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

[alsa-devel] HD-audio generic parser merged to linux-next
by Takashi Iwai 13 Feb '13

13 Feb '13

Hi, as most of feedbacks about the new generic parser migration were positive, I decided to take a risk: now all commit that have been in test/hda-migrate branch in sound-unstable git tree are merged to for-next branch of sound git tree for better test coverage. It means that all changes are merged to linux-next, i.e. will be included in 3.9 kernel. For the early adapter people, I'm going to keep test/hda-migrate branch of sound-unstable git tree updated for a while, but the branch will be abandoned some time later. If you encounter any regression by the recent update, please let me know. thanks, Takashi

3 14

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 13 Feb '13

13 Feb '13

From: Jeeja KP <jeeja.kp(a)intel.com> this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v3: - Change back to ioctl struct with padding - adding states for next_track and metadata for proper transistion v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation --- Documentation/sound/alsa/compress_offload.txt | 46 ++++++++++ include/sound/compress_driver.h | 8 ++ include/uapi/sound/compress_offload.h | 23 +++++- sound/core/compress_offload.c | 116 +++++++++++++++++++++++++ 4 files changed, 192 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..ff6c741 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..8903fe4 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -122,6 +122,19 @@ struct snd_compr_codec_caps { }; /** + * struct snd_compr_metadata: compressed stream metadata + * @encoder_delay: no of samples inserted by the encoder at the beginning + * of the track + * @encoder_padding: no of samples appended by the encoder at the end + * of the track + */ +struct snd_compr_metadata { + __u32 encoder_delay; + __u32 encoder_padding; + __u32 reserved[12]; +}; + +/** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec @@ -145,6 +158,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOW('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +169,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..ba2f0ad 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -514,6 +514,69 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + retval = stream->ops->get_metadata(stream, metadata); + if (retval != 0) + goto out; + + if (copy_to_user((void __user *)arg, metadata, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + +out: + kfree(metadata); + return retval; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, + sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + pr_debug("metadata encoder delay=%x encoder padding=%x\n", + metadata->encoder_delay, metadata->encoder_padding); + + retval = stream->ops->set_metadata(stream, metadata); + stream->metadata_set = true; + +out: + kfree(metadata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +657,46 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) + pr_err("Partial drain returned failure\n"); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +726,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +753,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

3 5

[alsa-devel] [PATCH] ASoC: omap-dmic: Clean up with devm_request_and_ioremap
by Sebastien Guiriec 13 Feb '13

13 Feb '13

Clean up dmic code with devm_request_and_ioremap function. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-dmic.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index ba49ccd..77e9e7e 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -493,16 +493,9 @@ static int asoc_dmic_probe(struct platform_device *pdev) goto err_put_clk; } - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(dmic->dev, "memory region already claimed\n"); - ret = -ENODEV; - goto err_put_clk; - } - - dmic->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); + dmic->io_base = devm_request_and_ioremap(&pdev->dev, res); if (!dmic->io_base) { + dev_err(&pdev->dev, "cannot remap\n"); ret = -ENOMEM; goto err_put_clk; } -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Clean up with devm_* function
by Sebastien Guiriec 13 Feb '13

13 Feb '13

Clean up McPDM driver with devm_ function. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 5ca11bd..079f277 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -369,7 +369,7 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(mcpdm->dev); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM", (void *)mcpdm); pm_runtime_put_sync(mcpdm->dev); @@ -389,7 +389,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); return 0; @@ -465,14 +464,11 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) if (res == NULL) return -ENOMEM; - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), "McPDM")) - return -EBUSY; - - mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!mcpdm->io_base) + mcpdm->io_base = devm_request_and_ioremap(&pdev->dev, res); + if (!mcpdm->io_base) { + dev_err(&pdev->dev, "cannot remap\n"); return -ENOMEM; + } mcpdm->irq = platform_get_irq(pdev, 0); if (mcpdm->irq < 0) -- 1.7.10.4

