March 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH RFC v5 0/8] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 08 Mar '16

08 Mar '16

There is very little change to this series since the RFC v4, but because of the recent interest on the hdmi-codec patch I decided to send another version of the series. Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI There was also some comments about the DT ports binding for tda998x by Jean-Francois Moine. However I did not have time to come up with alternative approach. All in all my tda998x patches should be considered more as a proof of concept for the hdmi-codec part, rather than a serious attempt to get those patches in. However, all comments to those patch are more than welcome as they will help me to come up with something that could finally get merged. Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 The binding for tda998x is taken from Jean Francois' patch series[1] on the same subject. The implementation of the of-node parsing has some minor changes from my self. Here is what I think at least could or should still be done, but non of that stuff does not sounds critical right now. Missing from tda998x driver side - hdmi_codec_ops audio_startup() implementation for audio abort support - multi channel audio support (I would need specs and preferably some HW to test for this). Missing from ASoC side generic implementation: - channel_allocation handling is completely left for the video side driver, see if ASoC side could help in any way - snd_soc_jack functionality to handle hdmi cable plug/unplug events [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095596.html Best regards, Jyri Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports Jyri Sarha (7): ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Remove include/sound/tda998x.h and fix graph parsing drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 51 +++ arch/arm/boot/dts/am335x-boneblack.dts | 90 ++++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 300 ++++++++++++--- include/drm/i2c/tda998x.h | 24 +- include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 52 ++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 411 +++++++++++++++++++++ 11 files changed, 964 insertions(+), 79 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

