From f193957b0fbbba397c8bddedf158b3bf7e4850fc Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 7 Mar 2024 11:02:27 +0000 Subject: [PATCH 01/74] ASoC: wm_adsp: Fix missing mutex_lock in wm_adsp_write_ctl() wm_adsp_write_ctl() must hold the pwr_lock mutex when calling cs_dsp_get_ctl(). This was previously partially fixed by commit 781118bc2fc1 ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()") but this only put locking around the call to cs_dsp_coeff_write_ctrl(), missing the call to cs_dsp_get_ctl(). Signed-off-by: Richard Fitzgerald Fixes: 781118bc2fc1 ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()") Link: https://msgid.link/r/20240307110227.41421-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 36ea0dcdc7ab..9cb9068c0462 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -683,11 +683,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); + struct cs_dsp_coeff_ctl *cs_ctl; struct wm_coeff_ctl *ctl; int ret; mutex_lock(&dsp->cs_dsp.pwr_lock); + cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); mutex_unlock(&dsp->cs_dsp.pwr_lock); From fb9f8125ed9d9b8e11f309a7dbfbe7b40de48fba Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:07:58 +0200 Subject: [PATCH 02/74] ASoC: SOF: Add dsp_max_burst_size_in_ms member to snd_sof_pcm_stream The dsp_max_burst_size_in_ms can be used to save the length of the maximum burst size in ms the host DMA will use. Platform code can place constraint using this to avoid user space requesting too small ALSA buffer which will result xruns. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 9ea2ac5adac7..04e5cb2c70a7 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -322,6 +322,7 @@ struct snd_sof_pcm_stream { struct work_struct period_elapsed_work; struct snd_soc_dapm_widget_list *list; /* list of connected DAPM widgets */ bool d0i3_compatible; /* DSP can be in D0I3 when this pcm is opened */ + unsigned int dsp_max_burst_size_in_ms; /* The maximum size of the host DMA burst in ms */ /* * flag to indicate that the DSP pipelines should be kept * active or not while suspending the stream From 842bb8b62cc6f3546d61eb63115b32ebc6dd4a87 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:07:59 +0200 Subject: [PATCH 03/74] ASoC: SOF: ipc4-topology: Save the DMA maximum burst size for PCMs When setting up the pcm widget, save the DSP buffer size (in ms) for platform code to place a constraint on playback. On playback the DMA will fill the buffer on start and if the period size is smaller it will immediately overrun. On capture the DMA will move data in 1ms bursts. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index da4a83afb87a..bb4cf6dd1e18 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -412,8 +412,9 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) struct sof_ipc4_available_audio_format *available_fmt; struct snd_soc_component *scomp = swidget->scomp; struct sof_ipc4_copier *ipc4_copier; + struct snd_sof_pcm *spcm; int node_type = 0; - int ret; + int ret, dir; ipc4_copier = kzalloc(sizeof(*ipc4_copier), GFP_KERNEL); if (!ipc4_copier) @@ -447,6 +448,25 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) } dev_dbg(scomp->dev, "host copier '%s' node_type %u\n", swidget->widget->name, node_type); + spcm = snd_sof_find_spcm_comp(scomp, swidget->comp_id, &dir); + if (!spcm) + goto skip_gtw_cfg; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + struct snd_sof_pcm_stream *sps = &spcm->stream[dir]; + + sof_update_ipc_object(scomp, &sps->dsp_max_burst_size_in_ms, + SOF_COPIER_DEEP_BUFFER_TOKENS, + swidget->tuples, + swidget->num_tuples, sizeof(u32), 1); + /* Set default DMA buffer size if it is not specified in topology */ + if (!sps->dsp_max_burst_size_in_ms) + sps->dsp_max_burst_size_in_ms = SOF_IPC4_MIN_DMA_BUFFER_SIZE; + } else { + /* Capture data is copied from DSP to host in 1ms bursts */ + spcm->stream[dir].dsp_max_burst_size_in_ms = 1; + } + skip_gtw_cfg: ipc4_copier->gtw_attr = kzalloc(sizeof(*ipc4_copier->gtw_attr), GFP_KERNEL); if (!ipc4_copier->gtw_attr) { From fe76d2e75a6da97edd2b4ec5cfb9efd541be087a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:00 +0200 Subject: [PATCH 04/74] ASoC: SOF: Intel: hda-pcm: Use dsp_max_burst_size_in_ms to place constraint If the PCM have the dsp_max_burst_size_in_ms set then place a constraint to limit the minimum buffer time to avoid xruns caused by DMA bursts spinning on the ALSA buffer. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-pcm.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 18f07364d219..69fefcd1abc5 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -259,6 +259,27 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32); + /* + * The dsp_max_burst_size_in_ms is the length of the maximum burst size + * of the host DMA in the ALSA buffer. + * + * On playback start the DMA will transfer dsp_max_burst_size_in_ms + * amount of data in one initial burst to fill up the host DMA buffer. + * Consequent DMA burst sizes are shorter and their length can vary. + * To make sure that userspace allocate large enough ALSA buffer we need + * to place a constraint on the buffer time. + * + * On capture the DMA will transfer 1ms chunks. + * + * Exact dsp_max_burst_size_in_ms constraint is racy, so set the + * constraint to a minimum of 2x dsp_max_burst_size_in_ms. + */ + if (spcm->stream[direction].dsp_max_burst_size_in_ms) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_TIME, + spcm->stream[direction].dsp_max_burst_size_in_ms * USEC_PER_MSEC * 2, + UINT_MAX); + /* binding pcm substream to hda stream */ substream->runtime->private_data = &dsp_stream->hstream; return 0; From 67b182bea08a8d1092b91b57aefdfe420fce1634 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:01 +0200 Subject: [PATCH 05/74] ASoC: SOF: Intel: hda: Implement get_stream_position (Linear Link Position) When the Linear Link Position is not available in firmware SRAM window we use the host accessible position registers to read it. The address of the PPLCLLPL/U registers depend on the number of streams (playback+capture). At probe time the pplc_addr is calculated for each stream and we can use it to read the LLP without the need of address re-calculation. Set the get_stream_position callback in sof_hda_common_ops for all platforms: The callback is used for IPC4 delay calculations only but the register is a generic HDA register, not tied to any specific IPC version. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-common-ops.c | 2 ++ sound/soc/sof/intel/hda-stream.c | 32 ++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 3 +++ 3 files changed, 37 insertions(+) diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 2b385cddc385..80a69599a8c3 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -57,6 +57,8 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_pointer = hda_dsp_pcm_pointer, .pcm_ack = hda_dsp_pcm_ack, + .get_stream_position = hda_dsp_get_stream_llp, + /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index b387b1a69d7e..48ea187f7230 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1063,3 +1063,35 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, return pos; } + +/** + * hda_dsp_get_stream_llp - Retrieve the LLP (Linear Link Position) of the stream + * @sdev: SOF device + * @component: ASoC component + * @substream: PCM substream + * + * Returns the raw Linear Link Position value + */ +u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + u32 llp_l, llp_u; + + /* + * The pplc_addr have been calculated during probe in + * hda_dsp_stream_init(): + * pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + + * SOF_HDA_PPLC_BASE + + * SOF_HDA_PPLC_MULTI * total_stream + + * SOF_HDA_PPLC_INTERVAL * stream_index + * + * Use this pre-calculated address to avoid repeated re-calculation. + */ + llp_l = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); + llp_u = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); + + return ((u64)llp_u << 32) | llp_l; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index b36eb7c78913..9d26cad785fe 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -662,6 +662,9 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, int direction, bool can_sleep); +u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); From 4374f698d7d9f849b66f3fa8f7a64f0bc1a53d7f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:02 +0200 Subject: [PATCH 06/74] ASoC: SOF: Intel: mtl/lnl: Use the generic get_stream_position callback Drop the MTL mtl_dsp_get_stream_hda_link_position() function and related defines since it can only work on platforms which have 19 streams because of the use of 0x948 as base offset for the LLP registers. The generic hda_dsp_get_stream_hda_link_position() takes the number of streams into consideration when reading the LLP registers for the stream and can handle different HDA configurations. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 2 -- sound/soc/sof/intel/mtl.c | 14 -------------- sound/soc/sof/intel/mtl.h | 10 ---------- 3 files changed, 26 deletions(-) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 7ae017a00184..d1c73d407e68 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -134,8 +134,6 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) sof_lnl_ops.runtime_resume = lnl_hda_dsp_runtime_resume; } - sof_lnl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - /* dsp core get/put */ sof_lnl_ops.core_get = mtl_dsp_core_get; sof_lnl_ops.core_put = mtl_dsp_core_put; diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index df05dc77b8d5..060c34988e90 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -626,18 +626,6 @@ static int mtl_dsp_disable_interrupts(struct snd_sof_dev *sdev) return mtl_enable_interrupts(sdev, false); } -u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - struct hdac_stream *hstream = substream->runtime->private_data; - u32 llp_l, llp_u; - - llp_l = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPL(hstream->index)); - llp_u = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, MTL_PPLCLLPU(hstream->index)); - return ((u64)llp_u << 32) | llp_l; -} - int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core) { const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; @@ -707,8 +695,6 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev) sof_mtl_ops.core_get = mtl_dsp_core_get; sof_mtl_ops.core_put = mtl_dsp_core_put; - sof_mtl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - sdev->private = kzalloc(sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index cc5a1f46fd09..ea8c1b83f712 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -6,12 +6,6 @@ * Copyright(c) 2020-2022 Intel Corporation. All rights reserved. */ -/* HDA Registers */ -#define MTL_PPLCLLPL_BASE 0x948 -#define MTL_PPLCLLPU_STRIDE 0x10 -#define MTL_PPLCLLPL(x) (MTL_PPLCLLPL_BASE + (x) * MTL_PPLCLLPU_STRIDE) -#define MTL_PPLCLLPU(x) (MTL_PPLCLLPL_BASE + 0x4 + (x) * MTL_PPLCLLPU_STRIDE) - /* DSP Registers */ #define MTL_HFDSSCS 0x1000 #define MTL_HFDSSCS_SPA_MASK BIT(16) @@ -103,9 +97,5 @@ int mtl_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id); void mtl_ipc_dump(struct snd_sof_dev *sdev); -u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream); - int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core); int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core); From ce2faa9a180c1984225689b6b1cb26045f8b7470 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:03 +0200 Subject: [PATCH 07/74] ASoC: SOF: Introduce a new callback pair to be used for PCM delay reporting For delay calculation we need two information: Number of bytes transferred between the DSP and host memory (ALSA buffer) Number of frames transferred between the DSP and external device (link/codec/DMIC/etc). The reason for the different units (bytes vs frames) on host and dai side is that the format on the dai side is decided by the firmware and might not be the same as on the host side, thus the expectation is that the counter reflects the number of frames. The kernel know the host side format and in there we have access to the DMA position which is in bytes. In a simplified way, the DSP caused delay is the difference between the two counters. The existing get_stream_position callback is defined to retrieve the frame counter on the DAI side but it's name is too generic to be intuitive and makes it hard to define a callback for the host side. This patch introduces a new set of callbacks to replace the get_stream_position and define the host side equivalent: get_dai_frame_counter get_host_byte_counter Subsequent patches will remove the old callback. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ops.h | 24 ++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 21 +++++++++++++++++++++ 2 files changed, 45 insertions(+) diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 6cf21e829e07..d83cd771015c 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -533,6 +533,30 @@ static inline u64 snd_sof_pcm_get_stream_position(struct snd_sof_dev *sdev, return 0; } +static inline u64 +snd_sof_pcm_get_dai_frame_counter(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->get_dai_frame_counter) + return sof_ops(sdev)->get_dai_frame_counter(sdev, component, + substream); + + return 0; +} + +static inline u64 +snd_sof_pcm_get_host_byte_counter(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->get_host_byte_counter) + return sof_ops(sdev)->get_host_byte_counter(sdev, component, + substream); + + return 0; +} + /* machine driver */ static inline int snd_sof_machine_register(struct snd_sof_dev *sdev, void *pdata) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index d453a4ce3b21..91043f177dfa 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -270,6 +270,27 @@ struct snd_sof_dsp_ops { struct snd_soc_component *component, struct snd_pcm_substream *substream); /* optional */ + /* + * optional callback to retrieve the number of frames left/arrived from/to + * the DSP on the DAI side (link/codec/DMIC/etc). + * + * The callback is used when the firmware does not provide this information + * via the shared SRAM window and it can be retrieved by host. + */ + u64 (*get_dai_frame_counter)(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); /* optional */ + + /* + * Optional callback to retrieve the number of bytes left/arrived from/to + * the DSP on the host side (bytes between host ALSA buffer and DSP). + * + * The callback is needed for ALSA delay reporting. + */ + u64 (*get_host_byte_counter)(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); /* optional */ + /* host read DSP stream data */ int (*ipc_msg_data)(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sps, From fd6f6a0632bc891673490bf4a92304172251825c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:04 +0200 Subject: [PATCH 08/74] ASoC: SOF: Intel: Set the dai/host get frame/byte counter callbacks Add implementation for reading the LDP (Linear DMA Position) to be used as get_host_byte_counter(). The LDP is counting the number of bytes moved between the DSP and host memory. Set the get_dai_frame_counter to hda_dsp_get_stream_llp, which is counting the frames on the link side of the DSP. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-common-ops.c | 2 ++ sound/soc/sof/intel/hda-stream.c | 31 ++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 3 +++ 3 files changed, 36 insertions(+) diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 80a69599a8c3..4d7ea18604ee 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -58,6 +58,8 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_ack = hda_dsp_pcm_ack, .get_stream_position = hda_dsp_get_stream_llp, + .get_dai_frame_counter = hda_dsp_get_stream_llp, + .get_host_byte_counter = hda_dsp_get_stream_ldp, /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 48ea187f7230..8504a4f27b60 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1095,3 +1095,34 @@ u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, return ((u64)llp_u << 32) | llp_l; } + +/** + * hda_dsp_get_stream_ldp - Retrieve the LDP (Linear DMA Position) of the stream + * @sdev: SOF device + * @component: ASoC component + * @substream: PCM substream + * + * Returns the raw Linear Link Position value + */ +u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + u32 ldp_l, ldp_u; + + /* + * The pphc_addr have been calculated during probe in + * hda_dsp_stream_init(): + * pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + + * SOF_HDA_PPHC_BASE + + * SOF_HDA_PPHC_INTERVAL * stream_index + * + * Use this pre-calculated address to avoid repeated re-calculation. + */ + ldp_l = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPL); + ldp_u = readl(hext_stream->pphc_addr + AZX_REG_PPHCLDPU); + + return ((u64)ldp_u << 32) | ldp_l; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 9d26cad785fe..81a1d4606d3c 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -665,6 +665,9 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, struct snd_soc_component *component, struct snd_pcm_substream *substream); +u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev, + struct snd_soc_component *component, + struct snd_pcm_substream *substream); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); From 37679a1bd372c8308a3faccf3438c9df642565b3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:05 +0200 Subject: [PATCH 09/74] ASoC: SOF: ipc4-pcm: Use the snd_sof_pcm_get_dai_frame_counter() for pcm_delay Switch to the new callback to retrieve the DAI (link) frame counter. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-9-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 0f332c8cdbe6..d0795f77cc15 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -897,11 +897,12 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, } /* - * HDaudio links don't support the LLP counter reported by firmware - * the link position is read directly from hardware registers. + * If the LLP counter is not reported by firmware in the SRAM window + * then read the dai (link) position via host accessible means if + * available. */ if (!time_info->llp_offset) { - tmp_ptr = snd_sof_pcm_get_stream_position(sdev, component, substream); + tmp_ptr = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); if (!tmp_ptr) return 0; } else { From 4ab6c38c664442c1fc9911eb3c5c6953d3dbcca5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:06 +0200 Subject: [PATCH 10/74] ASoC: SOF: Intel: hda-common-ops: Do not set the get_stream_position callback The get_stream_position has been replaced by get_dai_frame_counter, it should not be set to allow it to be dropped from core code. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-10-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-common-ops.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 4d7ea18604ee..d71bb66b9991 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -57,7 +57,6 @@ struct snd_sof_dsp_ops sof_hda_common_ops = { .pcm_pointer = hda_dsp_pcm_pointer, .pcm_ack = hda_dsp_pcm_ack, - .get_stream_position = hda_dsp_get_stream_llp, .get_dai_frame_counter = hda_dsp_get_stream_llp, .get_host_byte_counter = hda_dsp_get_stream_ldp, From 07007b8ac42cffc23043d00e56b0f67a75dc4b22 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:07 +0200 Subject: [PATCH 11/74] ASoC: SOF: Remove the get_stream_position callback The get_stream_position has been replaced by get_dai_frame_counter and all related code can be dropped form the core. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-11-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ops.h | 10 ---------- sound/soc/sof/sof-priv.h | 9 --------- 2 files changed, 19 deletions(-) diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index d83cd771015c..3cd748e13460 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -523,16 +523,6 @@ static inline int snd_sof_pcm_platform_ack(struct snd_sof_dev *sdev, return 0; } -static inline u64 snd_sof_pcm_get_stream_position(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - if (sof_ops(sdev) && sof_ops(sdev)->get_stream_position) - return sof_ops(sdev)->get_stream_position(sdev, component, substream); - - return 0; -} - static inline u64 snd_sof_pcm_get_dai_frame_counter(struct snd_sof_dev *sdev, struct snd_soc_component *component, diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 91043f177dfa..d3c436f82604 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -261,15 +261,6 @@ struct snd_sof_dsp_ops { /* pcm ack */ int (*pcm_ack)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */ - /* - * optional callback to retrieve the link DMA position for the substream - * when the position is not reported in the shared SRAM windows but - * instead from a host-accessible hardware counter. - */ - u64 (*get_stream_position)(struct snd_sof_dev *sdev, - struct snd_soc_component *component, - struct snd_pcm_substream *substream); /* optional */ - /* * optional callback to retrieve the number of frames left/arrived from/to * the DSP on the DAI side (link/codec/DMIC/etc). From 31d2874d083ba6cc2a4f4b26dab73c3be1c92658 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:08 +0200 Subject: [PATCH 12/74] ASoC: SOF: ipc4-pcm: Move struct sof_ipc4_timestamp_info definition locally The sof_ipc4_timestamp_info is only used by ipc4-pcm.c internally, it should not be in a generic header implying that it might be used elsewhere. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-12-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 14 ++++++++++++++ sound/soc/sof/ipc4-priv.h | 14 -------------- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index d0795f77cc15..2d7295221884 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -15,6 +15,20 @@ #include "ipc4-topology.h" #include "ipc4-fw-reg.h" +/** + * struct sof_ipc4_timestamp_info - IPC4 timestamp info + * @host_copier: the host copier of the pcm stream + * @dai_copier: the dai copier of the pcm stream + * @stream_start_offset: reported by fw in memory window + * @llp_offset: llp offset in memory window + */ +struct sof_ipc4_timestamp_info { + struct sof_ipc4_copier *host_copier; + struct sof_ipc4_copier *dai_copier; + u64 stream_start_offset; + u32 llp_offset; +}; + static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state, struct ipc4_pipeline_set_state_data *trigger_list) { diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index f3b908b093f9..afed618a15f0 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -92,20 +92,6 @@ struct sof_ipc4_fw_data { struct mutex pipeline_state_mutex; /* protect pipeline triggers, ref counts and states */ }; -/** - * struct sof_ipc4_timestamp_info - IPC4 timestamp info - * @host_copier: the host copier of the pcm stream - * @dai_copier: the dai copier of the pcm stream - * @stream_start_offset: reported by fw in memory window - * @llp_offset: llp offset in memory window - */ -struct sof_ipc4_timestamp_info { - struct sof_ipc4_copier *host_copier; - struct sof_ipc4_copier *dai_copier; - u64 stream_start_offset; - u32 llp_offset; -}; - extern const struct sof_ipc_fw_loader_ops ipc4_loader_ops; extern const struct sof_ipc_tplg_ops ipc4_tplg_ops; extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops; From 55ca6ca227bfc5a8d0a0c2c5d6e239777226a604 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:09 +0200 Subject: [PATCH 13/74] ASoC: SOF: ipc4-pcm: Combine the SOF_IPC4_PIPE_PAUSED cases in pcm_trigger The SNDRV_PCM_TRIGGER_PAUSE_PUSH does not need to be a separate case, it can be handled along with STOP and SUSPEND Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-13-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 2d7295221884..4e41b16d3205 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -478,14 +478,12 @@ static int sof_ipc4_pcm_trigger(struct snd_soc_component *component, /* determine the pipeline state */ switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - state = SOF_IPC4_PIPE_PAUSED; - break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: state = SOF_IPC4_PIPE_RUNNING; break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: state = SOF_IPC4_PIPE_PAUSED; From 3ce3bc36d91510389955b47e36ea4c4e94fcbdd3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:10 +0200 Subject: [PATCH 14/74] ASoC: SOF: ipc4-pcm: Invalidate the stream_start_offset in PAUSED state When the final state is SOF_IPC4_PIPE_PAUSED, it is possible that the stream will be restarted (resume or start) in which case we need to update the offset from the firmware. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-14-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 4e41b16d3205..905dbc4852b1 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -437,8 +437,19 @@ static int sof_ipc4_trigger_pipelines(struct snd_soc_component *component, } /* return if this is the final state */ - if (state == SOF_IPC4_PIPE_PAUSED) + if (state == SOF_IPC4_PIPE_PAUSED) { + struct sof_ipc4_timestamp_info *time_info; + + /* + * Invalidate the stream_start_offset to make sure that it is + * going to be updated if the stream resumes + */ + time_info = spcm->stream[substream->stream].private; + if (time_info) + time_info->stream_start_offset = SOF_IPC4_INVALID_STREAM_POSITION; + goto free; + } skip_pause_transition: /* else set the RUNNING/RESET state in the DSP */ ret = sof_ipc4_set_multi_pipeline_state(sdev, state, trigger_list); From 77165bd955d55114028b06787a530b8f9220e4b0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:11 +0200 Subject: [PATCH 15/74] ASoC: SOF: sof-pcm: Add pointer callback to sof_ipc_pcm_ops The IPC specific pointer callback can be used when additional or custom handling is needed during the pointer calculation, like executing a delay calculation at the same time to minimize drift between the reported pointer and the calculated delay. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-15-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 8 ++++++++ sound/soc/sof/sof-audio.h | 8 +++++++- 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 33d576b17647..f03cee94bce6 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -388,13 +388,21 @@ static snd_pcm_uframes_t sof_pcm_pointer(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm); struct snd_sof_pcm *spcm; snd_pcm_uframes_t host, dai; + int ret = -EOPNOTSUPP; /* nothing to do for BE */ if (rtd->dai_link->no_pcm) return 0; + if (pcm_ops && pcm_ops->pointer) + ret = pcm_ops->pointer(component, substream, &host); + + if (ret != -EOPNOTSUPP) + return ret ? ret : host; + /* use dsp ops pointer callback directly if set */ if (sof_ops(sdev)->pcm_pointer) return sof_ops(sdev)->pcm_pointer(sdev, substream); diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 04e5cb2c70a7..86bbb531e142 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -103,7 +103,10 @@ struct snd_sof_dai_config_data { * additional memory in the SOF PCM stream structure * @pcm_free: Function pointer for PCM free that can be used for freeing any * additional memory in the SOF PCM stream structure - * @delay: Function pointer for pcm delay calculation + * @pointer: Function pointer for pcm pointer + * Note: the @pointer callback may return -EOPNOTSUPP which should be + * handled in a same way as if the callback is not provided + * @delay: Function pointer for pcm delay reporting * @reset_hw_params_during_stop: Flag indicating whether the hw_params should be reset during the * STOP pcm trigger * @ipc_first_on_start: Send IPC before invoking platform trigger during @@ -124,6 +127,9 @@ struct sof_ipc_pcm_ops { int (*dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); int (*pcm_setup)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm); void (*pcm_free)(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm); + int (*pointer)(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + snd_pcm_uframes_t *pointer); snd_pcm_sframes_t (*delay)(struct snd_soc_component *component, struct snd_pcm_substream *substream); bool reset_hw_params_during_stop; From 0ea06680dfcb4464ac6c05968433d060efb44345 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:12 +0200 Subject: [PATCH 16/74] ASoC: SOF: ipc4-pcm: Correct the delay calculation This patch improves the delay calculation by relying on the LLP (Linear Link Position) on the DAI side and the LDP (Linear Data Pointer) on the host side. The LDP provides the same DMA position as LPIB, but with a linear count instead of a position in the ALSA ring buffer. The LDP values are provided in bytes and must be converted to frames. The difference in units means that the host counter will wrap earlier than the LLP. We need to wrap the LLP at the same boundary as the host counter. The ASoC framework relies on separate pointer and delay callback. Measurement errors can be reduced by processing all the counter values in the pointer callback. The delay value is stored, and will be reported to higher levels in the delay callback. For playback, the firmware provides a stream_start offset to handle mixing/pause usages, where the DAI might have started earlier than the PCM device. The delay calculation must be special-cased when the link counter has not reached the start offset value, i.e. no valid audio has left the DSP. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-16-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 161 +++++++++++++++++++++++++++++++-------- 1 file changed, 128 insertions(+), 33 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 905dbc4852b1..e915f9f87a6c 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -19,14 +19,22 @@ * struct sof_ipc4_timestamp_info - IPC4 timestamp info * @host_copier: the host copier of the pcm stream * @dai_copier: the dai copier of the pcm stream - * @stream_start_offset: reported by fw in memory window + * @stream_start_offset: reported by fw in memory window (converted to frames) + * @stream_end_offset: reported by fw in memory window (converted to frames) * @llp_offset: llp offset in memory window + * @boundary: wrap boundary should be used for the LLP frame counter + * @delay: Calculated and stored in pointer callback. The stored value is + * returned in the delay callback. */ struct sof_ipc4_timestamp_info { struct sof_ipc4_copier *host_copier; struct sof_ipc4_copier *dai_copier; u64 stream_start_offset; + u64 stream_end_offset; u32 llp_offset; + + u64 boundary; + snd_pcm_sframes_t delay; }; static int sof_ipc4_set_multi_pipeline_state(struct snd_sof_dev *sdev, u32 state, @@ -726,6 +734,10 @@ static int sof_ipc4_pcm_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm if (abi_version < SOF_IPC4_FW_REGS_ABI_VER) support_info = false; + /* For delay reporting the get_host_byte_counter callback is needed */ + if (!sof_ops(sdev) || !sof_ops(sdev)->get_host_byte_counter) + support_info = false; + for_each_pcm_streams(stream) { pipeline_list = &spcm->stream[stream].pipeline_list; @@ -858,7 +870,6 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, struct sof_ipc4_copier *host_copier = time_info->host_copier; struct sof_ipc4_copier *dai_copier = time_info->dai_copier; struct sof_ipc4_pipeline_registers ppl_reg; - u64 stream_start_position; u32 dai_sample_size; u32 ch, node_index; u32 offset; @@ -875,38 +886,51 @@ static int sof_ipc4_get_stream_start_offset(struct snd_sof_dev *sdev, if (ppl_reg.stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) return -EINVAL; - stream_start_position = ppl_reg.stream_start_offset; ch = dai_copier->data.out_format.fmt_cfg; ch = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(ch); dai_sample_size = (dai_copier->data.out_format.bit_depth >> 3) * ch; - /* convert offset to sample count */ - do_div(stream_start_position, dai_sample_size); - time_info->stream_start_offset = stream_start_position; + + /* convert offsets to frame count */ + time_info->stream_start_offset = ppl_reg.stream_start_offset; + do_div(time_info->stream_start_offset, dai_sample_size); + time_info->stream_end_offset = ppl_reg.stream_end_offset; + do_div(time_info->stream_end_offset, dai_sample_size); + + /* + * Calculate the wrap boundary need to be used for delay calculation + * The host counter is in bytes, it will wrap earlier than the frames + * based link counter. + */ + time_info->boundary = div64_u64(~((u64)0), + frames_to_bytes(substream->runtime, 1)); + /* Initialize the delay value to 0 (no delay) */ + time_info->delay = 0; return 0; } -static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, - struct snd_pcm_substream *substream) +static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + snd_pcm_uframes_t *pointer) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_timestamp_info *time_info; struct sof_ipc4_llp_reading_slot llp; - snd_pcm_uframes_t head_ptr, tail_ptr; + snd_pcm_uframes_t head_cnt, tail_cnt; struct snd_sof_pcm_stream *stream; + u64 dai_cnt, host_cnt, host_ptr; struct snd_sof_pcm *spcm; - u64 tmp_ptr; int ret; spcm = snd_sof_find_spcm_dai(component, rtd); if (!spcm) - return 0; + return -EOPNOTSUPP; stream = &spcm->stream[substream->stream]; time_info = stream->private; if (!time_info) - return 0; + return -EOPNOTSUPP; /* * stream_start_offset is updated to memory window by FW based on @@ -916,46 +940,116 @@ static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, if (time_info->stream_start_offset == SOF_IPC4_INVALID_STREAM_POSITION) { ret = sof_ipc4_get_stream_start_offset(sdev, substream, stream, time_info); if (ret < 0) - return 0; + return -EOPNOTSUPP; } + /* For delay calculation we need the host counter */ + host_cnt = snd_sof_pcm_get_host_byte_counter(sdev, component, substream); + host_ptr = host_cnt; + + /* convert the host_cnt to frames */ + host_cnt = div64_u64(host_cnt, frames_to_bytes(substream->runtime, 1)); + /* * If the LLP counter is not reported by firmware in the SRAM window - * then read the dai (link) position via host accessible means if + * then read the dai (link) counter via host accessible means if * available. */ if (!time_info->llp_offset) { - tmp_ptr = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); - if (!tmp_ptr) - return 0; + dai_cnt = snd_sof_pcm_get_dai_frame_counter(sdev, component, substream); + if (!dai_cnt) + return -EOPNOTSUPP; } else { sof_mailbox_read(sdev, time_info->llp_offset, &llp, sizeof(llp)); - tmp_ptr = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l; + dai_cnt = ((u64)llp.reading.llp_u << 32) | llp.reading.llp_l; } + dai_cnt += time_info->stream_end_offset; - /* In two cases dai dma position is not accurate + /* In two cases dai dma counter is not accurate * (1) dai pipeline is started before host pipeline - * (2) multiple streams mixed into one. Each stream has the same dai dma position + * (2) multiple streams mixed into one. Each stream has the same dai dma + * counter * - * Firmware calculates correct stream_start_offset for all cases including above two. - * Driver subtracts stream_start_offset from dai dma position to get accurate one + * Firmware calculates correct stream_start_offset for all cases + * including above two. + * Driver subtracts stream_start_offset from dai dma counter to get + * accurate one */ - tmp_ptr -= time_info->stream_start_offset; - /* Calculate the delay taking into account that both pointer can wrap */ - div64_u64_rem(tmp_ptr, substream->runtime->boundary, &tmp_ptr); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - head_ptr = substream->runtime->status->hw_ptr; - tail_ptr = tmp_ptr; + /* + * On stream start the dai counter might not yet have reached the + * stream_start_offset value which means that no frames have left the + * DSP yet from the audio stream (on playback, capture streams have + * offset of 0 as we start capturing right away). + * In this case we need to adjust the distance between the counters by + * increasing the host counter by (offset - dai_counter). + * Otherwise the dai_counter needs to be adjusted to reflect the number + * of valid frames passed on the DAI side. + * + * The delay is the difference between the counters on the two + * sides of the DSP. + */ + if (dai_cnt < time_info->stream_start_offset) { + host_cnt += time_info->stream_start_offset - dai_cnt; + dai_cnt = 0; } else { - head_ptr = tmp_ptr; - tail_ptr = substream->runtime->status->hw_ptr; + dai_cnt -= time_info->stream_start_offset; } - if (head_ptr < tail_ptr) - return substream->runtime->boundary - tail_ptr + head_ptr; + /* Wrap the dai counter at the boundary where the host counter wraps */ + div64_u64_rem(dai_cnt, time_info->boundary, &dai_cnt); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + head_cnt = host_cnt; + tail_cnt = dai_cnt; + } else { + head_cnt = dai_cnt; + tail_cnt = host_cnt; + } + + if (head_cnt < tail_cnt) { + time_info->delay = time_info->boundary - tail_cnt + head_cnt; + goto out; + } + + time_info->delay = head_cnt - tail_cnt; + +out: + /* + * Convert the host byte counter to PCM pointer which wraps in buffer + * and it is in frames + */ + div64_u64_rem(host_ptr, snd_pcm_lib_buffer_bytes(substream), &host_ptr); + *pointer = bytes_to_frames(substream->runtime, host_ptr); + + return 0; +} + +static snd_pcm_sframes_t sof_ipc4_pcm_delay(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct sof_ipc4_timestamp_info *time_info; + struct snd_sof_pcm_stream *stream; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return 0; + + stream = &spcm->stream[substream->stream]; + time_info = stream->private; + /* + * Report the stored delay value calculated in the pointer callback. + * In the unlikely event that the calculation was skipped/aborted, the + * default 0 delay returned. + */ + if (time_info) + return time_info->delay; + + /* No delay information available, report 0 as delay */ + return 0; - return head_ptr - tail_ptr; } const struct sof_ipc_pcm_ops ipc4_pcm_ops = { @@ -965,6 +1059,7 @@ const struct sof_ipc_pcm_ops ipc4_pcm_ops = { .dai_link_fixup = sof_ipc4_pcm_dai_link_fixup, .pcm_setup = sof_ipc4_pcm_setup, .pcm_free = sof_ipc4_pcm_free, + .pointer = sof_ipc4_pcm_pointer, .delay = sof_ipc4_pcm_delay, .ipc_first_on_start = true, .platform_stop_during_hw_free = true, From f9eeb6bb13fb5d7af1ea5b74a10b1f8ead962540 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:13 +0200 Subject: [PATCH 17/74] ALSA: hda: Add pplcllpl/u members to hdac_ext_stream The pplcllpl/u can be used to save the Link Connection Linear Link Position register value to be used for compensation of the LLP register value in case the counter is not reset (after pause/resume or stop/start without closing the stream). The LLP can be used along with PPHCLDP to calculate delay caused by the DSP processing for HDA links. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-17-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- include/sound/hdaudio_ext.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index a8bebac1e4b2..957295364a5e 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -56,6 +56,9 @@ struct hdac_ext_stream { u32 pphcldpl; u32 pphcldpu; + u32 pplcllpl; + u32 pplcllpu; + bool decoupled:1; bool link_locked:1; bool link_prepared; From 1abc2642588e06f6180b3fbb21968cf5d0ba9e5f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Mar 2024 15:08:14 +0200 Subject: [PATCH 18/74] ASoC: SOF: Intel: hda: Compensate LLP in case it is not reset During pause/reset or stop/start the LLP counter is not reset, which will result broken delay reporting. Read the LLP value on STOP/PAUSE trigger and use it in LLP reading to normalize the LLP from the register. Cc: stable@vger.kernel.org # 6.8 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240321130814.4412-18-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai-ops.c | 11 +++++++++++ sound/soc/sof/intel/hda-pcm.c | 8 ++++++++ sound/soc/sof/intel/hda-stream.c | 9 ++++++++- 3 files changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index c50ca9e72d37..b073720b4cf4 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -7,6 +7,7 @@ #include #include +#include #include #include #include @@ -362,6 +363,16 @@ static int hda_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_hdac_ext_stream_clear(hext_stream); + + /* + * Save the LLP registers in case the stream is + * restarting due PAUSE_RELEASE, or START without a pcm + * close/open since in this case the LLP register is not reset + * to 0 and the delay calculation will return with invalid + * results. + */ + hext_stream->pplcllpl = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); + hext_stream->pplcllpu = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); break; default: dev_err(sdev->dev, "unknown trigger command %d\n", cmd); diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 69fefcd1abc5..d7b446f3f973 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -282,6 +282,14 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, /* binding pcm substream to hda stream */ substream->runtime->private_data = &dsp_stream->hstream; + + /* + * Reset the llp cache values (they are used for LLP compensation in + * case the counter is not reset) + */ + dsp_stream->pplcllpl = 0; + dsp_stream->pplcllpu = 0; + return 0; } diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 8504a4f27b60..0c189d3b19c1 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1064,6 +1064,8 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, return pos; } +#define merge_u64(u32_u, u32_l) (((u64)(u32_u) << 32) | (u32_l)) + /** * hda_dsp_get_stream_llp - Retrieve the LLP (Linear Link Position) of the stream * @sdev: SOF device @@ -1093,7 +1095,12 @@ u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, llp_l = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPL); llp_u = readl(hext_stream->pplc_addr + AZX_REG_PPLCLLPU); - return ((u64)llp_u << 32) | llp_l; + /* Compensate the LLP counter with the saved offset */ + if (hext_stream->pplcllpl || hext_stream->pplcllpu) + return merge_u64(llp_u, llp_l) - + merge_u64(hext_stream->pplcllpu, hext_stream->pplcllpl); + + return merge_u64(llp_u, llp_l); } /** From c61115b37ff964d63191dbf4a058f481daabdf57 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 22 Mar 2024 13:25:04 +0200 Subject: [PATCH 19/74] ASoC: SOF: Intel: hda-dsp: Skip IMR boot on ACE platforms in case of S3 suspend SoCs with ACE architecture are tailored to use s2idle instead deep (S3) suspend state and the IMR content is lost when the system is forced to enter even to S3. When waking up from S3 state the IMR boot will fail as the content is lost. Set the skip_imr_boot flag to make sure that we don't try IMR in this case. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Liam Girdwood Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240322112504.4192-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 31ffa1a8f2ac..ef5c915db8ff 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -681,17 +681,27 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; struct hdac_bus *bus = sof_to_bus(sdev); + bool imr_lost = false; int ret, j; /* - * The memory used for IMR boot loses its content in deeper than S3 state - * We must not try IMR boot on next power up (as it will fail). - * + * The memory used for IMR boot loses its content in deeper than S3 + * state on CAVS platforms. + * On ACE platforms due to the system architecture the IMR content is + * lost at S3 state already, they are tailored for s2idle use. + * We must not try IMR boot on next power up in these cases as it will + * fail. + */ + if (sdev->system_suspend_target > SOF_SUSPEND_S3 || + (chip->hw_ip_version >= SOF_INTEL_ACE_1_0 && + sdev->system_suspend_target == SOF_SUSPEND_S3)) + imr_lost = true; + + /* * In case of firmware crash or boot failure set the skip_imr_boot to true * as well in order to try to re-load the firmware to do a 'cold' boot. */ - if (sdev->system_suspend_target > SOF_SUSPEND_S3 || - sdev->fw_state == SOF_FW_CRASHED || + if (imr_lost || sdev->fw_state == SOF_FW_CRASHED || sdev->fw_state == SOF_FW_BOOT_FAILED) hda->skip_imr_boot = true; From e2d7ad717a6b0880843dbc60855a5b97ad0395f8 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 25 Mar 2024 14:44:50 +0000 Subject: [PATCH 20/74] ASoC: cs-amp-lib: Check for no firmware controls when writing calibration When a wmfw file has not been loaded the firmware control descriptions necessary to write a stored calibration are not present. In this case print a more descriptive error message. The message is logged at info level because it is not fatal, and does not necessarily imply that anything is broken. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Link: https://msgid.link/r/20240325144450.293630-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs-amp-lib.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 01ef4db5407d..287ac01a3873 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -56,6 +56,11 @@ static int _cs_amp_write_cal_coeffs(struct cs_dsp *dsp, dev_dbg(dsp->dev, "Calibration: Ambient=%#x, Status=%#x, CalR=%d\n", data->calAmbient, data->calStatus, data->calR); + if (list_empty(&dsp->ctl_list)) { + dev_info(dsp->dev, "Calibration disabled due to missing firmware controls\n"); + return -ENOENT; + } + ret = cs_amp_write_cal_coeff(dsp, controls, controls->ambient, data->calAmbient); if (ret) return ret; From 708181c50b7763c689ecaba5db8075c2d03719c4 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Fri, 22 Mar 2024 13:27:03 +0200 Subject: [PATCH 21/74] ASoC: SOF: mtrace: rework mtrace timestamp setting MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The original timestamp is built base on windows epoch time which is not fit for Linux system and difficult to be used for kernel debugging. This patch adopts syslog timestamp so that we can simply use dmesg to check the timestamp between fw and kernel. Signed-off-by: Rander Wang Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Signed-off-by: Peter Ujfalusi Link: https://msgid.link/r/20240322112703.4549-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-mtrace.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/ipc4-mtrace.c b/sound/soc/sof/ipc4-mtrace.c index 9f1e33ee8826..0e04bea9432d 100644 --- a/sound/soc/sof/ipc4-mtrace.c +++ b/sound/soc/sof/ipc4-mtrace.c @@ -4,6 +4,7 @@ #include #include +#include #include #include "sof-priv.h" #include "ipc4-priv.h" @@ -412,7 +413,6 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev) const struct sof_ipc_ops *iops = sdev->ipc->ops; struct sof_ipc4_msg msg; u64 system_time; - ktime_t kt; int ret; if (priv->mtrace_state != SOF_MTRACE_DISABLED) @@ -424,9 +424,12 @@ static int ipc4_mtrace_enable(struct snd_sof_dev *sdev) msg.primary |= SOF_IPC4_MOD_INSTANCE(SOF_IPC4_MOD_INIT_BASEFW_INSTANCE_ID); msg.extension = SOF_IPC4_MOD_EXT_MSG_PARAM_ID(SOF_IPC4_FW_PARAM_SYSTEM_TIME); - /* The system time is in usec, UTC, epoch is 1601-01-01 00:00:00 */ - kt = ktime_add_us(ktime_get_real(), FW_EPOCH_DELTA * USEC_PER_SEC); - system_time = ktime_to_us(kt); + /* + * local_clock() is used to align with dmesg, so both kernel and firmware logs have + * the same base and a minor delta due to the IPC. system time is in us format but + * local_clock() returns the time in ns, so convert to ns. + */ + system_time = div64_u64(local_clock(), NSEC_PER_USEC); msg.data_size = sizeof(system_time); msg.data_ptr = &system_time; ret = iops->set_get_data(sdev, &msg, msg.data_size, true); From 051e0840ffa8ab25554d6b14b62c9ab9e4901457 Mon Sep 17 00:00:00 2001 From: Duoming Zhou Date: Tue, 26 Mar 2024 17:42:38 +0800 Subject: [PATCH 22/74] ALSA: sh: aica: reorder cleanup operations to avoid UAF bugs The dreamcastcard->timer could schedule the spu_dma_work and the spu_dma_work could also arm the dreamcastcard->timer. When the snd_pcm_substream is closing, the aica_channel will be deallocated. But it could still be dereferenced in the worker thread. The reason is that del_timer() will return directly regardless of whether the timer handler is running or not and the worker could be rescheduled in the timer handler. As a result, the UAF bug will happen. The racy situation is shown below: (Thread 1) | (Thread 2) snd_aicapcm_pcm_close() | ... | run_spu_dma() //worker | mod_timer() flush_work() | del_timer() | aica_period_elapsed() //timer kfree(dreamcastcard->channel) | schedule_work() | run_spu_dma() //worker ... | dreamcastcard->channel-> //USE In order to mitigate this bug and other possible corner cases, call mod_timer() conditionally in run_spu_dma(), then implement PCM sync_stop op to cancel both the timer and worker. The sync_stop op will be called from PCM core appropriately when needed. Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Suggested-by: Takashi Iwai Signed-off-by: Duoming Zhou Message-ID: <20240326094238.95442-1-duoming@zju.edu.cn> Signed-off-by: Takashi Iwai --- sound/sh/aica.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 320ac792c7fe..3182c634464d 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -278,7 +278,8 @@ static void run_spu_dma(struct work_struct *work) dreamcastcard->clicks++; if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER)) dreamcastcard->clicks %= AICA_PERIOD_NUMBER; - mod_timer(&dreamcastcard->timer, jiffies + 1); + if (snd_pcm_running(dreamcastcard->substream)) + mod_timer(&dreamcastcard->timer, jiffies + 1); } } @@ -290,6 +291,8 @@ static void aica_period_elapsed(struct timer_list *t) /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; + if (!snd_pcm_running(substream)) + return; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; /* Have we played out an additional period? */ @@ -350,12 +353,19 @@ static int snd_aicapcm_pcm_open(struct snd_pcm_substream return 0; } +static int snd_aicapcm_pcm_sync_stop(struct snd_pcm_substream *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + + del_timer_sync(&dreamcastcard->timer); + cancel_work_sync(&dreamcastcard->spu_dma_work); + return 0; +} + static int snd_aicapcm_pcm_close(struct snd_pcm_substream *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; - flush_work(&(dreamcastcard->spu_dma_work)); - del_timer(&dreamcastcard->timer); dreamcastcard->substream = NULL; kfree(dreamcastcard->channel); spu_disable(); @@ -401,6 +411,7 @@ static const struct snd_pcm_ops snd_aicapcm_playback_ops = { .prepare = snd_aicapcm_pcm_prepare, .trigger = snd_aicapcm_pcm_trigger, .pointer = snd_aicapcm_pcm_pointer, + .sync_stop = snd_aicapcm_pcm_sync_stop, }; /* TO DO: set up to handle more than one pcm instance */ From 56ebbd19c2989f7450341f581e2724a149d0f08e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 10:54:34 +0000 Subject: [PATCH 23/74] ASoC: cs42l43: Correct extraction of data pointer in suspend/resume The current code is pulling the wrong pointer causing it to disable the wrong IRQ. Correct the code to pull the correct cs42l43 core data pointer. Fixes: 64353af49fec ("ASoC: cs42l43: Add system suspend ops to disable IRQ") Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240326105434.852907-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 860d5cda67bf..94685449f0f4 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2364,7 +2364,8 @@ static int cs42l43_codec_runtime_resume(struct device *dev) static int cs42l43_codec_suspend(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); @@ -2373,7 +2374,8 @@ static int cs42l43_codec_suspend(struct device *dev) static int cs42l43_codec_suspend_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2382,7 +2384,8 @@ static int cs42l43_codec_suspend_noirq(struct device *dev) static int cs42l43_codec_resume(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2391,7 +2394,8 @@ static int cs42l43_codec_resume(struct device *dev) static int cs42l43_codec_resume_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); From 7590ac2249ebfa6a40db9055fa62d349e9c8e6a6 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 26 Mar 2024 23:38:07 +0100 Subject: [PATCH 24/74] ALSA: aoa: avoid false-positive format truncation warning clang warns about what it interprets as a truncated snprintf: sound/aoa/soundbus/i2sbus/core.c:171:6: error: 'snprintf' will always be truncated; specified size is 6, but format string expands to at least 7 [-Werror,-Wformat-truncation-non-kprintf] The actual problem here is that it does not understand the special %pOFn format string and assumes that it is a pointer followed by the string "OFn", which would indeed not fit. Slightly increasing the size of the buffer to its natural alignment avoids the warning, as it is now long enough for the correct and the incorrect interprations. Fixes: b917d58dcfaa ("ALSA: aoa: Convert to using %pOFn instead of device_node.name") Signed-off-by: Arnd Bergmann Message-ID: <20240326223825.4084412-9-arnd@kernel.org> Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 3f49a9e28bfc..e627ffffa1f2 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -158,7 +158,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *child, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; - char node_name[6]; + char node_name[8]; static const char *rnames[] = { "i2sbus: %pOFn (control)", "i2sbus: %pOFn (tx)", "i2sbus: %pOFn (rx)" }; From ae065d0ce9e36ca4efdfb9b96ce3395bd1c19372 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Tue, 26 Mar 2024 17:18:45 +0100 Subject: [PATCH 25/74] ALSA: hda/tas2781: remove digital gain kcontrol The "Speaker Digital Gain" kcontrol controls the TAS2781_DVC_LVL (0x1A) register. Unfortunately the tas2563 does not have DVC_LVL, but has INT_MASK0 in 0x1A, which has been misused so far. Since commit c1947ce61ff4 ("ALSA: hda/realtek: tas2781: enable subwoofer volume control") the volume of the tas2781 amplifiers can be controlled by the master volume, so this digital gain kcontrol is not needed. Remove it. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Message-ID: <741fc21db994efd58f83e7aef38931204961e5b2.1711469583.git.soyer@irl.hu> Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 37 +-------------------------------- 1 file changed, 1 insertion(+), 36 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 4475cea8e9f7..5acb475c10a7 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -89,7 +89,7 @@ struct tas2781_hda { struct snd_kcontrol *dsp_prog_ctl; struct snd_kcontrol *dsp_conf_ctl; struct snd_kcontrol *prof_ctl; - struct snd_kcontrol *snd_ctls[3]; + struct snd_kcontrol *snd_ctls[2]; }; static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) @@ -294,27 +294,6 @@ static int tasdevice_config_put(struct snd_kcontrol *kcontrol, return ret; } -/* - * tas2781_digital_getvol - get the volum control - * @kcontrol: control pointer - * @ucontrol: User data - * Customer Kcontrol for tas2781 is primarily for regmap booking, paging - * depends on internal regmap mechanism. - * tas2781 contains book and page two-level register map, especially - * book switching will set the register BXXP00R7F, after switching to the - * correct book, then leverage the mechanism for paging to access the - * register. - */ -static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - return tasdevice_digital_getvol(tas_priv, ucontrol, mc); -} - static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -325,17 +304,6 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, return tasdevice_amp_getvol(tas_priv, ucontrol, mc); } -static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - /* The check of the given value is in tasdevice_digital_putvol. */ - return tasdevice_digital_putvol(tas_priv, ucontrol, mc); -} - static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -381,9 +349,6 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv), - ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, - 0, 0, 200, 1, tas2781_digital_getvol, - tas2781_digital_putvol, dvc_tlv), ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tas2781_force_fwload_get, tas2781_force_fwload_put), }; From 15bc3066d2378eef1b45254be9df23b0dd7f1667 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Tue, 26 Mar 2024 17:18:46 +0100 Subject: [PATCH 26/74] ALSA: hda/tas2781: add locks to kcontrols The rcabin.profile_cfg_id, cur_prog, cur_conf, force_fwload_status variables are acccessible from multiple threads and therefore require locking. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Message-ID: Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 50 +++++++++++++++++++++++++++++++-- 1 file changed, 48 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 5acb475c10a7..9a43f563bb9e 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -185,8 +185,12 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -200,11 +204,15 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, val = clamp(nr_profile, 0, max); + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->rcabin.profile_cfg_id != val) { tas_priv->rcabin.profile_cfg_id = val; ret = 1; } + mutex_unlock(&tas_priv->codec_lock); + return ret; } @@ -241,8 +249,12 @@ static int tasdevice_program_get(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -257,11 +269,15 @@ static int tasdevice_program_put(struct snd_kcontrol *kcontrol, val = clamp(nr_program, 0, max); + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->cur_prog != val) { tas_priv->cur_prog = val; ret = 1; } + mutex_unlock(&tas_priv->codec_lock); + return ret; } @@ -270,8 +286,12 @@ static int tasdevice_config_get(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -286,11 +306,15 @@ static int tasdevice_config_put(struct snd_kcontrol *kcontrol, val = clamp(nr_config, 0, max); + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->cur_conf != val) { tas_priv->cur_conf = val; ret = 1; } + mutex_unlock(&tas_priv->codec_lock); + return ret; } @@ -300,8 +324,15 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret; - return tasdevice_amp_getvol(tas_priv, ucontrol, mc); + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_amp_getvol(tas_priv, ucontrol, mc); + + mutex_unlock(&tas_priv->codec_lock); + + return ret; } static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, @@ -310,9 +341,16 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret; + + mutex_lock(&tas_priv->codec_lock); /* The check of the given value is in tasdevice_amp_putvol. */ - return tasdevice_amp_putvol(tas_priv, ucontrol, mc); + ret = tasdevice_amp_putvol(tas_priv, ucontrol, mc); + + mutex_unlock(&tas_priv->codec_lock); + + return ret; } static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, @@ -320,10 +358,14 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, { struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + mutex_lock(&tas_priv->codec_lock); + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); + mutex_unlock(&tas_priv->codec_lock); + return 0; } @@ -333,6 +375,8 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); bool change, val = (bool)ucontrol->value.integer.value[0]; + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->force_fwload_status == val) change = false; else { @@ -342,6 +386,8 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); + mutex_unlock(&tas_priv->codec_lock); + return change; } From 26c04a8a3c05dc280fa961e79b5b3fcb66ac4625 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Tue, 26 Mar 2024 17:18:47 +0100 Subject: [PATCH 27/74] ALSA: hda/tas2781: add debug statements to kcontrols Sometimes it is useful to examine the timing of kcontrol events. Add debug statements to each kcontrol. Signed-off-by: Gergo Koteles Message-ID: <18ff4b0caab90a2dacf907e62346fd5079a9eb1a.1711469583.git.soyer@irl.hu> Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 35 +++++++++++++++++++++++++++++---- 1 file changed, 31 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 9a43f563bb9e..f495caee38e1 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -189,6 +189,9 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->rcabin.profile_cfg_id); + mutex_unlock(&tas_priv->codec_lock); return 0; @@ -206,6 +209,10 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, + tas_priv->rcabin.profile_cfg_id, val); + if (tas_priv->rcabin.profile_cfg_id != val) { tas_priv->rcabin.profile_cfg_id = val; ret = 1; @@ -253,6 +260,9 @@ static int tasdevice_program_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = tas_priv->cur_prog; + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->cur_prog); + mutex_unlock(&tas_priv->codec_lock); return 0; @@ -271,6 +281,9 @@ static int tasdevice_program_put(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, tas_priv->cur_prog, val); + if (tas_priv->cur_prog != val) { tas_priv->cur_prog = val; ret = 1; @@ -290,6 +303,9 @@ static int tasdevice_config_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = tas_priv->cur_conf; + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->cur_conf); + mutex_unlock(&tas_priv->codec_lock); return 0; @@ -308,6 +324,9 @@ static int tasdevice_config_put(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, tas_priv->cur_conf, val); + if (tas_priv->cur_conf != val) { tas_priv->cur_conf = val; ret = 1; @@ -330,6 +349,9 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, ret = tasdevice_amp_getvol(tas_priv, ucontrol, mc); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %ld\n", + __func__, kcontrol->id.name, ucontrol->value.integer.value[0]); + mutex_unlock(&tas_priv->codec_lock); return ret; @@ -345,6 +367,9 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: -> %ld\n", + __func__, kcontrol->id.name, ucontrol->value.integer.value[0]); + /* The check of the given value is in tasdevice_amp_putvol. */ ret = tasdevice_amp_putvol(tas_priv, ucontrol, mc); @@ -361,8 +386,8 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; - dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, - tas_priv->force_fwload_status ? "ON" : "OFF"); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d\n", + __func__, kcontrol->id.name, tas_priv->force_fwload_status); mutex_unlock(&tas_priv->codec_lock); @@ -377,14 +402,16 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, mutex_lock(&tas_priv->codec_lock); + dev_dbg(tas_priv->dev, "%s: kcontrol %s: %d -> %d\n", + __func__, kcontrol->id.name, + tas_priv->force_fwload_status, val); + if (tas_priv->force_fwload_status == val) change = false; else { change = true; tas_priv->force_fwload_status = val; } - dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, - tas_priv->force_fwload_status ? "ON" : "OFF"); mutex_unlock(&tas_priv->codec_lock); From 1506d96119eb9454d64f5ae80ab8d04c1594ac25 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Tue, 26 Mar 2024 17:18:48 +0100 Subject: [PATCH 28/74] ALSA: hda/tas2781: remove useless dev_dbg from playback_hook The debug message "Playback action not supported: action" is not useful, because the action was previously printed, and the list of supported actions are intentional. Remove the debug statement from the default switch case. Signed-off-by: Gergo Koteles Message-ID: <8b9546db6c92dea4476a7247a88d56248c2ba8c2.1711469583.git.soyer@irl.hu> Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index f495caee38e1..48dae3339305 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -161,8 +161,6 @@ static void tas2781_hda_playback_hook(struct device *dev, int action) pm_runtime_put_autosuspend(dev); break; default: - dev_dbg(tas_hda->dev, "Playback action not supported: %d\n", - action); break; } } From 4af565de9f8c74b9f6035924ce0d40adec211246 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 27 Mar 2024 16:16:53 +0530 Subject: [PATCH 29/74] ASoC: amd: acp: fix for acp pdm configuration check ACP PDM configuration has to be verified for all combinations. Remove FLAG_AMD_LEGACY_ONLY_DMIC check. Fixes: 3a94c8ad0aae ("ASoC: amd: acp: add code for scanning acp pdm controller") Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240327104657.3537664-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-pci.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 8c8b1dcac628..440b91d4f261 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -133,11 +133,9 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id } } - if (flag == FLAG_AMD_LEGACY_ONLY_DMIC) { - ret = check_acp_pdm(pci, chip); - if (ret < 0) - goto skip_pdev_creation; - } + ret = check_acp_pdm(pci, chip); + if (ret < 0) + goto skip_pdev_creation; chip->flag = flag; memset(&pdevinfo, 0, sizeof(pdevinfo)); From daf6c4681a74034d5723e2fb761e0d7f3a1ca18f Mon Sep 17 00:00:00 2001 From: Christoffer Sandberg Date: Thu, 28 Mar 2024 11:27:57 +0100 Subject: [PATCH 30/74] ALSA: hda/realtek - Fix inactive headset mic jack This patch adds the existing fixup to certain TF platforms implementing the ALC274 codec with a headset jack. It fixes/activates the inactive microphone of the headset. Signed-off-by: Christoffer Sandberg Signed-off-by: Werner Sembach Cc: Message-ID: <20240328102757.50310-1-wse@tuxedocomputers.