3 2

Re: [alsa-devel] [RFC]Connection list of hda widget
by Takashi Iwai 13 Feb '13

13 Feb '13

At Thu, 07 Feb 2013 16:09:35 +0100, David Henningsson wrote: > > On 02/07/2013 06:38 AM, Raymond Yau wrote: > > > > Treat vendor defined widget different from audio selector/audio input > > widget since hda_proc.c query the connection list of vendor defined widget > > > > ADI codecs have vendor defined widgets ( BIAS Power-Down and VREF > > Power-Down) which have connection list > > > > static void print_conn_list(struct snd_info_buffer *buffer, > > struct hda_codec *codec, hda_nid_t nid, > > unsigned int wid_type, hda_nid_t *conn, > > int conn_len) > > { > > int c, curr = -1; > > > > if (conn_len > 1 && > > wid_type != AC_WID_AUD_MIX && > > wid_type != AC_WID_VOL_KNB && > > + wide_type != AC_WID_VENDOR && > > wid_type != AC_WID_POWER) > > curr = snd_hda_codec_read(codec, nid, 0, > > AC_VERB_GET_CONNECT_SEL, 0); > > snd_iprintf(buffer, " Connection: %d\n", conn_len); > > > > I think there are some Sigmatel/IDT (or was it Analog Devices?) codecs > that actually have sensible connections for their vendor specific nodes, > so I would prefer not to change this. Right. The widget connection longer than 32 itself is OK. It's just ugly. That being said, it's basically a bug that we restrict the max connection size in the core code. So I applied the patch below. This removes the limit while we still keep the efficiency (except for proc output, which isn't too critical). The patch became bigger than I wanted originally, but it's life. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Remove limit of widget connections Currently we set the max number of connections to be 32, but there seems codec that gives longer connection lists like AD1988, and we see errors in proc output and else. (Though, in the case of AD1988, it's a list of all codecs connected to a single vendor widget, so this must be something fishy, but it's still valid from the h/w design POV.) This patch tries to remove this restriction. For efficiency, we still use the fixed size array in the parser, but takes a dynamic array when the size is reported to be greater than that. Now the fixed array size is found only in patch_hdmi.c, but it should be fine, as the codec itself can't support so many pins. Reported-by: Raymond Yau <superquad.vortex2(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 62 ++++++++++++++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 4 +-- sound/pci/hda/hda_proc.c | 19 +++++++++++--- sound/pci/hda/patch_hdmi.c | 3 +++ 4 files changed, 60 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f82a64d..3c92514 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -382,13 +382,23 @@ static void remove_conn_list(struct hda_codec *codec) /* read the connection and add to the cache */ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) { - hda_nid_t list[HDA_MAX_CONNECTIONS]; + hda_nid_t list[32]; + hda_nid_t *result = list; int len; len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list)); - if (len < 0) - return len; - return snd_hda_override_conn_list(codec, nid, len, list); + if (len == -ENOSPC) { + len = snd_hda_get_num_raw_conns(codec, nid); + result = kmalloc(sizeof(hda_nid_t) * len, GFP_KERNEL); + if (!result) + return -ENOMEM; + len = snd_hda_get_raw_connections(codec, nid, result, len); + } + if (len >= 0) + len = snd_hda_override_conn_list(codec, nid, len, result); + if (result != list) + kfree(result); + return len; } /** @@ -466,6 +476,27 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_get_connections); +/* return CONNLIST_LEN parameter of the given widget */ +static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!(wcaps & AC_WCAP_CONN_LIST) && + get_wcaps_type(wcaps) != AC_WID_VOL_KNB) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); + if (parm == -1) + parm = 0; + return parm; +} + +int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid) +{ + return get_num_conns(codec, nid) & AC_CLIST_LENGTH; +} + /** * snd_hda_get_raw_connections - copy connection list without cache * @codec: the HDA codec @@ -483,19 +514,16 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; int i, conn_len, conns; unsigned int shift, num_elems, mask; - unsigned int wcaps; hda_nid_t prev_nid; int null_count = 0; if (snd_BUG_ON(!conn_list || max_conns <= 0)) return -EINVAL; - wcaps = get_wcaps(codec, nid); - if (!(wcaps & AC_WCAP_CONN_LIST) && - get_wcaps_type(wcaps) != AC_WID_VOL_KNB) + parm = get_num_conns(codec, nid); + if (!parm) return 0; - parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); if (parm & AC_CLIST_LONG) { /* long form */ shift = 16; @@ -552,21 +580,13 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, continue; } for (n = prev_nid + 1; n <= val; n++) { - if (conns >= max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - conns, nid); - return -EINVAL; - } + if (conns >= max_conns) + return -ENOSPC; conn_list[conns++] = n; } } else { - if (conns >= max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - conns, nid); - return -EINVAL; - } + if (conns >= max_conns) + return -ENOSPC; conn_list[conns++] = val; } prev_nid = val; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6c59272..0be1826 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -551,9 +551,6 @@ enum { AC_JACK_PORT_BOTH, }; -/* max. connections to a widget */ -#define HDA_MAX_CONNECTIONS 32 - /* max. codec address */ #define HDA_MAX_CODEC_ADDRESS 0x0f @@ -958,6 +955,7 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) { return snd_hda_get_connections(codec, nid, NULL, 0); } +int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid); int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 5e02f26..97cdbef 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -623,7 +623,7 @@ static void print_codec_info(struct snd_info_entry *entry, snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); unsigned int wid_type = get_wcaps_type(wid_caps); - hda_nid_t conn[HDA_MAX_CONNECTIONS]; + hda_nid_t *conn = NULL; int conn_len = 0; snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, @@ -660,9 +660,18 @@ static void print_codec_info(struct snd_info_entry *entry, if (wid_type == AC_WID_VOL_KNB) wid_caps |= AC_WCAP_CONN_LIST; - if (wid_caps & AC_WCAP_CONN_LIST) - conn_len = snd_hda_get_raw_connections(codec, nid, conn, - HDA_MAX_CONNECTIONS); + if (wid_caps & AC_WCAP_CONN_LIST) { + conn_len = snd_hda_get_num_raw_conns(codec, nid); + if (conn_len > 0) { + conn = kmalloc(sizeof(hda_nid_t) * conn_len, + GFP_KERNEL); + if (!conn) + return; + if (snd_hda_get_raw_connections(codec, nid, conn, + conn_len) < 0) + conn_len = 0; + } + } if (wid_caps & AC_WCAP_IN_AMP) { snd_iprintf(buffer, " Amp-In caps: "); @@ -735,6 +744,8 @@ static void print_codec_info(struct snd_info_entry *entry, if (codec->proc_widget_hook) codec->proc_widget_hook(buffer, codec, nid); + + kfree(conn); } snd_hda_power_down(codec); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 85236da..899c4fb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -64,6 +64,9 @@ struct hdmi_spec_per_cvt { unsigned int maxbps; }; +/* max. connections to a widget */ +#define HDA_MAX_CONNECTIONS 32 + struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; -- 1.8.1.2