7 27

Re: [alsa-devel] Testing SAI with mainline
by Fabio Estevam 08 Mar '16

08 Mar '16

On Tue, Mar 8, 2016 at 12:03 AM, Zidan Wang <zidan.wang(a)nxp.com> wrote: > IMX_DMATYPE_SAI is 24. Thanks for the clarification. I have submitted the patches adding SAI support for mx6sx. Now I am trying to get SAI to work on mx7d-sdb. Please see the consolidated changes below. After running audio playback the system hangs. Any ideas? Thanks diff --git a/arch/arm/boot/dts/imx7d-sdb.dts b/arch/arm/boot/dts/imx7d-sdb.dts index b267f79..17320a1 100644 --- a/arch/arm/boot/dts/imx7d-sdb.dts +++ b/arch/arm/boot/dts/imx7d-sdb.dts @@ -95,6 +95,30 @@ regulator-max-microvolt = <1800000>; }; }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "mx7d-sdb-sound-card"; + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&dailink_master>; + simple-audio-card,frame-master = <&dailink_master>; + + simple-audio-card,cpu { + sound-dai = <&sai1>; + }; + + dailink_master: simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clks IMX7D_AUDIO_MCLK_ROOT_CLK>; + }; + }; +}; + +&clks { + assigned-clocks = <&clks IMX7D_AUDIO_MCLK_ROOT_SRC>, + <&clks IMX7D_AUDIO_MCLK_ROOT_CLK>; + assigned-clock-parents = <&clks IMX7D_PLL_AUDIO_POST_DIV>; + assigned-clock-rates = <0>, <12288000>; }; &adc1 { @@ -264,14 +288,23 @@ status = "okay"; codec: wm8960@1a { + #sound-dai-cells = <0>; compatible = "wlf,wm8960"; reg = <0x1a>; - clocks = <&clks IMX7D_AUDIO_MCLK_ROOT_CLK>; - clock-names = "mclk"; wlf,shared-lrclk; }; }; +&sai1 { + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai1>; + assigned-clocks = <&clks IMX7D_SAI1_ROOT_SRC>, + <&clks IMX7D_SAI1_ROOT_CLK>; + assigned-clock-parents = <&clks IMX7D_PLL_AUDIO_POST_DIV>; + assigned-clock-rates = <0>, <36864000>; + status = "okay"; +}; + &uart1 { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_uart1>; @@ -390,6 +423,16 @@ >; }; + pinctrl_sai1: sai1grp { + fsl,pins = < + MX7D_PAD_SAI1_MCLK__SAI1_MCLK 0x1f + MX7D_PAD_ENET1_RX_CLK__SAI1_TX_BCLK 0x1f + MX7D_PAD_ENET1_CRS__SAI1_TX_SYNC 0x1f + MX7D_PAD_ENET1_COL__SAI1_TX_DATA0 0x30 + MX7D_PAD_ENET1_TX_CLK__SAI1_RX_DATA0 0x1f + >; + }; + pinctrl_uart1: uart1grp { fsl,pins = < MX7D_PAD_UART1_TX_DATA__UART1_DCE_TX 0x79 diff --git a/arch/arm/boot/dts/imx7d.dtsi b/arch/arm/boot/dts/imx7d.dtsi index b5a50e0..f21b010 100644 --- a/arch/arm/boot/dts/imx7d.dtsi +++ b/arch/arm/boot/dts/imx7d.dtsi @@ -693,6 +693,20 @@ status = "disabled"; }; + sai1: sai@308a0000 { + #sound-dai-cells = <0>; + compatible = "fsl,imx7d-sai"; + reg = <0x308a0000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX7D_SAI1_IPG_CLK>, + <&clks IMX7D_SAI1_ROOT_CLK>, + <&clks IMX7D_CLK_DUMMY>, <&clks IMX7D_CLK_DUMMY>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "rx", "tx"; + dmas = <&sdma 8 24 0>, <&sdma 9 24 0>; + status = "disabled"; + }; + i2c1: i2c@30a20000 { #address-cells = <1>; #size-cells = <0>; @@ -884,6 +898,17 @@ status = "disabled"; }; + sdma: sdma@30bd0000 { + compatible = "fsl,imx7d-sdma"; + reg = <0x30bd0000 0x10000>; + interrupts = <GIC_SPI 2 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX7D_SDMA_CORE_CLK>, + <&clks IMX7D_AHB_CHANNEL_ROOT_CLK>; + clock-names = "ipg", "ahb"; + #dma-cells = <3>; + fsl,sdma-ram-script-name = "imx/sdma/sdma-imx7d.bin"; + }; + fec1: ethernet@30be0000 { compatible = "fsl,imx7d-fec", "fsl,imx6sx-fec"; reg = <0x30be0000 0x10000>; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..6e69bf55 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -258,6 +258,7 @@ CONFIG_SND_SOC_IMX_MC13783=y CONFIG_SND_SOC_FSL_ASOC_CARD=y CONFIG_SND_SOC_CS42XX8_I2C=y CONFIG_SND_SOC_TLV320AIC3X=y +CONFIG_SND_SOC_WM8960=y CONFIG_SND_SIMPLE_CARD=y CONFIG_USB=y CONFIG_USB_EHCI_HCD=y diff --git a/drivers/clk/imx/clk-imx7d.c b/drivers/clk/imx/clk-imx7d.c index fbb6a8c..c2f0c85 100644 --- a/drivers/clk/imx/clk-imx7d.c +++ b/drivers/clk/imx/clk-imx7d.c @@ -22,6 +22,10 @@ #include "clk.h" +static u32 share_count_sai1; +static u32 share_count_sai2; +static u32 share_count_sai3; + static struct clk *clks[IMX7D_CLK_END]; static const char *arm_a7_sel[] = { "osc", "pll_arm_main_clk", "pll_enet_500m_clk", "pll_dram_main_clk", @@ -777,9 +781,12 @@ static void __init imx7d_clocks_init(struct device_node *ccm_node) clks[IMX7D_MIPI_DSI_ROOT_CLK] = imx_clk_gate2("mipi_dsi_root_clk", "mipi_dsi_post_div", base + 0x4650, 