From ef61d4e6d88da80dc2b417cf8ad3c77aa94f0c8f Mon Sep 17 00:00:00 2001 From: Manoj Iyer Date: Fri, 3 Dec 2010 18:43:55 -0600 Subject: [PATCH 1/5] ALSA: hda - Enable jack sense for Thinkpad Edge 13 Added a quirk to cxt5066_cfg_tbl to enable jack sense for ThinkPad Edge 13. Reference: http://launchpad.net/bugs/685015 Signed-off-by: Manoj Iyer Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e652b34b1bd9..4ab5ea9f0530 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3110,6 +3110,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21c8, "Thinkpad Edge 11", CXT5066_IDEAPAD), From dd5a089edfa51a74692604b4b427953d8e16bc35 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 5 Dec 2010 08:43:14 -0500 Subject: [PATCH 2/5] ALSA: hda: Use position_fix=1 for Acer Aspire 5538 to enable capture on internal mic BugLink: https://launchpad.net/bugs/685161 The reporter of the bug states that he must use position_fix=1 to enable capture for the internal microphone, so set it for his machine's PCI SSID. Verified using 2.6.35 and the 2010-12-04 alsa-driver build. Reported-and-tested-by: Ralph Wabel Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21aa9b0e28f6..b030c8eba21f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2296,6 +2296,7 @@ static int azx_dev_free(struct snd_device *device) */ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), From 4b0dbdb17f846a8887e5f7fbeea2deb0703236bd Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Tue, 7 Dec 2010 18:41:35 +0200 Subject: [PATCH 3/5] ALSA: hda - Do not wrongly restrict min_channels based on ELD Commit bbbe33900d1f3c added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, it wrongly assumes that the bits 0-2 of the first byte of CEA Short Audio Descriptors mean a supported number of channels. In reality, they mean the maximum number of channels (as per CEA-861-D 7.5.2). This means that the channel count can only be used to restrict max_channels, not min_channels. Restricting min_channels causes us to deny opening the device in stereo mode if the sink only has SADs that declare larger numbers of channels (like Primare SP32 AV Processor does). Fix that by not restricting min_channels based on ELD information. Signed-off-by: Anssi Hannula Reported-by: Jean-Yves Avenard Tested-by: Jean-Yves Avenard Cc: stable@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 4 ---- sound/pci/hda/patch_hdmi.c | 1 - 2 files changed, 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index cb0c23a6b473..47ef8aa4a844 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -601,13 +601,10 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, pcm->rates = 0; pcm->formats = 0; pcm->maxbps = 0; - pcm->channels_min = -1; pcm->channels_max = 0; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; - if (a->channels < pcm->channels_min) - pcm->channels_min = a->channels; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { @@ -635,7 +632,6 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, /* restrict the parameters by the values the codec provides */ pcm->rates &= codec_pars->rates; pcm->formats &= codec_pars->formats; - pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min); pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max); pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d3e49aa5b9ec..31df7747990d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -834,7 +834,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, return -ENODEV; } else { /* fallback to the codec default */ - hinfo->channels_min = codec_pars->channels_min; hinfo->channels_max = codec_pars->channels_max; hinfo->rates = codec_pars->rates; hinfo->formats = codec_pars->formats; From 3dc86429032910bdf762adeb2969112bb303924c Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Tue, 7 Dec 2010 20:56:19 +0200 Subject: [PATCH 4/5] ALSA: hda - Always allow basic audio irrespective of ELD info Commit bbbe33900d1f3c added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, according to CEA-861-D no SAD is needed for basic audio (32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a basic audio flag in the CEA EDID Extension. The flag is not present in ELD. However, as all audio capable sinks are required to support basic audio, we can assume it to be always available. Fix allowed audio formats with sinks that have SADs (Short Audio Descriptors) which do not completely overlap with the basic audio formats (there are no reports of affected devices so far) by always assuming that basic audio is supported. Reported-by: Stephen Warren Signed-off-by: Anssi Hannula Cc: stable@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 47ef8aa4a844..009031fae2ba 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -598,21 +598,19 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, { int i; - pcm->rates = 0; - pcm->formats = 0; - pcm->maxbps = 0; - pcm->channels_max = 0; + /* assume basic audio support (the basic audio flag is not in ELD; + * however, all audio capable sinks are required to support basic + * audio) */ + pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; + pcm->formats = SNDRV_PCM_FMTBIT_S16_LE; + pcm->maxbps = 16; + pcm->channels_max = 2; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { - if (a->sample_bits & AC_SUPPCM_BITS_16) { - pcm->formats |= SNDRV_PCM_FMTBIT_S16_LE; - if (pcm->maxbps < 16) - pcm->maxbps = 16; - } if (a->sample_bits & AC_SUPPCM_BITS_20) { pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE; if (pcm->maxbps < 20) From 0bbaee3a58c379c4f7bab9635c71d7bad9c422a2 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Tue, 7 Dec 2010 21:19:23 +0200 Subject: [PATCH 5/5] ALSA: hda - Reset sample sizes and max bitrates when reading ELD When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc() is called for every SAD (Short Audio Descriptor) in the ELD data. For LPCM coding type SAD defines the supported sample sizes. For several other coding types (such as AC-3), a maximum bitrate is defined. The maximum bitrate and sample size fields are not always cleared. Therefore, if a device is unplugged and a different one is plugged in, and the coding types of some SAD positions differ between the devices, the old max_bitrate or sample_bits values will persist if the new SADs do not define those values. The leftover max_bitrate and sample_bits do not cause any issues other than wrongly showing up in eld#X.Y procfs file and kernel log. Fix that by always clearing sample_bits and max_bitrate when reading SADs. Signed-off-by: Anssi Hannula Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 009031fae2ba..4a663471dadc 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -189,6 +189,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->channels = GRAB_BITS(buf, 0, 0, 3); a->channels++; + a->sample_bits = 0; + a->max_bitrate = 0; + a->format = GRAB_BITS(buf, 0, 3, 4); switch (a->format) { case AUDIO_CODING_TYPE_REF_STREAM_HEADER: @@ -198,7 +201,6 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, case AUDIO_CODING_TYPE_LPCM: val = GRAB_BITS(buf, 2, 0, 3); - a->sample_bits = 0; for (i = 0; i < 3; i++) if (val & (1 << i)) a->sample_bits |= cea_sample_sizes[i + 1];