January 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] HD audio problem (fwd)
by Russ Whitaker 22 Jan '14

22 Jan '14

Still not heard from you. WHY? ---------- Forwarded message ---------- Date: Mon, 30 Dec 2013 20:20:31 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem (fwd) Have not heard back from you. What should I do? ---------- Forwarded message ---------- Date: Thu, 19 Dec 2013 23:48:19 -0800 (PST) From: Russ Whitaker <russ(a)ashlandhome.net> To: alsa-devel(a)alsa-project.org Subject: HD audio problem Hi Bought a new mini-itx mb: ECS KBN-I On board sound chip: VT1705CF Installed kernel linux-3.10.24 Otherwise distro is Slackware64 14.1 mplayer plays .mp3 files just fine. No sound from video clips that use Adobe flashplayer, and I get this error message: ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave I think the problem is: card0 and card1 are interchanged as card0 is listed as the HDMI video output. How to fixit, please. Thank you, Russ

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[alsa-devel] [PATCH 2/2] ASoC: fsl_ssi: Do not print 'baud clock' error message all the time
by Fabio Estevam 21 Jan '14

21 Jan '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently everytime we get the following message on boot: fsl-ssi-dai 202c000.ssi: could not get baud clock: -2 This is not really useful information to get on every boot, so make it a debug message instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7b1772d..3155611 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1407,7 +1407,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); if (IS_ERR(ssi_private->baudclk)) - dev_warn(&pdev->dev, "could not get baud clock: %ld\n", + dev_dbg(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); else clk_prepare_enable(ssi_private->baudclk); -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: fsl: use snd_soc_dai_init_dma_data()
by Xiubo Li 21 Jan '14

21 Jan '14

Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 4d075f1..8f36f49 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -911,8 +911,8 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) { struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai); - dai->playback_dma_data = &spdif_private->dma_params_tx; - dai->capture_dma_data = &spdif_private->dma_params_rx; + snd_soc_dai_init_dma_data(dai, &spdif_private->dma_params_tx, + &spdif_private->dma_params_rx); snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls)); -- 1.8.4

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[alsa-devel] [PATCH 1/3] ASoC: wm5102: Improve EQ coefficient controls
by Charles Keepax 21 Jan '14

21 Jan '14

The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a08e8bf..ac19f62 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,15 +685,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -705,6 +698,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -716,6 +711,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -727,6 +724,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: wm5110: Extend SYSCLK patch file for rev D
by Charles Keepax 21 Jan '14

21 Jan '14

Latest evaluation of the the device has given some patch file additions for improved performance. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 48 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 48 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 34fbb03..92f1b40 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -43,6 +43,54 @@ static const struct reg_default wm5110_sysclk_revd_patch[] = { { 0x3133, 0x1201 }, { 0x3183, 0x1501 }, { 0x31D3, 0x1401 }, + { 0x0049, 0x01ea }, + { 0x004a, 0x01f2 }, + { 0x0057, 0x01e7 }, + { 0x0058, 0x01fb }, + { 0x33ce, 0xc4f5 }, + { 0x33cf, 0x1361 }, + { 0x33d0, 0x0402 }, + { 0x33d1, 0x4700 }, + { 0x33d2, 0x026d }, + { 0x33d3, 0xff00 }, + { 0x33d4, 0x026d }, + { 0x33d5, 0x0101 }, + { 0x33d6, 0xc4f5 }, + { 0x33d7, 0x0361 }, + { 0x33d8, 0x0402 }, + { 0x33d9, 0x6701 }, + { 0x33da, 0xc4f5 }, + { 0x33db, 0x136f }, + { 0x33dc, 0xc4f5 }, + { 0x33dd, 0x134f }, + { 0x33de, 0xc4f5 }, + { 0x33df, 0x131f }, + { 0x33e0, 0x026d }, + { 0x33e1, 0x4f01 }, + { 0x33e2, 0x026d }, + { 0x33e3, 0xf100 }, + { 0x33e4, 0x026d }, + { 0x33e5, 0x0001 }, + { 0x33e6, 0xc4f5 }, + { 0x33e7, 0x0361 }, + { 0x33e8, 0x0402 }, + { 0x33e9, 0x6601 }, + { 0x33ea, 0xc4f5 }, + { 0x33eb, 0x136f }, + { 0x33ec, 0xc4f5 }, + { 0x33ed, 0x134f }, + { 0x33ee, 0xc4f5 }, + { 0x33ef, 0x131f }, + { 0x33f0, 0x026d }, + { 0x33f1, 0x4e01 }, + { 0x33f2, 0x026d }, + { 0x33f3, 0xf000 }, + { 0x33f6, 0xc4f5 }, + { 0x33f7, 0x1361 }, + { 0x33f8, 0x0402 }, + { 0x33f9, 0x4600 }, + { 0x33fa, 0x026d }, + { 0x33fb, 0xfe00 }, }; static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: wm_adsp: Add info print to note that the DSP has shutdown
by Charles Keepax 21 Jan '14

