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

[alsa-devel] [PATCH] ALSA: echoaudio: cleanup of unnecessary messages
by Sudip Mukherjee 05 Nov '14

05 Nov '14

commit "b5b4a41b392960010fccf1f9ccf8334d612bd450" was dereferencing chip after it has been freed. This patch fixes that and at the same time removes some debugging messages, which are unnecessary, as they are just printing information about entry and exit from a function. This information we can easily get from ftrace. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/echoaudio/darla20_dsp.c | 2 -- sound/pci/echoaudio/darla24_dsp.c | 2 -- sound/pci/echoaudio/echo3g_dsp.c | 2 -- sound/pci/echoaudio/echoaudio.c | 25 ------------------------- sound/pci/echoaudio/echoaudio_3g.c | 1 - sound/pci/echoaudio/echoaudio_dsp.c | 12 ------------ sound/pci/echoaudio/gina20_dsp.c | 4 ---- sound/pci/echoaudio/gina24_dsp.c | 4 ---- sound/pci/echoaudio/indigo_dsp.c | 2 -- sound/pci/echoaudio/indigodj_dsp.c | 2 -- sound/pci/echoaudio/indigodjx_dsp.c | 2 -- sound/pci/echoaudio/indigoio_dsp.c | 2 -- sound/pci/echoaudio/indigoiox_dsp.c | 2 -- sound/pci/echoaudio/layla20_dsp.c | 5 ----- sound/pci/echoaudio/layla24_dsp.c | 4 ---- sound/pci/echoaudio/mona_dsp.c | 3 --- 16 files changed, 74 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index c94e92e..febee5b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20)) return -ENODEV; @@ -58,7 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index b1272f88..7b4f6fd 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24)) return -ENODEV; @@ -57,7 +56,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index bc37168..ae11ce1 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -46,7 +46,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) int err; local_irq_enable(); - dev_dbg(chip->card->dev, "init_hw() - Echo3G\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G)) return -ENODEV; @@ -99,7 +98,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1ef29e5..146ffc5 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -96,7 +96,6 @@ static void free_firmware_cache(struct echoaudio *chip) dev_dbg(chip->card->dev, "release_firmware(%d)\n", i); } - dev_dbg(chip->card->dev, "firmware_cache released\n"); #endif } @@ -354,7 +353,6 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err; - dev_dbg(chip->card->dev, "pcm_analog_in_open\n"); if ((err = pcm_open(substream, num_analog_busses_in(chip) - substream->number)) < 0) return err; @@ -389,7 +387,6 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) #else max_channels = num_analog_busses_out(chip); #endif - dev_dbg(chip->card->dev, "pcm_analog_out_open\n"); if ((err = pcm_open(substream, max_channels - substream->number)) < 0) return err; if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, @@ -422,7 +419,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_in_open\n"); max_channels = num_digital_busses_in(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -464,7 +460,6 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_out_open\n"); max_channels = num_digital_busses_out(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -511,7 +506,6 @@ static int pcm_close(struct snd_pcm_substream *substream) /* Nothing to do here. Audio is already off and pipe will be * freed by its callback */ - dev_dbg(chip->card->dev, "pcm_close\n"); atomic_dec(&chip->opencount); oc = atomic_read(&chip->opencount); @@ -620,7 +614,6 @@ static int init_engine(struct snd_pcm_substream *substream, spin_lock_irq(&chip->lock); set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den); spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_params ok\n"); return 0; } @@ -691,7 +684,6 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) } spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_freed\n"); snd_pcm_lib_free_pages(substream); return 0; } @@ -763,10 +755,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - dev_dbg(chip->card->dev, "pcm_trigger resume\n"); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dev_dbg(chip->card->dev, "pcm_trigger start\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -788,9 +778,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) chip->pipe_cyclic_mask); break; case SNDRV_PCM_TRIGGER_SUSPEND: - dev_dbg(chip->card->dev, "pcm_trigger suspend\n"); case SNDRV_PCM_TRIGGER_STOP: - dev_dbg(chip->card->dev, "pcm_trigger stop\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -800,7 +788,6 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = stop_transport(chip, channelmask); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev_dbg(chip->card->dev, "pcm_trigger pause\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -1881,10 +1868,8 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) static int snd_echo_free(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "Stop DSP...\n"); if (chip->comm_page) rest_in_peace(chip); - dev_dbg(chip->card->dev, "Stopped.\n"); if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1898,14 +1883,12 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->iores) release_and_free_resource(chip->iores); - dev_dbg(chip->card->dev, "MMIO freed.\n"); pci_disable_device(chip->pci); /* release chip data */ free_firmware_cache(chip); kfree(chip); - dev_dbg(chip->card->dev, "Chip freed.\n"); return 0; } @@ -1915,7 +1898,6 @@ static int snd_echo_dev_free(struct snd_device *device) { struct echoaudio *chip = device->device_data; - dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n"); return snd_echo_free(chip); } @@ -2038,7 +2020,6 @@ static int snd_echo_probe(struct pci_dev *pci, return -ENOENT; } - dev_dbg(&pci->dev, "Echoaudio driver starting...