2 6

[alsa-devel] Channel map for Logitech Web Camera
by Raymond Yau 13 Feb '13

13 Feb '13

It seem that channel map is still missing for Logitech Web Camera since I can find the corresponding entries of hda-intel and the patched au8830 in asound.state except the Logitech Web Camera state.U0x46d0x992 { control.1 { iface MIXER name 'Mic Capture Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.2 { iface MIXER name 'Mic Capture Volume' value 3041 comment { access 'read write' type INTEGER count 1 range '0 - 3072' dbmin 1800 dbmax 3000 dbvalue.0 2987 } } } Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 2 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 2 Streaming bFunctionProtocol 0 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 38 bInCollection 1 baInterfaceNr( 0) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 1 wChannelConfig 0x0000 iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 1 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 1 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 16000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0044 1x 68 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined

1 0

[alsa-devel] [RFC] Fix for conflict of HDMI and SPDIF IEC958 controls
by Takashi Iwai 12 Feb '13

12 Feb '13

Hi, there is a long-standing problem in HD-audio regarding IEC958 controls. When both an SPDIF and an HDMI are created on the same card (e.g. one from analog codec and one from graphics chip), the driver assigns the IEC958 controls just with new indices, 0, 1, 2... The problem is that there is no way to connect between this index and the actual PCM device. Currently, alsa-lib HDA-Intel.conf has a fixed configuration: spdif -> IEC958 xxx index=0, PCM dev=1 hdmi,0 -> IEC958 xxx index=0, PCM dev=3 hdmi,1 -> IEC958 xxx index=1, PCM dev=7 hdmi,2 -> IEC958 xxx index=2, PCM dev=8 hdmi,3 -> IEC958 xxx index=3, PCM dev=9 So obviously spdif and the first hdmi conflict. Basically this can be fixed by reassigning each IEC958 control with the same device number corresponding to the PCM device. That is, for SPDIF, assign a control element with device=1, for HDMI, device=3,7... However, this obviously breaks the old configuration unless user upgrades the alsa-lib configuration. Thus this is no-go. Now here is a compromise: the IEC958 control for SPDIF is reassigned to device=1 *only* when SPDIF and HDMI coexist. Since the configuration is anyway broken as is now in such a case, it's no big deal to fix one side in an incompatible way. (The reason why SPDIF is re-assigned is that I guess majority of user require more HDMI than SPDIF in such a configuration.) In addition, we need a fix in alsa-lib. Also not for breaking the compatibility with older kernel, we need some fallback check. I did a quick hack to alsa-lib conf code and added "skip_rest" option to the hook element, so that only the first matching element is taken. Long story short, I cooked up three patches. One patch is for kernel, to add the workaround above, and the two are for alsa-lib, one clean up and one to introduce the skip_rest option (and application to HDA-Intel.conf). My plan is to merge this to 3.8 tree, so it's no urgent issue. But it's of course always good to fix something. thanks, Takashi

5 29

Re: [alsa-devel] Xonar DG pci sound card driver incomplete
by Clemens Ladisch 12 Feb '13

