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https://github.com/FEX-Emu/linux.git
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[media] em28xx-audio: don't hardcode audio URB calculus
The current code hardcodes the number of audio URBs, the number of packets per URB and the maximum URB size. This is not a good idea, as it: - wastes more bandwidth than necessary, by using a very large number of packets; - those constants are bound to an specific scenario, with a bandwidth of 48 kHz; - don't take the maximum endpoint size into account; - with urb->interval = 1 on xHCI, those constraints cause a "funny" setup: URBs with 64 packets inside, with only 24 bytes total. E. g. a complete waste of space. Change the code to do dynamic URB audio calculus and allocation. For now, use the same constraints as used before this patch, to avoid regressions. A good scenario (tested) seems to use those defines, instead: #define EM28XX_MAX_AUDIO_BUFS 8 #define EM28XX_MIN_AUDIO_PACKETS 2 But let's not do such change here, letting the optimization to happen on latter patches, after more tests. Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
This commit is contained in:
Mauro Carvalho Chehab
parent
439c491c4c
commit
1b3fd2d342
@ -50,6 +50,8 @@ static int debug;
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module_param(debug, int, 0644);
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MODULE_PARM_DESC(debug, "activates debug info");
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#define EM28XX_MAX_AUDIO_BUFS 5
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#define EM28XX_MIN_AUDIO_PACKETS 64
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#define dprintk(fmt, arg...) do { \
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if (debug) \
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printk(KERN_INFO "em28xx-audio %s: " fmt, \
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@ -63,7 +65,7 @@ static int em28xx_deinit_isoc_audio(struct em28xx *dev)
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int i;
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dprintk("Stopping isoc\n");
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for (i = 0; i < EM28XX_AUDIO_BUFS; i++) {
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for (i = 0; i < dev->adev.num_urb; i++) {
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struct urb *urb = dev->adev.urb[i];
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if (!irqs_disabled())
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@ -168,7 +170,7 @@ static int em28xx_init_audio_isoc(struct em28xx *dev)
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dprintk("Starting isoc transfers\n");
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/* Start streaming */
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for (i = 0; i < EM28XX_AUDIO_BUFS; i++) {
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for (i = 0; i < dev->adev.num_urb; i++) {
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memset(dev->adev.transfer_buffer[i], 0x80,
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dev->adev.urb[i]->transfer_buffer_length);
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@ -598,21 +600,35 @@ static void em28xx_audio_free_urb(struct em28xx *dev)
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{
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int i;
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for (i = 0; i < EM28XX_AUDIO_BUFS; i++) {
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for (i = 0; i < dev->adev.num_urb; i++) {
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struct urb *urb = dev->adev.urb[i];
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if (!dev->adev.urb[i])
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if (!urb)
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continue;
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usb_free_coherent(dev->udev,
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urb->transfer_buffer_length,
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dev->adev.transfer_buffer[i],
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urb->transfer_dma);
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if (dev->adev.transfer_buffer[i])
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usb_free_coherent(dev->udev,
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urb->transfer_buffer_length,
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dev->adev.transfer_buffer[i],
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urb->transfer_dma);
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usb_free_urb(urb);
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dev->adev.urb[i] = NULL;
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dev->adev.transfer_buffer[i] = NULL;
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}
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kfree(dev->adev.urb);
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kfree(dev->adev.transfer_buffer);
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dev->adev.num_urb = 0;
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}
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/* high bandwidth multiplier, as encoded in highspeed endpoint descriptors */
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static int em28xx_audio_ep_packet_size(struct usb_device *udev,
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struct usb_endpoint_descriptor *e)
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{
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int size = le16_to_cpu(e->wMaxPacketSize);
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if (udev->speed == USB_SPEED_HIGH)
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return (size & 0x7ff) * (1 + (((size) >> 11) & 0x03));
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return size & 0x7ff;
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}
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static int em28xx_audio_init(struct em28xx *dev)
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@ -623,9 +639,8 @@ static int em28xx_audio_init(struct em28xx *dev)
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struct usb_interface *intf;
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struct usb_endpoint_descriptor *e, *ep = NULL;
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static int devnr;
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int err, i;
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const int sb_size = EM28XX_NUM_AUDIO_PACKETS *
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EM28XX_AUDIO_MAX_PACKET_SIZE;
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int err, i, ep_size, interval, num_urb, npackets;
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int urb_size, bytes_per_transfer;
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u8 alt;
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if (!dev->has_alsa_audio || dev->audio_ifnum < 0) {
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@ -648,6 +663,9 @@ static int em28xx_audio_init(struct em28xx *dev)
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return err;
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spin_lock_init(&adev->slock);
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adev->sndcard = card;
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adev->udev = dev->udev;
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err = snd_pcm_new(card, "Em28xx Audio", 0, 0, 1, &pcm);
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if (err < 0) {
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snd_card_free(card);
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@ -712,25 +730,92 @@ static int em28xx_audio_init(struct em28xx *dev)
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return -ENODEV;
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}
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/* Alloc URB and transfer buffers */
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for (i = 0; i < EM28XX_AUDIO_BUFS; i++) {
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ep_size = em28xx_audio_ep_packet_size(dev->udev, ep);
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interval = 1 << (ep->bInterval - 1);
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em28xx_info("Endpoint 0x%02x %s on intf %d alt %d interval = %d, size %d\n",
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EM28XX_EP_AUDIO, usb_speed_string(dev->udev->speed),
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dev->audio_ifnum, alt,
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interval,
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ep_size);
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/* Calculate the number and size of URBs to better fit the audio samples */
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/*
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* Estimate the number of bytes per DMA transfer.
