From 4819d06292c9b57eabdd6d1603e49a27baf183be Mon Sep 17 00:00:00 2001 From: Jon Hunter Date: Fri, 24 May 2019 09:54:04 +0100 Subject: [PATCH 01/12] ASoC: simple-card: Restore original configuration of DAI format Revert commit 069d037aea98 ("ASoC: simple-card: Fix configuration of DAI format"). During further review, it turns out that the actual issue was caused by an incorrectly formatted device-tree node describing the soundcard. The following is incorrect because the simple-audio-card 'bitclock-master' and 'frame-master' properties should not reference the actual codec phandle ... sound { compatible = "simple-audio-card"; ... => simple-audio-card,bitclock-master = <&codec>; => simple-audio-card,frame-master = <&codec>; ... simple-audio-card,cpu { sound-dai = <&xxx>; }; simple-audio-card,codec { => sound-dai = <&codec>; }; }; Rather, these properties should reference the phandle to the 'simple-audio-card,codec' property as shown below ... sound { compatible = "simple-audio-card"; ... => simple-audio-card,bitclock-master = <&codec>; => simple-audio-card,frame-master = <&codec>; ... simple-audio-card,cpu { sound-dai = <&xxx>; }; => codec: simple-audio-card,codec { /* simple-card wants here */ sound-dai = <&xxx>; /* not here */ }; }; Signed-off-by: Jon Hunter Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c2c8dcbcf795..9b568f578bcd 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -283,6 +283,11 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, codec_dai = dai_props->codec_dai = &priv->dais[li->dais++]; + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); + if (ret < 0) + goto dai_link_of_err; + simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu); @@ -293,11 +298,6 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_parse_daifmt(dev, node, dai_link->codecs->of_node, - prefix, &dai_link->dai_fmt); - if (ret < 0) - goto dai_link_of_err; - ret = asoc_simple_parse_platform(plat, dai_link); if (ret < 0) goto dai_link_of_err; From 34ac3c3eb8f0c07252ceddf0a22dd240e5c91ccb Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Thu, 23 May 2019 10:12:01 -0700 Subject: [PATCH 02/12] ASoC: core: lock client_mutex while removing link components Removing link components results in topology unloading. So, acquire the client_mutex before removing components in soc_remove_link_components. This will prevent the lockdep warning seen when dai links are removed during topology removal. Signed-off-by: Ranjani Sridharan Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2403bec2fccf..7c9415987ac7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1005,12 +1005,14 @@ static void soc_remove_link_components(struct snd_soc_card *card, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; + mutex_lock(&client_mutex); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; if (component->driver->remove_order == order) soc_remove_component(component); } + mutex_unlock(&client_mutex); } static void soc_remove_dai_links(struct snd_soc_card *card) From 29040d1ac569606fece70966179de272cfc0d4db Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 27 May 2019 16:51:34 +0900 Subject: [PATCH 03/12] ASoC: soc-core: fixup references at soc_cleanup_card_resources() commit 53e947a0e1f7 ("ASoC: soc-core: merge card resources cleanup method") merged cleanup method of snd_soc_instantiate_card() and soc_cleanup_card_resources(). But, after this commit, if user uses unbind/bind to Component factor drivers, Kernel might indicates refcount error at soc_cleanup_card_resources(). The 1st reason is card->snd_card is still exist even though snd_card_free() was called, but it is already cleaned. We need to set NULL to it. 2nd is card->dapm and card create debugfs, but its dentry is still exist even though it was removed. We need to set NULL to it. Fixes: 53e947a0e1f7 ("ASoC: soc-core: merge card resources cleanup method") Cc: stable@vger.kernel.org # for v5.1 Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 7 ++++++- sound/soc/soc-dapm.c | 3 +++ 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7c9415987ac7..46042d41b79b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -228,7 +228,10 @@ static void soc_init_card_debugfs(struct snd_soc_card *card) static void soc_cleanup_card_debugfs(struct snd_soc_card *card) { + if (!card->debugfs_card_root) + return; debugfs_remove_recursive(card->debugfs_card_root); + card->debugfs_card_root = NULL; } static void snd_soc_debugfs_init(void) @@ -2039,8 +2042,10 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) static int soc_cleanup_card_resources(struct snd_soc_card *card) { /* free the ALSA card at first; this syncs with pending operations */ - if (card->snd_card) + if (card->snd_card) { snd_card_free(card->snd_card); + card->snd_card = NULL; + } /* remove and free each DAI */ soc_remove_dai_links(card); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 62e27defce56..5fc57af9cb6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2192,7 +2192,10 @@ static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) { + if (!dapm->debugfs_dapm) + return; debugfs_remove_recursive(dapm->debugfs_dapm); + dapm->debugfs_dapm = NULL; } #else From b85459aafae63f250606bd406d4f7537fda33b51 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Fri, 24 May 2019 14:09:17 -0500 Subject: [PATCH 04/12] ASoC: SOF: core: remove DSP after unregistering machine driver snd_sof_remove() disables the DSP and unmaps the DSP BAR. Removing topology after disabling the DSP results in a kernel panic while unloading the pipeline widget. This is because pipeline widget unload attempts to power down the core it is scheduled on by accessing the DSP registers. So, the suggested fix here is to unregister the machine driver first to remove the topology and then disable the DSP to avoid the situation described above. Note that the kernel panic only happens in cases where the HDaudio link is not managed by the hdac library, e.g. no codec or when HDMI is not supported. When the hdac library is used, snd_sof_remove() calls snd_hdac_ext_bus_device_remove() to remove the codec which unregisters the component driver thereby also removing the topology before the DSP is disabled. Fixes: c16211d6226 ("ASoC: SOF: Add Sound Open Firmware driver core") Reviewed-by: Takashi Iwai Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 32105e0fabe8..0bc4a8472c10 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -484,7 +484,6 @@ int snd_sof_device_remove(struct device *dev) snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); snd_sof_free_trace(sdev); - snd_sof_remove(sdev); /* * Unregister machine driver. This will unbind the snd_card which @@ -494,6 +493,14 @@ int snd_sof_device_remove(struct device *dev) if (!IS_ERR_OR_NULL(pdata->pdev_mach)) platform_device_unregister(pdata->pdev_mach); + /* + * Unregistering the machine driver results in unloading the topology. + * Some widgets, ex: scheduler, attempt to power down the core they are + * scheduled on, when they are unloaded. Therefore, the DSP must be + * removed only after the topology has been unloaded. + */ + snd_sof_remove(sdev); + /* release firmware */ release_firmware(pdata->fw); pdata->fw = NULL; From 13931ae31b67a8a26a4cd417088fc43e3d4a8591 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Fri, 24 May 2019 14:09:18 -0500 Subject: [PATCH 05/12] ASoC: SOF: core: remove snd_soc_unregister_component in case of error No need to call snd_soc_unregister_component in case of error because the component device is resource-managed. Fixes: c16211d6226 ("ASoC: SOF: Add Sound Open Firmware driver core") Reviewed-by: Takashi Iwai Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 0bc4a8472c10..693ad83bffc9 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -382,7 +382,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) if (IS_ERR(plat_data->pdev_mach)) { ret = PTR_ERR(plat_data->pdev_mach); - goto comp_err; + goto fw_run_err; } dev_dbg(sdev->dev, "created machine %s\n", @@ -393,8 +393,6 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) return 0; -comp_err: - snd_soc_unregister_component(sdev->dev); fw_run_err: snd_sof_fw_unload(sdev); fw_load_err: From 0bce512e784d137700275f7839c4547eddbd4b6a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 24 May 2019 14:09:19 -0500 Subject: [PATCH 06/12] ASoC: SOF: core: fix error handling with the probe workqueue In some configurations, it's a requirement to split the probe in two, with a second part handled in a workqueue (e.g. for HDMI support which depends on the DRM modules). SOF already handles these configurations but the error flow is incorrect. When an error occurs in the workqueue, the probe has technically already completed. If we release the resources on errors, this generates kernel oops/use-after-free when the resources are released a second time on module removal. GitHub issue: https://github.com/thesofproject/linux/issues/945 Fixes: c16211d6226 ("ASoC: SOF: Add Sound Open Firmware driver core") Reviewed-by: Takashi Iwai Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 693ad83bffc9..5beda47cdf9f 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -393,6 +393,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) return 0; +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE) fw_run_err: snd_sof_fw_unload(sdev); fw_load_err: @@ -401,6 +402,21 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) snd_sof_free_debug(sdev); dbg_err: snd_sof_remove(sdev); +#else + + /* + * when the probe_continue is handled in a work queue, the + * probe does not fail so we don't release resources here. + * They will be released with an explicit call to + * snd_sof_device_remove() when the PCI/ACPI device is removed + */ + +fw_run_err: +fw_load_err: +ipc_err: +dbg_err: + +#endif return ret; } From 04ea642ff62a2b2da3d5844283991a41328f49b1 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 24 May 2019 14:09:20 -0500 Subject: [PATCH 07/12] ASoC: SOF: pcm: clear hw_params_upon_resume flag correctly sof_pcm_hw_params() can only be called once to setup the FW hw_params. So after calling sof_pcm_hw_params(), hw_params_upon_resume flag must be cleared to avoid multiple invoking sof_pcm_hw_params() by prepare. For example, after resume, there is an xrun happened, prepare() will be called. As the hw_params_upon_resume flag is not cleared, sof_pcm_hw_params() will be called and this will cause IPC timeout. This patch fixes such issues. Fixes: 868bd00f495 ("ASoC: SOF: Add PCM operations support") Reviewed-by: Takashi Iwai Signed-off-by: Libin Yang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 649968841dad..d6dc9a7df0f4 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -214,6 +214,9 @@ static int sof_pcm_hw_params(struct snd_pcm_substream *substream, INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work, sof_pcm_period_elapsed_work); + /* clear hw_params_upon_resume flag */ + spcm->hw_params_upon_resume[substream->stream] = 0; + return ret; } @@ -429,9 +432,6 @@ static int sof_pcm_open(struct snd_pcm_substream *substream) dev_dbg(sdev->dev, "pcm: open stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); - /* clear hw_params_upon_resume flag */ - spcm->hw_params_upon_resume[substream->stream] = 0; - caps = &spcm->pcm.caps[substream->stream]; ret = pm_runtime_get_sync(sdev->dev); From fab4edf42d2d68d0aa67822650174dcd0ee25ffa Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 24 May 2019 14:09:21 -0500 Subject: [PATCH 08/12] ASoC: SOF: pcm: remove warning - initialize workqueue on open If the SOF hw_params() fail, typically with an IPC error thrown by the firmware, the period_elapsed workqueue is not initialized, but we still cancel it in hw_free(), which results in a kernel warning. Move the initialization