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

[alsa-devel] [PATCH] ASoC: Intel: fix possible acpi enumeration panic
by Kevin Strasser 16 Dec '14

16 Dec '14

A crash can occur on some platforms where adsp is enumerated but codec is not matched. Check that the codec_id string is valid before attempting to match. Signed-off-by: Kevin Strasser <kevin.strasser(a)intel.com> --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 31124aa..dd72e58 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -236,7 +236,7 @@ static struct sst_machines *sst_acpi_find_machine( struct sst_machines *mach; bool found = false; - for (mach = machines; mach->codec_id; mach++) + for (mach = machines; mach->codec_id[0]; mach++) if (ACPI_SUCCESS(acpi_get_devices(mach->codec_id, sst_acpi_mach_match, &found, NULL)) && found) -- 1.9.1

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[alsa-devel] Streaming wake-up buffer from DSP to SoC over SPI
by Dylan Reid 16 Dec '14

16 Dec '14

What is the right way to model a bursty, SPI based audio transfer? There could be hours between samples, then several seconds of audio will be ready. The codec, an rt5677, is attached to the SoC with both an IIS bus and a SPI bus. The IIS bus is used for normal audio playback and record. The SPI bus is used for transferring audio that caused a voice wake, "OK Google" in this case. When the codec detects the wake up phrase, the SoC will read out two plus seconds of buffered data over the SPI bus and continue to stream audio data over SPI for the rest of the query. Nexus 9 has a custom ioctl to access rt5677-spi and some user-space smarts to poll for more data. I was thinking of making a separate PCM device. The odd things are that it would have to originate from soc/codecs, the codec would own the codec and cpu end of the link, and the hw_ptr position would need to be queried over SPI. But it would make userspace access to it more standard looking. Thanks for any opinions or thoughts. Dylan

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[alsa-devel] Audio USB disappears suddenly
by mobidyc 16 Dec '14

16 Dec '14

Hello, don't know if I need to report my problem in alsa-dev list for the driver or linux-dev list for the usb interface, please correct me if I post in wrong mailing. sometimes, my USB audio device disappears from the system, the only way to make it appears again is to disconnect/reconnect the usb wire. Here are the traces I took : USB DAC suddenly disappears, from dmesg I can see: [13083.161157] usb 1-1.3: 1:2: usb_set_interface failed (-110) [13083.316344] usb 1-1.3: 1:2: usb_set_interface failed (-71) [13083.471927] usb 1-1.3: 1:2: usb_set_interface failed (-71) [13083.789810] usb 1-1.3: 1:2: usb_set_interface failed (-71) rmmod/modprobe snd-usb-audio: [13882.343950] usbcore: deregistering interface driver snd-usb-audio [13891.187513] usb 1-1.3: parse_audio_format_rates_v2(): unable to find clock source (clock -71) [13891.200234] usb 1-1.3: parse_audio_format_rates_v2(): unable to find clock source (clock -71) [13891.242722] usb 1-1.3: cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 [13891.242728] usb 1-1.3: 10:0: cannot get min/max values for control 2 (id 10) [13891.255466] usb 1-1.3: cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 [13891.255471] usb 1-1.3: 10:0: cannot get min/max values for control 2 (id 10) [13891.255737] usbcore: registered new interface driver snd-usb-audio [13891.276723] usb 1-1.3: cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0xa00, type = 1 [13891.284222] usb 1-1.3: cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0xa00, type = 1 [13891.290842] usb 1-1.3: cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0xa00, type = 1 [13891.297210] usb 1-1.3: cannot get ctl value: req = 0x81, wValue = 0x101, wIndex = 0xa00, type = 1 from "aplay -l" the card is seen as : card 1: U0x249c0x9313 [USB Device 0x249c:0x9313], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 after disconnected/reconnected the usb wire: card 1: UAC2 [YoungDSD UAC2], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 [16005.411254] usb 1-1.3: USB disconnect, device number 4 [16007.132442] usb 1-1.3: new high-speed USB device number 6 using ehci-pci [16007.250087] input: M2Tech YoungDSD UAC2 as /devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.3/1-1.3:1.3/0003:249C:9313.0006/input/input22 [16007.250233] hid-generic 0003:249C:9313.0006: input,hidraw2: USB HID v1.10 Device [M2Tech YoungDSD UAC2] on usb-0000:00:1a.0-1.3/input3 Is there any debug i could turn on to get more informations? Additional informations: # ./aadebug ALSA Audio Debug v0.2.0 - Wed Dec 10 19:56:00 CET 2014 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/agpl-3.0.txt Kernel ---------------------------------------------------- Linux TEVERE 3.16.3 #1 SMP Sun Nov 23 16:17:25 CET 2014 x86_64 GNU/Linux Advanced Linux Sound Architecture Driver Version k3.16.3. Loaded Modules -------------------------------------------- snd_usb_audio 105962 1 snd_hda_codec_hdmi 31141 1 snd_hda_codec_realtek 47854 1 snd_hda_codec_generic 39920 1 snd_hda_codec_realtek snd_usbmidi_lib 16134 1 snd_usb_audio snd_rawmidi 14042 1 snd_usbmidi_lib snd_hda_intel 18271 0 snd_hda_controller 15146 1 snd_hda_intel snd_hda_codec 73127 5 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_codec_generic,snd_hda_intel,snd_hda_controller snd_hwdep 4949 2 snd_usb_audio,snd_hda_codec snd_pcm 59473 6 snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel,snd_hda_controller Proc Asound ----------------------------------------------- 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7c10000 irq 45 1 [UAC2 ]: USB-Audio - YoungDSD UAC2 M2Tech YoungDSD UAC2 at usb-0000:00:1a.0-1.3, high speed 2: [ 1] : control 3: [ 1- 0]: digital audio playback 4: [ 0] : control 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0- 1]: digital audio playback 8: [ 0- 2]: digital audio capture 9: [ 0- 3]: digital audio playback 10: [ 0- 7]: digital audio playback 11: [ 0- 2]: hardware dependent 12: [ 0- 3]: hardware dependent 33: : timer 00-02: HDA Codec 2 00-03: HDA Codec 3 00-00: ALC892 Analog : ALC892 Analog : playback 1 : capture 1 00-01: ALC892 Digital : ALC892 Digital : playback 1 00-02: ALC892 Alt Analog : ALC892 Alt Analog : capture 1 00-03: HDMI 0 : HDMI 0 : playback 1 00-07: HDMI 1 : HDMI 1 : playback 1 01-00: USB Audio : USB Audio : playback 1 cat: can't open '/proc/asound/seq/clients': No such file or directory Dev Snd --------------------------------------------------- total 0 drwxr-xr-x 2 root root 60 Dec 10 18:28 by-id drwxr-xr-x 2 root root 80 Dec 10 18:28 by-path crw-rw---- 1 root audio 116, 4 Dec 10 14:01 controlC0 crw-rw---- 1 root audio 116, 2 Dec 10 18:28 controlC1 crw-rw---- 1 root audio 116, 11 Dec 10 14:01 hwC0D2 crw-rw---- 1 root audio 116, 12 Dec 10 14:01 hwC0D3 crw-rw---- 1 root audio 116, 6 Dec 10 14:01 pcmC0D0c crw-rw---- 1 root audio 116, 5 Dec 10 19:27 pcmC0D0p crw-rw---- 1 root audio 116, 7 Dec 10 14:01 pcmC0D1p crw-rw---- 1 root audio 116, 8 Dec 10 14:01 pcmC0D2c crw-rw---- 1 root audio 116, 9 Dec 10 14:01 pcmC0D3p crw-rw---- 1 root audio 116, 10 Dec 10 14:01 pcmC0D7p crw-rw---- 1 root audio 116, 3 Dec 10 19:29 pcmC1D0p crw-rw---- 1 root audio 116, 33 Dec 10 14:01 timer CPU ------------------------------------------------------- model name : Intel(R) Celeron(R) CPU G470 @ 2.00GHz cpu MHz : 1711.718 model name : Intel(R) Celeron(R) CPU G470 @ 2.00GHz cpu MHz : 1697.812 RAM ------------------------------------------------------- MemTotal: 1859156 kB SwapTotal: 0 kB Hardware -------------------------------------------------- 00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09) 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller (rev 04) Interupts ------------------------------------------------- CPU0 CPU1 0: 14 0 IO-APIC-edge timer 1: 2 0 IO-APIC-edge i8042 8: 1 0 IO-APIC-edge rtc0 9: 3 0 IO-APIC-fasteoi acpi 12: 4 0 IO-APIC-edge i8042 16: 15139141 0 IO-APIC-fasteoi ehci_hcd:usb1 23: 37 0 IO-APIC-fasteoi ehci_hcd:usb2 40: 386062 0 PCI-MSI-edge i915 41: 9653 0 PCI-MSI-edge ahci 42: 815667 0 PCI-MSI-edge eth0 43: 0 0 PCI-MSI-edge eth1 44: 0 0 PCI-MSI-edge xhci_hcd 45: 19140 0 PCI-MSI-edge snd_hda_intel NMI: 48 32 Non-maskable interrupts LOC: 4562425 1085862 Local timer interrupts SPU: 0 0 Spurious interrupts PMI: 48 32 Performance monitoring interrupts IWI: 0 0 IRQ work interrupts RTR: 1 0 APIC ICR read retries RES: 21236 17958 Rescheduling interrupts CAL: 136 121 Function call interrupts TLB: 2095 2460 TLB shootdowns ERR: 0 MIS: 0 Thanks for helping me understand this problem. -- Mobidyc

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[alsa-devel] [RFC PATCH 0/6] Those patches is used for dw_hdmi audio.
by Yakir Yang 16 Dec '14

16 Dec '14

rk3288 use Designware HDMI IP, Designware HDMI supports four interfaces to config hdmi audio(I2S, S/PDIF, Generic Parallel Audio, AHB Audio DMA), but rk3288 only support two ways to config hdmi audio(I2S, S/PDIF), So we take I2S as hdmi audio operation interfaces, and then treat hdmi audio as an standard ALSA devices(only got platback function). Four things we have done: 1. modify the audio part of dw_hdmi drivers for rk3288 platform specific. 2. add codec driver for hdmi audio, and creat codec device dynamically in hdmi driver probe function, using platform_register_device_full way, so that hdmi codec driver can connected with hdmi driver. 3. add sound driver for hdmi audio, creat hdmi audio sound card. 4. add dt-bings file and add hdmi_audio node to corresponding dt file. [Explanation]: those patch is depend on patch by Andy Yan drm: bridge/dw_hdmi: add rockchip rk3288 support see https://patchwork.kernel.org/patch/5442361/ Yakir Yang (6): drm: bridge/dw_hdmi: add hdmi audio config interfaces drm: bridge/bridge: add support for rk3288 hdmi-audio ASoC: dw-hdmi-audio: add codec driver for dw hdmi audio ASoC: rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for Rockchip hdmi-audio ARM: dts: rockchip: add hdmi audio nodes .../sound/rockchip,rockchip-hdmi-audio.txt | 16 + arch/arm/boot/dts/rk3288-evb.dtsi | 11 + arch/arm/boot/dts/rk3288.dtsi | 8 + drivers/gpu/drm/bridge/dw_hdmi.c | 32 +- include/drm/bridge/dw_hdmi.h | 12 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 371 +++++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 82 +++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 208 ++++++++++++ 12 files changed, 756 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-hdmi-audio.txt create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

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[alsa-devel] [PATCH] ASoC: Intel: Use proper print format for unsigned integers
by Rastislav Barlik 16 Dec '14

16 Dec '14

Use %u instead of %d format for printing unsigned integers. Also use unsigned integer for count and pages variables. Signed-off-by: Rastislav Barlik <barlik(a)zoho.com> --- sound/soc/intel/sst-haswell-dsp.c | 11 ++++++----- sound/soc/intel/sst-haswell-ipc.c | 30 +++++++++++++++--------------- sound/soc/intel/sst-haswell-pcm.c | 21 +++++++++++---------- 3 files changed, 32 insertions(+), 30 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 4b6c163..96c7201 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -88,7 +88,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, struct sst_module *mod; struct sst_module_data block_data; struct sst_module_template template; - int count; + u32 count; void __iomem *ram; /* TODO: allowed module types need to be configurable */ @@ -127,7 +127,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, if (block->size <= 0) { dev_err(dsp->dev, - "error: block %d size invalid\n", count); + "error: block %u size invalid\n", count); sst_module_free(mod); return -EINVAL; } @@ -156,7 +156,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, block_data.data = (void *)block + sizeof(*block); block_data.data_offset = block_data.data - fw->dma_buf; - dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + dev_dbg(dsp->dev, "copy firmware block %u type 0x%x " "size 0x%x ==> ram %p offset 0x%x\n", count, block->type, block->size, ram, block->ram_offset); @@ -175,7 +175,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) struct fw_module_header *module; struct sst_dsp *dsp = sst_fw->dsp; struct sst_hsw *hsw = sst_fw->private; - int ret, count; + int ret; + u32 count; /* Read the header information from the data pointer */ header = (struct fw_header *)sst_fw->dma_buf; @@ -198,7 +199,7 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) /* module */ ret = hsw_parse_module(dsp, sst_fw, module); if (ret < 0) { - dev_err(dsp->dev, "error: invalid module %d\n", count); + dev_err(dsp->dev, "error: invalid module %u\n", count); return ret; } module = (void *)module + sizeof(*module) + module->mod_size; diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index b629151..440cadd 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -577,7 +577,7 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) offset = (header & 0x1FFFFFFF) << 3; - dev_dbg(hsw->dev, "ipc: DSP is ready 0x%8.8x offset %d\n", + dev_dbg(hsw->dev, "ipc: DSP is ready 0x%8.8x offset %u\n", header, offset); /* copy data from the DSP FW ready offset */ @@ -626,7 +626,7 @@ static void hsw_notification_work(struct work_struct *work) stream->reply.stream_hw_id); sst_dsp_inbox_read(hsw->dsp, glitch, sizeof(*glitch)); - dev_err(hsw->dev, "glitch %d pos 0x%x write pos 0x%x\n", + dev_err(hsw->dev, "glitch %u pos 0x%x write pos 0x%x\n", glitch->glitch_type, glitch->present_pos, glitch->write_pos); break; @@ -878,7 +878,7 @@ static int hsw_process_notification(struct sst_hsw *hsw) case IPC_GLB_MAX_IPC_MESSAGE_TYPE: case IPC_GLB_RESTORE_CONTEXT: case IPC_GLB_SHORT_REPLY: - dev_err(hsw->dev, "error: message type %d header 0x%x\n", + dev_err(hsw->dev, "error: message type %u header 0x%x\n", type, header); break; case IPC_GLB_STREAM_MESSAGE: @@ -888,7 +888,7 @@ static int hsw_process_notification(struct sst_hsw *hsw) handled = hsw_log_message(hsw, header); break; default: - dev_err(hsw->dev, "error: unexpected type %d hdr 0x%8.8x\n", + dev_err(hsw->dev, "error: unexpected type %u hdr 0x%8.8x\n", type, header); break; } @@ -977,7 +977,7 @@ int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", + dev_err(hsw->dev, "error: can't unmute stream %u channel %u\n", stream->reply.stream_hw_id, channel); return ret; } @@ -996,7 +996,7 @@ int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, stream->mute_volume[channel]); if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", + dev_err(hsw->dev, "error: can't unmute stream %u channel %u\n", stream->reply.stream_hw_id, channel); return ret; } @@ -1076,7 +1076,7 @@ int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", + dev_err(hsw->dev, "error: failed to unmute mixer channel %u\n", channel); return ret; } @@ -1092,7 +1092,7 @@ int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, hsw->mixer_info.volume_register_address[channel]); if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", + dev_err(hsw->dev, "error: failed to unmute mixer channel %u\n", channel); return ret; } @@ -1219,7 +1219,7 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) ret = ipc_tx_message_wait(hsw, header, &stream->free_req, sizeof(stream->free_req), NULL, 0); if (ret < 0) { - dev_err(hsw->dev, "error: free stream %d failed\n", + dev_err(hsw->dev, "error: free stream %u failed\n", stream->free_req.stream_id); return -EAGAIN; } @@ -1517,7 +1517,7 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE, stream->reply.stream_hw_id, wait); if (ret < 0) - dev_err(hsw->dev, "error: failed to pause stream %d\n", + dev_err(hsw->dev, "error: failed to pause stream %u\n", stream->reply.stream_hw_id); return ret; @@ -1533,7 +1533,7 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME, stream->reply.stream_hw_id, wait); if (ret < 0) - dev_err(hsw->dev, "error: failed to resume stream %d\n", + dev_err(hsw->dev, "error: failed to resume stream %u\n", stream->reply.stream_hw_id); return ret; @@ -1551,7 +1551,7 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) while (stream->running && tries--) msleep(1); if (!tries) { - dev_err(hsw->dev, "error: reset stream %d still running\n", + dev_err(hsw->dev, "error: reset stream %u still running\n", stream->reply.stream_hw_id); return -EINVAL; } @@ -1561,7 +1561,7 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) ret = sst_hsw_stream_operations(hsw, IPC_STR_RESET, stream->reply.stream_hw_id, 1); if (ret < 0) - dev_err(hsw->dev, "error: failed to reset stream %d\n", + dev_err(hsw->dev, "error: failed to reset stream %u\n", stream->reply.stream_hw_id); return ret; } @@ -1609,7 +1609,7 @@ int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, sizeof(stream->wpos)); if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", + dev_err(hsw->dev, "error: stream %u set position %u failed\n", stream->reply.stream_hw_id, position); return ret; @@ -1670,7 +1670,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, dx->mem_info[item].size, dx->mem_info[item].source); } - dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", + dev_dbg(hsw->dev, "ipc: got %u entry numbers for state %d\n", dx->entries_no, state); memcpy(&hsw->dx, dx, sizeof(*dx)); diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 4df867c..7367326 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -278,11 +278,12 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, unsigned char *dma_area, size_t size, int pcm) { struct snd_dma_buffer *dmab = snd_pcm_get_dma_buf(substream); - int i, pages, stream = substream->stream; + int stream = substream->stream; + unsigned int i, pages; pages = snd_sgbuf_aligned_pages(size); - dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %u\n", dma_area, size, pages); for (i = 0; i < pages; i++) { @@ -290,7 +291,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, u32 pfn = snd_sgbuf_get_addr(dmab, i * PAGE_SIZE) >> PAGE_SHIFT; u32 *pg_table; - dev_dbg(rtd->dev, "pfn i %i idx %d pfn %x\n", i, idx, pfn); + dev_dbg(rtd->dev, "pfn i %u idx %u pfn %x\n", i, idx, pfn); pg_table = (u32 *)(pdata->dmab[pcm][stream].area + idx); @@ -388,7 +389,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); ret = sst_hsw_stream_set_rate(hsw, pcm_data->stream, rate); if (ret < 0) { - dev_err(rtd->dev, "error: could not set rate %d\n", rate); + dev_err(rtd->dev, "error: could not set rate %u\n", rate); return ret; } @@ -417,14 +418,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, ret = sst_hsw_stream_set_bits(hsw, pcm_data->stream, bits); if (ret < 0) { - dev_err(rtd->dev, "error: could not set bits %d\n", bits); + dev_err(rtd->dev, "error: could not set bits %u\n", bits); return ret; } /* we only support stereo atm */ channels = params_channels(params); if (channels != 2) { - dev_err(rtd->dev, "error: invalid channels %d\n", channels); + dev_err(rtd->dev, "error: invalid channels %u\n", channels); return -EINVAL; } @@ -434,14 +435,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, ret = sst_hsw_stream_set_channels(hsw, pcm_data->stream, channels); if (ret < 0) { - dev_err(rtd->dev, "error: could not set channels %d\n", + dev_err(rtd->dev, "error: could not set channels %u\n", channels); return ret; } ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); if (ret < 0) { - dev_err(rtd->dev, "error: could not allocate %d bytes for PCM %d\n", + dev_err(rtd->dev, "error: could not allocate %u bytes for PCM %d\n", params_buffer_bytes(params), ret); return ret; } @@ -558,7 +559,7 @@ static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data) pos = frames_to_bytes(runtime, (runtime->control->appl_ptr % runtime->buffer_size)); - dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + dev_dbg(rtd->dev, "PCM: App pointer %u bytes\n", pos); /* let alsa know we have play a period */ snd_pcm_period_elapsed(substream); @@ -580,7 +581,7 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream) offset = bytes_to_frames(runtime, position); ppos = sst_hsw_get_dsp_presentation_position(hsw, pcm_data->stream); - dev_dbg(rtd->dev, "PCM: DMA pointer %du bytes, pos %llu\n", + dev_dbg(rtd->dev, "PCM: DMA pointer %u bytes, pos %llu\n", position, ppos); return offset; } -- 2.1.3