21 Jan '14

It can be useful for debugging purposes to see at what point the DSP has powered down, so add a message to inform us of this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4fbcab6..a15f75b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1637,6 +1637,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_del(&alg_region->list); kfree(alg_region); } + + adsp_info(dsp, "Shutdown complete\n"); break; default: -- 1.7.2.5

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[alsa-devel] [alsa-plugins] [PATCH 2/2] Allow to specify a client name as parameters
by Corfu, ValentinX 21 Jan '14

21 Jan '14

0001-Modify-jack-client-name.patch is spitted in: 0001-Allow-number-of-channels.patch and 0002-Allow-to-specify-a-client-name-as-parameters-in-the-.patch >From 8d6d397306bdd09c6c3fae83543e124c89443e94 Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Tue, 21 Jan 2014 11:32:51 -0500 Subject: [PATCH 2/2] Allow to specify a client name as parameters The current jack client name contains the process id of the application providing the audio samples. This leads to unpredictable jack client names which makes handling of the connections by a controlling application very hard. This modification now, allows to specify a client name as parameters in the configuration file. The implementation is backward compatible and simply adds a new configuration option to the plugin. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 4306a13..61ac197 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -32,6 +32,11 @@ typedef enum _jack_format { SND_PCM_JACK_FORMAT_RAW } snd_pcm_jack_format_t; +typedef enum _naming_scheme { + SND_PCM_JACK_STD_NAME, + SND_PCM_JACK_USER_DEFINED_NAME +} snd_pcm_jack_naming_t; + typedef struct { snd_pcm_ioplug_t io; @@ -325,7 +330,8 @@ static int make_nonblock(int fd) static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, snd_config_t *playback_conf, snd_config_t *capture_conf, - snd_pcm_stream_t stream, int mode) + snd_pcm_stream_t stream, int mode, + snd_pcm_jack_naming_t naming) { snd_pcm_jack_t *jack; int err; @@ -356,11 +362,19 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return -EINVAL; } - if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, - stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) - >= (int)sizeof(jack_client_name)) { - fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", - __func__, jack_client_name, (int)strlen(jack_client_name)); + if (naming == SND_PCM_JACK_STD_NAME ) { + if (snprintf(jack_client_name, sizeof(jack_client_name), "alsa-jack.%s%s.%d.%d", name, + stream == SND_PCM_STREAM_PLAYBACK ? "P" : "C", getpid(), num++) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } + } else { + if (snprintf(jack_client_name, sizeof(jack_client_name), "%s", name) + >= (int)sizeof(jack_client_name)) { + fprintf(stderr, "%s: WARNING: JACK client name '%s' truncated to %d characters, might not be unique\n", + __func__, jack_client_name, (int)strlen(jack_client_name)); + } } jack->client = jack_client_new(jack_client_name); @@ -417,6 +431,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) int err; long num_channels = 0; long chan; + const char *local_name = name; + snd_pcm_jack_naming_t naming = SND_PCM_JACK_STD_NAME; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -425,6 +441,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "name") == 0) { + snd_config_get_string(n, &local_name); + naming = SND_PCM_JACK_USER_DEFINED_NAME; + continue; + } if (strcmp(id, "nchan") == 0) { snd_config_get_integer(n, &num_channels); continue; @@ -449,7 +470,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - if ( (num_channels != 0) && (playback_conf == NULL) && (capture_conf == NULL) ) { + if ( (naming == SND_PCM_JACK_USER_DEFINED_NAME) + && (num_channels != 0) + && (playback_conf == NULL) + && (capture_conf == NULL) ) { err = snd_config_make_compound(&playback_conf, "playback_ports", 0); if (err) { SNDERR("Error during snd_config_make_compound(p): %s\n", strerror(err)); @@ -486,7 +510,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return err; } } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, local_name, playback_conf, capture_conf, stream, mode, naming); if (err) { SNDERR("Error during snd_pcm_jack_open: %s\n", strerror(err)); return err; @@ -513,7 +537,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return err; } } else { - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + err = snd_pcm_jack_open(pcmp, local_name, playback_conf, capture_conf, stream, mode, naming); } return err; -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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[alsa-devel] [alsa-plugins] [PATCH 1/2] Allow number of channels
by Corfu, ValentinX 21 Jan '14

21 Jan '14

0001-Modify-jack-client-name.patch is spitted in: 0001-Allow-number-of-channels.patch and 0002-Allow-to-specify-a-client-name-as-parameters-in-the-.patch >From d247e4d675dd68c6f683a8c09ff792172396e870 Mon Sep 17 00:00:00 2001 From: Valentin Corfu <valentinx.corfu(a)intel.com> Date: Tue, 21 Jan 2014 10:10:52 -0500 Subject: [PATCH 1/2] Allow number of channels The modification allows to specify the number of channels as parameters in the configuration file The given number of channels are then created but not connected. Signed-off-by: Valentin Corfu <valentinx.corfu(a)intel.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index c51fe84..4306a13 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -415,6 +415,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) snd_config_t *playback_conf = NULL; snd_config_t *capture_conf = NULL; int err; + long num_channels = 0; + long chan; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -423,6 +425,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) continue; if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0 || strcmp(id, "hint") == 0) continue; + if (strcmp(id, "nchan") == 0) { + snd_config_get_integer(n, &num_channels); + continue; + } if (strcmp(id, "playback_ports") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); @@ -443,7 +449,72 @@ SND_PCM_PLUGIN_DEFINE_FUNC(jack) return -EINVAL; } - err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + if ( (num_channels != 0) && (playback_conf == NULL) && (capture_conf == NULL) ) { + err = snd_config_make_compound(&playback_conf, "playback_ports", 0); + if (err) { + SNDERR("Error during snd_config_make_compound(p): %s\n", strerror(err)); + return err; + } + err = snd_config_make_compound(&capture_conf, "capture_ports", 0); + if (err) { + SNDERR("Error during snd_config_make_compound(c): %s\n", strerror(err)); + return err; + } + char value[4]; + for (chan=0; chan<num_channels; ++chan) { + snprintf(value, sizeof(value), "%d", chan); + snd_config_t *newLeafPlayback; + snd_config_t *newLeafCapture; + err = snd_config_imake_integer(&newLeafPlayback, value, chan); + if (err) { + SNDERR("Error during snd_config_imake_string(p): %s\n", strerror(err)); + return err; + } + err = snd_config_add(playback_conf, newLeafPlayback); + if (err) { + SNDERR("Error during snd_config_add(p): %s\n", strerror(err)); + return err; + } + err = snd_config_imake_integer(&newLeafCapture, value, chan); + if (err) { + SNDERR("Error during snd_config_imake_string(c): %s\n", strerror(err)); + return err; + } + err = snd_config_add(capture_conf, newLeafCapture); + if (err) { + SNDERR("Error during snd_config_add(c): %s\n", strerror(err)); + return err; + } + } + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + if (err) { + SNDERR("Error during snd_pcm_jack_open: %s\n", strerror(err)); + return err; + } + + err = snd_config_delete_compound_members(playback_conf); + if (err) { + SNDERR("Error during snd_config_delete_compound_members(p): %s\n", strerror(err)); + return err; + } + err = snd_config_delete_compound_members(capture_conf); + if (err) { + SNDERR("Error during snd_config_delete_compound_members(c): %s\n", strerror(err)); + return err; + } + err = snd_config_delete(playback_conf); + if (err) { + SNDERR("Error during snd_config_delete(p): %s\n", strerror(err)); + return err; + } + err = snd_config_delete(capture_conf); + if (err) { + SNDERR("Error during snd_config_delete(c): %s\n", strerror(err)); + return err; + } + } else { + err = snd_pcm_jack_open(pcmp, name, playback_conf, capture_conf, stream, mode); + } return err; } -- 1.8.1.4 -------------------------------------------------------------- Intel Shannon Limited Registered in Ireland Registered Office: Collinstown Industrial Park, Leixlip, County Kildare Registered Number: 308263 Business address: Dromore House, East Park, Shannon, Co. Clare This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

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[alsa-devel] [PATCH] ALSA: usb-audio: Resume mixer values properly
by Takashi Iwai 21 Jan '14

21 Jan '14

Implement reset_resume callback so that the mixer values are properly restored. Still no boot quirks are called, so it might not work well on some devices. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 18 ++++++++-- sound/usb/mixer.c | 99 ++++++++++++++++++++++++++++++++++++++++++++----------- sound/usb/mixer.h | 7 ++-- 3 files changed, 99 insertions(+), 25 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d979050e6a6a..025224136129 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -691,12 +691,12 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) } list_for_each_entry(mixer, &chip->mixer_list, list) - snd_usb_mixer_inactivate(mixer); + snd_usb_mixer_suspend(mixer); return 0; } -static int usb_audio_resume(struct usb_interface *intf) +static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct usb_mixer_interface *mixer; @@ -711,7 +711,7 @@ static int usb_audio_resume(struct usb_interface *intf) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_activate(mixer); + err = snd_usb_mixer_resume(mixer, reset_resume); if (err < 0) goto err_out; } @@ -723,9 +723,20 @@ static int usb_audio_resume(struct usb_interface *intf) err_out: return err; } + +static int usb_audio_resume(struct usb_interface *intf) +{ + return __usb_audio_resume(intf, false); +} + +static int usb_audio_reset_resume(struct usb_interface *intf) +{ + return __usb_audio_resume(intf, true); +} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL +#define usb_audio_reset_resume NULL #endif /* CONFIG_PM */ static struct usb_device_id usb_audio_ids [] = { @@ -747,6 +758,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, + .reset_resume = usb_audio_reset_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 44b0ba4feab3..aa9bc19aae68 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2299,26 +2299,6 @@ requeue: } } -/* stop any bus activity of a mixer */ -void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) -{ - usb_kill_urb(mixer->urb); - usb_kill_urb(mixer->rc_urb); -} - -int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) -{ - int err; - - if (mixer->urb) { - err = usb_submit_urb(mixer->urb, GFP_NOIO); - if (err < 0) - return err; - } - - return 0; -} - /* create the handler for the optional status interrupt endpoint */ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) { @@ -2417,3 +2397,82 @@ void snd_usb_mixer_disconnect(struct list_head *p) usb_kill_urb(mixer->urb); usb_kill_urb(mixer->rc_urb); } + +#ifdef CONFIG_PM +/* stop any bus activity of a mixer */ +static void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) +{ + usb_kill_urb(mixer->urb); + usb_kill_urb(mixer->rc_urb); +} + +static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) +{ + int err; + + if (mixer->urb) { + err = usb_submit_urb(mixer->urb, GFP_NOIO); + if (err < 0) + return err; + } + + return 0; +} + +int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) +{ + snd_usb_mixer_inactivate(mixer); + return 0; +} + +static int restore_mixer_value(struct usb_mixer_elem_info *cval) +{ + int c, err, idx; + + if (cval->cmask) { + idx = 0; + for (c = 0; c < MAX_CHANNELS; c++) { + if (!(cval->cmask & (1 << c))) + continue; + if (cval->cached & (1 << c)) { + err = set_cur_mix_value(cval, c + 1, idx, + cval->cache_val[idx]); + if (err < 0) + return err; + } + idx++; + } + } else { + /* master */ + if (cval->cached) { + err = set_cur_mix_value(cval, 0, 0, *cval->cache_val); + if (err < 0) + return err; + } + } + + return 0; +} + +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +{ + struct usb_mixer_elem_info *cval; + int id, err; + + /* FIXME: any mixer quirks? */ + + if (reset_resume) { + /* restore cached mixer values */ + for (id = 0; id < MAX_ID_ELEMS; id++) { + for (cval = mixer->id_elems[id]; cval; + cval = cval->next_id_elem) { + err = restore_mixer_value(cval); + if (err < 0) + return err; + } + } + } + + return snd_usb_mixer_activate(mixer); +} +#endif diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index aab80df201bd..73b1f649447b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -63,8 +63,6 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); -void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer); -int snd_usb_mixer_activate(struct usb_mixer_interface *mixer); int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, struct snd_kcontrol *kctl); @@ -72,4 +70,9 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv); +#ifdef CONFIG_PM +int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +#endif + #endif /* __USBMIXER_H */ -- 1.8.5.2

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Re: [alsa-devel] [PATCH] ASoC: da9055: Partially fix device
by Opensource [Adam Thomson] 21 Jan '14

21 Jan '14

On Wed, 8 Jan 2014 20:38:55 +0000, Mark Brown wrote: > It has never been possible to even load the da9055 driver since both the > MFD and CODEC drivers register the I2C device "da9055". Fix this by using > the usual pattern for MFDs with multiple I2C addresses and having the CODEC > driver reference an I2C dummy registered by the MFD. > Since I don't know which I2C address to use for the CODEC a FIXME has been > left in the MFD, this doesn't make anything any worse since the device has > never been able to load in the first place. What you're saying here is incorrect, as previously discussed at length in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-September/066027.… The two drivers did work together fine until someone in 3.12 changed the PMIC I2C Id back to just 'da9055' and broke it again. Also, I wanted to add a simple patch to tidy up the I2C Id names and avoid this but you would not agree. The codec part can be standalone and by doing what you've done here means that customers would need to now add additional, unnecessary code to be able to use just the codec driver. This is not right to me.

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