to the .open callback. Tested on Broadwell (Samus) and IceLake. Fixes: e2803e610ae ("ASoC: SOF: PCM: add period_elapsed work to fix race condition in interrupt context") GitHub issue: https://github.com/thesofproject/linux/issues/932 Reviewed-by: Takashi Iwai Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index d6dc9a7df0f4..dace6c4cd91e 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -211,9 +211,6 @@ static int sof_pcm_hw_params(struct snd_pcm_substream *substream, /* save pcm hw_params */ memcpy(&spcm->params[substream->stream], params, sizeof(*params)); - INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work, - sof_pcm_period_elapsed_work); - /* clear hw_params_upon_resume flag */ spcm->hw_params_upon_resume[substream->stream] = 0; @@ -432,6 +429,9 @@ static int sof_pcm_open(struct snd_pcm_substream *substream) dev_dbg(sdev->dev, "pcm: open stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); + INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work, + sof_pcm_period_elapsed_work); + caps = &spcm->pcm.caps[substream->stream]; ret = pm_runtime_get_sync(sdev->dev); From 5661ad9490ee4abdb27295880e524acc656c89e7 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Fri, 24 May 2019 14:09:22 -0500 Subject: [PATCH 09/12] ASoC: SOF: control: correct the copy size for bytes kcontrol put The size for the bytes kcontrol should include the abi header, that is, data->size + sizeof(*data), it is also aligned with get method after this change. Fixes: c3078f53970 ("ASoC: SOF: Add Sound Open Firmware KControl support") Reviewed-by: Takashi Iwai Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/control.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 11762c4580f1..84e2cbfbbcbb 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -349,6 +349,7 @@ int snd_sof_bytes_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; struct sof_abi_hdr *data = cdata->data; + size_t size = data->size + sizeof(*data); int ret, err; if (be->max > sizeof(ucontrol->value.bytes.data)) { @@ -358,10 +359,10 @@ int snd_sof_bytes_put(struct snd_kcontrol *kcontrol, return -EINVAL; } - if (data->size > be->max) { + if (size > be->max) { dev_err_ratelimited(sdev->dev, - "error: size too big %d bytes max is %d\n", - data->size, be->max); + "error: size too big %zu bytes max is %d\n", + size, be->max); return -EINVAL; } @@ -375,7 +376,7 @@ int snd_sof_bytes_put(struct snd_kcontrol *kcontrol, } /* copy from kcontrol */ - memcpy(data, ucontrol->value.bytes.data, data->size); + memcpy(data, ucontrol->value.bytes.data, size); /* notify DSP of byte control updates */ snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, From 1183e9a634db06825da7faba566bce50afde4357 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 24 May 2019 14:09:23 -0500 Subject: [PATCH 10/12] ASoC: SOF: ipc: fix a race, leading to IPC timeouts Currently on all supported platforms the IPC IRQ thread first signals the sender when an IPC response is received from the DSP, then unmasks the IPC interrupt. Those actions are performed without holding any locks, so the thread can be interrupted between them. IPC timeouts have been observed in such scenarios: if the sender is woken up and it proceeds with sending the next message without unmasking the IPC interrupt, it can miss the next response. This patch takes a spin-lock to prevent the IRQ thread from being preempted at that point. It also makes sure, that the next IPC transmission by the host cannot take place before the IRQ thread has finished updating all the required IPC registers. Fixes: 53e0c72d98b ("ASoC: SOF: Add support for IPC IO between DSP and Host") Reviewed-by: Takashi Iwai Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/bdw.c | 9 ++++----- sound/soc/sof/intel/byt.c | 10 +++++----- sound/soc/sof/intel/cnl.c | 4 ++++ sound/soc/sof/intel/hda-ipc.c | 17 ++++++++++++++--- sound/soc/sof/ipc.c | 13 ------------- 5 