From d83901e82010cb3b25e69a9bbe991e9fbd940725 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Sun, 4 Jan 2015 09:15:04 +0800 Subject: ASoC: Intel: Don't change offset of block allocator during fixed allocate The offset of block allocator, ba->offset, should not be changed during fixed address allocating, for the caller may treat it as the offset of allocated memory and use it. In the case that we allocate more than 1 blocks, we should make sure this offset is correct. Here introduces a temp allocator for the later continuous allocating. Signed-off-by: Jie Yang Signed-off-by: Mark Brown diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index ef2e8b5..b3f9489 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -706,6 +706,7 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba struct list_head *block_list) { struct sst_mem_block *block, *tmp; + struct sst_block_allocator ba_tmp = *ba; u32 end = ba->offset + ba->size, block_end; int err; @@ -730,9 +731,9 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba if (ba->offset >= block->offset && ba->offset < block_end) { /* align ba to block boundary */ - ba->size -= block_end - ba->offset; - ba->offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; @@ -767,10 +768,10 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba list_move(&block->list, &dsp->used_block_list); list_add(&block->module_list, block_list); /* align ba to block boundary */ - ba->size -= block_end - ba->offset; - ba->offset = block_end; + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; -- cgit v0.10.2 From ae6f636b8b2ea9d297a07fc7ef8ae54707d67b36 Mon Sep 17 00:00:00 2001 From: Andrew Jackson Date: Wed, 31 Dec 2014 16:20:37 +0000 Subject: ASoC: adi: Add missing return statement. The probe routine was disabling the clock even if the system was configured successfully. Add a return statement to leave clocks enabled. Signed-off-by: Andrew Jackson Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 7752860..4c23381 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; + return 0; + err_clk_disable: clk_disable_unprepare(i2s->clk); return ret; -- cgit v0.10.2 From f81677b4d1acc0e7cd74a43bfd9900d9512b90ae Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Wed, 7 Jan 2015 22:07:05 +0800 Subject: ASoC: Intel: Add NULL checks for the stream pointer We should not send IPC stream commands to FW when the stream is NULL, dereference the NULL pointer may also occur without precheck. Here add NULL pointer checks for these stream APIs. Signed-off-by: Jie Yang Signed-off-by: Mark Brown diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c482..5bf1404 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1228,6 +1228,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) struct sst_dsp *sst = hsw->dsp; unsigned long flags; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to free, ignore it.\n"); + return 0; + } + /* dont free DSP streams that are not commited */ if (!stream->commited) goto out; @@ -1415,6 +1420,16 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) u32 header; int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to commit, ignore it.\n"); + return 0; + } + + if (stream->commited) { + dev_warn(hsw->dev, "warning: stream is already committed, ignore it.\n"); + return 0; + } + trace_ipc_request("stream alloc", stream->host_id); header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM); @@ -1519,6 +1534,11 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to pause, ignore it.\n"); + return 0; + } + trace_ipc_request("stream pause", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE, @@ -1535,6 +1555,11 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to resume, ignore it.\n"); + return 0; + } + trace_ipc_request("stream resume", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME, @@ -1550,6 +1575,11 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) { int ret, tries = 10; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to reset, ignore it.\n"); + return 0; + } + /* dont reset streams that are not commited */ if (!stream->commited) return 0; -- cgit v0.10.2 From a12d159d06317420c1a6941f5657b2918a02bf74 Mon Sep 17 00:00:00 2001 From: Jianqun Xu Date: Thu, 8 Jan 2015 10:49:59 +0800 Subject: ASoC: rockchip: i2s: applys rate symmetry for CPU DAI This patch applys rate symmetry for rockchip i2s DAI. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 26ec511..deced0e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -335,6 +335,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { SNDRV_PCM_FMTBIT_S24_LE), }, .ops = &rockchip_i2s_dai_ops, + .symmetric_rates = 1, }; static const struct snd_soc_component_driver rockchip_i2s_component = { -- cgit v0.10.2 From 64aa5f5843ab12455f6984928058a267f385a82c Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 7 Jan 2015 19:45:40 -0200 Subject: ASoC: fsl_ssi: Fix irq error check Commit 2ffa531078037a0 ("ASoC: fsl_ssi: Fix module unbound") changed the way to retrieve the irq number from irq_of_parse_and_map() to platform_get_irq(), but missed to updated the irq error check accordingly. We should test for negative irq number and propagate it in the case of error. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a65f17d..059496e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1362,9 +1362,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->irq = platform_get_irq(pdev, 0); - if (!ssi_private->irq) { + if (ssi_private->irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - return -ENXIO; + return ssi_private->irq; } /* Are the RX and the TX clocks locked? */ -- cgit v0.10.2 From 0984f3410089a773e408a0a76f719df109436cf1 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 14 Jan 2015 11:56:22 -0200 Subject: ASoC: fsl: imx-wm8962: Set the card owner field The following crash happens when trying to unload the snd_soc_imx_wm8962 module while playback is active: [ 208.666868] Unable to handle kernel paging request at virtc [ 208.674110] pgd = 80004000 [ 208.676867] [7f06541c] *pgd=4c334811, *pte=00000000, *ppte=00000000 [ 208.683211] Internal error: Oops: 80000007 [#1] SMP ARM [ 208.688445] Modules linked in: snd_soc_wm8962 snd_soc_fsl_ssi snd_soc_imx_audmux imx_pcm_fiq evbug] ... In order to avoid such problem, fill the card owner field as suggested by Lars-Peter Clausen: "But looking at the source it seems that this is a core feature of ALSA and at least for the card module itself it will do the ref-counting when a stream is started/stopped. And we even support setting the owner of a card in ASoC. It's just that pretty much no ASoC card driver bothers to set the owner field in the snd_soc_card struct. So this particular problem can be fixed by updating the imx-wm8962 driver to set the owner field." By doing as suggested, we no longer see the crash when attempting to unload the snd_soc_imx_wm8962 module while playback is active: $ modprobe -r snd_soc_imx_wm8962 modprobe: can't unload module snd_soc_imx_wm8962: Resource temporarily unavailable Reported-by: Jiada Wang Suggested-by: Lars-Peter Clausen Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 4caacb0..cd146d4 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -257,6 +257,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) if (ret) goto clk_fail; data->card.num_links = 1; + data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; data->card.dapm_widgets = imx_wm8962_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); -- cgit v0.10.2 From 45437fa58587dd31523cb2d78183088fb69cdeec Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 15 Jan 2015 10:49:25 +0800 Subject: ASoC: rt286: set the same format for dac and adc There is only one I2S I/F, AD/DA path must operate to the same format. Signed-off-by: Bard Liao Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2cd4fe4..1d1c7f8 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -861,10 +861,8 @@ static int rt286_hw_params(struct snd_pcm_substream *substream, RT286_I2S_CTRL1, 0x0018, d_len_code << 3); dev_dbg(codec->dev, "format val = 0x%x\n", val); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); - else - snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); return 0; } -- cgit v0.10.2 From d3268a40d4b19ff7bee23f52eabbc4e96bb685e8 Mon Sep 17 00:00:00 2001 From: Qais Yousef Date: Wed, 14 Jan 2015 08:47:29 +0000 Subject: ASoC: soc-compress.c: fix NULL dereference In soc_new_compress() when rtd->dai_link->dynamic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by using rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture as playback_count and capture_count to snd_pcm_new_internal(). Signed-off-by: Qais Yousef Signed-off-by: Mark Brown Cc: stable@vger.kernel.org diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f..025c38f 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + else if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- cgit v0.10.2 From 20602e34cd33dd452bc1836fa7c9b59978f75db0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 16 Jan 2015 11:20:25 +0200 Subject: ASoC: omap-mcbsp: Correct CBM_CFS dai format configuration We should select FSR also to be driven by McBSP, not only FSX. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown Cc: stable@vger.kernel.org diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8b79cafa..c7eb9dd 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -434,7 +434,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBM_CFS: /* McBSP slave. FS clock as output */ regs->srgr2 |= FSGM; - regs->pcr0 |= FSXM; + regs->pcr0 |= FSXM | FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: /* McBSP slave */ -- cgit v0.10.2