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a17c36a36aa5..c31e9be257a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10403,6 +10403,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1387, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), From 2d0401ee38d43ab0e4cdd02dfc9d402befb2b5c8 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Thu, 28 Mar 2024 12:13:55 +0000 Subject: [PATCH 31/74] ALSA: hda: cs35l56: Add ACPI device match tables Adding the ACPI HIDs to the match table triggers the cs35l56-hda modules to be loaded on boot so that Serial Multi Instantiate can add the devices to the bus and begin the driver init sequence. Signed-off-by: Simon Trimmer Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Message-ID: <20240328121355.18972-1-simont@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l56_hda_i2c.c | 13 +++++++++++-- sound/pci/hda/cs35l56_hda_spi.c | 13 +++++++++++-- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index 13beee807308..40f2f97944d5 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -56,10 +56,19 @@ static const struct i2c_device_id cs35l56_hda_i2c_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct i2c_driver cs35l56_hda_i2c_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_i2c_id, .probe = cs35l56_hda_i2c_probe, diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index a3b2fa76663d..7f02155fe61e 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -56,10 +56,19 @@ static const struct spi_device_id cs35l56_hda_spi_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct spi_driver cs35l56_hda_spi_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_spi_id, .probe = cs35l56_hda_spi_probe, From 310a5caa4e861616a27a83c3e8bda17d65026fa8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:12 -0500 Subject: [PATCH 32/74] ASoC: rt5682-sdw: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: 02fb23d72720 ("ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index e67c2e19cb1a..1fdbef5fd6cb 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -763,12 +763,12 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt5682->disable_irq_lock); if (rt5682->disable_irq == true) { - mutex_lock(&rt5682->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); rt5682->disable_irq = false; - mutex_unlock(&rt5682->disable_irq_lock); } + mutex_unlock(&rt5682->disable_irq_lock); goto regmap_sync; } From ee287771644394d071e6a331951ee8079b64f9a7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:13 -0500 Subject: [PATCH 33/74] ASoC: rt711-sdca: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: 23adeb7056ac ("ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdca-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 935e597022d3..b8471b2d8f4f 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -438,13 +438,13 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt711->disable_irq_lock); if (rt711->disable_irq == true) { - mutex_lock(&rt711->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt711->disable_irq = false; - mutex_unlock(&rt711->disable_irq_lock); } + mutex_unlock(&rt711->disable_irq_lock); goto regmap_sync; } From aae86cfd8790bcc7693a5a0894df58de5cb5128c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:14 -0500 Subject: [PATCH 34/74] ASoC: rt711-sdw: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: b69de265bd0e ("ASoC: rt711: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 3f5773310ae8..988451f24a75 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -536,12 +536,12 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt711->disable_irq_lock); if (rt711->disable_irq == true) { - mutex_lock(&rt711->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); rt711->disable_irq = false; - mutex_unlock(&rt711->disable_irq_lock); } + mutex_unlock(&rt711->disable_irq_lock); goto regmap_sync; } From c8b2e5c1b959d100990e4f0cbad38e7d047bb97c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:15 -0500 Subject: [PATCH 35/74] ASoC: rt712-sdca-sdw: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: 7a8735c1551e ("ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt712-sdca-sdw.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 01ac555cd79b..36d0dd532b8d 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -438,13 +438,14 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt712->disable_irq_lock); if (rt712->disable_irq == true) { - mutex_lock(&rt712->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt712->disable_irq = false; - mutex_unlock(&rt712->disable_irq_lock); } + mutex_unlock(&rt712->disable_irq_lock); goto regmap_sync; } From adb354bbc231b23d3a05163ce35c1d598512ff64 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:16 -0500 Subject: [PATCH 36/74] ASoC: rt722-sdca-sdw: fix locking sequence The disable_irq_lock protects the 'disable_irq' value, we need to lock before testing it. Fixes: a0b7c59ac1a9 ("ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://msgid.link/r/20240325221817.206465-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index eb76f4c675b6..65d584c1886e 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -467,13 +467,13 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev) return 0; if (!slave->unattach_request) { + mutex_lock(&rt722->disable_irq_lock); if (rt722->disable_irq == true) { - mutex_lock(&rt722->disable_irq_lock); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt722->disable_irq = false; - mutex_unlock(&rt722->disable_irq_lock); } + mutex_unlock(&rt722->disable_irq_lock); goto regmap_sync; } From f892e66fcabc6161cd38c0fc86e769208174b840 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:18:17 -0500 Subject: [PATCH 37/74] ASoC: rt-sdw*: add __func__ to all error logs The drivers for Realtek SoundWire codecs use similar logs, which is problematic to analyze problems reported by CI tools, e.g. "Failed to get private value: 752001 => 0000 ret=-5". It's not uncommon to have several Realtek devices on the same platform, having the same log thrown makes support difficult. This patch adds __func__ to all error logs which didn't already include it. No functionality change, only error logs are modified. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://msgid.link/r/20240325221817.206465-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1316-sdw.c | 8 +++--- sound/soc/codecs/rt1318-sdw.c | 8 +++--- sound/soc/codecs/rt5682-sdw.c | 12 ++++---- sound/soc/codecs/rt700.c | 16 +++++------ sound/soc/codecs/rt711-sdca-sdw.c | 2 +- sound/soc/codecs/rt711-sdca.c | 18 ++++++------ sound/soc/codecs/rt711-sdw.c | 4 +-- sound/soc/codecs/rt711.c | 16 +++++------ sound/soc/codecs/rt712-sdca-dmic.c | 24 +++++++++------- sound/soc/codecs/rt712-sdca-sdw.c | 2 +- sound/soc/codecs/rt712-sdca.c | 20 ++++++------- sound/soc/codecs/rt715-sdca-sdw.c | 2 +- sound/soc/codecs/rt715-sdca.c | 46 +++++++++++++++--------------- sound/soc/codecs/rt715-sdw.c | 4 +-- sound/soc/codecs/rt715.c | 24 ++++++++-------- sound/soc/codecs/rt722-sdca.c | 21 +++++++------- 16 files changed, 115 insertions(+), 112 deletions(-) diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 47511f70119a..0b3bf920bcab 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -537,7 +537,7 @@ static int rt1316_sdw_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt1316->sdw_slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -577,12 +577,12 @@ static int rt1316_sdw_parse_dt(struct rt1316_sdw_priv *rt1316, struct device *de if (rt1316->bq_params_cnt) { rt1316->bq_params = devm_kzalloc(dev, rt1316->bq_params_cnt, GFP_KERNEL); if (!rt1316->bq_params) { - dev_err(dev, "Could not allocate bq_params memory\n"); + dev_err(dev, "%s: Could not allocate bq_params memory\n", __func__); ret = -ENOMEM; } else { ret = device_property_read_u8_array(dev, "realtek,bq-params", rt1316->bq_params, rt1316->bq_params_cnt); if (ret < 0) - dev_err(dev, "Could not read list of realtek,bq-params\n"); + dev_err(dev, "%s: Could not read list of realtek,bq-params\n", __func__); } } @@ -759,7 +759,7 @@ static int __maybe_unused rt1316_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT1316_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c index ff364bde4a08..462c9a4b1be5 100644 --- a/sound/soc/codecs/rt1318-sdw.c +++ b/sound/soc/codecs/rt1318-sdw.c @@ -606,7 +606,7 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt1318->sdw_slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -631,8 +631,8 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT1318_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } @@ -835,7 +835,7 @@ static int __maybe_unused rt1318_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT1318_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 1fdbef5fd6cb..f9ee42c13dba 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -132,7 +132,7 @@ static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt5682->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -315,8 +315,8 @@ static int rt5682_sdw_init(struct device *dev, struct regmap *regmap, &rt5682_sdw_indirect_regmap); if (IS_ERR(rt5682->regmap)) { ret = PTR_ERR(rt5682->regmap); - dev_err(dev, "Failed to allocate register map: %d\n", - ret); + dev_err(dev, "%s: Failed to allocate register map: %d\n", + __func__, ret); return ret; } @@ -400,7 +400,7 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) } if (val != DEVICE_ID) { - dev_err(dev, "Device with ID register %x is not rt5682\n", val); + dev_err(dev, "%s: Device with ID register %x is not rt5682\n", __func__, val); ret = -ENODEV; goto err_nodev; } @@ -648,7 +648,7 @@ static int rt5682_bus_config(struct sdw_slave *slave, ret = rt5682_clock_config(&slave->dev); if (ret < 0) - dev_err(&slave->dev, "Invalid clk config"); + dev_err(&slave->dev, "%s: Invalid clk config", __func__); return ret; } @@ -775,7 +775,7 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 0ebf344a1b60..434b926f96c8 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -37,8 +37,8 @@ static int rt700_index_write(struct regmap *regmap, ret = regmap_write(regmap, addr, value); if (ret < 0) - pr_err("Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -52,8 +52,8 @@ static int rt700_index_read(struct regmap *regmap, *value = 0; ret = regmap_read(regmap, addr, value); if (ret < 0) - pr_err("Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -930,14 +930,14 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream, port_config.num += 2; break; default: - dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id); return -EINVAL; } retval = sdw_stream_add_slave(rt700->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -945,8 +945,8 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream, /* bit 3:0 Number of Channel */ val |= (params_channels(params) - 1); } else { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index b8471b2d8f4f..2636c2eea4bc 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -451,7 +451,7 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 447154cb6010..1e8dbfc3ecd9 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -36,8 +36,8 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt711->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -52,8 +52,8 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt711->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -1293,13 +1293,13 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt711->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 16) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -1318,8 +1318,8 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT711_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 988451f24a75..0d3b43dd22e6 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -408,7 +408,7 @@ static int rt711_bus_config(struct sdw_slave *slave, ret = rt711_clock_config(&slave->dev); if (ret < 0) - dev_err(&slave->dev, "Invalid clk config"); + dev_err(&slave->dev, "%s: Invalid clk config", __func__); return ret; } @@ -548,7 +548,7 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); return -ETIMEDOUT; } diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 66eaed13b0d6..5446f9506a16 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -37,8 +37,8 @@ static int rt711_index_write(struct regmap *regmap, ret = regmap_write(regmap, addr, value); if (ret < 0) - pr_err("Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -52,8 +52,8 @@ static int rt711_index_read(struct regmap *regmap, *value = 0; ret = regmap_read(regmap, addr, value); if (ret < 0) - pr_err("Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -428,7 +428,7 @@ static void rt711_jack_init(struct rt711_priv *rt711) RT711_HP_JD_FINAL_RESULT_CTL_JD12); break; default: - dev_warn(rt711->component->dev, "Wrong JD source\n"); + dev_warn(rt711->component->dev, "%s: Wrong JD source\n", __func__); break; } @@ -1020,7 +1020,7 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt711->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -1028,8 +1028,8 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream, /* bit 3:0 Number of Channel */ val |= (params_channels(params) - 1); } else { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c index 0926b26619bd..012b79e72cf6 100644 --- a/sound/soc/codecs/rt712-sdca-dmic.c +++ b/sound/soc/codecs/rt712-sdca-dmic.c @@ -139,8 +139,8 @@ static int rt712_sdca_dmic_index_write(struct rt712_sdca_dmic_priv *rt712, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -155,8 +155,8 @@ static int rt712_sdca_dmic_index_read(struct rt712_sdca_dmic_priv *rt712, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -317,7 +317,8 @@ static int rt712_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol, for (i = 0; i < p->count; i++) { err = regmap_write(rt712->mbq_regmap, p->reg_base + i, gain_val[i]); if (err < 0) - dev_err(&rt712->slave->dev, "0x%08x can't be set\n", p->reg_base + i); + dev_err(&rt712->slave->dev, "%s: 0x%08x can't be set\n", + __func__, p->reg_base + i); } return changed; @@ -667,13 +668,13 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt712->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 4) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -698,8 +699,8 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT712_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } @@ -923,7 +924,8 @@ static int __maybe_unused rt712_sdca_dmic_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", + __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 36d0dd532b8d..4e9ab3ef135b 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -452,7 +452,7 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c index 6954fbe7ec5f..b503de9fda80 100644 --- a/sound/soc/codecs/rt712-sdca.c +++ b/sound/soc/codecs/rt712-sdca.c @@ -34,8 +34,8 @@ static int rt712_sdca_index_write(struct rt712_sdca_priv *rt712, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -50,8 +50,8 @@ static int rt712_sdca_index_read(struct rt712_sdca_priv *rt712, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt712->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -1060,13 +1060,13 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt712->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 16) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -1085,8 +1085,8 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT712_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } @@ -1106,7 +1106,7 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate); break; default: - dev_err(component->dev, "Wrong DAI id\n"); + dev_err(component->dev, "%s: Wrong DAI id\n", __func__); return -EINVAL; } diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index ab54a67a27eb..ee450126106f 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -237,7 +237,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->enumeration_complete, msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Enumeration not complete, timed out\n"); + dev_err(&slave->dev, "%s: Enumeration not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 4533eedd7e18..3fb7b9adb61d 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -41,8 +41,8 @@ static int rt715_sdca_index_write(struct rt715_sdca_priv *rt715, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt715->slave->dev, - "Failed to set private value: %08x <= %04x %d\n", - addr, value, ret); + "%s: Failed to set private value: %08x <= %04x %d\n", + __func__, addr, value, ret); return ret; } @@ -59,8 +59,8 @@ static int rt715_sdca_index_read(struct rt715_sdca_priv *rt715, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt715->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -152,8 +152,8 @@ static int rt715_sdca_set_amp_gain_put(struct snd_kcontrol *kcontrol, mc->shift); ret = regmap_write(rt715->mbq_regmap, mc->reg + i, gain_val); if (ret != 0) { - dev_err(component->dev, "Failed to write 0x%x=0x%x\n", - mc->reg + i, gain_val); + dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n", + __func__, mc->reg + i, gain_val); return ret; } } @@ -188,8 +188,8 @@ static int rt715_sdca_set_amp_gain_4ch_put(struct snd_kcontrol *kcontrol, ret = regmap_write(rt715->mbq_regmap, reg_base + i, gain_val); if (ret != 0) { - dev_err(component->dev, "Failed to write 0x%x=0x%x\n", - reg_base + i, gain_val); + dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n", + __func__, reg_base + i, gain_val); return ret; } } @@ -224,8 +224,8 @@ static int rt715_sdca_set_amp_gain_8ch_put(struct snd_kcontrol *kcontrol, reg = i < 7 ? reg_base + i : (reg_base - 1) | BIT(15); ret = regmap_write(rt715->mbq_regmap, reg, gain_val); if (ret != 0) { - dev_err(component->dev, "Failed to write 0x%x=0x%x\n", - reg, gain_val); + dev_err(component->dev, "%s: Failed to write 0x%x=0x%x\n", + __func__, reg, gain_val); return ret; } } @@ -246,8 +246,8 @@ static int rt715_sdca_set_amp_gain_get(struct snd_kcontrol *kcontrol, for (i = 0; i < 2; i++) { ret = regmap_read(rt715->mbq_regmap, mc->reg + i, &val); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - mc->reg + i, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, mc->reg + i, ret); return ret; } ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, mc->shift); @@ -271,8 +271,8 @@ static int rt715_sdca_set_amp_gain_4ch_get(struct snd_kcontrol *kcontrol, for (i = 0; i < 4; i++) { ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - reg_base + i, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, reg_base + i, ret); return ret; } ucontrol->value.integer.value[i] = rt715_sdca_get_gain(val, gain_sft); @@ -297,8 +297,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol, for (i = 0; i < 8; i += 2) { ret = regmap_read(rt715->mbq_regmap, reg_base + i, &val_l); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - reg_base + i, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, reg_base + i, ret); return ret; } ucontrol->value.integer.value[i] = (val_l >> gain_sft) / 10; @@ -306,8 +306,8 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol, reg = (i == 6) ? (reg_base - 1) | BIT(15) : reg_base + 1 + i; ret = regmap_read(rt715->mbq_regmap, reg, &val_r); if (ret < 0) { - dev_err(component->dev, "Failed to read 0x%x, ret=%d\n", - reg, ret); + dev_err(component->dev, "%s: Failed to read 0x%x, ret=%d\n", + __func__, reg, ret); return ret; } ucontrol->value.integer.value[i + 1] = (val_r >> gain_sft) / 10; @@ -834,15 +834,15 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream, 0xaf00); break; default: - dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id); return -EINVAL; } retval = sdw_stream_add_slave(rt715->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(component->dev, "Unable to configure port, retval:%d\n", - retval); + dev_err(component->dev, "%s: Unable to configure port, retval:%d\n", + __func__, retval); return retval; } @@ -893,8 +893,8 @@ static int rt715_sdca_pcm_hw_params(struct snd_pcm_substream *substream, val = 0xf; break; default: - dev_err(component->dev, "Unsupported sample rate %d\n", - params_rate(params)); + dev_err(component->dev, "%s: Unsupported sample rate %d\n", + __func__, params_rate(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 21f37babd148..7e13868ff99f 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -482,7 +482,7 @@ static int rt715_bus_config(struct sdw_slave *slave, ret = rt715_clock_config(&slave->dev); if (ret < 0) - dev_err(&slave->dev, "Invalid clk config"); + dev_err(&slave->dev, "%s: Invalid clk config", __func__); return 0; } @@ -554,7 +554,7 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "Initialization not complete, timed out\n"); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 9f732a5abd53..299c9b12377c 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -40,8 +40,8 @@ static int rt715_index_write(struct regmap *regmap, unsigned int reg, ret = regmap_write(regmap, addr, value); if (ret < 0) { - pr_err("Failed to set private value: %08x <= %04x %d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %08x <= %04x %d\n", + __func__, addr, value, ret); } return ret; @@ -55,8 +55,8 @@ static int rt715_index_write_nid(struct regmap *regmap, ret = regmap_write(regmap, addr, value); if (ret < 0) - pr_err("Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + pr_err("%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -70,8 +70,8 @@ static int rt715_index_read_nid(struct regmap *regmap, *value = 0; ret = regmap_read(regmap, addr, value); if (ret < 0) - pr_err("Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + pr_err("%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -862,14 +862,14 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, rt715_index_write(rt715->regmap, RT715_SDW_INPUT_SEL, 