0); clks[IMX7D_MIPI_CSI_ROOT_CLK] = imx_clk_gate2("mipi_csi_root_clk", "mipi_csi_post_div", base + 0x4640, 0); clks[IMX7D_MIPI_DPHY_ROOT_CLK] = imx_clk_gate2("mipi_dphy_root_clk", "mipi_dphy_post_div", base + 0x4660, 0); - clks[IMX7D_SAI1_ROOT_CLK] = imx_clk_gate2("sai1_root_clk", "sai1_post_div", base + 0x48c0, 0); - clks[IMX7D_SAI2_ROOT_CLK] = imx_clk_gate2("sai2_root_clk", "sai2_post_div", base + 0x48d0, 0); - clks[IMX7D_SAI3_ROOT_CLK] = imx_clk_gate2("sai3_root_clk", "sai3_post_div", base + 0x48e0, 0); + clks[IMX7D_SAI1_ROOT_CLK] = imx_clk_gate2_shared("sai1_root_clk", "sai1_post_div", base + 0x48c0, 0, &share_count_sai1); + clks[IMX7D_SAI1_IPG_CLK] = imx_clk_gate2_shared("sai1_ipg_clk", "ipg_root_clk", base + 0x48c0, 0, &share_count_sai1); + clks[IMX7D_SAI2_ROOT_CLK] = imx_clk_gate2_shared("sai2_root_clk", "sai2_post_div", base + 0x48d0, 0, &share_count_sai2); + clks[IMX7D_SAI2_IPG_CLK] = imx_clk_gate2_shared("sai2_ipg_clk", "ipg_root_clk", base + 0x48d0, 0, &share_count_sai2); + clks[IMX7D_SAI3_ROOT_CLK] = imx_clk_gate2_shared("sai3_root_clk", "sai3_post_div", base + 0x48e0, 0, &share_count_sai3); + clks[IMX7D_SAI3_IPG_CLK] = imx_clk_gate2_shared("sai3_ipg_clk", "ipg_root_clk", base + 0x48e0, 0, &share_count_sai3); clks[IMX7D_SPDIF_ROOT_CLK] = imx_clk_gate2("spdif_root_clk", "spdif_post_div", base + 0x44d0, 0); clks[IMX7D_ENET1_REF_ROOT_CLK] = imx_clk_gate2("enet1_ref_root_clk", "enet1_ref_post_div", base + 0x44e0, 0); clks[IMX7D_ENET1_TIME_ROOT_CLK] = imx_clk_gate2("enet1_time_root_clk", "enet1_time_post_div", base + 0x44f0, 0); diff --git a/drivers/dma/imx-sdma.c b/drivers/dma/imx-sdma.c index 0f6fd42..620849f 100644 --- a/drivers/dma/imx-sdma.c +++ b/drivers/dma/imx-sdma.c @@ -477,6 +477,24 @@ static struct sdma_driver_data sdma_imx6q = { .script_addrs = &sdma_script_imx6q, }; +static struct sdma_script_start_addrs sdma_script_imx7d = { + .ap_2_ap_addr = 644, + .uart_2_mcu_addr = 819, + .mcu_2_app_addr = 749, + .uartsh_2_mcu_addr = 1034, + .mcu_2_shp_addr = 962, + .app_2_mcu_addr = 685, + .shp_2_mcu_addr = 893, + .spdif_2_mcu_addr = 1102, + .mcu_2_spdif_addr = 1136, +}; + +static struct sdma_driver_data sdma_imx7d = { + .chnenbl0 = SDMA_CHNENBL0_IMX35, + .num_events = 48, + .script_addrs = &sdma_script_imx7d, +}; + static const struct platform_device_id sdma_devtypes[] = { { .name = "imx25-sdma", @@ -497,6 +515,9 @@ static const struct platform_device_id sdma_devtypes[] = { .name = "imx6q-sdma", .driver_data = (unsigned long)&sdma_imx6q, }, { + .name = "imx7d-sdma", + .driver_data = (unsigned long)&sdma_imx7d, + }, { /* sentinel */ } }; @@ -509,6 +530,7 @@ static const struct of_device_id sdma_dt_ids[] = { { .compatible = "fsl,imx35-sdma", .data = &sdma_imx35, }, { .compatible = "fsl,imx31-sdma", .data = &sdma_imx31, }, { .compatible = "fsl,imx25-sdma", .data = &sdma_imx25, }, + { .compatible = "fsl,imx7d-sdma", .data = &sdma_imx7d, }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, sdma_dt_ids); @@ -1384,6 +1406,7 @@ static void sdma_issue_pending(struct dma_chan *chan) #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V1 34 #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V2 38 #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V3 41 +#define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V4 42 static void sdma_add_scripts(struct sdma_engine *sdma, const struct sdma_script_start_addrs *addr) @@ -1433,6 +1456,9 @@ static void sdma_load_firmware(const struct firmware *fw, void *context) case 3: sdma->script_number = SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V3; break; + case 4: + sdma->script_number = SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V4; + break; default: dev_err(sdma->dev, "unknown firmware version\n"); goto err_firmware; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a..62d041c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -909,7 +909,8 @@ config SND_SOC_WM8955 tristate config SND_SOC_WM8960 - tristate +tristate "Wolfson Microelectronics WM8960 CODEC" + depends on I2C config SND_SOC_WM8961 tristate diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..b60497c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -797,7 +797,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx7d-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -898,6 +899,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx7d-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids);

