Regards

Marcel

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Moin,

On Fri, 19 Nov 2004 19:56:40 +0100, Marcel Holtmann <[email protected]> w= rote:

Hmm, since we're dealing with a software and have quite some computational power the encoder will probably support any possible configuration. I'd just take the values from Table 4.7 on page 22 of the A2DP spec as a set of reasonable default values where the only choice left is middle vs. high quality.

--=20
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Hi Henryk,

from my current knowledge of ALSA we can set the frequency and the number of channels. So supporting all 4 frequencies and mono (1 channel) and joint stereo (2 channels) from the interface point of view is a reasonable way? Should we really do joint stereo instead of stereo?

Any thoughts about this are welcome.

Regards

Marcel

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Moin,

Am Thu, 18 Nov 2004 14:27:26 +0100 schrieb Henryk Plötz:

Ok, catched it. I reimplemented the SBC analyzer and synthesizer in Octave which is easier to play around with and found that my C synthesizer was working fine - on wrong subband samples. The bitstream unpacker had a bug which made it shift the incoming bits once to often.

So sbc-test from the CVS is now able to decode 4 and 8 subband mono SBC files (there's something wrong with stereo, I'll look into that shortly).

I also moved all my static const tables to a separate header file and replaced my cos() calls with precomputed tables. The decoder is still about 5 times slower than the reference one, though.

Henryk Plötz
Grüße aus Berlin
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Hi Henryk,

I was only able to decode 01, 03, 07 and 09. All other mono test files are failing. However good work, because these four sound perfect.

Regards

Marcel

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Hi Henryk,

I tested again all example files before you moved the bits array calculation into an own function. All of them sounded good.

With the latest code you still overwrite the stack at some point, because the pcm_sample[][] array is only 2x128 and you return 2x256 samples from sbc_synthesize_audio() function.

I also started to write a simple SBC play program using the libao. It works actually quite good linking your sbc.o file, but unfortunately with 2 channel files the left channel is silent.

We should start a new thread and discuss a SBC library.

Regards

Marcel

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Moin,

Am Mon, 22 Nov 2004 04:18:35 +0100 schrieb Marcel Holtmann:

So did I, this time. :-)

Ahh, thanks. I broke it when I moved the superfluous (I thought) assignment of current_sample=0 out of the channel for loop. Should work now again.

If that was a result of the abovementioned bug it should work now again.

Sure. I should have the bitstream packer working sometime soon and will then post again.

Henryk Plötz
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Hi Henryk,

yes. Everything works now like it should.

I re-arranged some stuff in the CVS again and the sbc directory is now also using autoconf/automake. However the sbc.h is now sbc_internal.h and the new sbc.h is the header file of the new SBC library. The sbcdec program is using libao to play the SBC file on your soundcard. However the libao library can also write the data to a file in AU or WAV format, but this is not implemented at the moment.

Can't wait to write the sbcenc program. Keep on with the good work.

And we need to add a clear license statement to the source files. I like to put the SBC library under LGPL and the programs under GPL. Is this fine with you?

Regards

Marcel

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Moin,

Am Mon, 22 Nov 2004 14:28:22 +0100 schrieb Marcel Holtmann:

Absolutely, that's what I wanted to do anyways.

Henryk Plötz
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Hi Henryk,

I added the license statement to all files now.

Regards

Marcel

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Henryk,

I didn't reindent your sbc stuff because I've been seeing you update there recently.

FYI, I just ran /usr/src/linux/scripts/Lindent on the other files, but I'll let you do your stuff so we don't have conflicts.

Brad

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you can acces to assigned numbers with this web page
http://www.bluetooth.org/assigned-numbers/
(I think that is what you are looking for)

Regards
Sebastien

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