March 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [patch] [ALSA] seq_oss_event: missing range checks
by Dan Carpenter 04 Mar '13

04 Mar '13

The "dev" variable could be out of bounds. Calling snd_seq_oss_synth_is_valid() checks that it is is a valid device which has been opened. We check this inside set_note_event() so this function can't succeed without a valid "dev". But we need to do the check earlier to prevent invalid dereferences and memory corruption. One call tree where "dev" could be out of bounds is: -> snd_seq_oss_oob_user() -> snd_seq_oss_process_event() -> extended_event() -> note_on_event() Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c index 066f5f3..c390886 100644 --- a/sound/core/seq/oss/seq_oss_event.c +++ b/sound/core/seq/oss/seq_oss_event.c @@ -285,7 +285,12 @@ local_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev static int note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) { @@ -340,7 +345,12 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st static int note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) {

2 1

[alsa-devel] [PATCH v4 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- v4: change subjuct to include "ALSA: ASoC: " sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

3 5

[alsa-devel] [PATCH v3] davinci-mcasp: don't overwrite DIT settings
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- v3: don't set mask in davinci_hw_dit_param() v2: fix checkpatch.pl warnings sound/soc/davinci/davinci-mcasp.c | 25 +++++++++++++------------ 1 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 90a9b99..dcd56f2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -643,16 +643,20 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); + if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(rotate), TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXROT(rotate), RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, + mask); + } + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); return 0; } @@ -772,9 +776,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* TXMASK for 24 bits */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); - /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, -- 1.7.7

3 3

[alsa-devel] Issue with Dmix
by mateen 04 Mar '13

04 Mar '13

HI We have enabled DMIX plugin for sample rate conversion and mixing of 2 pcm streams in our alsa package (Alsa lib 1.0.22 and alsa driver 1.0.22.1) to 48K output Sample rate.However we are seeing a noise issue with 44100 Sample rate input streams or Unsigned Stereo 48K stream. This is how our current design is: We call snd_pcm_lib_malloc_page() in which alsa driver internally allocates a dma buffer and map it when Dmix is invoked. We currently reads from that mmaped buffer to our hardware buffer inside a transfer_ack_begin() function which is called from snd_pcm_period_elapsed(). However we are getting a problem in understanding how do we know how much data has been written by dmix lib in mapped buffer? As we see no ioctl or corresponding hardware function call where dmix may pass information on Size of data being written to our driver. We do see appl_ptr and hw_ptr in runtime->mmap->control and status struct. Out architecture doesn’t support defining and mapping of these status and control structure at hardware level.Thus Dmix lib allocates them and copy them to runtime substream via ioctl. As far as I could understand both appl_ptr and hw_ptr are initialised to 0 and then appl_ptr is the one adjusted by lib after every write. Should we tap this value to update hw_ptr value in status information? Will difference of hw_ptr and appl_ptr will let us know amount of frames been written? Can anyone please help us to understand criteria and concept of data copy from intermediate to hardware buffer? Any information on this would be of immense help. Regards, Shaikh Mateen

1 1

[alsa-devel] [PATCH] davinci-mcasp: don't overwrite DIT settings
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- sound/soc/davinci/davinci-mcasp.c | 27 +++++++++++++++++---------- 1 files changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 90a9b99..de59075 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -643,16 +643,23 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + } + else + { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), + TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), + RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + } return 0; } -- 1.7.7

3 2

[alsa-devel] [PATCH v2] davinci-mcasp: don't overwrite DIT settings
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- v2: fix checkpatch.pl warnings sound/soc/davinci/davinci-mcasp.c | 22 +++++++++++++--------- 1 files changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 90a9b99..ce32b75 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -643,16 +643,20 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); + if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(rotate), TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXROT(rotate), RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, + mask); + } + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); return 0; } -- 1.7.7

1 0

[alsa-devel] [PATCH 09/12] ASoC: dmaengine_pcm: support use of generic DMA helper
by Shawn Guo 04 Mar '13