1 0

[alsa-devel] [PATCH v2 2/3] Provide sequencer sound card number / PID via alsa-lib
by Martin Koegler 08 Mar '16

08 Mar '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This changes adds support for exposing the card number for kernel clients as well as the PID for user client. It supports kernels with and without the required support. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- include/seq.h | 2 ++ include/sound/asequencer.h | 6 ++++-- src/seq/seq.c | 26 ++++++++++++++++++++++++++ src/seq/seq_hw.c | 12 +++++++++++- 4 files changed, 43 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index 9576822..d05940e 100644 --- a/include/seq.h +++ b/include/seq.h @@ -143,6 +143,8 @@ snd_seq_client_type_t snd_seq_client_info_get_type(const snd_seq_client_info_t * const char *snd_seq_client_info_get_name(snd_seq_client_info_t *info); int snd_seq_client_info_get_broadcast_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..3ac70fd 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -24,7 +24,7 @@ /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +356,9 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card; /* RO: card number[kernel] */ + int pid; /* RO: pid[user] */ + char reserved[56]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..4405e68 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,6 +1522,32 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** + * \brief Get the sound card number. + * \param info client_info container + * \return card number or -1 if value is not available. + * + * Only available for SND_SEQ_KERNEL_CLIENT clients. + */ +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) +{ + assert(info); + return info->card; +} + +/** + * \brief Get the owning PID. + * \param info client_info container + * \return pid or -1 if value is not available. + * + * Only available for SND_SEQ_USER_CLIENT clients. + */ +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info) +{ + assert(info); + return info->pid; +} + +/** * \brief (DEPRECATED) Get the event filter bitmap of a client_info container * \param info client_info container * \return NULL if no event filter, or pointer to event filter bitmap diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..24350b6 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int micro_version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,10 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -368,6 +373,10 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -480,6 +489,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->micro_version = SNDRV_PROTOCOL_MICRO(ver); if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { -- 2.1.4