12 Feb '13

kernel kernel wrote: > Requesting guidance on how to implement the missing mic input support > for this Asus Xonar card. I've downloaded the relevant datasheets but > am unsure how to proceed. Rip out the card, and look (or eletrically trace) how the CMI8788's GPIOs and the CS4245's inputs/outputs are connected. Test how to configure and set the GPIOs so that the mic works. Also check the register configuration of the CS4245. Regards, Clemens

1 0

Re: [alsa-devel] [PATCH 2/3] ALSA: hda - no-primary-hp is a quirk for model ALC889 not ALC882
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:54:16 +0900, Fernando Luis Vázquez Cao wrote: > > Substitute ALC889 for ALC882 in macro and function names. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> This fixup can work generically to all ALC882 variants, including ALC889. Unless you are really sure that this must be used exclusively only for ALC889, no reason to change the prefix. thanks, Takashi > --- > > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-09 22:52:40.301209823 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-09 22:58:54.695063675 +0900 > @@ -5030,7 +5030,7 @@ enum { > ALC889_FIXUP_MBP_VREF, > ALC889_FIXUP_IMAC91_VREF, > ALC882_FIXUP_INV_DMIC, > - ALC882_FIXUP_NO_PRIMARY_HP, > + ALC889_FIXUP_NO_PRIMARY_HP, > }; > > static void alc889_fixup_coef(struct hda_codec *codec, > @@ -5156,7 +5156,7 @@ static void alc889_fixup_imac91_vref(str > * Strangely, the speaker output doesn't work on Vaio Z and some Vaio > * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 > */ > -static void alc882_fixup_no_primary_hp(struct hda_codec *codec, > +static void alc889_fixup_no_primary_hp(struct hda_codec *codec, > const struct alc_fixup *fix, int action) > { > struct alc_spec *spec = codec->spec; > @@ -5350,9 +5350,9 @@ static const struct alc_fixup alc882_fix > .type = ALC_FIXUP_FUNC, > .v.func = alc_fixup_inv_dmic_0x12, > }, > - [ALC882_FIXUP_NO_PRIMARY_HP] = { > + [ALC889_FIXUP_NO_PRIMARY_HP] = { > .type = ALC_FIXUP_FUNC, > - .v.func = alc882_fixup_no_primary_hp, > + .v.func = alc889_fixup_no_primary_hp, > }, > }; > > @@ -5388,8 +5388,8 @@ static const struct snd_pci_quirk alc882 > SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), > SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), > SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), > - SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), > - SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC889_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC889_FIXUP_NO_PRIMARY_HP), > > /* All Apple entries are in codec SSIDs */ > SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), > @@ -5432,7 +5432,7 @@ static const struct alc_model_fixup alc8 > {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, > {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, > {.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"}, > - {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, > + {.id = ALC889_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, > {} > }; > > >

1 0

Re: [alsa-devel] [PATCH 2/3] ALSA: hda - update documentation for no-primary-hp fixup
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:49:46 +0900, Fernando Luis Vázquez Cao wrote: > > The problem addressed by this fixup is not specific to Vaio Z, affecting > some Vaio all-in-one desktop PCs too. Update the code comments accordingly. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> I applied this now to sound git tree but dropped Cc to stable because the patch is no longer cleanly applicable due to the recent code change. And this is just some minor documentation update, thus we can still live peacefully without this fix. thanks, Takashi > --- > > diff -urNp linux-3.7.6-orig/Documentation/sound/alsa/HD-Audio-Models.txt linux-3.7.6/Documentation/sound/alsa/HD-Audio-Models.txt > --- linux-3.7.6-orig/Documentation/sound/alsa/HD-Audio-Models.txt 2012-12-11 12:30:57.000000000 +0900 > +++ linux-3.7.6/Documentation/sound/alsa/HD-Audio-Models.txt 2013-02-09 22:44:28.614772054 +0900 > @@ -53,7 +53,7 @@ ALC882/883/885/888/889 > acer-aspire-8930g Acer Aspire 8330G/6935G > acer-aspire Acer Aspire others > inv-dmic Inverted internal mic workaround > - no-primary-hp VAIO Z workaround (for fixed speaker DAC) > + no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) > > ALC861/660 > ========== > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-09 22:39:11.745202168 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-09 22:50:10.924470402 +0900 > @@ -5153,7 +5153,8 @@ static void alc889_fixup_imac91_vref(str > } > > /* Don't take HP output as primary > - * strangely, the speaker output doesn't work on VAIO Z through DAC 0x05 > + * Strangely, the speaker output doesn't work on Vaio Z and some Vaio > + * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 > */ > static void alc882_fixup_no_primary_hp(struct hda_codec *codec, > const struct alc_fixup *fix, int action) > >

1 0