0xa000); break; default: - dev_err(component->dev, "Invalid DAI id %d\n", dai->id); + dev_err(component->dev, "%s: Invalid DAI id %d\n", __func__, dai->id); return -EINVAL; } retval = sdw_stream_add_slave(rt715->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } @@ -883,8 +883,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, val |= 0x0 << 8; break; default: - dev_err(component->dev, "Unsupported sample rate %d\n", - params_rate(params)); + dev_err(component->dev, "%s: Unsupported sample rate %d\n", + __func__, params_rate(params)); return -EINVAL; } @@ -892,8 +892,8 @@ static int rt715_pcm_hw_params(struct snd_pcm_substream *substream, /* bit 3:0 Number of Channel */ val |= (params_channels(params) - 1); } else { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index 0e1c65a20392..e0ea3a23f7cc 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -35,8 +35,8 @@ int rt722_sdca_index_write(struct rt722_sdca_priv *rt722, ret = regmap_write(regmap, addr, value); if (ret < 0) dev_err(&rt722->slave->dev, - "Failed to set private value: %06x <= %04x ret=%d\n", - addr, value, ret); + "%s: Failed to set private value: %06x <= %04x ret=%d\n", + __func__, addr, value, ret); return ret; } @@ -51,8 +51,8 @@ int rt722_sdca_index_read(struct rt722_sdca_priv *rt722, ret = regmap_read(regmap, addr, value); if (ret < 0) dev_err(&rt722->slave->dev, - "Failed to get private value: %06x => %04x ret=%d\n", - addr, *value, ret); + "%s: Failed to get private value: %06x => %04x ret=%d\n", + __func__, addr, *value, ret); return ret; } @@ -663,7 +663,8 @@ static int rt722_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol, for (i = 0; i < p->count; i++) { err = regmap_write(rt722->mbq_regmap, p->reg_base + i, gain_val[i]); if (err < 0) - dev_err(&rt722->slave->dev, "%#08x can't be set\n", p->reg_base + i); + dev_err(&rt722->slave->dev, "%s: %#08x can't be set\n", + __func__, p->reg_base + i); } return changed; @@ -1211,13 +1212,13 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream, retval = sdw_stream_add_slave(rt722->slave, &stream_config, &port_config, 1, sdw_stream); if (retval) { - dev_err(dai->dev, "Unable to configure port\n"); + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); return retval; } if (params_channels(params) > 16) { - dev_err(component->dev, "Unsupported channels %d\n", - params_channels(params)); + dev_err(component->dev, "%s: Unsupported channels %d\n", + __func__, params_channels(params)); return -EINVAL; } @@ -1236,8 +1237,8 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream, sampling_rate = RT722_SDCA_RATE_192000HZ; break; default: - dev_err(component->dev, "Rate %d is not supported\n", - params_rate(params)); + dev_err(component->dev, "%s: Rate %d is not supported\n", + __func__, params_rate(params)); return -EINVAL; } From fc563aa900659a850e2ada4af26b9d7a3de6c591 Mon Sep 17 00:00:00 2001 From: Stephen Lee Date: Mon, 25 Mar 2024 18:01:31 -0700 Subject: [PATCH 38/74] ASoC: ops: Fix wraparound for mask in snd_soc_get_volsw In snd_soc_info_volsw(), mask is generated by figuring out the index of the most significant bit set in max and converting the index to a bitmask through bit shift 1. Unintended wraparound occurs when max is an integer value with msb bit set. Since the bit shift value 1 is treated as an integer type, the left shift operation will wraparound and set mask to 0 instead of all 1's. In order to fix this, we type cast 1 as `1ULL` to prevent the wraparound. Fixes: 7077148fb50a ("ASoC: core: Split ops out of soc-core.c") Signed-off-by: Stephen Lee Link: https://msgid.link/r/20240326010131.6211-1-slee08177@gmail.com Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d25748ca706..b27e89ff6a16 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -263,7 +263,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; int min = mc->min; int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; + unsigned int mask = (1ULL << fls(max)) - 1; unsigned int invert = mc->invert; int val; int ret; From 831ec5e3538e989c7995137b5c5c661991a09504 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Thu, 28 Mar 2024 23:47:37 +0100 Subject: [PATCH 39/74] ASoC: tas2781: mark dvc_tlv with __maybe_unused Since we put dvc_tlv static variable to a header file it's copied to each module that includes the header. But not all of them are actually used it. Fix this W=1 build warning: include/sound/tas2781-tlv.h:18:35: warning: 'dvc_tlv' defined but not used [-Wunused-const-variable=] Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202403290354.v0StnRpc-lkp@intel.com/ Fixes: ae065d0ce9e3 ("ALSA: hda/tas2781: remove digital gain kcontrol") Signed-off-by: Gergo Koteles Message-ID: <0e461545a2a6e9b6152985143e50526322e5f76b.1711665731.git.soyer@irl.hu> Signed-off-by: Takashi Iwai --- include/sound/tas2781-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 4038dd421150..1dc59005d241 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); #endif From 2c603a4947a1247102ccb008d5eb6f37a4043c98 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 29 Mar 2024 11:08:12 +0530 Subject: [PATCH 40/74] ASoC: amd: acp: fix for acp_init function error handling If acp_init() fails, acp pci driver probe should return error. Add acp_init() function return value check logic. Fixes: e61b415515d3 ("ASoC: amd: acp: refactor the acp init and de-init sequence") Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20240329053815.2373979-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-pci.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 440b91d4f261..5f35b90eab8d 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -115,7 +115,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id goto unregister_dmic_dev; } - acp_init(chip); + ret = acp_init(chip); + if (ret) + goto unregister_dmic_dev; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; From c33f0d4fcfe072adbbb7f3cf93f1b146e181bf3b Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Fri, 29 Mar 2024 11:28:03 +0000 Subject: [PATCH 41/74] ALSA: hda/realtek: Add quirks for ASUS Laptops using CS35L56 These ASUS laptops use the Realtek HDA codec combined with a number of CS35L56 amplifiers. The SSID of the GA403U matches a previous ASUS laptop - we can tell them apart because they use different codecs. Signed-off-by: Simon Trimmer Message-ID: <20240329112803.23897-1-simont@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 53 ++++++++++++++++++++++++++++++++++- 1 file changed, 52 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c31e9be257a9..e866b8a75cda 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6875,11 +6875,38 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st comp_generic_fixup(cdc, action, "i2c", "CLSA0101", "-%s:00-cs35l41-hda.%d", 2); } +static void cs35l56_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + +static void cs35l56_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); +} + +static void cs35l56_fixup_spi_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + static void cs35l56_fixup_spi_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); } +static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + /* + * The same SSID has been re-used in different hardware, they have + * different codecs and the newer GA403U has a ALC285. + */ + if (cdc->core.vendor_id == 0x10ec0285) + cs35l56_fixup_i2c_two(cdc, fix, action); + else + alc_fixup_inv_dmic(cdc, fix, action); +} + static void tas2781_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7436,6 +7463,10 @@ enum { ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, ALC256_FIXUP_HEADPHONE_AMP_VOL, ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX, + ALC285_FIXUP_CS35L56_SPI_2, + ALC285_FIXUP_CS35L56_I2C_2, + ALC285_FIXUP_CS35L56_I2C_4, + ALC285_FIXUP_ASUS_GA403U, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9643,6 +9674,22 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_spectre_x360_eu0xxx, }, + [ALC285_FIXUP_CS35L56_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_spi_two, + }, + [ALC285_FIXUP_CS35L56_I2C_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_two, + }, + [ALC285_FIXUP_CS35L56_I2C_4] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_four, + }, + [ALC285_FIXUP_ASUS_GA403U] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_asus_ga403u, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10096,7 +10143,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), - SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U), SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2), @@ -10104,6 +10151,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_CS35L56_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), @@ -10115,11 +10163,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606", ALC285_FIXUP_CS35L56_I2C_4), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), + SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), From 755795cd3da053b0565085d9950c44d7b6cba5c4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Fri, 29 Mar 2024 22:54:42 +0100 Subject: [PATCH 42/74] OSS: dmasound/paula: Mark driver struct with __refdata to prevent section mismatch MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit As described in the added code comment, a reference to .exit.text is ok for drivers registered via module_platform_driver_probe(). Make this explicit to prevent the following section mismatch warning WARNING: modpost: sound/oss/dmasound/dmasound_paula: section mismatch in reference: amiga_audio_driver+0x8 (section: .data) -> amiga_audio_remove (section: .exit.text) that triggers on an allmodconfig W=1 build. Signed-off-by: Uwe Kleine-König Message-ID: Signed-off-by: Takashi Iwai --- sound/oss/dmasound/dmasound_paula.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 0ba8f0c4cd99..3a593da09280 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -725,7 +725,13 @@ static void __exit amiga_audio_remove(struct platform_device *pdev) dmasound_deinit(); } -static struct platform_driver amiga_audio_driver = { +/* + * amiga_audio_remove() lives in .exit.text. For drivers registered via + * module_platform_driver_probe() this is ok because they cannot get unbound at + * runtime. So mark the driver struct with __refdata to prevent modpost + * triggering a section mismatch warning. + */ +static struct platform_driver amiga_audio_driver __refdata = { .remove_new = __exit_p(amiga_audio_remove), .driver = { .name = "amiga-audio", From 03f56ed4ead162551ac596c9e3076ff01f1c5836 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Mon, 1 Apr 2024 16:58:05 +0200 Subject: [PATCH 43/74] Revert "ALSA: emu10k1: fix synthesizer sample playback position and caching" As already anticipated in the original commit, playback was broken for very short samples. I just didn't expect it to be an actual problem, because we're talking about less than 1.5 milliseconds here. But clearly such wavetable samples do actually exist. The problem was that for such short samples we'd set the current position beyond the end of the loop, so we'd run off the end of the sample and play garbage. This is a bigger (more audible) problem than the original one, which was that we'd start playback with garbage (whatever was still in the cache), which would be mostly masked by the note's attack phase. So revert to the old behavior for now. We'll subsequently fix it properly with a bigger patch series. Note that this isn't a full revert - the dead code is not re-introduced, because that would be silly. Fixes: df335e9a8bcb ("ALSA: emu10k1: fix synthesizer sample playback position and caching") Link: https://bugzilla.kernel.org/show_bug.cgi?id=218625 Signed-off-by: Oswald Buddenhagen Message-ID: <20240401145805.528794-1-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_callback.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index d36234b88fb4..941bfbf812ed 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -255,7 +255,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, /* check if sample is finished playing (non-looping only) */ if (bp != best + V_OFF && bp != best + V_FREE && (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) { - val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64; + val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch); if (val >= vp->reg.loopstart) bp = best + V_OFF; } @@ -362,7 +362,7 @@ start_voice(struct snd_emux_voice *vp) map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - addr = vp->reg.start + 64; + addr = vp->reg.start; temp = vp->reg.parm.filterQ; ccca = (temp << 28) | addr; if (vp->apitch < 0xe400) @@ -430,9 +430,6 @@ start_voice(struct snd_emux_voice *vp) /* Q & current address (Q 4bit value, MSB) */ CCCA, ccca, - /* cache */ - CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64), - /* reset volume */ VTFT, vtarget | vp->ftarget, CVCF, vtarget | CVCF_CURRENTFILTER_MASK, From b67a7dc418aabbddec41c752ac29b6fa0250d0a8 Mon Sep 17 00:00:00 2001 From: Christian Bendiksen Date: Mon, 1 Apr 2024 12:26:10 +0000 Subject: [PATCH 44/74] ALSA: hda/realtek: Add sound quirks for Lenovo Legion slim 7 16ARHA7 models This fixes the sound not working from internal speakers on Lenovo Legion Slim 7 16ARHA7 models. The correct subsystem ID have been added to cs35l41_hda_property.c and patch_realtek.c. Signed-off-by: Christian Bendiksen Cc: Message-ID: <20240401122603.6634-1-christian@bendiksen.me> Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 4 ++++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 6 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 72ec872afb8d..d6ea3ab72f75 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -109,6 +109,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38A9", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38AB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38B4", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, @@ -497,6 +499,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, + { "CSC3551", "17AA3877", generic_dsd_config }, + { "CSC3551", "17AA3878", generic_dsd_config }, { "CSC3551", "17AA38A9", generic_dsd_config }, { "CSC3551", "17AA38AB", generic_dsd_config }, { "CSC3551", "17AA38B4", generic_dsd_config }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e866b8a75cda..405620bd543c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10384,6 +10384,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), + SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C), From 1576f263ee2147dc395531476881058609ad3d38 Mon Sep 17 00:00:00 2001 From: I Gede Agastya Darma Laksana Date: Tue, 2 Apr 2024 00:46:02 +0700 Subject: [PATCH 45/74] ALSA: hda/realtek: Update Panasonic CF-SZ6 quirk to support headset with microphone This patch addresses an issue with the Panasonic CF-SZ6's existing quirk, specifically its headset microphone functionality. Previously, the quirk used ALC269_FIXUP_HEADSET_MODE, which does not support the CF-SZ6's design of a single 3.5mm jack for both mic and audio output effectively. The device uses pin 0x19 for the headset mic without jack detection. Following verification on the CF-SZ6 and discussions with the original patch author, i determined that the update to ALC269_FIXUP_ASPIRE_HEADSET_MIC is the appropriate solution. This change is custom-designed for the CF-SZ6's unique hardware setup, which includes a single 3.5mm jack for both mic and audio output, connecting the headset microphone to pin 0x19 without the use of jack detection. Fixes: 0fca97a29b83 ("ALSA: hda/realtek - Add Panasonic CF-SZ6 headset jack quirk") Signed-off-by: I Gede Agastya Darma Laksana Cc: Message-ID: <20240401174602.14133-1-gedeagas22@gmail.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 405620bd543c..e1729d209922 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10210,7 +10210,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12f6, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), - SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), + SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP), From 0bfe105018bd2d7b1e4373193d9b55b37cf4458b Mon Sep 17 00:00:00 2001 From: "Luke D. Jones" Date: Tue, 2 Apr 2024 14:51:26 +1300 Subject: [PATCH 46/74] ALSA: hda/realtek: cs35l41: Support ASUS ROG G634JYR Fixes the realtek quirk to initialise the Cirrus amp correctly and adds related quirk for missing DSD properties. This model laptop has slightly updated internals compared to the previous version with Realtek Codec ID of 0x1caf. Signed-off-by: Luke D. Jones Cc: Message-ID: <20240402015126.21115-1-luke@ljones.dev> Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index d6ea3ab72f75..8fb688e41414 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -108,6 +108,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, @@ -498,6 +499,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, + { "CSC3551", "10433A60", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, { "CSC3551", "17AA3877", generic_dsd_config }, { "CSC3551", "17AA3878", generic_dsd_config }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e1729d209922..cdcb28aa9d7b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10184,7 +10184,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), From c4e51e424e2c772ce1836912a8b0b87cd61bc9d5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Apr 2024 08:36:25 +0200 Subject: [PATCH 47/74] ALSA: line6: Zero-initialize message buffers For shutting up spurious KMSAN uninit-value warnings, just replace kmalloc() calls with kzalloc() for the buffers used for communications. There should be no real issue with the original code, but it's still better to cover. Reported-by: syzbot+7fb05ccf7b3d2f9617b3@syzkaller.appspotmail.com Closes: https://lore.kernel.org/r/00000000000084b18706150bcca5@google.com Message-ID: <20240402063628.26609-1-tiwai@suse.de> Signed-off-by: Takashi Iwai --- sound/usb/line6/driver.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index b67617b68e50..f4437015d43a 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -202,7 +202,7 @@ int line6_send_raw_message_async(struct usb_line6 *line6, const char *buffer, struct urb *urb; /* create message: */ - msg = kmalloc(sizeof(struct message), GFP_ATOMIC); + msg = kzalloc(sizeof(struct message), GFP_ATOMIC); if (msg == NULL) return -ENOMEM; @@ -688,7 +688,7 @@ static int line6_init_cap_control(struct usb_line6 *line6) int ret; /* initialize USB buffers: */ - line6->buffer_listen = kmalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL); + line6->buffer_listen = kzalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL); if (!line6->buffer_listen) return -ENOMEM; @@ -697,7 +697,7 @@ static int line6_init_cap_control(struct usb_line6 *line6) return -ENOMEM; if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) { - line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); + line6->buffer_message = kzalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); if (!line6->buffer_message) return -ENOMEM; From 8a655cee6c9d4588570ad0cb099c5660f9a44a12 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Tue, 2 Apr 2024 14:20:40 +0800 Subject: [PATCH 48/74] ASoC: codecs: ES8326: Solve error interruption issue We got an error report about headphone type detection and button detection. We fixed the headphone type detection error by adjusting the debounce timer configuration. And we fixed the button detection error by disabling the button detection feature when the headphone are unplugged and enabling it when headphone are plugged in. Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240402062043.20608-2-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 15289dadafea..a6783fd6553d 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -843,6 +843,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x0a); regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x03); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); /* * Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event */ @@ -865,6 +866,8 @@ static void es8326_jack_detect_handler(struct work_struct *work) * set auto-check mode, then restart jack_detect_work after 400ms. * Don't report jack status. */ + regmap_write(es8326->regmap, ES8326_INT_SOURCE, + (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); @@ -987,7 +990,7 @@ static int es8326_resume(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E); regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0); usleep_range(10000, 15000); - regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xe9); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9); regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xcb); /* set headphone default type and detect pin */ regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83); @@ -1038,8 +1041,7 @@ static int es8326_resume(struct snd_soc_component *component) es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); - regmap_write(es8326->regmap, ES8326_INT_SOURCE, - (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, From 4581468d071b64a2e3c2ae333fff82dc0391a306 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Tue, 2 Apr 2024 14:20:41 +0800 Subject: [PATCH 49/74] ASoC: codecs: ES8326: modify clock table We got a digital microphone feature issue. And we fixed it by modifying the clock table. Also, we changed the marco ES8326_CLK_ON declaration Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240402062043.20608-3-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 22 +++++++++++----------- sound/soc/codecs/es8326.h | 2 +- 2 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a6783fd6553d..275db81d10d4 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -412,9 +412,9 @@ static const struct _coeff_div coeff_div_v3[] = { {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x8A, 0x0A, 0x27, 0x27}, {128, 8000, 1024000, 0x60, 0x00, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, - {128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, - {128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, - {128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x3F}, + {128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, + {128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x8A, 0x1B, 0x23, 0x47}, {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x8A, 0x1B, 0x23, 0x47}, {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, @@ -423,10 +423,10 @@ static const struct _coeff_div coeff_div_v3[] = { {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x27, 0x27}, - {256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, - {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, - {256, 44100, 11289600, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, - {256, 48000, 12288000, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x08, 0x19, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x08, 0x19, 0x1F, 0x3F}, + {256, 44100, 11289600, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, + {256, 48000, 12288000, 0xE0, 0x01, 0x01, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x23, 0x47}, {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, @@ -435,10 +435,10 @@ static const struct _coeff_div coeff_div_v3[] = { {400, 48000, 19200000, 0xE4, 0x04, 0x35, 0x6d, 0xCA, 0x0A, 0x1F, 0x1F}, {500, 48000, 24000000, 0xF8, 0x04, 0x3F, 0x6D, 0xCA, 0x0A, 0x1F, 0x1F}, - {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, - {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, - {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, - {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x08, 0x19, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x08, 0x19, 0x1F, 0x3F}, + {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, + {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0x48, 0x08, 0x1F, 0x1F}, {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, {768, 32000, 24576000, 0xE0, 0x02, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h index ee12caef8105..c3e52e7bdef5 100644 --- a/sound/soc/codecs/es8326.h +++ b/sound/soc/codecs/es8326.h @@ -104,7 +104,7 @@ #define ES8326_MUTE (3 << 0) /* ES8326_CLK_CTL */ -#define ES8326_CLK_ON (0x7e << 0) +#define ES8326_CLK_ON (0x7f << 0) #define ES8326_CLK_OFF (0 << 0) /* ES8326_CLK_INV */ From 6e5f5bf894eb9260f07ad0da4e2dd2efd616ed59 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Tue, 2 Apr 2024 14:20:42 +0800 Subject: [PATCH 50/74] ASoC: codecs: ES8326: Solve a headphone detection issue after suspend and resume We got a headphone detection issue after suspend and resume. And we fixed it by modifying the configuration at es8326_suspend and invoke es8326_irq at es8326_resume. Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240402062043.20608-4-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 275db81d10d4..fa809ab41a4a 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -1062,6 +1062,8 @@ static int es8326_resume(struct snd_soc_component *component) es8326->hp = 0; es8326->hpl_vol = 0x03; es8326->hpr_vol = 0x03; + + es8326_irq(es8326->irq, es8326); return 0; } @@ -1072,6 +1074,9 @@ static int es8326_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&es8326->jack_detect_work); es8326_disable_micbias(component); es8326->calibrated = false; + regmap_write(es8326->regmap, ES8326_CLK_MUX, 0x2d); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x00); + regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); regcache_cache_only(es8326->regmap, true); regcache_mark_dirty(es8326->regmap); From fec9c7f668ac5dd107f4da5a3b18379e07ec1a41 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Tue, 2 Apr 2024 14:20:43 +0800 Subject: [PATCH 51/74] ASoC: codecs: ES8326: Removing the control of ADC_SCALE We removed the configuration of ES8326_ADC_SCALE in es8326_jack_detect_handler because user changed the configuration by snd_controls Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240402062043.20608-5-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index fa809ab41a4a..17bd6b516077 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -835,7 +835,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "Report hp remove event\n"); snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); /* mute adc when mic path switch */ - regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33); regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326->hp = 0; @@ -894,7 +893,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); - regmap_write(es8326->regmap, ES8326_ADC_SCALE, 0x33); regmap_update_bits(es8326->regmap, ES8326_PGA_PDN, 0x08, 0x08); regmap_update_bits(es8326->regmap, ES8326_PGAGAIN, From d619b0b70dc4f160f2b95d95ccfed2631ab7ac3a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Tue, 2 Apr 2024 15:06:40 +0200 Subject: [PATCH 52/74] ASoC: Intel: avs: boards: Add modules description MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Modpost warns about missing module description, add it. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240402130640.3310999-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/da7219.c | 1 + sound/soc/intel/avs/boards/dmic.c | 1 + sound/soc/intel/avs/boards/es8336.c | 1 + sound/soc/intel/avs/boards/i2s_test.c | 1 + sound/soc/intel/avs/boards/max98357a.c | 1 + sound/soc/intel/avs/boards/max98373.c | 1 + sound/soc/intel/avs/boards/max98927.c | 1 + sound/soc/intel/avs/boards/nau8825.c | 1 + sound/soc/intel/avs/boards/probe.c | 1 + sound/soc/intel/avs/boards/rt274.c | 1 + sound/soc/intel/avs/boards/rt286.c | 1 + sound/soc/intel/avs/boards/rt298.c | 1 + sound/soc/intel/avs/boards/rt5514.c | 1 + sound/soc/intel/avs/boards/rt5663.c | 1 + sound/soc/intel/avs/boards/rt5682.c | 1 + sound/soc/intel/avs/boards/ssm4567.c | 1 + 16 files changed, 16 insertions(+) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index c018f84fe025..fc072dc58968 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -296,5 +296,6 @@ static struct platform_driver avs_da7219_driver = { module_platform_driver(avs_da7219_driver); +MODULE_DESCRIPTION("Intel da7219 machine driver"); MODULE_AUTHOR("Cezary Rojewski "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c index ba2bc7f689eb..d9e5e85f5233 100644 --- a/sound/soc/intel/avs/boards/dmic.c +++ b/sound/soc/intel/avs/boards/dmic.c @@ -96,4 +96,5 @@ static struct platform_driver avs_dmic_driver = { module_platform_driver(avs_dmic_driver); +MODULE_DESCRIPTION("Intel DMIC machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 1090082e7d5b..5c90a6007577 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -326,4 +326,5 @@ static struct platform_driver avs_es8336_driver = { module_platform_driver(avs_es8336_driver); +MODULE_DESCRIPTION("Intel es8336 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c index 28f254eb0d03..027373d6a16d 100644 --- a/sound/soc/intel/avs/boards/i2s_test.c +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -204,4 +204,5 @@ static struct platform_driver avs_i2s_test_driver = { module_platform_driver(avs_i2s_test_driver); +MODULE_DESCRIPTION("Intel i2s test machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c index a83b95f25129..1ff85e4d8e16 100644 --- a/sound/soc/intel/avs/boards/max98357a.c +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -154,4 +154,5 @@ static struct platform_driver avs_max98357a_driver = { module_platform_driver(avs_max98357a_driver) +MODULE_DESCRIPTION("Intel max98357a machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c index 3b980a025e6f..8d31586b73ea 100644 --- a/sound/soc/intel/avs/boards/max98373.c +++ b/sound/soc/intel/avs/boards/max98373.c @@ -211,4 +211,5 @@ static struct platform_driver avs_max98373_driver = { module_platform_driver(avs_max98373_driver) +MODULE_DESCRIPTION("Intel max98373 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c index 86dd2b228df3..572ec58073d0 100644 --- a/sound/soc/intel/avs/boards/max98927.c +++ b/sound/soc/intel/avs/boards/max98927.c @@ -208,4 +208,5 @@ static struct platform_driver avs_max98927_driver = { module_platform_driver(avs_max98927_driver) +MODULE_DESCRIPTION("Intel max98927 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 1c1e2083f474..55db75efae41 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -313,4 +313,5 @@ static struct platform_driver avs_nau8825_driver = { module_platform_driver(avs_nau8825_driver) +MODULE_DESCRIPTION("Intel nau8825 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/probe.c b/sound/soc/intel/avs/boards/probe.c index a9469b5ecb40..8be6887bbc6e 100644 --- a/sound/soc/intel/avs/boards/probe.c +++ b/sound/soc/intel/avs/boards/probe.c @@ -69,4 +69,5 @@ static struct platform_driver avs_probe_mb_driver = { module_platform_driver(avs_probe_mb_driver); +MODULE_DESCRIPTION("Intel probe machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index bfcb8845fd15..1cf524216087 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -276,4 +276,5 @@ static struct platform_driver avs_rt274_driver = { module_platform_driver(avs_rt274_driver); +MODULE_DESCRIPTION("Intel rt274 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index 28d7d86b1cc9..4740bba10570 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -247,4 +247,5 @@ static struct platform_driver avs_rt286_driver = { module_platform_driver(avs_rt286_driver); +MODULE_DESCRIPTION("Intel rt286 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 80f490b9e118..6e409e29f697 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -266,4 +266,5 @@ static struct platform_driver avs_rt298_driver = { module_platform_driver(avs_rt298_driver); +MODULE_DESCRIPTION("Intel rt298 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5514.c b/sound/soc/intel/avs/boards/rt5514.c index 60105f453ae2..097ae5f73241 100644 --- a/sound/soc/intel/avs/boards/rt5514.c +++ b/sound/soc/intel/avs/boards/rt5514.c @@ -192,4 +192,5 @@ static struct platform_driver avs_rt5514_driver = { module_platform_driver(avs_rt5514_driver); +MODULE_DESCRIPTION("Intel rt5514 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c index b4762c2a7bf2..1880c315cc4d 100644 --- a/sound/soc/intel/avs/boards/rt5663.c +++ b/sound/soc/intel/avs/boards/rt5663.c @@ -265,4 +265,5 @@ static struct platform_driver avs_rt5663_driver = { module_platform_driver(avs_rt5663_driver); +MODULE_DESCRIPTION("Intel rt5663 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 243f979fda98..594a971ded9e 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -341,5 +341,6 @@ static struct platform_driver avs_rt5682_driver = { module_platform_driver(avs_rt5682_driver) +MODULE_DESCRIPTION("Intel rt5682 machine driver"); MODULE_AUTHOR("Cezary Rojewski "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index 4a0e136835ff..d6f7f046c24e 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -200,4 +200,5 @@ static struct platform_driver avs_ssm4567_driver = { module_platform_driver(avs_ssm4567_driver) +MODULE_DESCRIPTION("Intel ssm4567 machine driver"); MODULE_LICENSE("GPL"); From 3f5eb32513e75eb321919a703800d4e13e9d3ba8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 14:18:39 +0300 Subject: [PATCH 53/74] ASoC: SOF: Intel: lnl: Disable DMIC/SSP offload on remove During probe the DMIC/SSP offload is enabled and it is not reversed on remove. Add a remove wrapper for LNL to disable the offload for DMIC and SSP similarly to what is done during probe. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://msgid.link/r/20240403111839.27259-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index d1c73d407e68..aeb4350cce6b 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -29,15 +29,17 @@ static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = { }; /* this helps allows the DSP to setup DMIC/SSP */ -static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus) +static int hdac_bus_offload_dmic_ssp(struct hdac_bus *bus, bool enable) { int ret; - ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_SSP, true); + ret = hdac_bus_eml_enable_offload(bus, true, + AZX_REG_ML_LEPTR_ID_INTEL_SSP, enable); if (ret < 0) return ret; - ret = hdac_bus_eml_enable_offload(bus, true, AZX_REG_ML_LEPTR_ID_INTEL_DMIC, true); + ret = hdac_bus_eml_enable_offload(bus, true, + AZX_REG_ML_LEPTR_ID_INTEL_DMIC, enable); if (ret < 0) return ret; @@ -52,7 +54,19 @@ static int lnl_hda_dsp_probe(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); +} + +static void lnl_hda_dsp_remove(struct snd_sof_dev *sdev) +{ + int ret; + + ret = hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), false); + if (ret < 0) + dev_warn(sdev->dev, + "Failed to disable offload for DMIC/SSP: %d\n", ret); + + hda_dsp_remove(sdev); } static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev) @@ -63,7 +77,7 @@ static int lnl_hda_dsp_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); } static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev) @@ -74,7 +88,7 @@ static int lnl_hda_dsp_runtime_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev)); + return hdac_bus_offload_dmic_ssp(sof_to_bus(sdev), true); } static int lnl_dsp_post_fw_run(struct snd_sof_dev *sdev) @@ -97,9 +111,11 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) /* common defaults */ memcpy(&sof_lnl_ops, &sof_hda_common_ops, sizeof(struct snd_sof_dsp_ops)); - /* probe */ - if (!sdev->dspless_mode_selected) + /* probe/remove */ + if (!sdev->dspless_mode_selected) { sof_lnl_ops.probe = lnl_hda_dsp_probe; + sof_lnl_ops.remove = lnl_hda_dsp_remove; + } /* shutdown */ sof_lnl_ops.shutdown = hda_dsp_shutdown; From b9846a386734e73a1414950ebfd50f04919f5e24 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 4 Apr 2024 09:47:13 +0530 Subject: [PATCH 54/74] ASoC: SOF: amd: fix for false dsp interrupts Before ACP firmware loading, DSP interrupts are not expected. Sometimes after reboot, it's observed that before ACP firmware is loaded false DSP interrupt is reported. Registering the interrupt handler before acp initialization causing false interrupts sometimes on reboot as ACP reset is not applied. Correct the sequence by invoking acp initialization sequence prior to registering interrupt handler. Fixes: 738a2b5e2cc9 ("ASoC: SOF: amd: Add IPC support for ACP IP block") Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240404041717.430545-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index be7dc1e02284..c12c7f820529 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -704,6 +704,10 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto unregister_dev; } + ret = acp_init(sdev); + if (ret < 0) + goto free_smn_dev; + sdev->ipc_irq = pci->irq; ret = request_threaded_irq(sdev->ipc_irq, acp_irq_handler, acp_irq_thread, IRQF_SHARED, "AudioDSP", sdev); @@ -713,10 +717,6 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto free_smn_dev; } - ret = acp_init(sdev); - if (ret < 0) - goto free_ipc_irq; - /* scan SoundWire capabilities exposed by DSDT */ ret = acp_sof_scan_sdw_devices(sdev, chip->sdw_acpi_dev_addr); if (ret < 0) { From 90f8917e7a15f6dd508779048bdf00ce119b6ca0 Mon Sep 17 00:00:00 2001 From: Chaitanya Kumar Borah Date: Thu, 4 Apr 2024 13:48:13 -0500 Subject: [PATCH 55/74] ASoC: SOF: Core: Add remove_late() to sof_init_environment failure path MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In cases where the sof driver is unable to find the firmware and/or topology file [1], it exits without releasing the i915 runtime pm wakeref [2]. This results in dmesg warnings[3] during suspend/resume or driver unbind. Add remove_late() to the failure path of sof_init_environment so that i915 wakeref is released appropriately [1] [ 8.990366] sof-audio-pci-intel-mtl 0000:00:1f.3: SOF firmware and/or topology file not found. [ 8.990396] sof-audio-pci-intel-mtl 0000:00:1f.3: Supported default profiles [ 8.990398] sof-audio-pci-intel-mtl 0000:00:1f.3: - ipc type 1 (Requested): [ 8.990399] sof-audio-pci-intel-mtl 0000:00:1f.3: Firmware file: intel/sof-ipc4/mtl/sof-mtl.ri [ 8.990401] sof-audio-pci-intel-mtl 0000:00:1f.3: Topology file: intel/sof-ace-tplg/sof-mtl-rt711-2ch.tplg [ 8.990402] sof-audio-pci-intel-mtl 0000:00:1f.3: Check if you have 'sof-firmware' package installed. [ 8.990403] sof-audio-pci-intel-mtl 0000:00:1f.3: Optionally it can be manually downloaded from: [ 8.990404] sof-audio-pci-intel-mtl 0000:00:1f.3: https://github.com/thesofproject/sof-bin/ [ 8.999088] sof-audio-pci-intel-mtl 0000:00:1f.3: error: sof_probe_work failed err: -2 [2] ref_tracker: 0000:00:02.0@ffff9b8511b6a378 has 1/5 users at track_intel_runtime_pm_wakeref.part.0+0x36/0x70 [i915] __intel_runtime_pm_get+0x51/0xb0 [i915] intel_runtime_pm_get+0x17/0x20 [i915] intel_display_power_get+0x2f/0x70 [i915] i915_audio_component_get_power+0x23/0x120 [i915] snd_hdac_display_power+0x89/0x130 [snd_hda_core] hda_codec_i915_init+0x3f/0x50 [snd_sof_intel_hda] hda_dsp_probe_early+0x170/0x250 [snd_sof_intel_hda_common] snd_sof_device_probe+0x224/0x320 [snd_sof] sof_pci_probe+0x15b/0x220 [snd_sof_pci] hda_pci_intel_probe+0x30/0x70 [snd_sof_intel_hda_common] local_pci_probe+0x4c/0xb0 pci_device_probe+0xcc/0x250 really_probe+0x18e/0x420 __driver_probe_device+0x7e/0x170 driver_probe_device+0x23/0xa0 [3] [ 484.105070] ------------[ cut here ]------------ [ 484.108238] thunderbolt 0000:00:0d.2: PM: pci_pm_suspend_late+0x0/0x50 returned 0 after 0 usecs [ 484.117106] i915 0000:00:02.0: i915 raw-wakerefs=1 wakelocks=1 on cleanup [ 484.792005] WARNING: CPU: 2 PID: 2405 at drivers/gpu/drm/i915/intel_runtime_pm.c:444 intel_runtime_pm_driver_release+0x6c/0x80 Tested-by: Rodrigo Vivi Reviewed-by: Rodrigo Vivi Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Signed-off-by: Chaitanya Kumar Borah Signed-off-by: Pierre-Louis Bossart Acked-by: Lucas De Marchi Link: https://github.com/thesofproject/linux/pull/4878 Signed-off-by: Rodrigo Vivi Link: https://msgid.link/r/20240404184813.134566-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 9b00ede2a486..cc84d4c81be9 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -339,8 +339,7 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = snd_sof_probe(sdev); if (ret < 0) { dev_err(sdev->dev, "failed to probe DSP %d\n", ret); - sof_ops_free(sdev); - return ret; + goto err_sof_probe; } /* check machine info */ @@ -358,15 +357,18 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = validate_sof_ops(sdev); if (ret < 0) { snd_sof_remove(sdev); + snd_sof_remove_late(sdev); return ret; } } + return 0; + err_machine_check: - if (ret) { - snd_sof_remove(sdev); - sof_ops_free(sdev); - } + snd_sof_remove(sdev); +err_sof_probe: + snd_sof_remove_late(sdev); + sof_ops_free(sdev); return ret; } From 84471d01c92c33b3f4cedfe319639ecf7f8fc4c5 Mon Sep 17 00:00:00 2001 From: Vitaly Rodionov Date: Fri, 5 Apr 2024 22:06:35 +0100 Subject: [PATCH 56/74] ALSA: hda/realtek: Add quirk for HP SnowWhite laptops Add support for HP SnowWhite laptops with CS35L51 amplifiers on I2C bus connected to Realtek codec. Signed-off-by: Vitaly Rodionov Message-ID: <20240405210635.22193-1-vitalyr@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cdcb28aa9d7b..d6940bc4ec39 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10084,6 +10084,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8ca7, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), From 0b6f0ff01a4a8c1b66c600263465976d57dcc1a3 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sat, 6 Apr 2024 21:20:09 +0800 Subject: [PATCH 57/74] ALSA: hda/tas2781: correct the register for pow calibrated data Calibrated data was written into an incorrect register, which cause speaker protection sometimes malfuctions Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Shenghao Ding Cc: Message-ID: <20240406132010.341-1-shenghao-ding@ti.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 48dae3339305..75f7674c66ee 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -514,10 +514,10 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { static const unsigned char page_array[CALIB_MAX] = { - 0x17, 0x18, 0x18, 0x0d, 0x18 + 0x17, 0x18, 0x18, 0x13, 0x18, }; static const unsigned char rgno_array[CALIB_MAX] = { - 0x74, 0x0c, 0x14, 0x3c, 0x7c + 0x74, 0x0c, 0x14, 0x70, 0x7c, }; unsigned char *data; int i, j, rc; From 72829b98ff3a22efb66e5b618bd0219111db1811 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:14 +0200 Subject: [PATCH 58/74] ALSA: emux: fix /proc teardown at module unload We forgot to remember the wavetable /proc entry, so we'd fail to free it at module unload. This matters only when only the synth module is unloaded, as unloading the card driver would tear down the sub-entry anyway. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-2-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/emux_proc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/synth/emux/emux_proc.c b/sound/synth/emux/emux_proc.c index 7993e6a01e54..820351f52551 100644 --- a/sound/synth/emux/emux_proc.c +++ b/sound/synth/emux/emux_proc.c @@ -102,6 +102,7 @@ void snd_emux_proc_init(struct snd_emux *emu, struct snd_card *card, int device) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; entry->c.text.read = snd_emux_proc_info_read; + emu->proc = entry; } void snd_emux_proc_free(struct snd_emux *emu) From 3f3e0dfc83d586fe9204936fccae771754a9dbc2 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:15 +0200 Subject: [PATCH 59/74] ALSA: emux: prune unused parameter from snd_soundfont_load_guspatch() The `client` parameter was not used, so eliminate it from the call chain. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-3-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- include/sound/soundfont.h | 2 +- sound/synth/emux/emux_hwdep.c | 3 +-- sound/synth/emux/emux_oss.c | 3 +-- sound/synth/emux/soundfont.c | 7 +++---- 4 files changed, 6 insertions(+), 9 deletions(-) diff --git a/include/sound/soundfont.h b/include/sound/soundfont.h index e445688a4f4f..98ed98d89d6d 100644 --- a/include/sound/soundfont.h +++ b/include/sound/soundfont.h @@ -89,7 +89,7 @@ struct snd_sf_list { int snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, long count, int client); int snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data, - long count, int client); + long count); int snd_soundfont_close_check(struct snd_sf_list *sflist, int client); struct snd_sf_list *snd_sf_new(struct snd_sf_callback *callback, diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index 81719bfb8ed7..fd8f978cde1c 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -27,8 +27,7 @@ snd_emux_hwdep_load_patch(struct snd_emux *emu, void __user *arg) if (patch.key == GUS_PATCH) return snd_soundfont_load_guspatch(emu->sflist, arg, - patch.len + sizeof(patch), - TMP_CLIENT_ID); + patch.len + sizeof(patch)); if (patch.type >= SNDRV_SFNT_LOAD_INFO && patch.type <= SNDRV_SFNT_PROBE_DATA) { diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index d8d32671f703..04df46b269d3 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -205,8 +205,7 @@ snd_emux_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, return -ENXIO; if (format == GUS_PATCH) - rc = snd_soundfont_load_guspatch(emu->sflist, buf, count, - SF_CLIENT_NO(p->chset.port)); + rc = snd_soundfont_load_guspatch(emu->sflist, buf, count); else if (format == SNDRV_OSS_SOUNDFONT_PATCH) { struct soundfont_patch_info patch; if (count < (int)sizeof(patch)) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 16f00097cb95..e1e47518ac92 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -941,8 +941,7 @@ int snd_sf_vol_table[128] = { /* load GUS patch */ static int -load_guspatch(struct snd_sf_list *sflist, const char __user *data, - long count, int client) +load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) { struct patch_info patch; struct snd_soundfont *sf; @@ -1122,11 +1121,11 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, /* load GUS patch */ int snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data, - long count, int client) + long count) { int rc; lock_preset(sflist); - rc = load_guspatch(sflist, data, count, client); + rc = load_guspatch(sflist, data, count); unlock_preset(sflist); return rc; } From 19061f35b3eaf4925960be44d870244b99df8d1d Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:16 +0200 Subject: [PATCH 60/74] ALSA: emux: fix validation of snd_emux.num_ports Both bounds had off-by-one errors. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-4-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/emux_seq.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index b227c7e0bc2a..1adaa75df2f6 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -65,11 +65,11 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) return -ENODEV; } - if (emu->num_ports < 0) { + if (emu->num_ports <= 0) { snd_printk(KERN_WARNING "seqports must be greater than zero\n"); emu->num_ports = 1; - } else if (emu->num_ports >= SNDRV_EMUX_MAX_PORTS) { - snd_printk(KERN_WARNING "too many ports." + } else if (emu->num_ports > SNDRV_EMUX_MAX_PORTS) { + snd_printk(KERN_WARNING "too many ports. " "limited max. ports %d\n", SNDRV_EMUX_MAX_PORTS); emu->num_ports = SNDRV_EMUX_MAX_PORTS; } From 877d1e81c7a4c47c69a098cd8b87756b2809e885 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:17 +0200 Subject: [PATCH 61/74] ALSA: emux: fix init of patch_info.truesize in load_data() The field is explicitly documented to be initialized by the driver (which it actually is). Also, using patch_info.size would be actually wrong for 16-bit data, as one field counts samples, while the other counts bytes. load_guspatch() already did it right. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-5-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/soundfont.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index e1e47518ac92..ad0231d7a39d 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -735,7 +735,7 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) sp->v = sample_info; sp->v.sf_id = sf->id; sp->v.dummy = 0; - sp->v.truesize = sp->v.size; + sp->v.truesize = 0; /* * If there is wave data then load it. From 1edeac6555e9df008b1729ca445868c1177baa8b Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:18 +0200 Subject: [PATCH 62/74] ALSA: emu10k1: prune vestiges of SNDRV_SFNT_SAMPLE_{BIDIR,REVERSE}_LOOP support This is required only to implement WAVE_BIDIR_LOOP and WAVE_LOOP_BACK in the GUS patch loader. It has not worked on emu10k1 since before ALSA hit mainline, yet nobody appears to have complained. And as it isn't super easy to implement, just admit defeat and clean up the code. If somebody wanted to resurrect the feature, the emu8k driver could serve as a template, but the code would be quite different. But arguably, this should be done in user space in the first place, as this doesn't represent a hardware feature (somewhat ironically, the actual GUS driver has no synth support, and therefore no GUS patch loader). Note that instead of properly rejecting affected samples, we continue to just pretend that the feature wasn't requested. This is extremely questionable behavior, but avoids that possibly unused instruments suddenly prevent loading the entire file, which would break backwards compatibility. But at least we log a warning now. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-6-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 73 ++++--------------------------- 1 file changed, 8 insertions(+), 65 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 89890f24509f..49214c226808 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -28,8 +28,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, { int offset; int truesize, size, blocksize; - __maybe_unused int loopsize; - int loopend, sampleend; unsigned int start_addr; struct snd_emu10k1 *emu; @@ -43,32 +41,24 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, return 0; } + if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP | SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) { + /* should instead return -ENOTSUPP; but compatibility */ + printk(KERN_WARNING "Emu10k1 wavetable patch %d with unsupported loop feature\n", + sp->v.sample); + } + /* recalculate address offset */ sp->v.end -= sp->v.start; sp->v.loopstart -= sp->v.start; sp->v.loopend -= sp->v.start; sp->v.start = 0; - /* some samples have invalid data. the addresses are corrected in voice info */ - sampleend = sp->v.end; - if (sampleend > sp->v.size) - sampleend = sp->v.size; - loopend = sp->v.loopend; - if (loopend > sampleend) - loopend = sampleend; - /* be sure loop points start < end */ if (sp->v.loopstart >= sp->v.loopend) swap(sp->v.loopstart, sp->v.loopend); /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; - loopsize = 0; -#if 0 /* not supported */ - if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) - loopsize = sp->v.loopend - sp->v.loopstart; - truesize += loopsize; -#endif if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) truesize += BLANK_LOOP_SIZE; @@ -96,8 +86,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, snd_emu10k1_synth_bzero(emu, sp->block, offset, size); offset += size; - /* copy start->loopend */ - size = loopend; + /* copy provided samples */ + size = sp->v.size; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; if (offset + size > blocksize) @@ -108,53 +98,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, return -EFAULT; } offset += size; - data += size; - -#if 0 /* not supported yet */ - /* handle reverse (or bidirectional) loop */ - if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) { - /* copy loop in reverse */ - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) { - int woffset; - unsigned short *wblock = (unsigned short*)block; - woffset = offset / 2; - if (offset + loopsize * 2 > blocksize) - return -EINVAL; - for (i = 0; i < loopsize; i++) - wblock[woffset + i] = wblock[woffset - i -1]; - offset += loopsize * 2; - } else { - if (offset + loopsize > blocksize) - return -EINVAL; - for (i = 0; i < loopsize; i++) - block[offset + i] = block[offset - i -1]; - offset += loopsize; - } - - /* modify loop pointers */ - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_BIDIR_LOOP) { - sp->v.loopend += loopsize; - } else { - sp->v.loopstart += loopsize; - sp->v.loopend += loopsize; - } - /* add sample pointer */ - sp->v.end += loopsize; - } -#endif - - /* loopend -> sample end */ - size = sp->v.size - loopend; - if (size < 0) - return -EINVAL; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - size *= 2; - if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) { - snd_emu10k1_synth_free(emu, sp->block); - sp->block = NULL; - return -EFAULT; - } - offset += size; /* clear rest of samples (if any) */ if (offset < blocksize) From de67aab120d4d5ba7d9e94ee5b25464ae0d1bd0e Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:19 +0200 Subject: [PATCH 63/74] ALSA: emux: centralize & improve patch info validation This does several closely related things: - Move the code from the drivers into the SoundFont loader, which de-duplicates it. - Sort of explain the weird "recalculate address offset" feature. Note that I don't think it actually makes any sense - the calling user space code should do that. The background is certainly that the source data (the SoundFont format) uses pointers into a single wave block (and the API allows doing the same for on-board ROM), but the API expects the wave data from user space to be pre-chopped into individual patches anyway. - Make sure that the specified offsets actually lie within the supplied wave data. Note that we don't validate ROM offsets, so one can play back anything within the sound card's address space. - In load_guspatch(), don't call the sample_new callback anymore when the patch size is zero, as was already the case in load_data(). The callbacks would instantly return in that case anyway; these checks are now removed. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-7-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000_patch.c | 13 ----------- sound/pci/emu10k1/emu10k1_patch.c | 16 ------------- sound/synth/emux/soundfont.c | 37 ++++++++++++++++++++++++++++++- 3 files changed, 36 insertions(+), 30 deletions(-) diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index 8c1e7f2bfc34..ab4f988f080d 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -148,13 +148,6 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, if (snd_BUG_ON(!sp)) return -EINVAL; - if (sp->v.size == 0) - return 0; - - /* be sure loop points start < end */ - if (sp->v.loopstart > sp->v.loopend) - swap(sp->v.loopstart, sp->v.loopend); - /* compute true data size to be loaded */ truesize = sp->v.size; if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) @@ -177,12 +170,6 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, return -EFAULT; } - /* recalculate address offset */ - sp->v.end -= sp->v.start; - sp->v.loopstart -= sp->v.start; - sp->v.loopend -= sp->v.start; - sp->v.start = 0; - /* dram position (in word) -- mem_offset is byte */ dram_offset = EMU8000_DRAM_OFFSET + (sp->block->offset >> 1); dram_start = dram_offset; diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 49214c226808..47d69a0e44bc 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -35,28 +35,12 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, if (snd_BUG_ON(!sp || !hdr)) return -EINVAL; - if (sp->v.size == 0) { - dev_dbg(emu->card->dev, - "emu: rom font for sample %d\n", sp->v.sample); - return 0; - } - if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP | SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) { /* should instead return -ENOTSUPP; but compatibility */ printk(KERN_WARNING "Emu10k1 wavetable patch %d with unsupported loop feature\n", sp->v.sample); } - /* recalculate address offset */ - sp->v.end -= sp->v.start; - sp->v.loopstart -= sp->v.start; - sp->v.loopend -= sp->v.start; - sp->v.start = 0; - - /* be sure loop points start < end */ - if (sp->v.loopstart >= sp->v.loopend) - swap(sp->v.loopstart, sp->v.loopend); - /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index ad0231d7a39d..6d6f0102ed5b 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -689,6 +689,21 @@ find_sample(struct snd_soundfont *sf, int sample_id) } +static int +validate_sample_info(struct soundfont_sample_info *si) +{ + if (si->end < 0 || si->end > si->size) + return -EINVAL; + if (si->loopstart < 0 || si->loopstart > si->end) + return -EINVAL; + if (si->loopend < 0 || si->loopend > si->end) + return -EINVAL; + /* be sure loop points start < end */ + if (si->loopstart > si->loopend) + swap(si->loopstart, si->loopend); + return 0; +} + /* * Load sample information, this can include data to be loaded onto * the soundcard. It can also just be a pointer into soundcard ROM. @@ -727,6 +742,21 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) return -EINVAL; } + if (sample_info.size > 0) { + if (sample_info.start < 0) + return -EINVAL; + + // Here we "rebase out" the start address, because the + // real start is the start of the provided sample data. + sample_info.end -= sample_info.start; + sample_info.loopstart -= sample_info.start; + sample_info.loopend -= sample_info.start; + sample_info.start = 0; + + if (validate_sample_info(&sample_info) < 0) + return -EINVAL; + } + /* Allocate a new sample structure */ sp = sf_sample_new(sflist, sf); if (!sp) @@ -974,6 +1004,11 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) smp->v.loopend = patch.loop_end; smp->v.size = patch.len; + if (validate_sample_info(&smp->v) < 0) { + sf_sample_delete(sflist, sf, smp); + return -EINVAL; + } + /* set up mode flags */ smp->v.mode_flags = 0; if (!(patch.mode & WAVE_16_BITS)) @@ -1011,7 +1046,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) /* * load wave data */ - if (sflist->callback.sample_new) { + if (smp->v.size > 0 && sflist->callback.sample_new) { rc = sflist->callback.sample_new (sflist->callback.private_data, smp, sflist->memhdr, data, count); From 89b32ccb12ae67e630c6453d778ec30a592a212f Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:20 +0200 Subject: [PATCH 64/74] ALSA: emux: improve patch ioctl data validation In load_data(), make the validation of and skipping over the main info block match that in load_guspatch(). In load_guspatch(), add checking that the specified patch length matches the actually supplied data, like load_data() already did. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-8-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/soundfont.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 6d6f0102ed5b..4edc693da8e7 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -716,7 +716,6 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) struct snd_soundfont *sf; struct soundfont_sample_info sample_info; struct snd_sf_sample *sp; - long off; /* patch must be opened */ sf = sflist->currsf; @@ -726,12 +725,16 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) if (is_special_type(sf->type)) return -EINVAL; + if (count < (long)sizeof(sample_info)) { + return -EINVAL; + } if (copy_from_user(&sample_info, data, sizeof(sample_info))) return -EFAULT; + data += sizeof(sample_info); + count -= sizeof(sample_info); - off = sizeof(sample_info); - - if (sample_info.size != (count-off)/2) + // SoundFont uses S16LE samples. + if (sample_info.size * 2 != count) return -EINVAL; /* Check for dup */ @@ -774,7 +777,7 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) int rc; rc = sflist->callback.sample_new (sflist->callback.private_data, sp, sflist->memhdr, - data + off, count - off); + data, count); if (rc < 0) { sf_sample_delete(sflist, sf, sp); return rc; @@ -986,10 +989,12 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) } if (copy_from_user(&patch, data, sizeof(patch))) return -EFAULT; - count -= sizeof(patch); data += sizeof(patch); + if ((patch.len << (patch.mode & WAVE_16_BITS ? 1 : 0)) != count) + return -EINVAL; + sf = newsf(sflist, SNDRV_SFNT_PAT_TYPE_GUS|SNDRV_SFNT_PAT_SHARED, NULL); if (sf == NULL) return -ENOMEM; From 6e36d4c2744e143625cd2fcbf9d38ff76cda5e2a Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:21 +0200 Subject: [PATCH 65/74] ALSA: emu10k1: move patch loader assertions into low-level functions Convert some checks in snd_emu10k1_sample_new() back into assertions (as they were prior to da3cec35dd (ALSA: Kill snd_assert() in sound/pci/*, 2008-08-08)), and move them into the low-level memory access functions they protect. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-9-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 4 ---- sound/pci/emu10k1/memory.c | 6 ++++++ 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 47d69a0e44bc..55bb60d31fe4 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -65,8 +65,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, size = BLANK_HEAD_SIZE; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; - if (offset + size > blocksize) - return -EINVAL; snd_emu10k1_synth_bzero(emu, sp->block, offset, size); offset += size; @@ -74,8 +72,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, size = sp->v.size; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; - if (offset + size > blocksize) - return -EINVAL; if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) { snd_emu10k1_synth_free(emu, sp->block); sp->block = NULL; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 20b07117574b..fc9444404151 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -574,6 +574,9 @@ int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk void *ptr; struct snd_emu10k1_memblk *p = (struct snd_emu10k1_memblk *)blk; + if (snd_BUG_ON(offset + size > p->mem.size)) + return -EFAULT; + offset += blk->offset & (PAGE_SIZE - 1); end_offset = offset + size; page = get_aligned_page(offset); @@ -604,6 +607,9 @@ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_me void *ptr; struct snd_emu10k1_memblk *p = (struct snd_emu10k1_memblk *)blk; + if (snd_BUG_ON(offset + size > p->mem.size)) + return -EFAULT; + offset += blk->offset & (PAGE_SIZE - 1); end_offset = offset + size; page = get_aligned_page(offset); From 38fc804a776ea66ca8ac8113022e445c587f5e01 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:22 +0200 Subject: [PATCH 66/74] ALSA: emu10k1: fix sample signedness issues in wavetable loader The hardware supports S16LE and U8 samples, while U16LE and S8 (which the driver implicitly claims to support) require sign flipping. Note that this matters only for the GUS patch loader, as the implemented SoundFont v2.01 spec is limited to S16LE. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-10-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 4 +-- sound/pci/emu10k1/emu10k1_patch.c | 30 ++++++++----------- sound/pci/emu10k1/memory.c | 49 +++++++++++++++++++++++++------ 3 files changed, 55 insertions(+), 28 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 1af9e6819392..9e3bd4f81460 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1882,8 +1882,8 @@ int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size, struct snd_dma_buffer *dmab); struct snd_util_memblk *snd_emu10k1_synth_alloc(struct snd_emu10k1 *emu, unsigned int size); int snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *blk); -int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size); -int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, const char __user *data, int size); +int snd_emu10k1_synth_memset(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size, u8 value); +int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, const char __user *data, int size, u32 xor); int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk); /* voice allocation */ diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 55bb60d31fe4..eb3d1ef8a33a 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -26,6 +26,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, struct snd_util_memhdr *hdr, const void __user *data, long count) { + u8 fill; + u32 xor; int offset; int truesize, size, blocksize; unsigned int start_addr; @@ -41,6 +43,14 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, sp->v.sample); } + if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS) { + fill = 0x80; + xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0 : 0x80808080; + } else { + fill = 0; + xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0x80008000 : 0; + } + /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) @@ -65,14 +75,14 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, size = BLANK_HEAD_SIZE; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; - snd_emu10k1_synth_bzero(emu, sp->block, offset, size); + snd_emu10k1_synth_memset(emu, sp->block, offset, size, fill); offset += size; /* copy provided samples */ size = sp->v.size; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; - if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) { + if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) { snd_emu10k1_synth_free(emu, sp->block); sp->block = NULL; return -EFAULT; @@ -81,7 +91,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* clear rest of samples (if any) */ if (offset < blocksize) - snd_emu10k1_synth_bzero(emu, sp->block, offset, blocksize - offset); + snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill); if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) { /* if no blank loop is attached in the sample, add it */ @@ -91,20 +101,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, } } -#if 0 /* not supported yet */ - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) { - /* unsigned -> signed */ - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) { - unsigned short *wblock = (unsigned short*)block; - for (i = 0; i < truesize; i++) - wblock[i] ^= 0x8000; - } else { - for (i = 0; i < truesize; i++) - block[i] ^= 0x80; - } - } -#endif - /* recalculate offset */ start_addr = BLANK_HEAD_SIZE * 2; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index fc9444404151..d29711777161 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -565,10 +565,10 @@ static inline void *offset_ptr(struct snd_emu10k1 *emu, int page, int offset) } /* - * bzero(blk + offset, size) + * memset(blk + offset, value, size) */ -int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, - int offset, int size) +int snd_emu10k1_synth_memset(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, + int offset, int size, u8 value) { int page, nextofs, end_offset, temp, temp1; void *ptr; @@ -588,20 +588,47 @@ int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk temp = temp1; ptr = offset_ptr(emu, page + p->first_page, offset); if (ptr) - memset(ptr, 0, temp); + memset(ptr, value, temp); offset = nextofs; page++; } while (offset < end_offset); return 0; } -EXPORT_SYMBOL(snd_emu10k1_synth_bzero); +EXPORT_SYMBOL(snd_emu10k1_synth_memset); + +// Note that the value is assumed to be suitably repetitive. +static void xor_range(void *ptr, int size, u32 value) +{ + if ((long)ptr & 1) { + *(u8 *)ptr ^= (u8)value; + ptr++; + size--; + } + if (size > 1 && ((long)ptr & 2)) { + *(u16 *)ptr ^= (u16)value; + ptr += 2; + size -= 2; + } + while (size > 3) { + *(u32 *)ptr ^= value; + ptr += 4; + size -= 4; + } + if (size > 1) { + *(u16 *)ptr ^= (u16)value; + ptr += 2; + size -= 2; + } + if (size > 0) + *(u8 *)ptr ^= (u8)value; +} /* - * copy_from_user(blk + offset, data, size) + * copy_from_user(blk + offset, data, size) ^ xor */ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, - int offset, const char __user *data, int size) + int offset, const char __user *data, int size, u32 xor) { int page, nextofs, end_offset, temp, temp1; void *ptr; @@ -620,8 +647,12 @@ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_me if (temp1 < temp) temp = temp1; ptr = offset_ptr(emu, page + p->first_page, offset); - if (ptr && copy_from_user(ptr, data, temp)) - return -EFAULT; + if (ptr) { + if (copy_from_user(ptr, data, temp)) + return -EFAULT; + if (xor) + xor_range(ptr, temp, xor); + } offset = nextofs; data += temp; page++; From bca5174b437307c9315e25768ed2b2bfcf6f561c Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:23 +0200 Subject: [PATCH 67/74] ALSA: emu10k1: fix playback of 8-bit wavetable samples Samples are byte-sized in this mode, and thus the offset calculation needs no shifting. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-11-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_callback.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 941bfbf812ed..5f6c47cbb809 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -310,6 +310,7 @@ start_voice(struct snd_emux_voice *vp) { unsigned int temp; int ch; + bool w_16; u32 psst, dsl, map, ccca, vtarget; unsigned int addr, mapped_offset; struct snd_midi_channel *chan; @@ -321,6 +322,7 @@ start_voice(struct snd_emux_voice *vp) if (snd_BUG_ON(ch < 0)) return -EINVAL; chan = vp->chan; + w_16 = !(vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS); emem = (struct snd_emu10k1_memblk *)vp->block; if (emem == NULL) @@ -330,7 +332,7 @@ start_voice(struct snd_emux_voice *vp) /* dev_err(hw->card->devK, "emu: cannot map!\n"); */ return -ENOMEM; } - mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1; + mapped_offset = snd_emu10k1_memblk_offset(emem) >> w_16; vp->reg.start += mapped_offset; vp->reg.end += mapped_offset; vp->reg.loopstart += mapped_offset; @@ -371,7 +373,7 @@ start_voice(struct snd_emux_voice *vp) unsigned int shift = (vp->apitch - 0xe000) >> 10; ccca |= shift << 25; } - if (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS) + if (!w_16) ccca |= CCCA_8BITSELECT; vtarget = (unsigned int)vp->vtarget << 16; From 93fd86a47de3097488611ffbfe12b4940933670d Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:24 +0200 Subject: [PATCH 68/74] ALSA: emu10k1: merge conditions in patch loader This de-duplicates the code slightly. But the real reason is that it moves the code up, which the next patch will depend on. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-12-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index eb3d1ef8a33a..281881f7d0a4 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -53,8 +53,14 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) + if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) { truesize += BLANK_LOOP_SIZE; + /* if no blank loop is attached in the sample, add it */ + if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_SINGLESHOT) { + sp->v.loopstart = sp->v.end + BLANK_LOOP_START; + sp->v.loopend = sp->v.end + BLANK_LOOP_END; + } + } /* try to allocate a memory block */ blocksize = truesize; @@ -93,14 +99,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, if (offset < blocksize) snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill); - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) { - /* if no blank loop is attached in the sample, add it */ - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_SINGLESHOT) { - sp->v.loopstart = sp->v.end + BLANK_LOOP_START; - sp->v.loopend = sp->v.end + BLANK_LOOP_END; - } - } - /* recalculate offset */ start_addr = BLANK_HEAD_SIZE * 2; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) From 392925791a5b6f41806d445ea71319a116e32295 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:25 +0200 Subject: [PATCH 69/74] ALSA: emu10k1: fix wavetable offset recalculation The offsets are counted in samples, not in bytes. While the code block is being rewritten, also move it up a bit, to avoid churn in a subsequent patch. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-13-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 281881f7d0a4..ad16de99b800 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -30,7 +30,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, u32 xor; int offset; int truesize, size, blocksize; - unsigned int start_addr; struct snd_emu10k1 *emu; emu = rec->hw; @@ -62,6 +61,12 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, } } + /* recalculate offset */ + sp->v.start += BLANK_HEAD_SIZE; + sp->v.end += BLANK_HEAD_SIZE; + sp->v.loopstart += BLANK_HEAD_SIZE; + sp->v.loopend += BLANK_HEAD_SIZE; + /* try to allocate a memory block */ blocksize = truesize; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) @@ -99,15 +104,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, if (offset < blocksize) snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill); - /* recalculate offset */ - start_addr = BLANK_HEAD_SIZE * 2; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - start_addr >>= 1; - sp->v.start += start_addr; - sp->v.end += start_addr; - sp->v.loopstart += start_addr; - sp->v.loopend += start_addr; - return 0; } From 80d7c3cccd546c16da2ef9d2e88eaf215498c1e1 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:26 +0200 Subject: [PATCH 70/74] ALSA: emu10k1: de-duplicate size calculations for 16-bit samples Instead of repeatedly checking the sample width, assign a size shift centrally. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-14-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index ad16de99b800..481fe03fef4d 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -28,6 +28,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, { u8 fill; u32 xor; + int shift; int offset; int truesize, size, blocksize; struct snd_emu10k1 *emu; @@ -43,9 +44,11 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, } if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS) { + shift = 0; fill = 0x80; xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0 : 0x80808080; } else { + shift = 1; fill = 0; xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0x80008000 : 0; } @@ -68,9 +71,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, sp->v.loopend += BLANK_HEAD_SIZE; /* try to allocate a memory block */ - blocksize = truesize; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - blocksize *= 2; + blocksize = truesize << shift; sp->block = snd_emu10k1_synth_alloc(emu, blocksize); if (sp->block == NULL) { dev_dbg(emu->card->dev, @@ -83,16 +84,12 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* write blank samples at head */ offset = 0; - size = BLANK_HEAD_SIZE; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - size *= 2; + size = BLANK_HEAD_SIZE << shift; snd_emu10k1_synth_memset(emu, sp->block, offset, size, fill); offset += size; /* copy provided samples */ - size = sp->v.size; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - size *= 2; + size = sp->v.size << shift; if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) { snd_emu10k1_synth_free(emu, sp->block); sp->block = NULL; From 65db949667b0b74f4534e96f762aff0e6687dc51 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:27 +0200 Subject: [PATCH 71/74] ALSA: emu10k1: improve cache behavior documentation Resulting from more reverse engineering in the course of debugging. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-15-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 28 ++++++++++++++++++---------- 1 file changed, 18 insertions(+), 10 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 9e3bd4f81460..12c7dc760724 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -598,17 +598,25 @@ SUB_REG(PEFE, FILTERAMOUNT, 0x000000ff) /* Filter envlope amount */ // In stereo mode, the two channels' caches are concatenated into one, // and hold the interleaved frames. // The cache holds 64 frames, so the upper half is not used in 8-bit mode. -// All registers mentioned below count in frames. -// The cache is a ring buffer; CCR_READADDRESS operates modulo 64. -// The cache is filled from (CCCA_CURRADDR - CCR_CACHEINVALIDSIZE) -// into (CCR_READADDRESS - CCR_CACHEINVALIDSIZE). +// All registers mentioned below count in frames. Shortcuts: +// CA = CCCA_CURRADDR, CRA = CCR_READADDRESS, +// CLA = CCR_CACHELOOPADDRHI:CLP_CACHELOOPADDR, +// CIS = CCR_CACHEINVALIDSIZE, LIS = CCR_LOOPINVALSIZE, +// CLF = CCR_CACHELOOPFLAG, LF = CCR_LOOPFLAG +// The cache is a ring buffer; CRA operates modulo 64. +// The cache is filled from (CA - CIS) into (CRA - CIS). // The engine has a fetch threshold of 32 bytes, so it tries to keep -// CCR_CACHEINVALIDSIZE below 8 (16-bit stereo), 16 (16-bit mono, -// 8-bit stereo), or 32 (8-bit mono). The actual transfers are pretty -// unpredictable, especially if several voices are running. -// Frames are consumed at CCR_READADDRESS, which is incremented afterwards, -// along with CCCA_CURRADDR and CCR_CACHEINVALIDSIZE. This implies that the -// actual playback position always lags CCCA_CURRADDR by exactly 64 frames. +// CIS below 8 (16-bit stereo), 16 (16-bit mono, 8-bit stereo), or +// 32 (8-bit mono). The actual transfers are pretty unpredictable, +// especially if several voices are running. +// Frames are consumed at CRA, which is incremented afterwards, +// along with CA and CIS. This implies that the actual playback +// position always lags CA by exactly 64 frames. +// When CA reaches DSL_LOOPENDADDR, LF is set for one frame's time. +// LF's rising edge causes the current values of CA and CIS to be +// copied into CLA and LIS, resp., and CLF to be set. +// If CLF is set, the first LIS of the CIS frames are instead +// filled from (CLA - LIS), and CLF is subsequently reset. #define CD0 0x20 /* Cache data registers 0 .. 0x1f */ #define PTB 0x40 /* Page table base register */ From d0440680a197bef3cfd725b0982518f5d05079a5 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:28 +0200 Subject: [PATCH 72/74] ALSA: emu10k1: fix wavetable playback position and caching, take 2 Compensate for the cache lag of 64 frames, and actually populate the cache. Without these, the playback would start with garbage (which would be (mostly?) masqueraded by the note's attack phase). Note that we set the starting address only 61 frames ahead, to compensate for the interpolator's epsilon. Unlike for PCM playback, we don't even need to manually silence-fill the first frames in the cache, because we insert some silence in front of each sample anyway. A challenge are extremely short samples with a loop end below the cache size, because a) we'd have to wrap the current address to be within the loop and b) automatic pre-filling of the cache with the right data does not work in this case. We could pre-fill the cache manually, but that's slow, requires additional code for each sample width, and is made even more complex by the driver's virtual address space having no contiguous mapping for the CPU. We could have the engine fill the cache piece-wise (which is really what happens when playback is running), but that would also be complex, and we'd need to wait for the engine to handle each piece, so it wouldn't be that much faster than the manual fill. For the case of requiring only one loop iteration prior to reaching the cache size, we could leverage the engine's looping mechanism around CCR_CACHELOOPFLAG, but this special case doesn't seem worth the complexity. So we just unroll the loop as far as necessary to be able to play back the sample without any fiddling. Pedantically, this would be incorrect for loop-until-release samples with a low loop end which are released very quickly, but that would be relatively harmless, is not a plausible use case in the first place, and SoundFont sample mode 3 isn't actually implemented anyway (it's conflated with mode 1, infinite looping). Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-16-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_callback.c | 7 ++-- sound/pci/emu10k1/emu10k1_patch.c | 53 +++++++++++++++++++++++++--- 2 files changed, 53 insertions(+), 7 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 5f6c47cbb809..ef26e4d3e2a3 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -255,7 +255,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, /* check if sample is finished playing (non-looping only) */ if (bp != best + V_OFF && bp != best + V_FREE && (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) { - val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch); + val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64 + 3; if (val >= vp->reg.loopstart) bp = best + V_OFF; } @@ -364,7 +364,7 @@ start_voice(struct snd_emux_voice *vp) map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - addr = vp->reg.start; + addr = vp->reg.start + 64 - 3; temp = vp->reg.parm.filterQ; ccca = (temp << 28) | addr; if (vp->apitch < 0xe400) @@ -432,6 +432,9 @@ start_voice(struct snd_emux_voice *vp) /* Q & current address (Q 4bit value, MSB) */ CCCA, ccca, + /* cache */ + CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64), + /* reset volume */ VTFT, vtarget | vp->ftarget, CVCF, vtarget | CVCF_CURRENTFILTER_MASK, diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 481fe03fef4d..2a13fb32c1d2 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -31,6 +31,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, int shift; int offset; int truesize, size, blocksize; + int loop_start, loop_end, loop_size, data_end, unroll; struct snd_emu10k1 *emu; emu = rec->hw; @@ -64,12 +65,35 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, } } + loop_start = sp->v.loopstart; + loop_end = sp->v.loopend; + loop_size = loop_end - loop_start; + if (!loop_size) + return -EINVAL; + data_end = sp->v.end; + /* recalculate offset */ sp->v.start += BLANK_HEAD_SIZE; sp->v.end += BLANK_HEAD_SIZE; sp->v.loopstart += BLANK_HEAD_SIZE; sp->v.loopend += BLANK_HEAD_SIZE; + // Automatic pre-filling of the cache does not work in the presence + // of loops (*), and we don't want to fill it manually, as that is + // fiddly and slow. So we unroll the loop until the loop end is + // beyond the cache size. + // (*) Strictly speaking, a single iteration is supported (that's + // how it works when the playback engine runs), but handling this + // special case is not worth it. + unroll = 0; + while (sp->v.loopend < 64) { + truesize += loop_size; + sp->v.loopstart += loop_size; + sp->v.loopend += loop_size; + sp->v.end += loop_size; + unroll++; + } + /* try to allocate a memory block */ blocksize = truesize << shift; sp->block = snd_emu10k1_synth_alloc(emu, blocksize); @@ -89,12 +113,26 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, offset += size; /* copy provided samples */ - size = sp->v.size << shift; - if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) { - snd_emu10k1_synth_free(emu, sp->block); - sp->block = NULL; - return -EFAULT; + if (unroll && loop_end <= data_end) { + size = loop_end << shift; + if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) + goto faulty; + offset += size; + + data += loop_start << shift; + while (--unroll > 0) { + size = loop_size << shift; + if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) + goto faulty; + offset += size; + } + + size = (data_end - loop_start) << shift; + } else { + size = data_end << shift; } + if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) + goto faulty; offset += size; /* clear rest of samples (if any) */ @@ -102,6 +140,11 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill); return 0; + +faulty: + snd_emu10k1_synth_free(emu, sp->block); + sp->block = NULL; + return -EFAULT; } /* From 62001ad1b4ee412c915120ee6ef2cfdc924bd007 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:29 +0200 Subject: [PATCH 73/74] ALSA: emu10k1: shrink blank space in front of wavetable samples There is no need for it to be 32 samples - 3 will do just fine (which is the interpolator's epsilon). The old size was presumably meant to compensate for the cache's presence, but we're now handling that properly. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-17-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 2a13fb32c1d2..dbfa89435ac2 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -16,7 +16,7 @@ #define BLANK_LOOP_START 4 #define BLANK_LOOP_END 8 #define BLANK_LOOP_SIZE 12 -#define BLANK_HEAD_SIZE 32 +#define BLANK_HEAD_SIZE 3 /* * allocate a sample block and copy data from userspace From 4c4cbe66828f185903d2127aed6ac9b7302e9d3a Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:30 +0200 Subject: [PATCH 74/74] ALSA: emux: simplify snd_sf_list.callback handling Both drivers provide both sample_new and sample_free, and it makes no sense to pretend that they could not. In fact, load_data() would already crash if sample_new was null. So remove the remaining null checks. Contrary to that, the emu10k1 driver actually has a null sample_reset, though I'm not convinced that this inconsistency is justified. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-18-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/emux.c | 6 ++---- sound/synth/emux/soundfont.c | 12 +++++------- 2 files changed, 7 insertions(+), 11 deletions(-) diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index a82af9374852..01444fc960d0 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -94,10 +94,8 @@ int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, ch /* create soundfont list */ memset(&sf_cb, 0, sizeof(sf_cb)); sf_cb.private_data = emu; - if (emu->ops.sample_new) - sf_cb.sample_new = sf_sample_new; - if (emu->ops.sample_free) - sf_cb.sample_free = sf_sample_free; + sf_cb.sample_new = sf_sample_new; + sf_cb.sample_free = sf_sample_free; if (emu->ops.sample_reset) sf_cb.sample_reset = sf_sample_reset; emu->sflist = snd_sf_new(&sf_cb, emu->memhdr); diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 4edc693da8e7..2373ed580bf8 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -1051,7 +1051,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) /* * load wave data */ - if (smp->v.size > 0 && sflist->callback.sample_new) { + if (smp->v.size > 0) { rc = sflist->callback.sample_new (sflist->callback.private_data, smp, sflist->memhdr, data, count); @@ -1416,9 +1416,8 @@ snd_sf_clear(struct snd_sf_list *sflist) } for (sp = sf->samples; sp; sp = nextsp) { nextsp = sp->next; - if (sflist->callback.sample_free) - sflist->callback.sample_free(sflist->callback.private_data, - sp, sflist->memhdr); + sflist->callback.sample_free(sflist->callback.private_data, + sp, sflist->memhdr); kfree(sp); } kfree(sf); @@ -1520,9 +1519,8 @@ snd_soundfont_remove_unlocked(struct snd_sf_list *sflist) nextsp = sp->next; sf->samples = nextsp; sflist->mem_used -= sp->v.truesize; - if (sflist->callback.sample_free) - sflist->callback.sample_free(sflist->callback.private_data, - sp, sflist->memhdr); + sflist->callback.sample_free(sflist->callback.private_data, + sp, sflist->memhdr); kfree(sp); } }