04 Mar '13

With generic DMA device tree binding and helper function dma_request_slave_channel() in place, dmaengine_pcm should support that in requesting DMA channel for users that support generic DMA device tree binding. Instead of inventing a new API, it defines the parameters needed by dma_request_slave_channel() into struct snd_dma_channel_params, interprets filter_data into snd_dma_channel_params, and calls the helper in case that dmaengine_pcm users pass in a NULL filter_fn. Then, dmaengine_pcm users can call snd_dmaengine_pcm_open() with NULL filter_fn and snd_dma_channel_params being filter_data to direct the API to request DMA channel using generic DMA helper. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- include/sound/dmaengine_pcm.h | 5 +++++ sound/soc/soc-dmaengine-pcm.c | 19 +++++++++++++------ 2 files changed, 18 insertions(+), 6 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b877334..6c0f795 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -18,6 +18,11 @@ #include <sound/pcm.h> #include <linux/dmaengine.h> +struct snd_dma_channel_params { + struct device *dev; + char *name; +}; + /** * snd_pcm_substream_to_dma_direction - Get dma_transfer_direction for a PCM * substream diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 111b7d92..7ef8034 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -247,12 +247,19 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd, dma_filter_fn filter_fn, void *filter_data) { - dma_cap_mask_t mask; - - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - dma_cap_set(DMA_CYCLIC, mask); - prtd->dma_chan = dma_request_channel(mask, filter_fn, filter_data); + if (filter_fn) { + dma_cap_mask_t mask; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_CYCLIC, mask); + prtd->dma_chan = dma_request_channel(mask, filter_fn, + filter_data); + } else { + struct snd_dma_channel_params *params = filter_data; + prtd->dma_chan = dma_request_slave_channel(params->dev, + params->name); + } if (!prtd->dma_chan) return -ENXIO; -- 1.7.9.5

3 4

[alsa-devel] [PATCH] davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

2 1

[alsa-devel] [PATCH 2/2] ALSA: snd-usb-caiaq: switch to dev_*() logging
by Daniel Mack 03 Mar '13