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*
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* This is given by the bit rate (for now, only 48000 Hz) multiplied
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* by 2 channels and 2 bytes/sample divided by the number of microframe
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* intervals and by the microframe rate (125 us)
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*/
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bytes_per_transfer = DIV_ROUND_UP(48000 * 2 * 2, 125 * interval);
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/*
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* Estimate the number of transfer URBs. Don't let it go past the
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* maximum number of URBs that is known to be supported by the device.
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*/
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num_urb = DIV_ROUND_UP(bytes_per_transfer, ep_size);
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if (num_urb > EM28XX_MAX_AUDIO_BUFS)
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num_urb = EM28XX_MAX_AUDIO_BUFS;
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/*
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* Now that we know the number of bytes per transfer and the number of
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* URBs, estimate the typical size of an URB, in order to adjust the
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* minimal number of packets.
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*/
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urb_size = bytes_per_transfer / num_urb;
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/*
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* Now, calculate the amount of audio packets to be filled on each
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* URB. In order to preserve the old behaviour, use a minimal
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* threshold for this value.
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*/
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npackets = EM28XX_MIN_AUDIO_PACKETS;
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if (urb_size > ep_size * npackets)
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npackets = DIV_ROUND_UP(urb_size, ep_size);
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em28xx_info("Number of URBs: %d, with %d packets and %d size",
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num_urb, npackets, urb_size);
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/* Allocate space to store the number of URBs to be used */
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dev->adev.transfer_buffer = kcalloc(num_urb,
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sizeof(*dev->adev.transfer_buffer),
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GFP_ATOMIC);
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if (!dev->adev.transfer_buffer) {
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snd_card_free(card);
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return -ENOMEM;
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}
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dev->adev.urb = kcalloc(num_urb, sizeof(*dev->adev.urb), GFP_ATOMIC);
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if (!dev->adev.urb) {
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snd_card_free(card);
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kfree(dev->adev.transfer_buffer);
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return -ENOMEM;
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}
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/* Alloc memory for each URB and for each transfer buffer */
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dev->adev.num_urb = num_urb;
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for (i = 0; i < num_urb; i++) {
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struct urb *urb;
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int j, k;
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void *buf;
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urb = usb_alloc_urb(EM28XX_NUM_AUDIO_PACKETS, GFP_ATOMIC);
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urb = usb_alloc_urb(npackets, GFP_ATOMIC);
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if (!urb) {
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em28xx_errdev("usb_alloc_urb failed!\n");
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em28xx_audio_free_urb(dev);
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snd_card_free(card);
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return -ENOMEM;
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}
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dev->adev.urb[i] = urb;
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buf = usb_alloc_coherent(dev->udev, sb_size, GFP_ATOMIC,
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buf = usb_alloc_coherent(dev->udev, npackets * ep_size, GFP_ATOMIC,
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&urb->transfer_dma);
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if (!buf) {
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em28xx_errdev("usb_alloc_coherent failed!\n");
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em28xx_audio_free_urb(dev);
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snd_card_free(card);
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return -ENOMEM;
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}
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dev->adev.transfer_buffer[i] = buf;
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@ -739,23 +824,15 @@ static int em28xx_audio_init(struct em28xx *dev)
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urb->context = dev;
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urb->pipe = usb_rcvisocpipe(dev->udev, EM28XX_EP_AUDIO);
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urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
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urb->transfer_buffer = dev->adev.transfer_buffer[i];
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urb->interval = 1 << (ep->bInterval - 1);
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urb->transfer_buffer = buf;
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urb->interval = interval;
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urb->complete = em28xx_audio_isocirq;
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urb->number_of_packets = EM28XX_NUM_AUDIO_PACKETS;
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urb->transfer_buffer_length = sb_size;
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urb->number_of_packets = npackets;
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urb->transfer_buffer_length = ep_size * npackets;
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if (!i)
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dprintk("Will use ep 0x%02x on intf %d alt %d interval = %d (rcv isoc pipe: 0x%08x)\n",
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EM28XX_EP_AUDIO, dev->audio_ifnum, alt,
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urb->interval,
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urb->pipe);
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for (j = k = 0; j < EM28XX_NUM_AUDIO_PACKETS;
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j++, k += EM28XX_AUDIO_MAX_PACKET_SIZE) {
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for (j = k = 0; j < npackets; j++, k += ep_size) {
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urb->iso_frame_desc[j].offset = k;
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urb->iso_frame_desc[j].length =
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EM28XX_AUDIO_MAX_PACKET_SIZE;
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urb->iso_frame_desc[j].length = ep_size;
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}
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}
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@ -765,8 +842,6 @@ static int em28xx_audio_init(struct em28xx *dev)
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snd_card_free(card);
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return err;
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}
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adev->sndcard = card;
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adev->udev = dev->udev;
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em28xx_info("Audio extension successfully initialized\n");
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return 0;
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@ -481,9 +481,6 @@ struct em28xx_eeprom {
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u8 string_idx_table;
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};
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#define EM28XX_AUDIO_BUFS 5
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#define EM28XX_NUM_AUDIO_PACKETS 64
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#define EM28XX_AUDIO_MAX_PACKET_SIZE 196 /* static value */
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#define EM28XX_CAPTURE_STREAM_EN 1
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/* em28xx extensions */
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@ -498,8 +495,9 @@ struct em28xx_eeprom {
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struct em28xx_audio {
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char name[50];
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char *transfer_buffer[EM28XX_AUDIO_BUFS];
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struct urb *urb[EM28XX_AUDIO_BUFS];
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unsigned num_urb;
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char **transfer_buffer;
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struct urb **urb;
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struct usb_device *udev;
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unsigned int capture_transfer_done;
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struct snd_pcm_substream *capture_pcm_substream;
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