\n"); i = 0; err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -2191,7 +2172,6 @@ static int snd_echo_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct echoaudio *chip = dev_get_drvdata(dev); - dev_dbg(dev, "suspend start\n"); snd_pcm_suspend_all(chip->analog_pcm); snd_pcm_suspend_all(chip->digital_pcm); @@ -2218,7 +2198,6 @@ static int snd_echo_suspend(struct device *dev) pci_save_state(pci); pci_disable_device(pci); - dev_dbg(dev, "suspend done\n"); return 0; } @@ -2232,7 +2211,6 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - dev_dbg(dev, "resume start\n"); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); if (commpage_bak == NULL) @@ -2247,7 +2225,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_free(chip); return err; } - dev_dbg(dev, "resume init OK\n"); /* Temporarily set chip->pipe_alloc_mask=0 otherwise * restore_dsp_settings() fails. @@ -2260,7 +2237,6 @@ static int snd_echo_resume(struct device *dev) kfree(commpage_bak); return err; } - dev_dbg(dev, "resume restore OK\n"); memcpy(&commpage->audio_format, &commpage_bak->audio_format, sizeof(commpage->audio_format)); @@ -2286,7 +2262,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif - dev_dbg(dev, "resume done\n"); return 0; } diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 50a2169..7258328 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -328,7 +328,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index ba9d4f1..1a9427a 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,7 +206,6 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) goto la_error; } - dev_dbg(chip->card->dev, "ASIC loaded\n"); free_firmware(fw, chip); return 0; @@ -473,7 +472,6 @@ static int load_dsp(struct echoaudio *chip, u16 *code) chip->dsp_code = code; /* Show which DSP code loaded */ chip->bad_board = FALSE; /* DSP OK */ - dev_dbg(chip->card->dev, "load_dsp: OK!\n"); return 0; } udelay(100); @@ -674,7 +672,6 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { int i, o, err; - dev_dbg(chip->card->dev, "restore_dsp_settings\n"); if ((err = check_asic_status(chip)) < 0) return err; @@ -771,7 +768,6 @@ static int restore_dsp_rettings(struct echoaudio *chip) if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) return -EIO; - dev_dbg(chip->card->dev, "restore_dsp_rettings done\n"); return 0; } @@ -865,7 +861,6 @@ Same thing for pause_ and stop_ -trasport below. */ static int start_transport(struct echoaudio *chip, u32 channel_mask, u32 cyclic_mask) { - dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -891,7 +886,6 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask, static int pause_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -918,7 +912,6 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask) static int stop_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -954,8 +947,6 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index) stopped and unallocated. */ static int rest_in_peace(struct echoaudio *chip) { - dev_dbg(chip->card->dev, - "rest_in_peace() open=%x\n", chip->pipe_alloc_mask); /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); @@ -1018,7 +1009,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) */ static int init_line_levels(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "init_line_levels\n"); memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); @@ -1099,7 +1089,6 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, it moves data. The DMA counter is in units of bytes, not samples. */ pipe->dma_counter = &chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; - dev_dbg(chip->card->dev, "allocate_pipes: ok\n"); return pipe_index; } @@ -1110,7 +1099,6 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe) u32 channel_mask; int i; - dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index))) return -EINVAL; if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED)) diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index a959eae..a8dcf1e 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -37,7 +37,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20)) return -ENODEV; @@ -63,7 +62,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -150,7 +148,6 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) static int set_input_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_input_clock:\n"); switch (clock) { case ECHO_CLOCK_INTERNAL: @@ -209,7 +206,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index c8ea576..165fe3d 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24)) return -ENODEV; @@ -79,7 +78,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -156,7 +154,6 @@ static int load_asic(struct echoaudio *chip) control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; err = write_control_reg(chip, control_reg, TRUE); } - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -238,7 +235,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index cdeb073..54edd67 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 133915c..2cf5cc0 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 26cdfcf..5252863 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 5e6df7c..4e81787 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 90cdd27..6de3f9b 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 7f0f6ea..637e72f 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20)) return -ENODEV; @@ -65,7 +64,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -190,7 +188,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) u16 clock; u32 rate; - dev_dbg(chip->card->dev, "set_input_clock:\n"); rate = 0; switch (clock_source) { case ECHO_CLOCK_INTERNAL: @@ -232,7 +229,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) static int set_output_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock); switch (clock) { case ECHO_CLOCK_SUPER: clock = LAYLA20_OUTPUT_CLOCK_SUPER; @@ -286,7 +282,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index eb8f218..93813a8 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24)) return -ENODEV; @@ -70,7 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -118,7 +116,6 @@ static int load_asic(struct echoaudio *chip) if (chip->asic_loaded) return 1; - dev_dbg(chip->card->dev, "load_asic\n"); /* Give the DSP a few milliseconds to settle down */ mdelay(10); @@ -152,7 +149,6 @@ static int load_asic(struct echoaudio *chip) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, TRUE); - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index cc46a8c..63a6a50 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mona\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA)) return -ENODEV; @@ -72,7 +71,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -302,7 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) u32 control_reg, clocks_from_dsp; int err; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Prevent two simultaneous calls to switch_asic() */ if (atomic_read(&chip->opencount)) -- 1.8.1.2