03 Mar '13

Get rid of the proprietary functions log() and debug() and use the generic dev_*() approach. A macro is needed to cast a cdev to a struct device *. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/caiaq/audio.c | 61 ++++++++++++++++++++++++++++++----------------- sound/usb/caiaq/control.c | 1 + sound/usb/caiaq/device.c | 42 +++++++++++++++++--------------- sound/usb/caiaq/device.h | 10 +------- sound/usb/caiaq/input.c | 8 +++++-- sound/usb/caiaq/midi.c | 9 +++---- 6 files changed, 75 insertions(+), 56 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 75d8ba9..67330af 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/spinlock.h> #include <linux/slab.h> #include <linux/init.h> @@ -101,8 +102,9 @@ all_substreams_zero(struct snd_pcm_substream **subs) static int stream_start(struct snd_usb_caiaqdev *cdev) { int i, ret; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, cdev); + dev_dbg(dev, "%s(%p)\n", __func__, cdev); if (cdev->streaming) return -EINVAL; @@ -118,7 +120,8 @@ static int stream_start(struct snd_usb_caiaqdev *cdev) for (i = 0; i < N_URBS; i++) { ret = usb_submit_urb(cdev->data_urbs_in[i], GFP_ATOMIC); if (ret) { - log("unable to trigger read #%d! (ret %d)\n", i, ret); + dev_err(dev, "unable to trigger read #%d! (ret %d)\n", + i, ret); cdev->streaming = 0; return -EPIPE; } @@ -130,8 +133,9 @@ static int stream_start(struct snd_usb_caiaqdev *cdev) static void stream_stop(struct snd_usb_caiaqdev *cdev) { int i; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, cdev); + dev_dbg(dev, "%s(%p)\n", __func__, cdev); if (!cdev->streaming) return; @@ -150,17 +154,21 @@ static void stream_stop(struct snd_usb_caiaqdev *cdev) static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); - debug("%s(%p)\n", __func__, substream); + struct device *dev = caiaqdev_to_dev(cdev); + + dev_dbg(dev, "%s(%p)\n", __func__, substream); substream->runtime->hw = cdev->pcm_info; snd_pcm_limit_hw_rates(substream->runtime); + return 0; } static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, substream); + dev_dbg(dev, "%s(%p)\n", __func__, substream); if (all_substreams_zero(cdev->sub_playback) && all_substreams_zero(cdev->sub_capture)) { /* when the last client has stopped streaming, @@ -175,14 +183,12 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, struct snd_pcm_hw_params *hw_params) { - debug("%s(%p)\n", __func__, sub); return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); } static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); - debug("%s(%p)\n", __func__, sub); deactivate_substream(cdev, sub); return snd_pcm_lib_free_pages(sub); } @@ -201,8 +207,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) int index = substream->number; struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, substream); + dev_dbg(dev, "%s(%p)\n", __func__, substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { int out_pos; @@ -283,8 +290,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_trigger(struct snd_pcm_substream *sub, int cmd) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p) cmd %d\n", __func__, sub, cmd); + dev_dbg(dev, "%s(%p) cmd %d\n", __func__, sub, cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -443,6 +451,7 @@ static void read_in_urb_mode3(struct snd_usb_caiaqdev *cdev, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; + struct device *dev = caiaqdev_to_dev(cdev); int stream, i; /* paranoia check */ @@ -479,8 +488,8 @@ static void read_in_urb_mode3(struct snd_usb_caiaqdev *cdev, if (usb_buf[i] != ((stream << 1) | c) && !cdev->first_packet) { if (!cdev->input_panic) - printk(" EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", - ((stream << 1) | c), usb_buf[i], c, stream, i); + dev_warn(dev, " EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", + ((stream << 1) | c), usb_buf[i], c, stream, i); cdev->input_panic = 1; } @@ -497,6 +506,8 @@ static void read_in_urb(struct snd_usb_caiaqdev *cdev, const struct urb *urb, const struct usb_iso_packet_descriptor *iso) { + struct device *dev = caiaqdev_to_dev(cdev); + if (!cdev->streaming) return; @@ -516,7 +527,7 @@ static void read_in_urb(struct snd_usb_caiaqdev *cdev, } if ((cdev->input_panic || cdev->output_panic) && !cdev->warned) { - debug("streaming error detected %s %s\n", + dev_warn(dev, "streaming error detected %s %s\n", cdev->input_panic ? "(input)" : "", cdev->output_panic ? "(output)" : ""); cdev->warned = 1; @@ -619,6 +630,7 @@ static void read_completed(struct urb *urb) { struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *cdev; + struct device *dev; struct urb *out = NULL; int i, frame, len, send_it = 0, outframe = 0; size_t offset = 0; @@ -627,6 +639,7 @@ static void read_completed(struct urb *urb) return; cdev = info->cdev; + dev = caiaqdev_to_dev(cdev); if (!cdev->streaming) return; @@ -639,7 +652,7 @@ static void read_completed(struct urb *urb) } if (!out) { - log("Unable to find an output urb to use\n"); + dev_err(dev, "Unable to find an output urb to use\n"); goto requeue; } @@ -708,6 +721,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) int i, frame; struct urb **urbs; struct usb_device *usb_dev = cdev->chip.dev; + struct device *dev = caiaqdev_to_dev(cdev); unsigned int pipe; pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? @@ -716,7 +730,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) urbs = kmalloc(N_URBS * sizeof(*urbs), GFP_KERNEL); if (!urbs) { - log("unable to kmalloc() urbs, OOM!?