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

[alsa-devel] [PATCH] salsa-lib: Fix mistakes after refactoring pcm_params.c
by Mihail Zenkov 20 Mar '13

20 Mar '13

Fix some regressions after refactoring pcm_params.c These mistakes don't allow snd_pcm_hw_param_set_near to establish correct value. --- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/src/pcm_params.c +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/pcm_params.c @@ -590,11 +590,9 @@ save = *params; err = hw_param_set_max(params, var, *val, dir ? *dir : 0); - if (!err) { - err = hw_param_update_var(pcm, params, var, err); - if (!err) - return _snd_pcm_hw_param_get_max(params, var, val, dir); - } + err = hw_param_update_var(pcm, params, var, err); + if (!err) + return _snd_pcm_hw_param_get_max(params, var, val, dir); *params = save; return err; } @@ -683,7 +681,7 @@ snd_pcm_hw_params_t save = *params; int err = hw_param_set(params, var, val, dir); err = hw_param_update_var(pcm, params, var, err); - if (err) + if (err < 0) *params = save; return err; } @@ -712,7 +710,7 @@ save = *params; i->integer = 1; err = hw_param_update_var(pcm, params, var, 1); - if (err) + if (err < 0) *params = save; return err; } @@ -827,7 +825,7 @@ else if (valdir == 0) maxdir = -1; else { - valdir = 1; + maxdir = 1; max--; } save = *params;

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Re: [alsa-devel] [PATCH] sound: soc: fix invalid free of devm_ allocated data
by Mark Brown 20 Mar '13

20 Mar '13

On Sat, Mar 16, 2013 at 01:45:34PM +0200, Silviu-Mihai Popescu wrote: > The objects allocated by devm_* APIs are managed by devres and are freed when > the device is detached. Hence there is no need to use kfree() explicitly. ...but I applied this anyway since it's clearly an improvement and a small leak is better than a crash.

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Re: [alsa-devel] [PATCH] sound: soc: fix invalid free of devm_ allocated data
by Mark Brown 20 Mar '13

20 Mar '13

On Sat, Mar 16, 2013 at 01:45:34PM +0200, Silviu-Mihai Popescu wrote: > The objects allocated by devm_* APIs are managed by devres and are freed when > the device is detached. Hence there is no need to use kfree() explicitly. In this case since we're in error handling paths which might not be fatal (though I suspect they probably should be normally) it looks like the better fix is to use devm_kfree() to free immediately. Otherwise the unused memory will end up hanging around for as long as the device is bound.

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[alsa-devel] [PATCH 1/2] ASoC: spear_pcm: Update to new pcm_new() API
by Lars-Peter Clausen 20 Mar '13

20 Mar '13

Commit 552d1ef6 ("ASoC: core - Optimise and refactor pcm_new() to pass only rtd") updated the pcm_new() callback to take the rtd as the only parameter. The spear PCM driver (which was merged much later) still uses the old API. This patch updates the driver to the new API. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Btw. where are the Kconfig and Makefile files for spear? --- sound/soc/spear/spear_pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9b76cc5..5e7aebe 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm) static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); -static int spear_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; int ret; if (!card->dev->dma_mask) @@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, spear_pcm_hardware.buffer_bytes_max); if (ret) return ret; } - if (dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_CAPTURE, spear_pcm_hardware.buffer_bytes_max); if (ret) -- 1.8.0

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[alsa-devel] [PATCH 0/3] ASoC: omap-mcpdm: Fix for full duplex audio
by Peter Ujfalusi 20 Mar '13

20 Mar '13

Hi, We had this issue for a long time but I had no time to think and fix it. With this series the full duplex audio will work correctly with McPDM. Under the hood we have HW limitation which prevents us to start up/downlink separately. Regards, Peter --- Peter Ujfalusi (3): ASoC: omap-mcpdm: Collect link direction configuration under a struct ASoC: omap-mcpdm: Fix for full duplex audio use case ASoC: omap-mcpdm: Remove leftower define for IO address sound/soc/omap/omap-mcpdm.c | 66 ++++++++++++++++++++++++++++++++------------- 1 file changed, 48 insertions(+), 18 deletions(-) -- 1.8.1.5