3 4

[alsa-devel] [PATCH v2 0/5] ASoC: Intel: ACPI support and machines
by Vinod Koul 05 Nov '14

05 Nov '14

This is second rev of patch series which fixes the modules split as discussed in ML. The core part is now module along with PCI and ACPI one. Added fix reported by Dan as well. Also added the BYTCR machine driver Subhransu S. Prusty (1): ASoC: Intel: add BYTCR machine driver with RT5640 Vinod Koul (4): ASoC: Intel: mrfld - remove unnecessary check for pointer ASoC: Intel: mrfld - create separate module for pci part ASoC: Intel: mrfld - add shim save restore ASoC: Intel: mrfld- add ACPI module sound/soc/intel/Kconfig | 22 ++- sound/soc/intel/Makefile | 2 + sound/soc/intel/bytcr_dpcm_rt5640.c | 258 +++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.c | 2 +- sound/soc/intel/sst/Makefile | 8 +- sound/soc/intel/sst/sst.c | 243 +++++----------------- sound/soc/intel/sst/sst.h | 9 +- sound/soc/intel/sst/sst_acpi.c | 386 +++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pci.c | 225 ++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 2 + 10 files changed, 958 insertions(+), 199 deletions(-) create mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_pci.c

4 18

[alsa-devel] [PATCH] ASoC: Intel: Fix the driver data not set issue
by Jie Yang 05 Nov '14

05 Nov '14

The priv_data is allocated again here wrongly, and it is not set to the driver data after assignment. This make the pdata->dev is NULL and oops occurs on the first call to hsw_volume_put. The resource has been allocated in driver probe callback hsw_pcm_dev_probe, so here just remove this sencond allocation is OK. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 21f1316..bc02a6e 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -866,7 +866,6 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) dev = platform->dev; dma_dev = pdata->dma_dev; - priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); priv_data->hsw = pdata->dsp; priv_data->dev = platform->dev; priv_data->pm_state = HSW_PM_STATE_D0; -- 1.9.1

3 3

[alsa-devel] [PATCH] ALSA: hda - fix mute led problem for three HP laptops
by Hui Wang 05 Nov '14

05 Nov '14

Without the fix, the mute led can't work on these three machines. After apply this fix, these three machines will fall back on the led control quirk as below, and through testing, the mute led works very well. PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED, ALC282_STANDARD_PINS, {0x12, 0x90a60140}, ... BugLink: https://bugs.launchpad.net/bugs/1389497 Tested-by: TieFu Chen <tienfu.chen(a)canonical.com> Cc: Kailang Yang <kailang(a)realtek.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ba72fff..1af917f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4744,9 +4744,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2246, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2247, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2248, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2249, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), -- 1.9.1