files changed, 27 insertions(+), 26 deletions(-) diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 065cb868bdfa..8ff3ee520aea 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -283,6 +283,8 @@ static irqreturn_t bdw_irq_thread(int irq, void *context) SHIM_IMRX, SHIM_IMRX_DONE, SHIM_IMRX_DONE); + spin_lock_irq(&sdev->ipc_lock); + /* * handle immediate reply from DSP core. If the msg is * found, set done bit in cmd_done which is called at the @@ -294,6 +296,8 @@ static irqreturn_t bdw_irq_thread(int irq, void *context) snd_sof_ipc_reply(sdev, ipcx); bdw_dsp_done(sdev); + + spin_unlock_irq(&sdev->ipc_lock); } ipcd = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_IPCD); @@ -485,7 +489,6 @@ static void bdw_get_reply(struct snd_sof_dev *sdev) { struct snd_sof_ipc_msg *msg = sdev->msg; struct sof_ipc_reply reply; - unsigned long flags; int ret = 0; /* @@ -501,8 +504,6 @@ static void bdw_get_reply(struct snd_sof_dev *sdev) /* get reply */ sof_mailbox_read(sdev, sdev->host_box.offset, &reply, sizeof(reply)); - spin_lock_irqsave(&sdev->ipc_lock, flags); - if (reply.error < 0) { memcpy(msg->reply_data, &reply, sizeof(reply)); ret = reply.error; @@ -521,8 +522,6 @@ static void bdw_get_reply(struct snd_sof_dev *sdev) } msg->reply_error = ret; - - spin_unlock_irqrestore(&sdev->ipc_lock, flags); } static void bdw_host_done(struct snd_sof_dev *sdev) diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 7bf9143d3106..9e4c07eb889b 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -329,6 +329,9 @@ static irqreturn_t byt_irq_thread(int irq, void *context) SHIM_IMRX, SHIM_IMRX_DONE, SHIM_IMRX_DONE); + + spin_lock_irq(&sdev->ipc_lock); + /* * handle immediate reply from DSP core. If the msg is * found, set done bit in cmd_done which is called at the @@ -340,6 +343,8 @@ static irqreturn_t byt_irq_thread(int irq, void *context) snd_sof_ipc_reply(sdev, ipcx); byt_dsp_done(sdev); + + spin_unlock_irq(&sdev->ipc_lock); } /* new message from DSP */ @@ -383,7 +388,6 @@ static void byt_get_reply(struct snd_sof_dev *sdev) { struct snd_sof_ipc_msg *msg = sdev->msg; struct sof_ipc_reply reply; - unsigned long flags; int ret = 0; /* @@ -399,8 +403,6 @@ static void byt_get_reply(struct snd_sof_dev *sdev) /* get reply */ sof_mailbox_read(sdev, sdev->host_box.offset, &reply, sizeof(reply)); - spin_lock_irqsave(&sdev->ipc_lock, flags); - if (reply.error < 0) { memcpy(msg->reply_data, &reply, sizeof(reply)); ret = reply.error; @@ -419,8 +421,6 @@ static void byt_get_reply(struct snd_sof_dev *sdev) } msg->reply_error = ret; - - spin_unlock_irqrestore(&sdev->ipc_lock, flags); } static void byt_host_done(struct snd_sof_dev *sdev) diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 08a1a3d3c08d..b2eba7adcad8 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -64,6 +64,8 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context) CNL_DSP_REG_HIPCCTL, CNL_DSP_REG_HIPCCTL_DONE, 0); + spin_lock_irq(&sdev->ipc_lock); + /* handle immediate reply from DSP core */ hda_dsp_ipc_get_reply(sdev); snd_sof_ipc_reply(sdev, msg); @@ -75,6 +77,8 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context) cnl_ipc_dsp_done(sdev); + spin_unlock_irq(&sdev->ipc_lock); + ret = IRQ_HANDLED; } diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 73ead7070cde..51b285103394 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -72,7 +72,6 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) struct snd_sof_ipc_msg *msg = sdev->msg; struct sof_ipc_reply reply; struct sof_ipc_cmd_hdr *hdr; - unsigned long flags; int ret = 0; /* @@ -84,7 +83,6 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) dev_warn(sdev->dev, "unexpected ipc interrupt raised!