\n"); + dev_err(dev, "unable to kmalloc() urbs, OOM!?\n"); *ret = -ENOMEM; return NULL; } @@ -724,7 +738,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) for (i = 0; i < N_URBS; i++) { urbs[i] = usb_alloc_urb(FRAMES_PER_URB, GFP_KERNEL); if (!urbs[i]) { - log("unable to usb_alloc_urb(), OOM!?\n"); + dev_err(dev, "unable to usb_alloc_urb(), OOM!?\n"); *ret = -ENOMEM; return urbs; } @@ -732,7 +746,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { - log("unable to kmalloc() transfer buffer, OOM!?\n"); + dev_err(dev, "unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } @@ -783,6 +797,7 @@ static void free_urbs(struct urb **urbs) int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) { int i, ret; + struct device *dev = caiaqdev_to_dev(cdev); cdev->n_audio_in = max(cdev->spec.num_analog_audio_in, cdev->spec.num_digital_audio_in) / @@ -792,12 +807,12 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) CHANNELS_PER_STREAM; cdev->n_streams = max(cdev->n_audio_in, cdev->n_audio_out); - debug("cdev->n_audio_in = %d\n", cdev->n_audio_in); - debug("cdev->n_audio_out = %d\n", cdev->n_audio_out); - debug("cdev->n_streams = %d\n", cdev->n_streams); + dev_dbg(dev, "cdev->n_audio_in = %d\n", cdev->n_audio_in); + dev_dbg(dev, "cdev->n_audio_out = %d\n", cdev->n_audio_out); + dev_dbg(dev, "cdev->n_streams = %d\n", cdev->n_streams); if (cdev->n_streams > MAX_STREAMS) { - log("unable to initialize device, too many streams.\n"); + dev_err(dev, "unable to initialize device, too many streams.\n"); return -EINVAL; } @@ -805,7 +820,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) cdev->n_audio_out, cdev->n_audio_in, &cdev->pcm); if (ret < 0) { - log("snd_pcm_new() returned %d\n", ret); + dev_err(dev, "snd_pcm_new() returned %d\n", ret); return ret; } @@ -880,7 +895,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *cdev) { - debug("%s(%p)\n", __func__, cdev); + struct device *dev = caiaqdev_to_dev(cdev); + + dev_dbg(dev, "%s(%p)\n", __func__, cdev); stream_stop(cdev); free_urbs(cdev->data_urbs_in); free_urbs(cdev->data_urbs_out); diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 2c51959..ae6b50f 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -17,6 +17,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/init.h> #include <linux/usb.h> #include <sound/control.h> diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 45c3853..d898f73 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -20,6 +20,7 @@ */ #include <linux/moduleparam.h> +#include <linux/device.h> #include <linux/interrupt.h> #include <linux/module.h> #include <linux/init.h> @@ -159,10 +160,11 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) { int ret; struct snd_usb_caiaqdev *cdev = urb->context; + struct device *dev = caiaqdev_to_dev(cdev); unsigned char *buf = urb->transfer_buffer; if (urb->status || !cdev) { - log("received EP1 urb->status = %i\n", urb->status); + dev_warn(dev, "received EP1 urb->status = %i\n", urb->status); return; } @@ -170,7 +172,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) case EP1_CMD_GET_DEVICE_INFO: memcpy(&cdev->spec, buf+1, sizeof(struct caiaq_device_spec)); cdev->spec.fw_version = le16_to_cpu(cdev->spec.fw_version); - debug("device spec (firmware %d): audio: %d in, %d out, " + dev_dbg(dev, "device spec (firmware %d): audio: %d in, %d out, " "MIDI: %d in, %d out, data alignment %d\n", cdev->spec.fw_version, cdev->spec.num_analog_audio_in, @@ -209,7 +211,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) cdev->ep1_in_urb.actual_length = 0; ret = usb_submit_urb(&cdev->ep1_in_urb, GFP_ATOMIC); if (ret < 0) - log("unable to submit urb. OOM!?\n"); + dev_err(dev, "unable to submit urb. OOM!?\n"); } int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, @@ -239,6 +241,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, { int ret; char tmp[5]; + struct device *dev = caiaqdev_to_dev(cdev); switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; @@ -259,7 +262,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, tmp[3] = bpp >> 8; tmp[4] = 1; /* packets per microframe */ - debug("setting audio params: %d Hz, %d bits, %d bpp\n", + dev_dbg(dev, "setting audio params: %d Hz, %d bits, %d bpp\n", rate, depth, bpp); cdev->audio_parm_answer = -1; @@ -274,7 +277,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, return -EPIPE; if (cdev->audio_parm_answer != 1) - debug("unable to set the device's audio params\n"); + dev_dbg(dev, "unable to set the device's audio params\n"); else cdev->bpp = bpp; @@ -293,6 +296,7 @@ static void setup_card(struct snd_usb_caiaqdev *cdev) { int ret; char val[4]; + struct device *dev = caiaqdev_to_dev(cdev); /* device-specific startup specials */ switch (cdev->chip.usb_id) { @@ -346,32 +350,32 @@ static void setup_card(struct snd_usb_caiaqdev *cdev) cdev->spec.num_digital_audio_in > 0) { ret = snd_usb_caiaq_audio_init(cdev); if (ret < 0) - log("Unable to set up audio system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up audio system (ret=%d)\n", ret); } if (cdev->spec.num_midi_in + cdev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(cdev); if (ret < 0) - log("Unable to set up MIDI system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up MIDI system (ret=%d)\n", ret); } #ifdef CONFIG_SND_USB_CAIAQ_INPUT ret = snd_usb_caiaq_input_init(cdev); if (ret < 0) - log("Unable to set up input system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up input system (ret=%d)\n", ret); #endif /* finally, register the card and all its sub-instances */ ret = snd_card_register(cdev->chip.card); if (ret < 0) { - log("snd_card_register() returned %d\n", ret); + dev_err(dev, "snd_card_register() returned %d\n", ret); snd_card_free(cdev->chip.card); } ret = snd_usb_caiaq_control_init(cdev); if (ret < 0) - log("Unable to set up control system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up control system (ret=%d)\n", ret); } static int create_card(struct usb_device *usb_dev, @@ -412,10 +416,11 @@ static int init_card(struct snd_usb_caiaqdev *cdev) char *c, usbpath[32]; struct usb_device *usb_dev = cdev->chip.dev; struct snd_card *card = cdev->chip.card; + struct device *dev = caiaqdev_to_dev(cdev); int err, len; if (usb_set_interface(usb_dev, 0, 1) != 0) { - log("can't set alt interface.\n"); + dev_err(dev, "can't set alt interface.\n"); return -EIO; } @@ -473,8 +478,7 @@ static int init_card(struct snd_usb_caiaqdev *cdev) } usb_make_path(usb_dev, usbpath, sizeof(usbpath)); - snprintf(card->longname, sizeof(card->longname), - "%s %s (%s)", + snprintf(card->longname, sizeof(card->longname), "%s %s (%s)", cdev->vendor_name, cdev->product_name, usbpath); setup_card(cdev); @@ -496,7 +500,7 @@ static int snd_probe(struct usb_interface *intf, usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { - log("unable to init card! (ret=%d)\n", ret); + dev_err(&usb_dev->dev, "unable to init card! (ret=%d)\n", ret); snd_card_free(card); return ret; } @@ -506,15 +510,16 @@ static int snd_probe(struct usb_interface *intf, static void snd_disconnect(struct usb_interface *intf) { - struct snd_usb_caiaqdev *cdev; struct snd_card *card = usb_get_intfdata(intf); - - debug("%s(%p)\n", __func__, intf); + struct snd_usb_caiaqdev *cdev = caiaqdev(card); + struct device *dev; if (!card) return; - cdev = caiaqdev(card); + dev = caiaqdev_to_dev(cdev); + dev_dbg(dev, "%s(%p)\n", __func__, intf); + snd_card_disconnect(card); #ifdef CONFIG_SND_USB_CAIAQ_INPUT @@ -539,4 +544,3 @@ static struct usb_driver snd_usb_driver = { }; module_usb_driver(snd_usb_driver); - diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index 7176a0e..ad102fa 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -25,16 +25,7 @@ #define CAIAQ_USB_STR_LEN 0xff #define MAX_STREAMS 32 -//#define SND_USB_CAIAQ_DEBUG - #define MODNAME "snd-usb-caiaq" -#define log(x...) snd_printk(KERN_WARNING MODNAME" log: " x) - -#ifdef SND_USB_CAIAQ_DEBUG -#define debug(x...) snd_printk(KERN_WARNING MODNAME " debug: " x) -#else -#define debug(x...) do { } while(0) -#endif #define EP1_CMD_GET_DEVICE_INFO 0x1 #define EP1_CMD_READ_ERP 0x2 @@ -129,6 +120,7 @@ struct snd_usb_caiaq_cb_info { }; #define caiaqdev(c) ((struct snd_usb_caiaqdev*)(c)->private_data) +#define caiaqdev_to_dev(d) (d->chip.card->dev) int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, int rate, int depth, int bbp); int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *cdev, int digital, int analog, int erp); diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index a32ad7c..fe8f4b4 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/gfp.h> #include <linux/init.h> #include <linux/usb.h> @@ -340,6 +341,8 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *cdev, const unsigned char *buf, unsigned int len) { + struct device *dev = caiaqdev_to_dev(cdev); + while (len) { unsigned int i, block_id = (buf[0] << 8) | buf[1]; @@ -466,7 +469,7 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *cdev, break; default: - debug("%s(): bogus block (id %d)\n", + dev_dbg(dev, "%s(): bogus block (id %d)\n", __func__, block_id); return; } @@ -500,6 +503,7 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) { struct snd_usb_caiaqdev *cdev = urb->context; + struct device *dev = caiaqdev_to_dev(cdev); unsigned char *buf = urb->transfer_buffer; int ret; @@ -535,7 +539,7 @@ requeue: cdev->ep4_in_urb->actual_length = 0; ret = usb_submit_urb(cdev->ep4_in_urb, GFP_ATOMIC); if (ret < 0) - log("unable to submit urb. OOM!?\n"); + dev_err(dev, "unable to submit urb. OOM!?\n"); } static int snd_usb_caiaq_input_open(struct input_dev *idev) diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 63c5a2c..2d75884 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/usb.h> #include <linux/gfp.h> #include <sound/rawmidi.h> @@ -65,6 +66,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *cdev, struct snd_rawmidi_substream *substream) { int len, ret; + struct device *dev = caiaqdev_to_dev(cdev); cdev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; cdev->midi_out_buf[1] = 0; /* port */ @@ -79,9 +81,9 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *cdev, ret = usb_submit_urb(&cdev->midi_out_urb, GFP_ATOMIC); if (ret < 0) - log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," - "ret=%d, len=%d\n", - substream, ret, len); + dev_err(dev, + "snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," + "ret=%d, len=%d\n", substream, ret, len); else cdev->midi_out_active = 1; } @@ -171,4 +173,3 @@ void snd_usb_caiaq_midi_output_done(struct urb* urb) snd_usb_caiaq_midi_send(cdev, cdev->midi_out_substream); } - -- 1.8.1.4