2 1

[alsa-devel] Compressed audio support over HDMI
by D K 05 Nov '14

05 Nov '14

Hi, I recently came across the compressed audio framework used with ALSA-SOC. Can the framework be used to send compressed audio to HDMI driver?.Like using tinycompress from usespace . thanks,

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Use specific r/w function for DSP mode
by Oder Chiou 04 Nov '14

04 Nov '14

In DSP mode, the register r/w should use the specific function to access that is invoked by address mapping of the DSP. The MX-65[1] is for switching DSP or codec mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 222 ++++++++++++++++++++++++++++++---------------- sound/soc/codecs/rt5677.h | 2 + 2 files changed, 149 insertions(+), 75 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 413bccb..74f65ff 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -539,53 +539,113 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } } +static int rt5677_hw_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + u8 r[1]; + u8 data[2]; + int ret; + + /* Write register */ + r[0] = reg & 0xff; + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = &r[0]; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return -EIO; + } + + *val = (data[0] << 8) | data[1]; + + dev_dbg(&client->dev, "read %02x => %04x\n", reg, *val); + + return 0; +} + +static int rt5677_hw_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + u8 data[3]; + int ret; + + data[0] = reg & 0xff; + data[1] = (0xff00 & val) >> 8; + data[2] = val & 0xff; + + dev_dbg(&client->dev, "write %02x = %04x\n", reg, val); + + ret = i2c_master_send(client, data, 3); + if (ret == 3) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + /** * rt5677_dsp_mode_i2c_write_addr - Write value to address on DSP mode. - * @codec: SoC audio codec device. + * @rt5677: Private Data. * @addr: Address index. * @value: Address data. * * * Returns 0 for success or negative error code. */ -static int rt5677_dsp_mode_i2c_write_addr(struct snd_soc_codec *codec, +static int rt5677_dsp_mode_i2c_write_addr(struct rt5677_priv *rt5677, unsigned int addr, unsigned int value, unsigned int opcode) { - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt5677->i2c; int ret; mutex_lock(&rt5677->dsp_cmd_lock); - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr msb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_LSB, addr & 0xffff); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr lsb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_DATA_MSB, value >> 16); if (ret < 0) { - dev_err(codec->dev, "Failed to set data msb value: %d\n", ret); + dev_err(&client->dev, "Failed to set data msb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_DATA_LSB, value & 0xffff); if (ret < 0) { - dev_err(codec->dev, "Failed to set data lsb value: %d\n", ret); + dev_err(&client->dev, "Failed to set data lsb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE, opcode); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_OP_CODE, opcode); if (ret < 0) { - dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + dev_err(&client->dev, "Failed to set op code value: %d\n", + ret); goto err; } @@ -597,42 +657,45 @@ err: /** * rt5677_dsp_mode_i2c_read_addr - Read value from address on DSP mode. - * @codec: SoC audio codec device. + * rt5677: Private Data. * @addr: Address index. * @value: Address data. * + * * Returns 0 for success or negative error code. */ static int rt5677_dsp_mode_i2c_read_addr( - struct snd_soc_codec *codec, unsigned int addr, unsigned int *value) + struct rt5677_priv *rt5677, unsigned int addr, unsigned int *value) { - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt5677->i2c; int ret; unsigned int msb, lsb; mutex_lock(&rt5677->dsp_cmd_lock); - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr msb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_LSB, addr & 0xffff); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr lsb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE , 0x0002); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_OP_CODE , 0x0002); if (ret < 0) { - dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + dev_err(&client->dev, "Failed to set op code value: %d\n", ret); goto err; } - regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, &msb); - regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, &lsb); + rt5677_hw_read(client, RT5677_DSP_I2C_DATA_MSB, &msb); + rt5677_hw_read(client, RT5677_DSP_I2C_DATA_LSB, &lsb); *value = (msb << 16) | lsb; err: @@ -643,17 +706,17 @@ err: /** * rt5677_dsp_mode_i2c_write - Write register on DSP mode. - * @codec: SoC audio codec device. + * rt5677: Private Data. * @reg: Register index. * @value: Register data. * * * Returns 0 for success or negative error code. */ -static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, +static int rt5677_dsp_mode_i2c_write(struct rt5677_priv *rt5677, unsigned int reg, unsigned int value) { - return rt5677_dsp_mode_i2c_write_addr(codec, 0x18020000 + reg * 2, + return rt5677_dsp_mode_i2c_write_addr(rt5677, 0x18020000 + reg * 