From 60902a2cb12c3c1682ee7a04ad7448ec16dc0c29 Mon Sep 17 00:00:00 2001 From: Sudhakar Rajashekhara Date: Thu, 21 May 2009 07:41:35 -0400 Subject: davinci: EDMA: multiple CCs, channel mapping and API changes - restructure to support multiple channel controllers by using additional struct resources for each CC - interface changes visible to EDMA clients Introduce macros to build IDs from controller and channel number, and to extract them. Modify the edma_alloc_slot function to take an extra argument for the controller. Also update ASoC drivers to use API. ASoC changes Acked-by: Mark Brown - Move queue related mappings to dm.c EDMA in DM355 and DM644x has two transfer controllers while DM646x has four transfer controllers. Moving the queue to tc mapping and queue priority mapping to dm.c will be helpful to probe these mappings from platform device so that the machine_is_* testing will be avoided. - add channel mapping logic Channel mapping logic is introduced in dm646x EDMA. This implies that there is no fixed association for a channel number to a parameter entry number. In other words, using the DMA channel mapping registers (DCHMAPn), a PaRAM entry can be mapped to any channel. While in the case of dm644x and dm355 there is a fixed mapping between the EDMA channel and Param entry number. Signed-off-by: Naresh Medisetty Signed-off-by: Sudhakar Rajashekhara Reviewed-by: David Brownell Signed-off-by: Kevin Hilman --- sound/soc/davinci/davinci-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a05996588489..3ee38b62766d 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -143,7 +143,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) prtd->master_lch = ret; /* Request parameter RAM reload slot */ - ret = edma_alloc_slot(EDMA_SLOT_ANY); + ret = edma_alloc_slot(EDMA_CTLR(prtd->master_lch), EDMA_SLOT_ANY); if (ret < 0) { edma_free_channel(prtd->master_lch); return ret; @@ -160,8 +160,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ edma_read_slot(prtd->slave_lch, &p_ram); - p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch); - p_ram.link_bcntrld = prtd->slave_lch << 5; + p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->master_lch)); + p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->slave_lch) << 5; edma_write_slot(prtd->slave_lch, &p_ram); return 0; -- cgit v1.2.1 From 79dfc9687661c13ef95eb4c2226f3db4ccab52c9 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Wed, 16 Sep 2009 20:25:08 -0400 Subject: ASoC: Blackfin AC97: add a few missing multichannel define handling Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was merged, but two places were missed (the probe/resume functions). Restore handling of this option so it gets initialized properly. Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 2758b9017a7f..e69322978739 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -277,7 +277,11 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -334,7 +338,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, goto sport_err; } /*SPORT works in TDM mode to simulate AC97 transfers*/ +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; -- cgit v1.2.1 From d75150d7c49db42021b8f966d2cbdc215a530208 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Wed, 16 Sep 2009 20:25:09 -0400 Subject: ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-sport.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 469ce7fab20c..99051ff0954e 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -326,7 +326,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport) int sport_tx_start(struct sport_device *sport) { - unsigned flags; + unsigned long flags; pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__, sport->tx_run, sport->rx_run); if (sport->tx_run) -- cgit v1.2.1 From 7d156a25bd3e8e6ff74faf02faecb5fc5fb4839e Mon Sep 17 00:00:00 2001 From: Barry Song Date: Wed, 16 Sep 2009 20:25:10 -0400 Subject: ASoC: fix typos in Blackfin headers Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.h | 2 +- sound/soc/blackfin/bf5xx-i2s.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 3f2a911fe0cb..a1f97dd809d6 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-ac97.h + * sound/soc/blackfin/bf5xx-ac97.h * * 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 diff --git a/sound/soc/blackfin/bf5xx-i2s.h b/sound/soc/blackfin/bf5xx-i2s.h index 7107d1a0b06b..264ecdcba35a 100644 --- a/sound/soc/blackfin/bf5xx-i2s.h +++ b/sound/soc/blackfin/bf5xx-i2s.