3 8

[alsa-devel] [PATCH 0/9] ASoC: wm*: Fix enum ctl accesses
by Takashi Iwai 08 Mar '16

08 Mar '16

Hi, here is a patchset for correcting the wrong access types to the enum ctl values in various wm* codec drivers. I didn't put Cc to stable. If they might be used for 64 bit architectures, it's a real problem. In that case, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (9): ASoC: wm8753: Fix enum ctl accesses in a wrong type ASoC: wm8904: Fix enum ctl accesses in a wrong type ASoC: wm8958: Fix enum ctl accesses in a wrong type ASoC: wm8983: Fix enum ctl accesses in a wrong type ASoC: wm8985: Fix enum ctl accesses in a wrong type ASoC: wm8994: Fix enum ctl accesses in a wrong type ASoC: wm8996: Fix enum ctl accesses in a wrong type ASoC: wm9081: Fix enum ctl accesses in a wrong type ASoC: wm_adsp: Fix enum ctl accesses in a wrong type sound/soc/codecs/wm8753.c | 6 +++--- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8958-dsp2.c | 8 ++++---- sound/soc/codecs/wm8983.c | 14 +++++++------- sound/soc/codecs/wm8985.c | 14 +++++++------- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 8 ++++---- sound/soc/codecs/wm_adsp.c | 8 ++++---- 9 files changed, 34 insertions(+), 34 deletions(-) -- 2.7.2

3 19

[alsa-devel] [RFC PATCH 0/4] MST DP audio support
by libin.yang＠linux.intel.com 08 Mar '16

08 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> The patchset are tmp patches. The API between audio and gfx is still in discussion. Libin Yang (4): ALSA: hda - codec add DP MST support for connection list ALSA: hda - add DP mst verb support ALSA - hda: add devid support in acom ALSA - hda: add DP MST support drivers/gpu/drm/i915/intel_audio.c | 2 +- include/drm/i915_component.h | 2 +- include/sound/hda_i915.h | 6 +- sound/hda/hdac_i915.c | 7 +- sound/pci/hda/hda_codec.c | 217 ++++++++++++++++++++++++++++++++++--- sound/pci/hda/hda_codec.h | 8 ++ sound/pci/hda/patch_hdmi.c | 172 ++++++++++++++++++++--------- 7 files changed, 340 insertions(+), 74 deletions(-) -- 1.9.1

3 12

Re: [alsa-devel] Testing SAI with mainline
by Fabio Estevam 08 Mar '16

08 Mar '16

Hi Zidan, On Mon, Mar 7, 2016 at 11:49 PM, Zidan Wang <zidan.wang(a)nxp.com> wrote: > You should change the sdma peripheral types of SAI to 24. Thanks! That did the trick. I have also tried with 25, which is what NXP BSP uses, but it did not work. Care to explain why 24 is needed here? I will prepare some patches and submit them tomorrow so that we have SAI working on this board. Thanks for your help

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[alsa-devel] [PATCH v5 22/22] sound/usb: Use Media Controller API to share media resources
by Shuah Khan 07 Mar '16