2, value, 0x0001); } @@ -661,57 +724,33 @@ static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, * rt5677_dsp_mode_i2c_read - Read register on DSP mode. * @codec: SoC audio codec device. * @reg: Register index. + * @value: Register data. * * - * Returns Register value. + * Returns 0 for success or negative error code. */ -static unsigned int rt5677_dsp_mode_i2c_read( - struct snd_soc_codec *codec, unsigned int reg) +static int rt5677_dsp_mode_i2c_read( + struct rt5677_priv *rt5677, unsigned int reg, unsigned int *value) { - unsigned int value = 0; + int ret = rt5677_dsp_mode_i2c_read_addr(rt5677, 0x18020000 + reg * 2, + value); - rt5677_dsp_mode_i2c_read_addr(codec, 0x18020000 + reg * 2, &value); + *value &= 0xffff; - return value; + return ret; } -/** - * rt5677_dsp_mode_i2c_update_bits - update register on DSP mode. - * @codec: audio codec - * @reg: register index. - * @mask: register mask - * @value: new value - * - * - * Returns 1 for change, 0 for no change, or negative error code. - */ -static int rt5677_dsp_mode_i2c_update_bits(struct snd_soc_codec *codec, - unsigned int reg, unsigned int mask, unsigned int value) +static void rt5677_set_dsp_mode(struct snd_soc_codec *codec, bool on) { - unsigned int old, new; - int change, ret; - - ret = rt5677_dsp_mode_i2c_read(codec, reg); - if (ret < 0) { - dev_err(codec->dev, "Failed to read reg: %d\n", ret); - goto err; - } + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = rt5677_dsp_mode_i2c_write(codec, reg, new); - if (ret < 0) { - dev_err(codec->dev, - "Failed to write reg: %d\n", ret); - goto err; - } + if (on) { + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x2); + rt5677->is_dsp_mode = true; + } else { + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x0); + rt5677->is_dsp_mode = false; } - return change; - -err: - return ret; } static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) @@ -733,9 +772,14 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_write(rt5677->regmap, RT5677_GLB_CLK2, 0x0080); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); - regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07ff); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); + rt5677_set_dsp_mode(codec, true); ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1, codec->dev); @@ -751,8 +795,7 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) release_firmware(rt5677->fw2); } - rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, - 0x0); + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x0); regcache_cache_bypass(rt5677->regmap, false); regcache_cache_only(rt5677->regmap, true); @@ -762,9 +805,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) regcache_cache_only(rt5677->regmap, false); regcache_cache_bypass(rt5677->regmap, true); - rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, - 0x1); - rt5677_dsp_mode_i2c_write(codec, RT5677_PWR_DSP1, 0x0001); + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x1); + rt5677_set_dsp_mode(codec, false); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x0001); regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); @@ -3804,6 +3847,32 @@ static int rt5677_resume(struct snd_soc_codec *codec) #define rt5677_resume NULL #endif +static int rt5677_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5677_priv *rt5677 = i2c_get_clientdata(client); + + if (rt5677->is_dsp_mode) + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + else + rt5677_hw_read(context, reg, val); + + return 0; +} + +static int rt5677_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5677_priv *rt5677 = i2c_get_clientdata(client); + + if (rt5677->is_dsp_mode) + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + else + rt5677_hw_write(context, reg, val); + + return 0; +} + #define RT5677_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5677_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) @@ -3938,6 +4007,8 @@ static const struct regmap_config rt5677_regmap = { .volatile_reg = rt5677_volatile_register, .readable_reg = rt5677_readable_register, + .reg_read = rt5677_read, + .reg_write = rt5677_write, .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5677_reg, @@ -4063,6 +4134,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, rt5677); + rt5677->i2c = i2c; if (pdata) rt5677->pdata = *pdata; @@ -4094,7 +4166,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, msleep(10); } - rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap); + rt5677->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5677_regmap); if (IS_ERR(rt5677->regmap)) { ret = PTR_ERR(rt5677->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index d2c743c..ab81659f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1599,6 +1599,7 @@ struct rt5677_priv { struct regmap *regmap; const struct firmware *fw1, *fw2; struct mutex dsp_cmd_lock; + struct i2c_client *i2c; int sysclk; int sysclk_src; @@ -1614,6 +1615,7 @@ struct rt5677_priv { #endif bool dsp_vad_en; struct regmap_irq_chip_data *irq_data; + bool is_dsp_mode; }; #endif /* __RT5677_H__ */ -- 1.8.1.1.439.g50a6b54