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-i2s.h + * sound/soc/blackfin/bf5xx-i2s.h * * 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 -- cgit v1.2.1 From fab19bae0c2951ed8bc517a53848b027fead293d Mon Sep 17 00:00:00 2001 From: Barry Song Date: Wed, 16 Sep 2009 20:25:11 -0400 Subject: ASoC: Blackfin I2S: add lost platform_device parameter to resume function Commit dc7d7b830ee1 trimmed the platform_device parameter from all of the suspend functions, but it also accidentally removed it from the resume function in the Blackfin I2S driver. So restore it. Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 876abade27e1..19539c68c053 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -227,7 +227,8 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } -static void bf5xx_i2s_remove(struct snd_soc_dai *dai) +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); peripheral_free_list(&sport_req[sport_num][0]); -- cgit v1.2.1 From ad80efc469f56d41f3f4adc1b2c86bf65689ebeb Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Wed, 16 Sep 2009 20:25:12 -0400 Subject: ASoC: Blackfin I2S: fix resuming when device hasn't been used If the sound system hasn't been utilized yet and we suspend, then we attempt to save/restore using state that doesn't exist. So use a global handle instead to reconfigure properly. Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 19539c68c053..1e9d161c76c4 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -237,36 +237,31 @@ static void bf5xx_i2s_remove(struct platform_device *pdev, #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; + if (dai->capture.active) - sport_rx_stop(sport); + sport_rx_stop(sport_handle); if (dai->playback.active) - sport_tx_stop(sport); + sport_tx_stop(sport_handle); return 0; } static int bf5xx_i2s_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; - ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; -- cgit v1.2.1 From 0c31cf3e4af79ea18bbd365b07ef0de207673894 Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Tue, 15 Sep 2009 18:13:29 -0400 Subject: ASoC: DaVinci: Fixes to McASP configuration McASP register settings are not correct for DSP mode of operation. There is a channel swap initally. This patch provides fixes to the register values for proper working. Tested on DA830/OMAP-L137 EVM, DM6467 EVM. Signed-off-by: Chaithrika U S Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index eca22d7829d2..7a06c0a86665 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -512,34 +512,49 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int channel_size) { u32 fmt = 0; + u32 mask, rotate; switch (channel_size) { case DAVINCI_AUDIO_WORD_8: fmt = 0x03; + rotate = 6; + mask = 0x000000ff; break; case DAVINCI_AUDIO_WORD_12: fmt = 0x05; + rotate = 5; + mask = 0x00000fff; break; case DAVINCI_AUDIO_WORD_16: fmt = 0x07; + rotate = 4; + mask = 0x0000ffff; break; case DAVINCI_AUDIO_WORD_20: fmt = 0x09; + rotate = 3; + mask = 0x000fffff; break; case DAVINCI_AUDIO_WORD_24: fmt = 0x0B; + rotate = 2; + mask = 0x00ffffff; break; case DAVINCI_AUDIO_WORD_28: fmt = 0x0D; + rotate = 1; + mask = 0x0fffffff; break; case DAVINCI_AUDIO_WORD_32: fmt = 0x0F; + rotate = 0; + mask = 0xffffffff; break; default: @@ -550,6 +565,13 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, 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; } @@ -638,7 +660,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "playback tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); } else { /* bit stream is MSB first with no delay */ @@ -655,7 +676,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "capture tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); } } -- cgit v1.2.1 From e454cea20bdcff10ee698d11b8882662a0153a47 Mon Sep 17 00:00:00 2001 From: Kay Sievers Date: Fri, 18 Sep 2009 23:01:12 +0200 Subject: Driver-Core: extend devnode callbacks to provide permissions This allows subsytems to provide devtmpfs with non-default permissions for the device node. Instead of the default mode of 0600, null, zero, random, urandom, full, tty, ptmx now have a mode of 0666, which allows non-privileged processes to access standard device nodes in case no other userspace process applies the expected permissions. This also fixes a wrong assignment in pktcdvd and a checkpatch.pl complain. Signed-off-by: Kay Sievers Signed-off-by: Greg Kroah-Hartman --- sound/sound_core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/sound_core.c b/sound/sound_core.c index bb4b88e606bb..49c998186592 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -29,7 +29,7 @@ MODULE_DESCRIPTION("Core sound module"); MODULE_AUTHOR("Alan Cox"); MODULE_LICENSE("GPL"); -static char *sound_nodename(struct device *dev) +static char *sound_devnode(struct device *dev, mode_t *mode) { if (MAJOR(dev->devt) == SOUND_MAJOR) return NULL; @@ -50,7 +50,7 @@ static int __init init_soundcore(void) return PTR_ERR(sound_class); } - sound_class->nodename = sound_nodename; + sound_class->devnode = sound_devnode; return 0; } -- cgit v1.2.1 From d3a7dcfeeb6856215412065bd0f774de83f43414 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Sep 2009 18:46:26 +0200 Subject: ALSA: pcm - Simplify snd_pcm_drain() implementation Simplify snd_pcm_drain() implementation and avoid unneeded array- allocation for waitqueues. Instead, one waitqueue is used for the first draining stream, and wait until all streams finished. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 73 ++++++++++++++----------------------------------- 1 file changed, 20 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 59e5fbe6af51..561d6d95a2d3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1387,11 +1387,6 @@ static struct action_ops snd_pcm_action_drain_init = { .post_action = snd_pcm_post_drain_init }; -struct drain_rec { - struct snd_pcm_substream *substream; - wait_queue_t wait; -}; - static int snd_pcm_drop(struct snd_pcm_substream *substream); /* @@ -1407,10 +1402,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, struct snd_card *card; struct snd_pcm_runtime *runtime; struct snd_pcm_substream *s; + wait_queue_t wait; int result = 0; - int i, num_drecs; int nonblock = 0; - struct drain_rec *drec, drec_tmp, *d; card = substream->pcm->card; runtime = substream->runtime; @@ -1433,38 +1427,10 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } else if (substream->f_flags & O_NONBLOCK) nonblock = 1; - if (nonblock) - goto lock; /* no need to allocate waitqueues */ - - /* allocate temporary record for drain sync */ down_read(&snd_pcm_link_rwsem); - if (snd_pcm_stream_linked(substream)) { - drec = kmalloc(substream->group->count * sizeof(*drec), GFP_KERNEL); - if (! drec) { - up_read(&snd_pcm_link_rwsem); - snd_power_unlock(card); - return -ENOMEM; - } - } else - drec = &drec_tmp; - - /* count only playback streams */ - num_drecs = 0; - snd_pcm_group_for_each_entry(s, substream) { - runtime = s->runtime; - if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { - d = &drec[num_drecs++]; - d->substream = s; - init_waitqueue_entry(&d->wait, current); - add_wait_queue(&runtime->sleep, &d->wait); - } - } - up_read(&snd_pcm_link_rwsem); - - lock: snd_pcm_stream_lock_irq(substream); /* resume pause */ - if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED) + if (runtime->status->state == SNDRV_PCM_STATE_PAUSED) snd_pcm_pause(substream, 0); /* pre-start/stop - all running streams are changed to DRAINING state */ @@ -1479,25 +1445,35 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, for (;;) { long tout; + struct snd_pcm_runtime *to_check; if (signal_pending(current)) { result = -ERESTARTSYS; break; } - /* all finished? */ - for (i = 0; i < num_drecs; i++) { - runtime = drec[i].substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) + /* find a substream to drain */ + to_check = NULL; + snd_pcm_group_for_each_entry(s, substream) { + if (s->stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; + runtime = s->runtime; + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { + to_check = runtime; break; + } } - if (i == num_drecs) - break; /* yes, all drained */ - + if (!to_check) + break; /* all drained */ + init_waitqueue_entry(&wait, current); + add_wait_queue(&to_check->sleep, &wait); set_current_state(TASK_INTERRUPTIBLE); snd_pcm_stream_unlock_irq(substream); + up_read(&snd_pcm_link_rwsem); snd_power_unlock(card); tout = schedule_timeout(10 * HZ); snd_power_lock(card); + down_read(&snd_pcm_link_rwsem); snd_pcm_stream_lock_irq(substream); + remove_wait_queue(&to_check->sleep, &wait); if (tout == 0) { if (substream->runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) result = -ESTRPIPE; @@ -1512,16 +1488,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, unlock: snd_pcm_stream_unlock_irq(substream); - - if (!nonblock) { - for (i = 0; i < num_drecs; i++) { - d = &drec[i]; - runtime = d->substream->runtime; - remove_wait_queue(&runtime->sleep, &d->wait); - } - if (drec != &drec_tmp) - kfree(drec); - } + up_read(&snd_pcm_link_rwsem); snd_power_unlock(card); return result; -- cgit v1.2.1 From 95eff499c96ac728307fd34cbf1a28ea20a5b615 Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Mon, 21 Sep 2009 15:00:22 +0200 Subject: ALSA: lx6464es - cleanup of rmh message bus function the rmh bus is not used asynchronously, so it is safe to remove the specific code pieces. Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx6464es.h | 1 - sound/pci/lx6464es/lx_core.c | 98 +------------------------------------------ 2 files changed, 1 insertion(+), 98 deletions(-) (limited to 'sound') diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h index 012c010c8c89..16442dad58d1 100644 --- a/sound/pci/lx6464es/lx6464es.h +++ b/sound/pci/lx6464es/lx6464es.h @@ -86,7 +86,6 @@ struct lx6464es { /* messaging */ spinlock_t msg_lock; /* message spinlock */ - atomic_t send_message_locked; struct lx_rmh rmh; /* configuration */ diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 5812780d6e89..3086b751da4a 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -314,98 +314,6 @@ static inline void lx_message_dump(struct lx_rmh *rmh) #define XILINX_POLL_NO_SLEEP 100 #define XILINX_POLL_ITERATIONS 150 -#if 0 /* not used now */ -static int lx_message_send(struct lx6464es *chip, struct lx_rmh *rmh) -{ - u32 reg = ED_DSP_TIMED_OUT; - int dwloop; - int answer_received; - - if (lx_dsp_reg_read(chip, eReg_CSM) & (Reg_CSM_MC | Reg_CSM_MR)) { - snd_printk(KERN_ERR LXP "PIOSendMessage eReg_CSM %x\n", reg); - return -EBUSY; - } - - /* write command */ - lx_dsp_reg_writebuf(chip, eReg_CRM1, rmh->cmd, rmh->cmd_len); - - snd_BUG_ON(atomic_read(&chip->send_message_locked) != 0); - atomic_set(&chip->send_message_locked, 1); - - /* MicoBlaze gogogo */ - lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC); - - /* wait for interrupt to answer */ - for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS; ++dwloop) { - answer_received = atomic_read(&chip->send_message_locked); - if (answer_received == 0) - break; - msleep(1); - } - - if (answer_received == 0) { - /* in Debug mode verify Reg_CSM_MR */ - snd_BUG_ON(!(lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR)); - - /* command finished, read status */ - if (rmh->dsp_stat == 0) - reg = lx_dsp_reg_read(chip, eReg_CRM1); - else - reg = 0; - } else { - int i; - snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! " - "Interrupts disabled?\n"); - - /* attente bit Reg_CSM_MR */ - for (i = 0; i != XILINX_POLL_ITERATIONS; i++) { - if ((lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR)) { - if (rmh->dsp_stat == 0) - reg = lx_dsp_reg_read(chip, eReg_CRM1); - else - reg = 0; - goto polling_successful; - } - - if (i > XILINX_POLL_NO_SLEEP) - msleep(1); - } - snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! " - "polling failed\n"); - -polling_successful: - atomic_set(&chip->send_message_locked, 0); - } - - if ((reg & ERROR_VALUE) == 0) { - /* read response */ - if (rmh->stat_len) { - snd_BUG_ON(rmh->stat_len >= (REG_CRM_NUMBER-1)); - - lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat, - rmh->stat_len); - } - } else - snd_printk(KERN_WARNING LXP "lx_message_send: error_value %x\n", - reg); - - /* clear Reg_CSM_MR */ - lx_dsp_reg_write(chip, eReg_CSM, 0); - - switch (reg) { - case ED_DSP_TIMED_OUT: - snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n"); - return -ETIMEDOUT; - - case ED_DSP_CRASHED: - snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n"); - return -EAGAIN; - } - - lx_message_dump(rmh); - return 0; -} -#endif /* not used now */ static int lx_message_send_atomic(struct lx6464es *chip, struct lx_rmh *rmh) { @@ -423,7 +331,7 @@ static int lx_message_send_atomic(struct lx6464es *chip, struct lx_rmh *rmh) /* MicoBlaze gogogo */ lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC); - /* wait for interrupt to answer */ + /* wait for device to answer */ for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS * 1000; ++dwloop) { if (lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR) { if (rmh->dsp_stat == 0) @@ -1175,10 +1083,6 @@ static int lx_interrupt_ack(struct lx6464es *chip, u32 *r_irqsrc, *r_async_escmd = 1; } - if (irqsrc & MASK_SYS_STATUS_CMD_DONE) - /* xilinx command notification */ - atomic_set(&chip->send_message_locked, 0); - if (irq_async) { /* snd_printd("interrupt: async event pending\n"); */ *r_async_pending = 1; -- cgit v1.2.1 From 8fdc9e870c07054bc34a62a4655cdb078953255d Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Mon, 21 Sep 2009 15:00:57 +0200 Subject: ALSA: lx6464es - remove unused struct member we cannot set the sampling rate of the device, but can only read it from the board, so we don't need the member for it. Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx6464es.h | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h index 16442dad58d1..51afc048961d 100644 --- a/sound/pci/lx6464es/lx6464es.h +++ b/sound/pci/lx6464es/lx6464es.h @@ -94,7 +94,6 @@ struct lx6464es { uint hardware_running[2]; u32 board_sample_rate; /* sample rate read from * board */ - u32 sample_rate; /* our sample rate */ u16 pcm_granularity; /* board blocksize */ /* dma */ -- cgit v1.2.1 From 786d8ca341a30296264bc6cebac52d37b0851647 Mon Sep 17 00:00:00 2001 From: "Robert P. J. Day" Date: Sat, 4 Jul 2009 20:44:04 -0400 Subject: trivial: Remove commented out usage of dead MODULE_PARM() in swarm_cs4297a Get rid of that commented usage of the now defunct MODULE_PARM macro. Signed-off-by: Robert P. J. Day Signed-off-by: Jiri Kosina --- sound/oss/swarm_cs4297a.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 1edab7b4ea83..3136c88eacdf 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -110,9 +110,6 @@ static void start_adc(struct cs4297a_state *s); // rather than 64k as some of the games work more responsively. // log base 2( buff sz = 32k). -//static unsigned long defaultorder = 3; -//MODULE_PARM(defaultorder, "i"); - // // Turn on/off debugging compilation by commenting out "#define CSDEBUG" // -- cgit v1.2.1 From a419aef8b858a2bdb98df60336063d28df4b272f Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Tue, 18 Aug 2009 11:18:35 -0700 Subject: trivial: remove unnecessary semicolons Signed-off-by: Joe Perches Signed-off-by: Jiri Kosina --- sound/oss/sys_timer.c | 3 --- sound/soc/codecs/wm9081.c | 2 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- 4 files changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c index 107534477a2f..8db6aefe15e4 100644 --- a/sound/oss/sys_timer.c +++ b/sound/oss/sys_timer.c @@ -100,9 +100,6 @@ def_tmr_open(int dev, int mode) curr_tempo = 60; curr_timebase = 100; opened = 1; - - ; - { def_tmr.expires = (1) + jiffies; add_timer(&def_tmr); diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index c64e55aa63b6..686e5aa97206 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1027,7 +1027,7 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, - wm9081->fs); for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) { cur_val = abs((wm9081->sysclk_rate / - clk_sys_rates[i].ratio) - wm9081->fs);; + clk_sys_rates[i].ratio) - wm9081->fs); if (cur_val < best_val) { best = i; best_val = cur_val; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 5b9ed6464789..d11a6d7e384a 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -351,7 +351,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, do_div(tmp, freq_out); val = tmp; - val = (val << 16) | 64;; + val = (val << 16) | 64; ssp_write_reg(ssp, SSACDD, val); ssacd |= (0x6 << 4); diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index 8e79a416db57..c215d32d6322 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -67,7 +67,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) { int ret = 0; #ifdef ENFORCE_RATES - struct snd_pcm_runtime *runtime = substream->runtime;; + struct snd_pcm_runtime *runtime = substream->runtime; #endif mutex_lock(&clk_lock); -- cgit v1.2.1 From 98235a4bb01f288c82f7b878996e507f6159fd69 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Mon, 21 Sep 2009 11:20:51 +0800 Subject: ASoC: some minor changes for AD1836 and AD1938 codec drivers 1. delete redundant assignment to bus field in spi_driver structure 2. fix lost assignment to set_bias_level entry in ad1938 codec dai 3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836" Signed-off-by: Barry Song Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 3 +-- sound/soc/codecs/ad1938.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 01343dc984fd..c48485f2c55d 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -251,8 +251,7 @@ static int __devexit ad1836_spi_remove(struct spi_device *spi) static struct spi_driver ad1836_spi_driver = { .driver = { - .name = "ad1836-spi", - .bus = &spi_bus_type, + .name = "ad1836", .owner = THIS_MODULE, }, .probe = ad1836_spi_probe, diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 9a049a1995a3..34b30efc3cb0 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -456,7 +456,6 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi) static struct spi_driver ad1938_spi_driver = { .driver = { .name = "ad1938", - .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = ad1938_spi_probe, @@ -515,6 +514,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->num_dai = 1; codec->write = ad1938_write_reg; codec->read = ad1938_read_reg_cache; + codec->set_bias_level = ad1938_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); -- cgit v1.2.1 From 877ae70763fa0b19ffbe25e7e119fc96f52fbbd3 Mon Sep 17 00:00:00 2001 From: Phil Vandry Date: Mon, 21 Sep 2009 11:36:08 -0400 Subject: ASoC: wm8753: fix mapping when MONOMIX is set to Stereo When MONOMIX is set to Stereo, Left PGA was not powered on but should be. Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue. Signed-off-by: Phil Vandry Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d80d414cfbbd..5ad677ce80da 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -595,6 +595,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Mono Capture mixer-mux */ {"Capture Right Mixer", "Stereo", "Capture Right Mux"}, + {"Capture Left Mixer", "Stereo", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"}, {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"}, -- cgit v1.2.1