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[alsa-devel] [PATCH] ALSA: hda/hdmi - apply Haswell fix-ups to Skylake display codec
by libin.yang＠intel.com 16 Dec '14

16 Dec '14

From: Libin Yang <libin.yang(a)intel.com> Skylake and Haswell have the same behavior on display audio. So this patch applys Haswell fix-ups to Skylake. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 95129c2..5f13d2d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -47,7 +47,9 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->vendor_id == 0x80862807) #define is_broadwell(codec) ((codec)->vendor_id == 0x80862808) -#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec)) +#define is_skylake(codec) ((codec)->vendor_id == 0x80862809) +#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ + || is_skylake(codec)) #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->vendor_id == 0x80862883) -- 1.9.1

2 1

[alsa-devel] [PATCH -next] ASoC: Intel: fix return value check in sst_acpi_probe()
by weiyj_lk＠163.com 15 Dec '14

15 Dec '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> In case of error, the function platform_device_register_data() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/intel/sst/sst_acpi.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 31124aa..f59972a 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -277,16 +277,16 @@ int sst_acpi_probe(struct platform_device *pdev) dev_dbg(dev, "ACPI device id: %x\n", dev_id); plat_dev = platform_device_register_data(dev, mach->pdata->platform, -1, NULL, 0); - if (plat_dev == NULL) { + if (IS_ERR(plat_dev)) { dev_err(dev, "Failed to create machine device: %s\n", mach->pdata->platform); - return -ENODEV; + return PTR_ERR(plat_dev); } /* Create platform device for sst machine driver */ mdev = platform_device_register_data(dev, mach->machine, -1, NULL, 0); - if (mdev == NULL) { + if (IS_ERR(mdev)) { dev_err(dev, "Failed to create machine device: %s\n", mach->machine); - return -ENODEV; + return PTR_ERR(mdev); } ret = sst_alloc_drv_context(&ctx, dev, dev_id);

2 1

[alsa-devel] [PATCH 1/3] ASoC: rt5677: add REGMAP_I2C and REGMAP_IRQ dependency
by Ben Zhang 15 Dec '14

15 Dec '14

The codec driver uses regmap to do i2c read/write. The codec driver started to use REGMAP_IRQ since: 5e3363ad1b7b2e1f197a3f56b01e21cb155ad454 ASoC: rt5677: add GPIO IRQ support Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6f21a76..60f9425 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -503,6 +503,8 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate + select REGMAP_I2C + select REGMAP_IRQ config SND_SOC_RT5677_SPI tristate -- 2.2.0.rc0.207.ga3a616c

3 7

[alsa-devel] [PATCH] ALSA: hda - add codec ID for Skylake display audio codec
by libin.yang＠intel.com 15 Dec '14

15 Dec '14

From: Libin Yang <libin.yang(a)intel.com> This patch adds codec ID (0x80862809) and module alias for Skylake display codec. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9dc9cf8..95129c2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3365,6 +3365,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862808, .name = "Broadwell HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862809, .name = "Skylake HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi }, @@ -3425,6 +3426,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:80862807"); MODULE_ALIAS("snd-hda-codec-id:80862808"); +MODULE_ALIAS("snd-hda-codec-id:80862809"); MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_ALIAS("snd-hda-codec-id:80862882"); MODULE_ALIAS("snd-hda-codec-id:80862883"); -- 1.9.1

2 1

[alsa-devel] [patch 1/2] ALSA: oxfw: some signedness bugs
by Dan Carpenter 15 Dec '14

15 Dec '14

This code tends to use unsigned variables by default and it causes signedness bugs when we use negative variables for error handling. The "i" and "j" variables are used to iterated over small positive values and so they should be type "int". The "len" variable doesn't *need* to be signed but it should be signed to make the code easier to read and audit. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/oxfw/oxfw-proc.c b/sound/firewire/oxfw/oxfw-proc.c index 604808e..8ba4f9f2 100644 --- a/sound/firewire/oxfw/oxfw-proc.c +++ b/sound/firewire/oxfw/oxfw-proc.c @@ -15,7 +15,7 @@ static void proc_read_formation(struct snd_info_entry *entry, struct snd_oxfw_stream_formation formation, curr; u8 *format; char flag; - unsigned int i, err; + int i, err; /* Show input. */ err = snd_oxfw_stream_get_current_formation(oxfw, diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index b77cf80..bda845a 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -61,7 +61,8 @@ static int set_stream_format(struct snd_oxfw *oxfw, struct amdtp_stream *s, u8 **formats; struct snd_oxfw_stream_formation formation; enum avc_general_plug_dir dir; - unsigned int i, err, len; + unsigned int len; + int i, err; if (s == &oxfw->tx_stream) { formats = oxfw->tx_stream_formats; diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9bc556b..67ade07 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -19,7 +19,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, .min = UINT_MAX, .max = 0, .integer = 1 }; struct snd_oxfw_stream_formation formation; - unsigned int i, err; + int i, err; for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { if (formats[i] == NULL) @@ -47,7 +47,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, const struct snd_interval *r = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_oxfw_stream_formation formation; - unsigned int i, j, err; + int i, j, err; unsigned int count, list[SND_OXFW_STREAM_FORMAT_ENTRIES] = {0}; count = 0; @@ -80,7 +80,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, static void limit_channels_and_rates(struct snd_pcm_hardware *hw, u8 **formats) { struct snd_oxfw_stream_formation formation; - unsigned int i, err; + int i, err; hw->channels_min = UINT_MAX; hw->channels_max = 0;

3 8

[alsa-devel] [PATCH v2 0/6] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 15 Dec '14

15 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Changes v1->v2 + Drop negative use count patch [Mark Brown] + Remove unnecessary debug print messages [Lars-Peter Clausen] + Rewrite iteration as for loop rather than do...while [Mark Brown] + Reorder patches to send fixes first [Mark Brown] + Simplify device tree code [Mark Brown] + Split device tree patch in two [Mark Brown] + Expand explanatory comment on channel configuration [Rajeev Kumar] Arnd: I've not forgotten about updating the spear entries Andrew Jackson (6): ASoC: dwc: Remove unnecessary debug messages and tests ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels ASoC: dwc: Reorder code in preparation for DT support ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Add documentation for I2S DT .../devicetree/bindings/sound/designware-i2s.txt | 32 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 294 ++++++++++++++------ 3 files changed, 248 insertions(+), 79 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

4 14

[alsa-devel] [PATCH v2] ASoC: rt5670: set platform data by dmi
by Bard Liao 15 Dec '14

