2009/6/10 Daniel Ribeiro <drwyrm@xxxxxxxxx>: > * Extend set_tdm_slot to allow the user to arbitrarily set the frame > width and active TX/RX slots. > * Reset SSCR0_EDSS and SSCR0_DSS on pxa_ssp_set_dai_fmt. > * Makes SSCR0_MOD optional. > * Automatically sets network mode when needed if set_tdm_slot was > never called. > * Clears SSCR1_RWOT case SSCR0_MOD is set. > * Updates magician.c and wm9081.c for the new set_tdm_slot() > > (Patch is based on Mark's for-2.6.32 branch) > > Signed-off-by: Daniel Ribeiro <drwyrm@xxxxxxxxx> > > --- > include/sound/soc-dai.h | 5 + > sound/soc/codecs/wm9081.c | 4 - > sound/soc/pxa/magician.c | 2 > sound/soc/pxa/pxa-ssp.c | 119 ++++++++++++++++++++++++++++------------------ > sound/soc/soc-core.c | 9 ++- > 5 files changed, 85 insertions(+), 54 deletions(-) > > diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h > index 352d7ee..f96cc36 100644 > --- a/include/sound/soc-dai.h > +++ b/include/sound/soc-dai.h > @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, > int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); > > int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, > - unsigned int mask, int slots); > + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width); > > int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); > > @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { > */ > int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); > int (*set_tdm_slot)(struct snd_soc_dai *dai, > - unsigned int mask, int slots); > + unsigned int tx_mask, unsigned int rx_mask, > + int slots, int frame_width); > int (*set_tristate)(struct snd_soc_dai *dai, int tristate); > > /* > diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c > index 86fc57e..85c720a 100644 > --- a/sound/soc/codecs/wm9081.c > +++ b/sound/soc/codecs/wm9081.c > @@ -1207,7 +1207,7 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, > } > > static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, > - unsigned int mask, int slots) > + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width) > { > struct snd_soc_codec *codec = dai->codec; > unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); > @@ -1219,7 +1219,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, > > aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; > > - switch (mask) { > + switch (tx_mask) { > case 1: > break; > case 2: > diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c > index c89a3cd..2345869 100644 > --- a/sound/soc/pxa/magician.c > +++ b/sound/soc/pxa/magician.c > @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, > if (ret < 0) > return ret; > > - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); > + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1, 1, 16); > if (ret < 0) > return ret; > > diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c > index 46d14f3..f0931e7 100644 > --- a/sound/soc/pxa/pxa-ssp.c > +++ b/sound/soc/pxa/pxa-ssp.c > @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, > * Set the active slots in TDM/Network mode > */ > static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, > - unsigned int mask, int slots) > + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width) > { > struct ssp_priv *priv = cpu_dai->private_data; > struct ssp_device *ssp = priv->dev.ssp; > u32 sscr0; > > - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); > + sscr0 = ssp_read_reg(ssp, SSCR0); > + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(7) | SSCR0_EDSS | SSCR0_DSS); > + > + /* set frame width */ > + if (frame_width > 16) > + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); > + else > + sscr0 |= SSCR0_DataSize(frame_width); > + > + if (slots > 1) { > + /* enable network mode */ > + sscr0 |= SSCR0_MOD; > Seems the frame_width has no such relations with sampe width, the sample width is defined on runtime by the audio files. Make a example sample with paly a often used 16bit sample width stereo aduio file. When you play 16bit audio sample with 2*16 bit frame , frame --|____________|------------------|_____________|--- data --<======>-<======>-<======>- here the datasize should be SSCR0_DataSize(16), the frame size is 32 also you can play 16bit audio sample with 2*32 frame, frame --|_________________________|------------------------------------|____ data --<======>-------------------<======>-------------------<== here the datasize is also DataSize(16), but the frame size is 64. Do I misunderstand you? > - /* set number of active slots */ > - sscr0 |= SSCR0_SlotsPerFrm(slots); > + /* set number of active slots */ > + sscr0 |= SSCR0_SlotsPerFrm(slots); > + > + /* set active slot mask */ > + ssp_write_reg(ssp, SSTSA, tx_mask); > + ssp_write_reg(ssp, SSRSA, rx_mask); > + } > ssp_write_reg(ssp, SSCR0, sscr0); > > - /* set active slot mask */ > - ssp_write_reg(ssp, SSTSA, mask); > - ssp_write_reg(ssp, SSRSA, mask); > return 0; > } > > @@ -439,8 +452,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, > } > > /* reset port settings */ > - sscr0 = ssp_read_reg(ssp, SSCR0) & > - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); > + sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | > + SSCR0_ACS | SSCR0_EDSS | SSCR0_DSS); > sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); > sspsp = 0; > > @@ -487,7 +500,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, > case SND_SOC_DAIFMT_DSP_A: > sspsp |= SSPSP_FSRT; > case SND_SOC_DAIFMT_DSP_B: > - sscr0 |= SSCR0_MOD | SSCR0_PSP; > + sscr0 |= SSCR0_PSP; > sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; > > switch (fmt & SND_SOC_DAIFMT_INV_MASK) { > @@ -537,48 +550,70 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, > struct ssp_priv *priv = cpu_dai->private_data; > struct ssp_device *ssp = priv->dev.ssp; > int chn = params_channels(params); > - u32 sscr0; > - u32 sspsp; > + u32 sscr0, sscr1, sspsp; > int width = snd_pcm_format_physical_width(params_format(params)); > - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; > + int frame_width; > + > + /* check if the user explicitly set a frame_width */ > + sscr0 = ssp_read_reg(ssp, SSCR0); > + > + if (sscr0 & (SSCR0_EDSS | SSCR0_DSS)) > + frame_width = (sscr0 & SSCR0_DSS) + > + (sscr0 & SSCR0_EDSS ? 17 : 1); > + else > + frame_width = width * chn; > > /* generate correct DMA params */ > if (cpu_dai->dma_data) > kfree(cpu_dai->dma_data); > > - /* Network mode with one active slot (ttsa == 1) can be used > - * to force 16-bit frame width on the wire (for S16_LE), even > - * with two channels. Use 16-bit DMA transfers for this case. > - */ > - cpu_dai->dma_data = ssp_get_dma_params(ssp, > - ((chn == 2) && (ttsa != 1)) || (width == 32), > + cpu_dai->dma_data = ssp_get_dma_params(ssp, frame_width > 16, > substream->stream == SNDRV_PCM_STREAM_PLAYBACK); > > /* we can only change the settings if the port is not in use */ > - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) > + if (sscr0 & SSCR0_SSE) > return 0; > > - /* clear selected SSP bits */ > - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); > - ssp_write_reg(ssp, SSCR0, sscr0); > - > - /* bit size */ > - sscr0 = ssp_read_reg(ssp, SSCR0); > - switch (params_format(params)) { > - case SNDRV_PCM_FORMAT_S16_LE: > + /* FIXME: What this is for? */ > #ifdef CONFIG_PXA3xx > - if (cpu_is_pxa3xx()) > - sscr0 |= SSCR0_FPCKE; > + if (width == 16 && cpu_is_pxa3xx()) > + sscr0 |= SSCR0_FPCKE; > #endif > - sscr0 |= SSCR0_DataSize(16); > - break; > - case SNDRV_PCM_FORMAT_S24_LE: > - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); > - break; > - case SNDRV_PCM_FORMAT_S32_LE: > - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); > - break; > + > + if (!(sscr0 & (SSCR0_EDSS | SSCR0_DSS))) { > + /* Frame width not set yet, we are not using network mode */ > + if (frame_width > 16) > + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); > + else > + sscr0 |= SSCR0_DataSize(frame_width); > + > + if (frame_width > 32) { > + /* > + * Network mode is needed to support this frame_width > + * We assume that the wire is not networked and setup > + * a "fake" network mode here. > + */ > + int slots = frame_width / 32; > + > + sscr0 |= SSCR0_MOD; > + sscr0 |= SSCR0_SlotsPerFrm(slots); > + > + /* > + * Set active slots. Only set an active TX slot > + * if we are going to use it. > + */ > + ssp_write_reg(ssp, SSRSA, slots - 1); > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > + ssp_write_reg(ssp, SSTSA, slots - 1); > + } > } > + > + /* If SSCR0_MOD is set we can't use SSCR1_RWOT */ > + if (sscr0 & SSCR0_MOD) { > + sscr1 = ssp_read_reg(ssp, SSCR1); > + ssp_write_reg(ssp, SSCR1, sscr1 & ~SSCR1_RWOT); > + } > + > ssp_write_reg(ssp, SSCR0, sscr0); > > switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > @@ -625,14 +660,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, > break; > } > > - /* When we use a network mode, we always require TDM slots > - * - complain loudly and fail if they've not been set up yet. > - */ > - if ((sscr0 & SSCR0_MOD) && !ttsa) { > - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); > - return -EINVAL; > - } > - > dump_registers(ssp); > > return 0; > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index e1a920c..69becf2 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); > /** > * snd_soc_dai_set_tdm_slot - configure DAI TDM. > * @dai: DAI > - * @mask: DAI specific mask representing used slots. > + * @tx_mask: bitmask representing active TX slots. > + * @rx_mask: bitmask representing active RX slots. > * @slots: Number of slots in use. > + * @frame_width: Width in bits for each slot. > * > * Configures a DAI for TDM operation. Both mask and slots are codec and DAI > * specific. > */ > int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, > - unsigned int mask, int slots) > + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width) > { > if (dai->ops && dai->ops->set_tdm_slot) > - return dai->ops->set_tdm_slot(dai, mask, slots); > + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, > + slots, frame_width); > else > return -EINVAL; > } > > -- > Daniel Ribeiro > -- Paul Shen _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel