From efd6947945fe5ebcf759e762f51c4e1d867edc49 Mon Sep 17 00:00:00 2001 From: Antonio Ospite Date: Fri, 18 Mar 2011 12:47:33 +0100 Subject: ASoC: zylonite: set .codec_dai_name in initializer Fix the initialization of .codec_dai_name in zylonite_dai initializer, do not mix it with the initialization of .codec_name which is set already a few lines above. Signed-off-by: Antonio Ospite Acked-by: Eric Miao Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/zylonite.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ac577263b3e3..b6445757fc54 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -167,7 +167,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97", - .codec_name = "wm9713-hifi", + .codec_dai_name = "wm9713-hifi", .init = zylonite_wm9713_init, }, { @@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97-aux", - .codec_name = "wm9713-aux", + .codec_dai_name = "wm9713-aux", }, { .name = "WM9713 Voice", @@ -184,7 +184,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { .codec_name = "wm9713-codec", .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", - .codec_name = "wm9713-voice", + .codec_dai_name = "wm9713-voice", .ops = &zylonite_voice_ops, }, }; -- cgit v1.2.3-59-g8ed1b From 97c96c076b8ef86533f2427ea145a9cd0ec5564f Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Sat, 2 Apr 2011 10:54:47 +0300 Subject: ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare pxa2xx_pcm_hw_free frees dma channel and sets prtd->dma_ch to -1, but does not set prtd->params to NULL, so if pxa2xx_pcm_hw_params will be called immediately, it leaves prtd->dma_ch initialized with -1, and it results in oops in __pxa2xx_pcm_prepare. This bug is triggered via SDL. This patch adds check for prtd->dma_ch to __pxa2xx_pcm_prepare and cleans prtd->params, so now it works properly. Signed-off-by: Vasily Khoruzhick Signed-off-by: Mark Brown --- sound/arm/pxa2xx-pcm-lib.c | 3 +++ sound/soc/pxa/pxa2xx-pcm.c | 1 + 2 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 8808b82311b1..76e0d5695075 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -140,6 +140,9 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) if (!prtd || !prtd->params) return 0; + if (prtd->dma_ch == -1) + return -EINVAL; + DCSR(prtd->dma_ch) &= ~DCSR_RUN; DCSR(prtd->dma_ch) = 0; DCMD(prtd->dma_ch) = 0; diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 02fb66416ddc..2ce0b2d891d5 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -65,6 +65,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) if (prtd->dma_ch >= 0) { pxa_free_dma(prtd->dma_ch); prtd->dma_ch = -1; + prtd->params = NULL; } return 0; -- cgit v1.2.3-59-g8ed1b From 00b317a41c5428b13eb7e5b4bbc691b1aa7afa80 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 1 Apr 2011 14:50:44 -0600 Subject: ASoC: format_register_str: Don't clip register values wordsize is used as the textual width of a register address. regsize is used as the textual width of a register value. The assignments to these values were swapped. In the case of WM8903, which has 8-bit register addresses and 16-bit register values, this caused the register values to be clipped to 2 digits instead of the full 4. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4dda58926bc5..b76b74db0968 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -92,8 +92,8 @@ static int min_bytes_needed(unsigned long val) static int format_register_str(struct snd_soc_codec *codec, unsigned int reg, char *buf, size_t len) { - int wordsize = codec->driver->reg_word_size * 2; - int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + int wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + int regsize = codec->driver->reg_word_size * 2; int ret; char tmpbuf[len + 1]; char regbuf[regsize + 1]; @@ -132,8 +132,8 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, size_t total = 0; loff_t p = 0; - wordsize = codec->driver->reg_word_size * 2; - regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + regsize = codec->driver->reg_word_size * 2; len = wordsize + regsize + 2 + 1; -- cgit v1.2.3-59-g8ed1b From b2cb1292b1c7c73abbdc0e07ef3aab056fc2615f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 5 Apr 2011 07:55:24 +0200 Subject: ALSA: HDA: Fix dock mic for Lenovo X220-tablet Without the "thinkpad" quirk, the dock mic in Lenovo X220 tablet edition won't work. BugLink: http://bugs.launchpad.net/bugs/751033 Cc: stable@kernel.org Tested-by: James Ferguson Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 69e33869a53e..ad97d937d3a8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3035,6 +3035,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} -- cgit v1.2.3-59-g8ed1b From 49c039f071d36586ba32da75996ef339e4ab8405 Mon Sep 17 00:00:00 2001 From: Tarek Soliman Date: Mon, 4 Apr 2011 09:23:53 -0500 Subject: ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable There are many USB MIDI cables out there that have buggy firmware that reports it can do more than 4 bytes in a packet when they can only properly handle 4 This patch adds the ID of yet another one of those cables Signed-off-by: Tarek Soliman Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b4b39c0b6c9e..f9289102886a 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1301,6 +1301,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */ case USB_ID(0x15ca, 0x1806): /* Textech USB Midi Cable */ case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */ + case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */ ep->max_transfer = 4; break; /* -- cgit v1.2.3-59-g8ed1b From 1f348522844bb1f6e7b10d50b9e8aa89a2511b09 