15 Dec '14

This patch set specific data according to dmi data. Signed-off-by: Jin, Yao <yao.jin(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * rename dmi_platform_data to dmi_platform_intel_braswell --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8bf3a56..09cf100 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2549,6 +2550,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_intel_braswell[] __initconst = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2580,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_intel_braswell)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH 1/3] ASoC: codecs: wm8904: add dt ids table
by Bo Shen 15 Dec '14

15 Dec '14

From: Alexander Morozov <linux(a)meltdown.ru> Signed-off-by: Alexander Morozov <linux(a)meltdown.ru> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..23831a5 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2255,10 +2255,19 @@ static const struct i2c_device_id wm8904_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); +static const struct of_device_id wm8904_of_match[] = { + { .compatible = "wlf,wm8904", }, + { .compatible = "wlf,wm8912", }, + { .compatible = "wlf,wm8918", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8904_of_match); + static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", .owner = THIS_MODULE, + .of_match_table = wm8904_of_match, }, .probe = wm8904_i2c_probe, .remove = wm8904_i2c_remove, -- 2.1.0.24.g4109c28

2 4

[alsa-devel] [patch 2/2] ALSA: oxfw: fix detect_loud_models() return value
by Dan Carpenter 14 Dec '14

14 Dec '14

This code causes a static checker warning: sound/firewire/oxfw/oxfw.c:46 detect_loud_models() warn: signedness bug returning '(-2)' The detect_loud_models() function should return false on falure, so that we don't try to set up the loud code for hardware that doesn't support it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cf1d0b5..60e5cad 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -43,7 +43,7 @@ static bool detect_loud_models(struct fw_unit *unit) err = fw_csr_string(unit->directory, CSR_MODEL, model, sizeof(model)); if (err < 0) - return err; + return false; for (i = 0; i < ARRAY_SIZE(models); i++) { if (strcmp(models[i], model) == 0)

3 2

[alsa-devel] [PATCH] ASoC: rt5670: set platform data by dmi
by Bard Liao 12 Dec '14

12 Dec '14

This patch set specific data according to dmi data. Signed-off-by: Jin Yao <yao.jin(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8bf3a56..6074f04 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2549,6 +2550,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_data[] __initconst = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2580,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_data)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] linux-next: Tree for Dec 12 (sound/soc/codecs/rt5640)
by Randy Dunlap 12 Dec '14

12 Dec '14

On 12/11/14 23:35, Stephen Rothwell wrote: > Hi all, > > Please do not add any code destined for v3.20 to your linux-next included > trees/branches until after v3.19-rc1 is released. > > Changes since 20141211: on i386: When CONFIG_ACPI is not enabled & CONFIG_I2C is not enabled: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH) selects SND_SST_IPC_ACPI which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) ../sound/soc/codecs/rt5640.c:2252:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(rt5640_i2c_driver); ^ ../sound/soc/codecs/rt5640.c:2252:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] ../sound/soc/codecs/rt5640.c:2252:1: warning: parameter names (without types) in function declaration [enabled by default] ../sound/soc/codecs/rt5640.c:2241:26: warning: 'rt5640_i2c_driver' defined but not used [-Wunused-variable] static struct i2c_driver rt5640_i2c_driver = { ^ cc1: some warnings being treated as errors Full randconfig file is attached. Can (should) SND_SOC_INTEL_BYTCR_RT5640_MACH depend on I2C? -- ~Randy

1 0

[alsa-devel] [PATCH] ASoC: soc-dapm: Fix comparison of pointers
by Rasmus Villemoes 12 Dec '14

12 Dec '14

The idiom 'return a - b;' often used in comparison functions is wrong unless one is certain the values being compared lie in a sufficiently small range. In this case dapm_seq_compare would also return 0 if the ->dapm pointers happened to differ by a multiple of 2^32. Signed-off-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> --- Notes: I don't know if the other occurences of the idiom in this function are ok; that depends on whether the values always lie in [0, 2^31-1]. sound/soc/soc-dapm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9cedbcd..a894a7c71557 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1255,8 +1255,10 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, } if (a->reg != b->reg) return a->reg - b->reg; - if (a->dapm != b->dapm) - return (unsigned long)a->dapm - (unsigned long)b->dapm; + if (a->dapm < b->dapm) + return -1; + if (a->dapm > b->dapm) + return 1; return 0; } -- 2.1.3

2 3

[alsa-devel] [PATCH] ASoC: wm8750: Make VREF a VMID instead of OUTPUT widget
by Lars-Peter Clausen 12 Dec '14

12 Dec '14

Commit 04489eeb02a4 ("ALSA: wm8750: add missing VREF output") added the previously missing VREF widget. And while adding the widget as OUTPUT widget silences the error of the missing widget it should have rather been a VMID widget. The VREF widget is only used internally as a source for the VREF input of the Out3 Mux. Selecting that input means that the path should be kept powered up at the VREF/VMID voltage, but that there is no actual audio signal. Using a VMID widget will do this, whereas using a OUTPUT widget will simply cause the path to completely power down. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8750.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index f6847fd..eb0a164 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -323,7 +323,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("ROUT2"), SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_OUTPUT("OUT3"), - SND_SOC_DAPM_OUTPUT("VREF"), + SND_SOC_DAPM_VMID("VREF"), SND_SOC_DAPM_INPUT("LINPUT1"), SND_SOC_DAPM_INPUT("LINPUT2"), -- 1.8.0

3 2

[alsa-devel] [PATCH] ASoC: samsung: Fix non-DT use of I2S controller
by Mark Brown 12 Dec '14