2 5

[alsa-devel] [PATCH 1/2] ASoC: max98090: Correct pclk divisor settings
by Dylan Reid 04 Nov '14

04 Nov '14

The Baytrail-based chromebooks have a 20MHz mclk, the code was setting the divisor incorrectly in this case. According to the 98090 datasheet, the divisor should be set to DIV1 for 10 <= mclk <= 20. Correct this and the surrounding clock ranges as well to match the datasheet. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98090.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index d519294..1229554 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1941,13 +1941,13 @@ static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, * 0x02 (when master clk is 20MHz to 40MHz).. * 0x03 (when master clk is 40MHz to 60MHz).. */ - if ((freq >= 10000000) && (freq < 20000000)) { + if ((freq >= 10000000) && (freq <= 20000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV1); - } else if ((freq >= 20000000) && (freq < 40000000)) { + } else if ((freq > 20000000) && (freq <= 40000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV2); - } else if ((freq >= 40000000) && (freq < 60000000)) { + } else if ((freq > 40000000) && (freq <= 60000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV4); } else { -- 2.1.2.330.g565301e

2 2

[alsa-devel] [PATCH v6] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 04 Nov '14

04 Nov '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 143 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 146 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6b59471..3ee031e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..51c6c4f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -906,6 +906,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d8..1db2168 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1461,7 +1461,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1473,7 +1474,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index cdc837e..68f6057 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -729,6 +729,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2664,6 +2694,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3142,6 +3175,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config += w->params_select; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3222,8 +3258,35 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3231,14 +3294,48 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = kcalloc(num_params, sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + w_param_text[count] = kmemdup((void *)(config->stream_name), + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3246,6 +3343,25 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + private_value = + (unsigned long) kmemdup((void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + template.kcontrol_news = kmemdup(&kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3253,15 +3369,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dpcm: Fix race between FE/BE updates and trigger
by Takashi Iwai 04 Nov '14