07 Mar '16

Change ALSA driver to use Media Controller API to share media resources with DVB and V4L2 drivers on a AU0828 media device. Media Controller specific initialization is done after sound card is registered. ALSA creates Media interface and entity function graph nodes for Control, Mixer, PCM Playback, and PCM Capture devices. snd_usb_hw_params() will call Media Controller enable source handler interface to request the media resource. If resource request is granted, it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is returned. Media specific cleanup is done in usb_audio_disconnect(). Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> --- Changes since v4: 1. Renamed media_* interfaces in sound/usb to media_snd* to avoid potential conflicts with the MC interfaces. This addresses Laurent Pinchart's comment. 2. Added Acked-by from Takashi Iwai sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 +++ sound/usb/card.h | 3 + sound/usb/media.c | 318 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 72 +++++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 ++++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 11 files changed, 448 insertions(+), 5 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index a452ad7..d14bf41 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,6 +15,7 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE + select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -22,6 +23,9 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. +config SND_USB_AUDIO_USE_MEDIA_CONTROLLER + bool + config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 2d2d122..8dca3c4 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,6 +15,8 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o +snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o + snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 1f09d95..258cf70 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,6 +66,7 @@ #include "format.h" #include "power.h" #include "stream.h" +#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -561,6 +562,11 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; + if (quirk->media_device) { + /* don't want to fail when media_snd_device_create() fails */ + media_snd_device_create(chip, intf); + } + usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -617,6 +623,14 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } + /* + * Nice to check quirk && quirk->media_device + * need some special handlings. Doesn't look like + * we have access to quirk here + * Acceses mixer_list + */ + media_snd_device_delete(chip); + /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 71778ca..34a0898 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,6 +105,8 @@ struct snd_usb_endpoint { struct list_head list; }; +struct media_ctl; + struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -156,6 +158,7 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ + struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c new file mode 100644 index 0000000..6c0dfd8 --- /dev/null +++ b/sound/usb/media.c @@ -0,0 +1,318 @@ +/* + * media.c - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ + +#include <linux/init.h> +#include <linux/list.h> +#include <linux/mutex.h> +#include <linux/slab.h> +#include <linux/usb.h> + +#include <sound/pcm.h> +#include <sound/core.h> + +#include "usbaudio.h" +#include "card.h" +#include "mixer.h" +#include "media.h" + +static int media_snd_enable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->enable_source) + return mctl->media_dev->enable_source(&mctl->media_entity, + &mctl->media_pipe); + return 0; +} + +static void media_snd_disable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->disable_source) + mctl->media_dev->disable_source(&mctl->media_entity); +} + +int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream) +{ + struct media_device *mdev; + struct media_ctl *mctl; + struct device *pcm_dev = &pcm->streams[stream].dev; + u32 intf_type; + int ret = 0; + u16 mixer_pad; + struct media_entity *entity; + + mdev = subs->stream->chip->media_dev; + if (!mdev) + return -ENODEV; + + if (subs->media_ctl) + return 0; + + /* allocate media_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; + mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; + mixer_pad = 1; + } else { + intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; + mctl->media_pad.flags = MEDIA_PAD_FL_SINK; + mixer_pad = 2; + } + mctl->media_entity.name = pcm->name; + media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) + goto err1; + + mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, + MAJOR(pcm_dev->devt), + MINOR(pcm_dev->devt)); + if (!mctl->intf_devnode) { + ret = -ENOMEM; + goto err2; + } + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &mctl->intf_devnode->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + ret = -ENOMEM; + goto err3; + } + + /* create link between mixer and audio */ + media_device_for_each_entity(entity, mdev) { + switch (entity->function) { + case MEDIA_ENT_F_AUDIO_MIXER: + ret = media_create_pad_link(entity, mixer_pad, + &mctl->media_entity, 0, + MEDIA_LNK_FL_ENABLED); + if (ret) + goto err4; + break; + } + } + + subs->media_ctl = mctl; + return 0; + +err4: + media_remove_intf_link(mctl->intf_link); +err3: + media_devnode_remove(mctl->intf_devnode); +err2: + media_device_unregister_entity(&mctl->media_entity); +err1: + kfree(mctl); + return ret; +} + +void media_snd_stream_delete(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl && mctl->media_dev) { + struct media_device *mdev; + + mdev = subs->stream->chip->media_dev; + if (mdev && media_devnode_is_registered(&mdev->devnode)) { + media_devnode_remove(mctl->intf_devnode); + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + subs->media_ctl = NULL; + } +} + +int media_snd_start_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + return media_snd_enable_source(mctl); + return 0; +} + +void media_snd_stop_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + media_snd_disable_source(mctl); +} + +int media_snd_mixer_init(struct snd_usb_audio *chip) +{ + struct device *ctl_dev = &chip->card->ctl_dev; + struct media_intf_devnode *ctl_intf; + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + struct media_mixer_ctl *mctl; + u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; + int ret; + + if (!mdev) + return -ENODEV; + + ctl_intf = chip->ctl_intf_media_devnode; + if (!ctl_intf) { + ctl_intf = media_devnode_create(mdev, intf_type, 0, + MAJOR(ctl_dev->devt), + MINOR(ctl_dev->devt)); + if (!ctl_intf) + return -ENOMEM; + chip->ctl_intf_media_devnode = ctl_intf; + } + + list_for_each_entry(mixer, &chip->mixer_list, list) { + + if (mixer->media_mixer_ctl) + continue; + + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; + mctl->media_entity.name = chip->card->mixername; + mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; + mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; + mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; + media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, + mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) { + kfree(mctl); + return