12 Dec '14

The changes in commit a5a56871f804e (ASoC: samsung: add support for exynos7 I2S controller) introduce a new variant_regs structure in the driver data which is now mandatory for accessing registers. Unfortunately this is only hooked up for DT platforms so non-DT platforms like my primary development platform for audio are broken by this change and crash on boot. Since the only non-DT user of these device is s3c64xx fix this by making the standard samsung-i2s device be of type I2Sv3 and add a new I2Sv4 name to the platform data section, currently using the I2Sv5 information which should be about right. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index e1ace5270a8b..25919e090983 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1418,10 +1418,13 @@ static const struct samsung_i2s_dai_data samsung_dai_type_sec = { static struct platform_device_id samsung_i2s_driver_ids[] = { { .name = "samsung-i2s", - .driver_data = (kernel_ulong_t)&samsung_dai_type_pri, + .driver_data = (kernel_ulong_t)&i2sv3_dai_type, }, { .name = "samsung-i2s-sec", .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, + }, { + .name = "samsung-i2sv4", + .driver_data = (kernel_ulong_t)&i2sv5_dai_type_i2s1, }, {}, }; -- 2.1.3

2 6

[alsa-devel] [BUG] dmix reports channel layouts that are not supported
by wm4 11 Dec '14

11 Dec '14

Opening a dmix device (i.e. "default" on most distros), and calling snd_pcm_query_chmaps() on it, returns some surround audio channel maps, for example 5.1. Trying to set it with snd_pcm_set_chmap() will fail with ENXIO. Only setting the stereo channel map succeeds, and I suppose dmix supports stereo only. I think this should be considered a bug. dmix probably merely passes through the channel layout of the underlying device, even though it supports stereo only - this behavior makes no sense at all. Any device should strictly report supported channel maps only. To make matters worse, if you intend to play 5.1, and dmix fails in the manner described above, then it will report "FL FR NA NA NA NA" as channel map. This would require the application to pad stereo with 4 channels of silence, which requires extra code and is generally just stupid.

1 0

Re: [alsa-devel] [Patch v2 02/11] ASoC: qcom: Add device tree binding docs
by Kenneth Westfield 11 Dec '14

11 Dec '14

On Tue, December 9, 2014 7:37 am, Mark Brown wrote: > On Mon, Dec 08, 2014 at 02:01:04PM -0800, Kenneth Westfield wrote: >> From: Kenneth Westfield <kwestfie(a)codeaurora.org> >> Add documentation to the sound directory of the >> device-tree bindings for IPQ806x audio drivers. > Please don't send multiple separate changes in a single patch. One change per patch. In an effort to reduce the number of patches, would it be possible to package the DT doc with its respective driver in one patch? Or does the DT doc need to be in a patch by itself? I've seen examples of both. Also, I will combine the partial addition of each driver to its respective Kconfig/Makefile entries. For this, I believe I need to start with an empty Kconfig/Makefile in the first patch? -- Kenneth Westfield Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH 1/4] sound: oss: Remove SoftOSS documentation
by Jarkko Nikula 11 Dec '14

11 Dec '14

SoftOSS has been removed before kernel git history but not this documentation part. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- Documentation/sound/oss/README.OSS | 46 -------------------------------------- 1 file changed, 46 deletions(-) diff --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS index 4be259428a1c..ad261dd602ce 100644 --- a/Documentation/sound/oss/README.OSS +++ b/Documentation/sound/oss/README.OSS @@ -516,52 +516,6 @@ Yamaha OPL3-SA1 use the standard MSS, MPU401 and OPL3 options with these chips and to activate the card using isapnptools. -4Front Technologies SoftOSS - - SoftOSS is a software based wave table emulation which works with - any 16 bit stereo sound card. Due to its nature a fast CPU is - required (P133 is minimum). Although SoftOSS does _not_ use MMX - instructions it has proven out that recent processors (which appear - to have MMX) perform significantly better with SoftOSS than earlier - ones. For example a P166MMX beats a PPro200. SoftOSS should not be used - on 486 or 386 machines. - - The amount of CPU load caused by SoftOSS can be controlled by - selecting the CONFIG_SOFTOSS_RATE and CONFIG_SOFTOSS_VOICES - parameters properly (they will be prompted by make config). It's - recommended to set CONFIG_SOFTOSS_VOICES to 32. If you have a - P166MMX or faster (PPro200 is not faster) you can set - CONFIG_SOFTOSS_RATE to 44100 (kHz). However with slower systems it - recommended to use sampling rates around 22050 or even 16000 kHz. - Selecting too high values for these parameters may hang your - system when playing MIDI files with hight degree of polyphony - (number of concurrently playing notes). It's also possible to - decrease CONFIG_SOFTOSS_VOICES. This makes it possible to use - higher sampling rates. However using fewer voices decreases - playback quality more than decreasing the sampling rate. - - SoftOSS keeps the samples loaded on the system's RAM so much RAM is - required. SoftOSS should never be used on machines with less than 16 MB - of RAM since this is potentially dangerous (you may accidentally run out - of memory which probably crashes the machine). - - SoftOSS implements the wave table API originally designed for GUS. For - this reason all applications designed for GUS should work (at least - after minor modifications). For example gmod/xgmod and playmidi -g are - known to work. - - To work SoftOSS will require GUS compatible - patch files to be installed on the system (in /dos/ultrasnd/midi). You - can use the public domain MIDIA patchset available from several ftp - sites. - - ********************************************************************* - IMPORTANT NOTICE! The original patch set distributed with the Gravis - Ultrasound card is not in public domain (even though it's available from - some FTP sites). You should contact Voice Crystal (www.voicecrystal.com) - if you like to use these patches with SoftOSS included in OSS/Free. - ********************************************************************* - PSS based cards (AD1848 + ADSP-2115 + Echo ESC614 ASIC) Analog Devices and Echo Speech have together defined a sound card architecture based on the above chips. The DSP chip is used -- 2.1.3