04 Nov '14

DPCM can update the FE/BE connection states totally asynchronously from the FE's PCM state. Most of FE/BE state changes are protected by mutex, so that they won't race, but there are still some actions that are uncovered. For example, suppose to switch a BE while a FE's stream is running. This would call soc_dpcm_runtime_update(), which sets FE's runtime_update flag, then sets up and starts BEs, and clears FE's runtime_update flag again. When a device emits XRUN during this operation, the PCM core triggers snd_pcm_stop(XRUN). Since the trigger action is an atomic ops, this isn't blocked by the mutex, thus it kicks off DPCM's trigger action. It eventually updates and clears FE's runtime_update flag while soc_dpcm_runtime_update() is running concurrently, and it results in confusion. Usually, for avoiding such a race, we take a lock. There is a PCM stream lock for that purpose. However, as already mentioned, the trigger action is atomic, and we can't take the lock for the whole soc_dpcm_runtime_update() or other operations that include the lengthy jobs like hw_params or prepare. This patch provides an alternative solution. This adds a way to defer the conflicting trigger callback to be executed at the end of FE/BE state changes. For doing it, two things are introduced: - Each runtime_update state change of FEs is protected via PCM stream lock. - The FE's trigger callback checks the runtime_update flag. If it's not set, the trigger action is executed there. If set, mark the pending trigger action and returns immediately. - At the exit of runtime_update state change, it checks whether the pending trigger is present. If yes, it executes the trigger action at this point. Reported-and-tested-by: Qiao Zhou <zhouqiao(a)marvell.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc-dpcm.h | 2 ++ sound/soc/soc-pcm.c | 72 +++++++++++++++++++++++++++++++++++++----------- 2 files changed, 58 insertions(+), 16 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 2883a7a6f9f3..98f2ade0266e 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -102,6 +102,8 @@ struct snd_soc_dpcm_runtime { /* state and update */ enum snd_soc_dpcm_update runtime_update; enum snd_soc_dpcm_state state; + + int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; /* can this BE stop and free */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311afdeaa..57277dd79e11 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1522,13 +1522,36 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); } +static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); + +/* Set FE's runtime_update state; the state is protected via PCM stream lock + * for avoiding the race with trigger callback. + * If the state is unset and a trigger is pending while the previous operation, + * process the pending trigger action here. + */ +static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, + int stream, enum snd_soc_dpcm_update state) +{ + struct snd_pcm_substream *substream = + snd_soc_dpcm_get_substream(fe, stream); + + snd_pcm_stream_lock_irq(substream); + if (state == SND_SOC_DPCM_UPDATE_NO && fe->dpcm[stream].trigger_pending) { + dpcm_fe_dai_do_trigger(substream, + fe->dpcm[stream].trigger_pending - 1); + fe->dpcm[stream].trigger_pending = 0; + } + fe->dpcm[stream].runtime_update = state; + snd_pcm_stream_unlock_irq(substream); +} + static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; struct snd_pcm_runtime *runtime = fe_substream->runtime; int stream = fe_substream->stream, ret = 0; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); ret = dpcm_be_dai_startup(fe, fe_substream->stream); if (ret < 0) { @@ -1550,13 +1573,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_runtime(fe_substream); snd_pcm_limit_hw_rates(runtime); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; unwind: dpcm_be_dai_startup_unwind(fe, fe_substream->stream); be_err: - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; } @@ -1603,7 +1626,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ dpcm_be_dai_shutdown(fe, substream->stream); @@ -1617,7 +1640,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; } @@ -1665,7 +1688,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) int err, stream = substream->stream; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); dev_dbg(fe->dev, "ASoC: hw_free FE %s\n", fe->dai_link->name); @@ -1680,7 +1703,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) err = dpcm_be_dai_hw_free(fe, stream); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); mutex_unlock(&fe->card->mutex); return 0; @@ -1773,7 +1796,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, int ret, stream = substream->stream; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); memcpy(&fe->dpcm[substream->stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); @@ -1796,7 +1819,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS; out: - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); mutex_unlock(&fe->card->mutex); return ret; } @@ -1910,7 +1933,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); -static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream, ret; @@ -1984,6 +2007,23 @@ out: return ret; } +static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream; + + /* if FE's runtime_update is already set, we're in race; + * process this trigger later at exit + */ + if (fe->dpcm[stream].runtime_update != SND_SOC_DPCM_UPDATE_NO) { + fe->dpcm[stream].trigger_pending = cmd + 1; + return 0; /* delayed, assuming it's successful */ + } + + /* we're alone, let's trigger */ + return dpcm_fe_dai_do_trigger(substream, cmd); +} + int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -2027,7 +2067,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) dev_dbg(fe->dev, "ASoC: prepare FE %s\n", fe->dai_link->name); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* there is no point preparing this FE if there are no BEs */ if (list_empty(&fe->dpcm[stream].be_clients)) { @@ -2054,7 +2094,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); mutex_unlock(&fe->card->mutex); return ret; @@ -2201,11 +2241,11 @@ static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream) { int ret; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); ret = dpcm_run_update_startup(fe, stream); if (ret < 0) dev_err(fe->dev, "ASoC: failed to startup some BEs\n"); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; } @@ -2214,11 +2254,11 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) { int ret; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); ret = dpcm_run_update_shutdown(fe, stream); if (ret < 0) dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n"); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; } -- 2.1.3

3 2

Re: [alsa-devel] ALSA: echoaudio: remove all snd_printk
by Dan Carpenter 04 Nov '14

04 Nov '14

Hello Sudip Mukherjee, The patch b5b4a41b3929: "ALSA: echoaudio: remove all snd_printk" from Nov 3, 2014, leads to the following static checker warning: sound/pci/echoaudio/echoaudio.c:1908 snd_echo_free() error: dereferencing freed memory 'chip' sound/pci/echoaudio/echoaudio.c 1905 /* release chip data */ 1906 free_firmware_cache(chip); 1907 kfree(chip); ^^^^^^^^^^^ 1908 dev_dbg(chip->card->dev, "Chip freed.\n"); ^^^^^^^^^^ We have lots of printks in this function. We could just delete them all and no one would care. These days you can get the same information from ftrace. 1909 return 0; 1910 } regards, dan carpenter

3 2