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[alsa-devel] [PATCH] ASoC: fsl: imx-pcm-fiq: Use 'unsigned int' for period
by Fabio Estevam 20 Mar '13

20 Mar '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Fix the following warning when building with W=1 option: sound/soc/fsl/imx-pcm-fiq.c: In function 'snd_hrtimer_callback': sound/soc/fsl/imx-pcm-fiq.c:76:12: warning: comparison between signed and unsigned integer expressions [-Wsign-compare] Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-pcm-fiq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 920f945..47228c0 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -34,7 +34,7 @@ #include "imx-ssi.h" struct imx_pcm_runtime_data { - int period; + unsigned int period; int periods; unsigned long offset; unsigned long last_offset; -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: snd-usb: mixer: propagate errors up the call chain
by Daniel Mack 20 Mar '13

20 Mar '13

In check_input_term() and parse_audio_feature_unit(), propagate the error value that has been returned by a failing function instead of -EINVAL. That helps cleaning up the error pathes in the mixer. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/mixer.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 638e7f7..db30d02 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -715,8 +715,9 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - if (check_input_term(state, d->baSourceID[0], term) < 0) - return -ENODEV; + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; term->type = d->bDescriptorSubtype << 16; /* virtual type */ term->id = id; term->name = uac_selector_unit_iSelector(d); @@ -1356,8 +1357,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void return err; /* determine the input source type and name */ - if (check_input_term(state, hdr->bSourceID, &iterm) < 0) - return -EINVAL; + err = check_input_term(state, hdr->bSourceID, &iterm); + if (err < 0) + return err; master_bits = snd_usb_combine_bytes(bmaControls, csize); /* master configuration quirks */ -- 1.8.1.4

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[alsa-devel] [PATCH] ALSA: usb: Parse UAC2 extension unit like for UAC1
by Torstein Hegge 20 Mar '13

20 Mar '13

UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in the same way when parsing the unit. Otherwise parse_audio_unit() fails when it sees an extension unit on a UAC2 device. UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- sound/usb/mixer.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 638e7f7..8eb84c0 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -725,7 +725,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC1_PROCESSING_UNIT: case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ - /* UAC2_EFFECT_UNIT */ { + /* UAC2_EFFECT_UNIT */ + case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; if (state->mixer->protocol == UAC_VERSION_2 && @@ -2052,6 +2053,8 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_extension_unit(state, unitid, p1); else /* UAC_VERSION_2 */ return parse_audio_processing_unit(state, unitid, p1); + case UAC2_EXTENSION_UNIT_V2: + return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); return -EINVAL; -- 1.7.10.4

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[alsa-devel] [TINYCOMPRESS] Remove local copies of kernel headers?
by Richard Fitzgerald 20 Mar '13

20 Mar '13

I'm assuming that the local copies of compress_offload.h and compress_params.h were there as a convenience during early development Can we remove these local copies now?

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[alsa-devel] ASoC: atmel-pcm-dma: Leaking memory in atmel_pcm_hw_params
by Lars-Peter Clausen 19 Mar '13

19 Mar '13

Hi, The dmaengine based pcm driver for atmel calls snd_dmaengine_pcm_open from it's hw_params callback. There is nothing preventing an application calling hw_params more than once. snd_dmaengine_pcm_open allocates a new prtd struct each time it gets called. So in case hw_params is called multiple times we leak memory here. Is there any specific reason why snd_dmaengine_pcm_open() needs to be called from the hw_params callback and why it can't be called from the open callback? The filter function depends on the data returned by snd_soc_dai_get_dma_data(), but as far as I can see the DAI driver calls snd_soc_dai_set_dma_data() from its startup() callback, so the data is available in the PCM drivers open callback. - Lars

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