\n"); return; } - spin_lock_irqsave(&sdev->ipc_lock, flags); hdr = msg->msg_data; if (hdr->cmd == (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CTX_SAVE)) { @@ -123,7 +121,6 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) out: msg->reply_error = ret; - spin_unlock_irqrestore(&sdev->ipc_lock, flags); } static bool hda_dsp_ipc_is_sof(uint32_t msg) @@ -172,6 +169,18 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) HDA_DSP_REG_HIPCCTL, HDA_DSP_REG_HIPCCTL_DONE, 0); + /* + * Make sure the interrupt thread cannot be preempted between + * waking up the sender and re-enabling the interrupt. Also + * protect against a theoretical race with sof_ipc_tx_message(): + * if the DSP is fast enough to receive an IPC message, reply to + * it, and the host interrupt processing calls this function on + * a different core from the one, where the sending is taking + * place, the message might not yet be marked as expecting a + * reply. + */ + spin_lock_irq(&sdev->ipc_lock); + /* handle immediate reply from DSP core - ignore ROM messages */ if (hda_dsp_ipc_is_sof(msg)) { hda_dsp_ipc_get_reply(sdev); @@ -187,6 +196,8 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) /* set the done bit */ hda_dsp_ipc_dsp_done(sdev); + spin_unlock_irq(&sdev->ipc_lock); + ret = IRQ_HANDLED; } diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 894e68cbd69d..10304a90cf25 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -308,19 +308,8 @@ EXPORT_SYMBOL(sof_ipc_tx_message); int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id) { struct snd_sof_ipc_msg *msg = &sdev->ipc->msg; - unsigned long flags; - - /* - * Protect against a theoretical race with sof_ipc_tx_message(): if the - * DSP is fast enough to receive an IPC message, reply to it, and the - * host interrupt processing calls this function on a different core - * from the one, where the sending is taking place, the message might - * not yet be marked as expecting a reply. - */ - spin_lock_irqsave(&sdev->ipc_lock, flags); if (msg->ipc_complete) { - spin_unlock_irqrestore(&sdev->ipc_lock, flags); dev_err(sdev->dev, "error: no reply expected, received 0x%x", msg_id); return -EINVAL; @@ -330,8 +319,6 @@ int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id) msg->ipc_complete = true; wake_up(&msg->waitq); - spin_unlock_irqrestore(&sdev->ipc_lock, flags); - return 0; } EXPORT_SYMBOL(snd_sof_ipc_reply); From be1b577d01787c67acc6dd1257588183386a08f4 Mon Sep 17 00:00:00 2001 From: Zhu Yingjiang Date: Fri, 24 May 2019 14:09:24 -0500 Subject: [PATCH 11/12] ASoC: SOF: Intel: hda: fix the hda init chip re-write hda_init_caps and remove the HDA reset, clean HDA streams and clear interrupt steps in hda_dsp_probe so the HDA init steps will not be called twice if the CONFIG_SND_SOC_SOF_HDA is true. Fixes: 8a300c8fb17 ("ASoC: SOF: Intel: Add HDA controller for Intel DSP") Reviewed-by: Takashi Iwai Signed-off-by: Zhu Yingjiang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ctrl.c | 102 ++++++++++++++++++++++++++++++--- sound/soc/sof/intel/hda.c | 101 +++++++------------------------- 2 files changed, 113 insertions(+), 90 deletions(-) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 2c3645736e1f..07bc123112c9 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -161,21 +161,105 @@ int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable) return 0; } -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) -/* - * While performing reset, controller may not come back properly and causing - * issues, so recommendation is to set CGCTL.MISCBDCGE to 0 then do reset - * (init chip) and then again set CGCTL.MISCBDCGE to 1 - */ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) { struct hdac_bus *bus = sof_to_bus(sdev); - int ret; + struct hdac_stream *stream; + int sd_offset, ret = 0; + + if (bus->chip_init) + return 0; hda_dsp_ctrl_misc_clock_gating(sdev, false); - ret = snd_hdac_bus_init_chip(bus, full_reset); + + if (full_reset) { + /* clear WAKESTS */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_WAKESTS, + SOF_HDA_WAKESTS_INT_MASK, + SOF_HDA_WAKESTS_INT_MASK); + + /* reset HDA controller */ + ret = hda_dsp_ctrl_link_reset(sdev, true); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to reset HDA controller\n"); + return ret; + } + + usleep_range(500, 1000); + + /* exit HDA controller reset */ + ret = hda_dsp_ctrl_link_reset(sdev, false); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to exit HDA controller reset\n"); + return ret; + } + + usleep_range(1000, 1200); + } + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* check to see if controller is ready */ + if (!snd_hdac_chip_readb(bus, GCTL)) { + dev_dbg(bus->dev, "controller not ready!