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[alsa-devel] usb: Endpoints sharing the same interface
by Mark Hills 03 Mar '13

03 Mar '13

Novation Twitch [1] is a USB device with audio capture and playback endpoints on the same bInterfaceNumber. This has thrown up some tricky cases in the USB audio driver, and I'd appreciate some advice. I'll follow up with my patches so far as an RFC. This device has endpoints for 2 channels capture (0x01), 4 channels playback (0x82), both on interface #0. Interface #1 is raw MIDI. I seem to be finding code that assumes pcm to always use the first endpoint on an interface. eg. problems with maxpacksize (see patch in thread) Now I'm in a position where playback _or_ capture works reliably. But to start capture stops playback, and vice-versa (or sometimes causes hangs) It seems like snd_pcm_prepare() concludes that need_setup_ep is required, but I think this may be based on something going on in set_sample_rate_v1() also looking to the first endpoint -- usb_sndctrlpipe(dev,0). Before I do any more work on this, are there any hints where to go with this? If we can overcome thse limitations then the same would likely apply to Saffire 6 USB and other Focusrite devices [2], and maybe some of the other devices with quirks that only support playback. Many thanks [1] http://uk.novationmusic.com/digital-dj/twitch/ [2] http://focusritedevelopmentteam.wordpress.com/category/drivers/ -- Mark --- Bus 005 Device 010: ID 1235:0018 Novation EMS Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1235 Novation EMS idProduct 0x0018 bcdDevice 1.00 iManufacturer 1 Focusrite A.E. Ltd iProduct 2 Twitch iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 64 bNumInterfaces 2 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 498mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x024c 1x 588 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 1 Device Status: 0x0000 (Bus Powered)

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