ret; + } + + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &ctl_intf->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + kfree(mctl); + return -ENOMEM; + } + mctl->intf_devnode = ctl_intf; + mixer->media_mixer_ctl = mctl; + } + return 0; +} + +static void media_snd_mixer_delete(struct snd_usb_audio *chip) +{ + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + + if (!mdev) + return; + + list_for_each_entry(mixer, &chip->mixer_list, list) { + struct media_mixer_ctl *mctl; + + mctl = mixer->media_mixer_ctl; + if (!mixer->media_mixer_ctl) + continue; + + if (media_devnode_is_registered(&mdev->devnode)) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + mixer->media_mixer_ctl = NULL; + } + if (media_devnode_is_registered(&mdev->devnode)) + media_devnode_remove(chip->ctl_intf_media_devnode); + chip->ctl_intf_media_devnode = NULL; +} + +int media_snd_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) +{ + struct media_device *mdev; + struct usb_device *usbdev = interface_to_usbdev(iface); + int ret; + + mdev = media_device_get_devres(&usbdev->dev); + if (!mdev) + return -ENOMEM; + if (!mdev->dev) { + /* register media device */ + mdev->dev = &usbdev->dev; + if (usbdev->product) + strlcpy(mdev->model, usbdev->product, + sizeof(mdev->model)); + if (usbdev->serial) + strlcpy(mdev->serial, usbdev->serial, + sizeof(mdev->serial)); + strcpy(mdev->bus_info, usbdev->devpath); + mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); + media_device_init(mdev); + } + if (!media_devnode_is_registered(&mdev->devnode)) { + ret = media_device_register(mdev); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't register media device. Error: %d\n", + ret); + return ret; + } + } + + /* save media device - avoid lookups */ + chip->media_dev = mdev; + + /* Create media entities for mixer and control dev */ + ret = media_snd_mixer_init(chip); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't create media mixer entities. Error: %d\n", + ret); + + /* clear saved media_dev */ + chip->media_dev = NULL; + + return ret; + } + return 0; +} + +void media_snd_device_delete(struct snd_usb_audio *chip) +{ + struct media_device *mdev = chip->media_dev; + + media_snd_mixer_delete(chip); + + if (mdev) { + if (media_devnode_is_registered(&mdev->devnode)) + media_device_unregister(mdev); + chip->media_dev = NULL; + } +} diff --git a/sound/usb/media.h b/sound/usb/media.h new file mode 100644 index 0000000..1dcdcdc --- /dev/null +++ b/sound/usb/media.h @@ -0,0 +1,72 @@ +/* + * media.h - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ +#ifndef __MEDIA_H + +#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER + +#include <media/media-device.h> +#include <media/media-entity.h> +#include <sound/asound.h> + +struct media_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad; + struct media_pipeline media_pipe; +}; + +/* + * One source pad each for SNDRV_PCM_STREAM_CAPTURE and + * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link + * to AUDIO Source +*/ +#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) + +struct media_mixer_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; + struct media_pipeline media_pipe; +}; + +int media_snd_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface); +void media_snd_device_delete(struct snd_usb_audio *chip); +int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream); +void media_snd_stream_delete(struct snd_usb_substream *subs); +int media_snd_start_pipeline(struct snd_usb_substream *subs); +void media_snd_stop_pipeline(struct snd_usb_substream *subs); +#else +static inline int media_snd_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) + { return 0; } +static inline void media_snd_device_delete(struct snd_usb_audio *chip) { } +static inline int media_snd_stream_init(struct snd_usb_substream *subs, + struct snd_pcm *pcm, int stream) + { return 0; } +static inline void media_snd_stream_delete(struct snd_usb_substream *subs) { } +static inline int media_snd_start_pipeline(struct snd_usb_substream *subs) + { return 0; } +static inline void media_snd_stop_pipeline(struct snd_usb_substream *subs) { } +#endif +#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 3417ef3..f378944 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,6 +3,8 @@ #include <sound/info.h> +struct media_mixer_ctl; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -22,6 +24,7 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; + struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9245f52..b0370d5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,6 +35,7 @@ #include "pcm.h" #include "clock.h" #include "power.h" +#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -715,10 +716,14 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; + ret = media_snd_start_pipeline(subs); + if (ret) + return ret; + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - return ret; + goto err_ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -732,22 +737,27 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - return -EINVAL; + ret = -EINVAL; + goto err_ret; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; + +err_ret: + media_snd_stop_pipeline(subs); + return ret; } /* @@ -759,6 +769,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; + media_snd_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1219,6 +1230,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; + int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1232,7 +1244,12 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - return setup_hw_info(runtime, subs); + ret = setup_hw_info(runtime, subs); + if (ret == 0) + ret = media_snd_stream_init(subs, as->pcm, direction); + if (ret) + snd_usb_autosuspend(subs->stream->chip); + return ret; } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1241,6 +1258,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); + media_snd_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c60a776..9d087b1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,6 +2886,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + .media_device = 1, \ } \ } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index c4dc577..51258a1 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,6 +36,7 @@ #include "format.h" #include "clock.h" #include "stream.h" +#include "media.h" /* * free a substream @@ -52,6 +53,7 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); + media_snd_stream_delete(subs); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b665d85..a161c7c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,6 +30,9 @@ * */ +struct media_device; +struct media_intf_devnode; + struct snd_usb_audio { int index; struct usb_device *dev; @@ -60,6 +63,8 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ + struct media_device *media_dev; + struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -110,6 +115,7 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; + bool media_device; const void *data; }; -- 2.5.0