\n"); + return -EBUSY; + } + + /* Accept unsolicited responses */ + snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, AZX_GCTL_UNSOL); + + /* detect codecs */ + if (!bus->codec_mask) { + bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); + } +#endif + + /* clear stream status */ + list_for_each_entry(stream, &bus->stream_list, list) { + sd_offset = SOF_STREAM_SD_OFFSET(stream); + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + sd_offset + + SOF_HDA_ADSP_REG_CL_SD_STS, + SOF_HDA_CL_DMA_SD_INT_MASK, + SOF_HDA_CL_DMA_SD_INT_MASK); + } + + /* clear WAKESTS */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_WAKESTS, + SOF_HDA_WAKESTS_INT_MASK, + SOF_HDA_WAKESTS_INT_MASK); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* clear rirb status */ + snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK); +#endif + + /* clear interrupt status register */ + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_ALL_STREAM); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* initialize the codec command I/O */ + snd_hdac_bus_init_cmd_io(bus); +#endif + + /* enable CIE and GIE interrupts */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* program the position buffer */ + if (bus->use_posbuf && bus->posbuf.addr) { + snd_hdac_chip_writel(bus, DPLBASE, (u32)bus->posbuf.addr); + snd_hdac_chip_writel(bus, DPUBASE, + upper_32_bits(bus->posbuf.addr)); + } +#endif + + bus->chip_init = true; + hda_dsp_ctrl_misc_clock_gating(sdev, true); return ret; } -#endif diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7e3980a2f7ba..e47f03dc62f0 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -264,9 +264,12 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev, return tplg_filename; } +#endif + static int hda_init_caps(struct snd_sof_dev *sdev) { struct hdac_bus *bus = sof_to_bus(sdev); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) struct hdac_ext_link *hlink; struct snd_soc_acpi_mach_params *mach_params; struct snd_soc_acpi_mach *hda_mach; @@ -274,8 +277,9 @@ static int hda_init_caps(struct snd_sof_dev *sdev) struct snd_soc_acpi_mach *mach; const char *tplg_filename; int codec_num = 0; - int ret = 0; int i; +#endif + int ret = 0; device_disable_async_suspend(bus->dev); @@ -283,6 +287,14 @@ static int hda_init_caps(struct snd_sof_dev *sdev) if (bus->ppcap) dev_dbg(sdev->dev, "PP capability, will probe DSP later.\n"); + ret = hda_dsp_ctrl_init_chip(sdev, true); + if (ret < 0) { + dev_err(bus->dev, "error: init chip failed with ret: %d\n", + ret); + return ret; + } + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); @@ -293,12 +305,6 @@ static int hda_init_caps(struct snd_sof_dev *sdev) return ret; } - ret = hda_dsp_ctrl_init_chip(sdev, true); - if (ret < 0) { - dev_err(bus->dev, "error: init chip failed with ret: %d\n", ret); - goto out; - } - /* codec detection */ if (!bus->codec_mask) { dev_info(bus->dev, "no hda codecs found!