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[alsa-devel] [PATCH v3 0/2] Disable the ThinkPad HW mute/level control when possible
by Andy Lutomirski 11 Dec '14

11 Dec '14

This series is being resurrected after several years, and it's quite different from before, so I'm restarting the version numbers :) ThinkPad volume and mute controls are a mess. For whatever reason, ThinkPads have mute buttons that send KEY_MUTE *and* control an invisible-to-ALSA mute switch. Some of them have volume controls that interact with this switch as well. This is a perennial source of problems. On most ThinkPads, if you press mute and then unmute using GUI controls, you have no sound, because userspace and ALSA state gets out of sync with the hardware switch. There's a separate "sound card" that exposes the hardware switch, but userspace code generally doesn't understand that. There are already a few _OSI(Linux) overrides to turn all the hardware buttons into regular buttons. Rather than quirking ACPI everywhere, just teach thinkpad-acpi to program the buttons for full software control and to disable hardware controls. That allows us to remove the ACPI quirks and have normal mute controls. This approach should be much simpler than adding even more kludgey ALSA integration for questionable gain. Tested on an X200s (with latching mute by default) and an X220 (with a mute light and toggle mute by default). Everything works as expected. Changes from v2: - Don't try to change the mute mode on IBM ThinkPads. - Further minor cleanups. - Try to restore the mode on module unload. Changes from v1: - Simplified the code a bit. - Improved suspend/hibernate behavior. Andy Lutomirski (2): thinkpad-acpi: Try to use full software mute control acpi: Remove _OSI(Linux) for ThinkPads drivers/acpi/blacklist.c | 54 ---------------- drivers/platform/x86/thinkpad_acpi.c | 116 ++++++++++++++++++++++++++++++++--- 2 files changed, 106 insertions(+), 64 deletions(-) -- 1.9.3

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Re: [alsa-devel] Laptop Asus Z99He internal speaker not working until I enable EAPD with HDA-Analizer
by Takashi Iwai 11 Dec '14

11 Dec '14

At Thu, 11 Dec 2014 11:53:16 +0300, Dmitry V. Zimin wrote: > > This patch will be applied only to the latest kernel version or to all versions of the long-term ? The commit contains a Cc to stable kernel, so that the stable kernel maintainers will pick it up. Which one to apply depends on the stable kernel maintainer. Takashi > > > Wed, 10 Dec 2014 14:45:58 +0100 от Takashi Iwai <tiwai(a)suse.de>: > >At Wed, 10 Dec 2014 15:28:09 +0300, > >Dmitry V. Zimin wrote: > >> > >> > >> Yes, it works. Thank you so much! > > > >Good to hear, the patch is queued now. > > > > > >Takashi > > > >> > >> > >> Tue, 09 Dec 2014 20:01:36 +0100 от Takashi Iwai < tiwai(a)suse.de >: > >> >At Tue, 09 Dec 2014 18:01:44 +0300, > >> >Dmitry V. Zimin wrote: > >> >> > >> >> Hello! > >> >> > >> >> I have laptop Asus Z99He and internal speaker doesn't work until I enable EAPD through HDA-Analyzer by hand. After connect and disconnect external headphones EAPD setting reset again. > >> >> How to enable it permanently? > >> >> model=eapd, laptop-eapd or many other does not change anything. > >> >> http://www.alsa-project.org/db/?f=df83e22e0c9a047db3778f394c1aa73376dfaf39 > >> >> > >> >> Also I try old Live CD with OpenELEC. He is working properly . http://www.alsa-project.org/db/?f=a5e75412c8230afe9516f0ebf6404afb18d9040c > >> > > >> >Could you try the patch below? > >> > > >> > > >> >Takashi > >> > > >> >-- 8< -- > >> >From: Takashi Iwai < tiwai(a)suse.de > > >> >Subject: [PATCH] ALSA: hda - Add EAPD fixup for ASUS Z99He laptop > >> > > >> >The same fixup to enable EAPD is needed for ASUS Z99He with AD1986A > >> >codec like another ASUS machine. > >> > > >> >Reported-by: Dmitry V. Zimin < pfzim(a)mail.ru > > >> >Cc: < stable(a)vger.kernel.org > > >> >Signed-off-by: Takashi Iwai < tiwai(a)suse.de > > >> >--- > >> > sound/pci/hda/patch_analog.c | 1 + > >> > 1 file changed, 1 insertion(+) > >> > > >> >diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c > >> >index 06275f8807a8..28815efeab06 100644 > >> >--- a/sound/pci/hda/patch_analog.c > >> >+++ b/sound/pci/hda/patch_analog.c > >> >@@ -332,6 +332,7 @@ static const struct hda_fixup ad1986a_fixups[] = { > >> > > >> > static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { > >> > SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), > >> >+ SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD), > >> > SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), > >> > SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), > >> > SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), > >> >-- > >> >2.2.0 > >> > > >> > >> > >> > >> [2 <text/html; utf-8 (base64)>] > >> > > >

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[alsa-devel] About SNDRV_PCM_HW_PARAM_TICK_TIME
by Lars-Peter Clausen 10 Dec '14

10 Dec '14

Hi, I was trying to figure out what exactly SNDRV_PCM_HW_PARAM_TICK_TIME does, but couldn't find too much. Is it safe to say that nowadays this is unused and deprecated and the only reason it is kept around is for ABI reasons? Thanks, - Lars

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