diff options
Diffstat (limited to 'pd/src/s_audio_alsa.c')
| -rw-r--r-- | pd/src/s_audio_alsa.c | 39 |
1 files changed, 24 insertions, 15 deletions
diff --git a/pd/src/s_audio_alsa.c b/pd/src/s_audio_alsa.c index a5034c8d..cf731109 100644 --- a/pd/src/s_audio_alsa.c +++ b/pd/src/s_audio_alsa.c @@ -90,9 +90,17 @@ static int alsaio_canmmap(t_alsa_dev *dev) return ((err1 < 0) && (err2 >= 0)); } +static void check_setup_error(int err, int out, const char *why) { + char bf[256]; + snprintf(bf, sizeof bf, "%s (%s)", why, out ? "output" : "input"); + check_error(err, bf); +} + static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate, int nfrags, int frag_size) { +#define CHECK_ERROR(why_) check_setup_error(err, out, why_) + int bufsizeforthis, err; snd_pcm_hw_params_t* hw_params; unsigned int tmp_uint; @@ -110,14 +118,14 @@ static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate, /* get the default params */ err = snd_pcm_hw_params_any(dev->a_handle, hw_params); - check_error(err, "snd_pcm_hw_params_any"); + CHECK_ERROR("snd_pcm_hw_params_any"); /* try to set interleaved access */ err = snd_pcm_hw_params_set_access(dev->a_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) return (-1); - check_error(err, "snd_pcm_hw_params_set_access"); + CHECK_ERROR("snd_pcm_hw_params_set_access"); #if 0 /* enable this to print out which formats are available */ { int i; @@ -141,7 +149,7 @@ static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate, "PD-ALSA: 32/24 bit format not available - using 16\n"); */ err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params, SND_PCM_FORMAT_S16); - check_error(err, "snd_pcm_hw_params_set_format"); + CHECK_ERROR("_pcm_hw_params_set_format"); dev->a_sampwidth = 2; } else dev->a_sampwidth = 3; @@ -154,22 +162,22 @@ static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate, /* set the subformat */ err = snd_pcm_hw_params_set_subformat(dev->a_handle, hw_params, SND_PCM_SUBFORMAT_STD); - check_error(err, "snd_pcm_hw_params_set_subformat"); + CHECK_ERROR("snd_pcm_hw_params_set_subformat"); /* set the number of channels */ tmp_uint = *channels; err = snd_pcm_hw_params_set_channels_min(dev->a_handle, hw_params, &tmp_uint); - check_error(err, "snd_pcm_hw_params_set_channels"); + CHECK_ERROR("snd_pcm_hw_params_set_channels"); if (tmp_uint != (unsigned)*channels) post("ALSA: set %s channels to %d", (out?"output":"input"), tmp_uint); *channels = tmp_uint; dev->a_channels = *channels; /* set the sampling rate */ - err = snd_pcm_hw_params_set_rate_min(dev->a_handle, hw_params, + err = snd_pcm_hw_params_set_rate_near(dev->a_handle, hw_params, (unsigned int *)rate, 0); - check_error(err, "snd_pcm_hw_params_set_rate_min (input)"); + CHECK_ERROR("snd_pcm_hw_params_set_rate_min"); #if 0 err = snd_pcm_hw_params_get_rate(hw_params, &subunitdir); post("input sample rate %d", err); @@ -185,7 +193,7 @@ static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate, err = snd_pcm_hw_params_set_period_size_near(dev->a_handle, hw_params, &tmp_snd_pcm_uframes, 0); #endif - check_error(err, "snd_pcm_hw_params_set_period_size_near (input)"); + CHECK_ERROR("snd_pcm_hw_params_set_period_size_near"); /* set the buffer size */ #ifdef ALSAAPI9 @@ -196,10 +204,10 @@ static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate, err = snd_pcm_hw_params_set_buffer_size_near(dev->a_handle, hw_params, &tmp_snd_pcm_uframes); #endif - check_error(err, "snd_pcm_hw_params_set_buffer_size_near (input)"); + CHECK_ERROR("snd_pcm_hw_params_set_buffer_size_near"); err = snd_pcm_hw_params(dev->a_handle, hw_params); - check_error(err, "snd_pcm_hw_params (input)"); + CHECK_ERROR("snd_pcm_hw_params"); /* set up the buffer */ bufsizeforthis = DEFDACBLKSIZE * dev->a_sampwidth * *channels; @@ -227,18 +235,19 @@ static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate, alsa_snd_bufsize = bufsizeforthis; } return (1); +#undef CHECK_ERROR } /* return 0 on success */ int alsa_open_audio(int naudioindev, int *audioindev, int nchindev, int *chindev, int naudiooutdev, int *audiooutdev, int nchoutdev, - int *choutdev, int rate) + int *choutdev, int rate, int blocksize) { int err, inchans = 0, outchans = 0, subunitdir; char devname[512]; snd_output_t* out; - int frag_size = (sys_blocksize ? sys_blocksize : ALSA_DEFFRAGSIZE); + int frag_size = (blocksize ? blocksize : ALSA_DEFFRAGSIZE); int nfrags, i, iodev, dev2; int wantinchans, wantoutchans, device; @@ -292,7 +301,7 @@ int alsa_open_audio(int naudioindev, int *audioindev, int nchindev, if (alsa_usemmap) { post("using mmap audio interface"); - if (alsamm_open_audio(rate)) + if (alsamm_open_audio(rate, blocksize)) goto blewit; else return (0); } @@ -469,7 +478,7 @@ int alsa_send_dacs(void) ((char *)(alsa_snd_buf))[3*j+1] = ((s>>8) & 255); ((char *)(alsa_snd_buf))[3*j+2] = ((s>>16) & 255); #else - fprintf(stderr("big endian 24-bit not supported"); + fprintf(stderr, "big endian 24-bit not supported"); #endif } for (; i < thisdevchans; i++, ch++) @@ -581,7 +590,7 @@ int alsa_send_dacs(void) * (1./ INT32_MAX); } #else - fprintf(stderr("big endian 24-bit not supported"); + fprintf(stderr, "big endian 24-bit not supported"); #endif } else |