\n"); @@ -339,8 +345,10 @@ static int hda_init_caps(struct snd_sof_dev *sdev) /* use local variable for readability */ tplg_filename = pdata->tplg_filename; tplg_filename = fixup_tplg_name(sdev, tplg_filename); - if (!tplg_filename) - goto out; + if (!tplg_filename) { + hda_codec_i915_exit(sdev); + return ret; + } pdata->tplg_filename = tplg_filename; } } @@ -364,34 +372,9 @@ static int hda_init_caps(struct snd_sof_dev *sdev) */ list_for_each_entry(hlink, &bus->hlink_list, list) snd_hdac_ext_bus_link_put(bus, hlink); - - return 0; - -out: - hda_codec_i915_exit(sdev); - return ret; -} - -#else - -static int hda_init_caps(struct snd_sof_dev *sdev) -{ - /* - * set CGCTL.MISCBDCGE to 0 during reset and set back to 1 - * when reset finished. - * TODO: maybe no need for init_caps? - */ - hda_dsp_ctrl_misc_clock_gating(sdev, 0); - - /* clear WAKESTS */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_WAKESTS, - SOF_HDA_WAKESTS_INT_MASK, - SOF_HDA_WAKESTS_INT_MASK); - - return 0; -} - #endif + return 0; +} static const struct sof_intel_dsp_desc *get_chip_info(struct snd_sof_pdata *pdata) @@ -409,9 +392,8 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) struct pci_dev *pci = to_pci_dev(sdev->dev); struct sof_intel_hda_dev *hdev; struct hdac_bus *bus; - struct hdac_stream *stream; const struct sof_intel_dsp_desc *chip; - int sd_offset, ret = 0; + int ret = 0; /* * detect DSP by checking class/subclass/prog-id information @@ -558,49 +540,6 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) if (ret < 0) goto free_ipc_irq; - /* reset HDA controller */ - ret = hda_dsp_ctrl_link_reset(sdev, true); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to reset HDA controller\n"); - goto free_ipc_irq; - } - - /* exit HDA controller reset */ - ret = hda_dsp_ctrl_link_reset(sdev, false); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to exit HDA controller reset\n"); - goto free_ipc_irq; - } - - /* clear stream status */ - list_for_each_entry(stream, &bus->stream_list, list) { - sd_offset = SOF_STREAM_SD_OFFSET(stream); - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, - sd_offset + - SOF_HDA_ADSP_REG_CL_SD_STS, - SOF_HDA_CL_DMA_SD_INT_MASK, - SOF_HDA_CL_DMA_SD_INT_MASK); - } - - /* clear WAKESTS */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_WAKESTS, - SOF_HDA_WAKESTS_INT_MASK, - SOF_HDA_WAKESTS_INT_MASK); - - /* clear interrupt status register */ - snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS, - SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_ALL_STREAM); - - /* enable CIE and GIE interrupts */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, - SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, - SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN); - - /* re-enable CGCTL.MISCBDCGE after reset */ - hda_dsp_ctrl_misc_clock_gating(sdev, true); - - device_disable_async_suspend(&pci->dev); - /* enable DSP features */ snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, SOF_HDA_PPCTL_GPROCEN, SOF_HDA_PPCTL_GPROCEN); From 1f2675f6655838aaf910f911fd0abc821e3ff3df Mon Sep 17 00:00:00 2001 From: Georgii Staroselskii Date: Tue, 28 May 2019 13:47:39 +0300 Subject: [PATCH 12/12] ASoC: sun4i-codec: fix first delay on Speaker Allwinner DAC seems to have a delay in the Speaker audio routing. When playing a sound for the first time, the sound gets chopped. On a second play the sound is played correctly. After some time (~5s) the issue gets back. This commit seems to be fixing the same issue as bf14da7 but for another codepath. This is the DTS that was used to debug the problem. &codec { allwinner,pa-gpios = <&r_pio 0 11 GPIO_ACTIVE_HIGH>; /* PL11 */ allwinner,audio-routing = "Speaker", "LINEOUT"; status = "okay"; } Signed-off-by: Georgii Staroselskii Reviewed-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-codec.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 15d08e343b47..28d2f7713f8d 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1329,6 +1329,15 @@ static int sun4i_codec_spk_event(struct snd_soc_dapm_widget *w, gpiod_set_value_cansleep(scodec->gpio_pa, !!SND_SOC_DAPM_EVENT_ON(event)); + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* + * Need a delay to wait for DAC to push the data. 700ms seems + * to be the best compromise not to feel this delay while + * playing a sound. + */ + msleep(700); + } + return 0; }