Mon Sep 17 00:00:00 2001 From: Aaron Plattner Date: Wed, 6 Apr 2011 17:19:04 -0700 Subject: ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums The MCP7x hardware computes the audio infoframe channel count automatically, but requires the audio driver to set the audio infoframe checksum manually via the Nv_VERB_SET_Info_Frame_Checksum control verb. When audio starts playing, nvhdmi_8ch_7x_pcm_prepare sets the checksum to (0x71 - chan - chanmask). For example, for 2ch audio, chan == 1 and chanmask == 0 so the checksum is set to 0x70. When audio playback finishes and the device is closed, nvhdmi_8ch_7x_pcm_close resets the channel formats, causing the channel count to revert to 8ch. Since the checksum is not reset, the hardware starts generating audio infoframes with invalid checksums. This causes some displays to blank the video. Fix this by updating the checksum and channel mask when the device is closed and also when it is first initialized. In addition, make sure that the channel mask is appropriate for an 8ch infoframe by setting it to 0x13 (FL FR LFE FC RL RR RLC RRC). Signed-off-by: Aaron Plattner Acked-by: Stephen Warren Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 70 +++++++++++++++++++++++++++++----------------- 1 file changed, 44 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 251773e45f61..715615a88a8d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1280,6 +1280,39 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec, + int channels) +{ + unsigned int chanmask; + int chan = channels ? (channels - 1) : 1; + + switch (channels) { + default: + case 0: + case 2: + chanmask = 0x00; + break; + case 4: + chanmask = 0x08; + break; + case 6: + chanmask = 0x0b; + break; + case 8: + chanmask = 0x13; + break; + } + + /* Set the audio infoframe channel allocation and checksum fields. The + * channel count is computed implicitly by the hardware. */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Channel_Allocation, chanmask); + + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Info_Frame_Checksum, + (0x71 - chan - chanmask)); +} + static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1298,6 +1331,10 @@ static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo, AC_VERB_SET_STREAM_FORMAT, 0); } + /* The audio hardware sends a channel count of 0x7 (8ch) when all the + * streams are disabled. */ + nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8); + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -1308,37 +1345,16 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { int chs; - unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id; + unsigned int dataDCC1, dataDCC2, channel_id; int i; mutex_lock(&codec->spdif_mutex); chs = substream->runtime->channels; - chan = chs ? (chs - 1) : 1; - switch (chs) { - default: - case 0: - case 2: - chanmask = 0x00; - break; - case 4: - chanmask = 0x08; - break; - case 6: - chanmask = 0x0b; - break; - case 8: - chanmask = 0x13; - break; - } dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT; dataDCC2 = 0x2; - /* set the Audio InforFrame Channel Allocation */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Channel_Allocation, chanmask); - /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) snd_hda_codec_write(codec, @@ -1413,10 +1429,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, } } - /* set the Audio Info Frame Checksum */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Info_Frame_Checksum, - (0x71 - chan - chanmask)); + nvhdmi_8ch_7x_set_info_frame_parameters(codec, chs); mutex_unlock(&codec->spdif_mutex); return 0; @@ -1512,6 +1525,11 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) spec->multiout.max_channels = 8; spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x; codec->patch_ops = nvhdmi_patch_ops_8ch_7x; + + /* Initialize the audio infoframe channel mask and checksum to something + * valid */ + nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8); + return 0; } -- cgit v1.2.3-59-g8ed1b From 262ac22d21ee2bf3e1655b2e5e45cc94b356e62f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 7 Apr 2011 11:43:00 +0200 Subject: ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E) In cases where there is only one internal mic connected to ADC 0x11, alc275_setup_dual_adc won't handle the case, so we need to add the ADC node to the array of candidates. Cc: stable@kernel.org BugLink: http://bugs.launchpad.net/bugs/752792 Reported-by: Vincenzo Pii Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12c6f4508c54..d9f1ef7dac2e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14124,7 +14124,7 @@ static hda_nid_t alc269vb_capsrc_nids[1] = { }; static hda_nid_t alc269_adc_candidates[] = { - 0x08, 0x09, 0x07, + 0x08, 0x09, 0x07, 0x11, }; #define alc269_modes alc260_modes -- cgit v1.2.3-59-g8ed1b From cd9abc7a22806f2340b853058a634b0e32a9f47c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 14:55:57 +0200 Subject: ALSA: hda - Don't query connections for widgets have no connections Fixes the kernel warnings with IDT codecs like hda_codec: connection list not available for 0x1e Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 05fcd60cc46f..a2f57e3581e0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3408,6 +3408,9 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t conn[HDA_MAX_NUM_INPUTS]; int i, nums; + if (!(get_wcaps(codec, mux) & AC_WCAP_CONN_LIST)) + return -1; + nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); for (i = 0; i < nums; i++) if (conn[i] == nid) -- cgit v1.2.3-59-g8ed1b