3 8

[alsa-devel] Testing SAI with mainline
by Fabio Estevam 07 Mar '16

07 Mar '16

Hi, I am trying to get SAI to work on a mx6sx sdb board. Here are the changes I made: http://pastebin.com/aq32B24v I can see the driver getting probed: [ 2.198276] fsl-asoc-card sound: wm8962 <-> 21d4000.sai mapping ok ,but when attempting a playback it fails: $ aplay clarinet.wav [ 110.843619] fsl-sai 21d4000.sai: isr: Transmit underrun detected [ 110.849841] fsl-sai 21d4000.sai: isr: Transmit underrun detected [ 110.855995] fsl-sai 21d4000.sai: isr: Transmit underrun detected (repeats forever). Any ideas as to how to get SAI working on this platform? Zidan, How have you tested SAI when you usptream the driver? Thanks

2 2

[alsa-devel] [PATCH] ALSA: hda - Fix unexpected resume through regmap code path
by Takashi Iwai 07 Mar '16

07 Mar '16

HD-audio driver has a mechanism to trigger the runtime resume automatically at accessing the verbs. This auto-resume, however, causes the mutex deadlock when invoked from the regmap handler since the regmap keeps the mutex while auto-resuming. For avoiding that, there is some tricky check in the HDA regmap handler to return -EAGAIN error to back-off when the codec is powered down. Then the caller of regmap r/w will retry after properly turning on the codec power. This works in most cases, but there seems a slight race between the codec power check and the actual on-demand auto-resume trigger. This resulted in the lockdep splat, eventually leading to a real deadlock. This patch tries to address the race window by getting the runtime PM refcount at the check time using pm_runtime_get_if_in_use(). With this call, we can keep the power on only when the codec has been already turned on, and back off if not. For keeping the code consistency, the code touching the runtime PM is stored in hdac_device.c although it's used only locally in hdac_regmap.c. Reported-by: Jiri Slaby <jslaby(a)suse.cz> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 2 ++ sound/hda/hdac_device.c | 16 ++++++++++++ sound/hda/hdac_regmap.c | 69 ++++++++++++++++++++++++++++++++----------------- 3 files changed, 64 insertions(+), 23 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index c21c38ce7450..93e63c56f48f 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -168,11 +168,13 @@ int snd_hdac_power_up(struct hdac_device *codec); int snd_hdac_power_down(struct hdac_device *codec); int snd_hdac_power_up_pm(struct hdac_device *codec); int snd_hdac_power_down_pm(struct hdac_device *codec); +int snd_hdac_keep_power_up(struct hdac_device *codec); #else static inline int snd_hdac_power_up(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_up_pm(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; } +static inline int snd_hdac_keep_power_up(struct hdac_device *codec) { return 0; } #endif /* diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index e361024eabb6..d1a4d6973330 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -611,6 +611,22 @@ int snd_hdac_power_up_pm(struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); +/* like snd_hdac_power_up_pm(), but only increment the pm count when + * already powered up. Returns -1 if not powered up, 1 if incremented + * or 0 if unchanged. Only used in hdac_regmap.c + */ +int snd_hdac_keep_power_up(struct hdac_device *codec) +{ + if (!atomic_inc_not_zero(&codec->in_pm)) { + int ret = pm_runtime_get_if_in_use(&codec->dev); + if (!ret) + return -1; + if (ret < 0) + return 0; + } + return 1; +} + /** * snd_hdac_power_down_pm - power down the codec * @codec: the codec object diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index eb8f7c30cb09..bdbcd6b75ff6 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -21,13 +21,16 @@ #include <sound/hdaudio.h> #include <sound/hda_regmap.h> -#ifdef CONFIG_PM -#define codec_is_running(codec) \ - (atomic_read(&(codec)->in_pm) || \ - !pm_runtime_suspended(&(codec)->dev)) -#else -#define codec_is_running(codec) true -#endif +static int codec_pm_lock(struct hdac_device *codec) +{ + return snd_hdac_keep_power_up(codec); +} + +static void codec_pm_unlock(struct hdac_device *codec, int lock) +{ + if (lock == 1) + snd_hdac_power_down_pm(codec); +} #define get_verb(reg) (((reg) >> 8) & 0xfff) @@ -238,20 +241,28 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) struct hdac_device *codec = context; int verb = get_verb(reg); int err; + int pm_lock = 0; - if (!codec_is_running(codec) && verb != AC_VERB_GET_POWER_STATE) - return -EAGAIN; + if (verb != AC_VERB_GET_POWER_STATE) { + pm_lock = codec_pm_lock(codec); + if (pm_lock < 0) + return -EAGAIN; + } reg |= (codec->addr << 28); - if (is_stereo_amp_verb(reg)) - return hda_reg_read_stereo_amp(codec, reg, val); - if (verb == AC_VERB_GET_PROC_COEF) - return hda_reg_read_coef(codec, reg, val); + if (is_stereo_amp_verb(reg)) { + err = hda_reg_read_stereo_amp(codec, reg, val); + goto out; + } + if (verb == AC_VERB_GET_PROC_COEF) { + err = hda_reg_read_coef(codec, reg, val); + goto out; + } if ((verb & 0x700) == AC_VERB_SET_AMP_GAIN_MUTE) reg &= ~AC_AMP_FAKE_MUTE; err = snd_hdac_exec_verb(codec, reg, 0, val); if (err < 0) - return err; + goto out; /* special handling for asymmetric reads */ if (verb == AC_VERB_GET_POWER_STATE) { if (*val & AC_PWRST_ERROR) @@ -259,7 +270,9 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) else /* take only the actual state */ *val = (*val >> 4) & 0x0f; } - return 0; + out: + codec_pm_unlock(codec, pm_lock); + return err; } static int hda_reg_write(void *context, unsigned int reg, unsigned int val) @@ -267,6 +280,7 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) struct hdac_device *codec = context; unsigned int verb; int i, bytes, err; + int pm_lock = 0; if (codec->caps_overwriting) return 0; @@ -275,14 +289,21 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) reg |= (codec->addr << 28); verb = get_verb(reg); - if (!codec_is_running(codec) && verb != AC_VERB_SET_POWER_STATE) - return codec->lazy_cache ? 0 : -EAGAIN; + if (verb != AC_VERB_SET_POWER_STATE) { + pm_lock = codec_pm_lock(codec); + if (pm_lock < 0) + return codec->lazy_cache ? 0 : -EAGAIN; + } - if (is_stereo_amp_verb(reg)) - return hda_reg_write_stereo_amp(codec, reg, val); + if (is_stereo_amp_verb(reg)) { + err = hda_reg_write_stereo_amp(codec, reg, val); + goto out; + } - if (verb == AC_VERB_SET_PROC_COEF) - return hda_reg_write_coef(codec, reg, val); + if (verb == AC_VERB_SET_PROC_COEF) { + err = hda_reg_write_coef(codec, reg, val); + goto out; + } switch (verb & 0xf00) { case AC_VERB_SET_AMP_GAIN_MUTE: @@ -319,10 +340,12 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) reg |= (verb + i) << 8 | ((val >> (8 * i)) & 0xff); err = snd_hdac_exec_verb(codec, reg, 0, NULL); if (err < 0) - return err; + goto out; } - return 0; + out: + codec_pm_unlock(codec, pm_lock); + return err; } static const struct regmap_config hda_regmap_cfg = { -- 2.7.2

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[alsa-devel] [PATCH v3 0/7] ALSA: hda: Move chmap helpers to core
by Subhransu S. Prusty 07 Mar '16

07 Mar '16

The HDMI channel map helper APIs and control registration can be reused by ASoC skylake HDMI drivers. A common chmap object is created, channel map control callback handlers, helper APIs and data structures are moved to the core. Driver object store an instance of this chmap object to register chmap controls and program channel allocation for HDMI infoframe. Driver registers ops for specific operations. changes in v3: - Namespace managed as per hda frmwrk requirement - Address review comments in v2 Subhransu S. Prusty (7): ALSA: hda - Create common chmap object ALSA: hda - Register chmap obj as priv data instead of codec ALSA: hda - Use hdac name space for CEA spk alloc structure ALSA: hda - Add hdmi chmap verb programming ops to chmap object ALSA: hda - chmap helper args modified to use generic hdac objs. ALSA: hda - Move chmap support helpers/ops to core ALSA: hda - Use snd_hdac namespace prefix for chmap exported APIs include/sound/hda_chmap.h | 76 +++++ sound/hda/Makefile | 2 +- sound/hda/hdmi_chmap.c | 791 +++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_eld.c | 31 +- sound/pci/hda/patch_hdmi.c | 824 +++++---------------------------------------- 5 files changed, 950 insertions(+), 774 deletions(-) create mode 100644 include/sound/hda_chmap.h create mode 100644 sound/hda/hdmi_chmap.c -- 1.9.1

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