diff options
Diffstat (limited to 'audio/alsaaudio.c')
| -rw-r--r-- | audio/alsaaudio.c | 88 |
1 files changed, 47 insertions, 41 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 77a08a1c58..43cfa258d7 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -86,9 +86,9 @@ static struct { }; struct alsa_params_req { - unsigned int freq; - audfmt_e fmt; - unsigned int nchannels; + int freq; + snd_pcm_format_t fmt; + int nchannels; unsigned int buffer_size; unsigned int period_size; }; @@ -96,6 +96,7 @@ struct alsa_params_req { struct alsa_params_obt { int freq; audfmt_e fmt; + int endianness; int nchannels; snd_pcm_uframes_t samples; }; @@ -143,7 +144,7 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len) return audio_pcm_sw_write (sw, buf, len); } -static int aud_to_alsafmt (audfmt_e fmt) +static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt) { switch (fmt) { case AUD_FMT_S8: @@ -173,7 +174,8 @@ static int aud_to_alsafmt (audfmt_e fmt) } } -static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness) +static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, + int *endianness) { switch (alsafmt) { case SND_PCM_FORMAT_S8: @@ -234,7 +236,6 @@ static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness) return 0; } -#if defined DEBUG_MISMATCHES || defined DEBUG static void alsa_dump_info (struct alsa_params_req *req, struct alsa_params_obt *obt) { @@ -248,7 +249,6 @@ static void alsa_dump_info (struct alsa_params_req *req, req->buffer_size, req->period_size); dolog ("obtained: samples %ld\n", obt->samples); } -#endif static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) { @@ -291,6 +291,7 @@ static int alsa_open (int in, struct alsa_params_req *req, unsigned int period_size, buffer_size; snd_pcm_uframes_t obt_buffer_size; const char *typ = in ? "ADC" : "DAC"; + snd_pcm_format_t obtfmt; freq = req->freq; period_size = req->period_size; @@ -327,9 +328,8 @@ static int alsa_open (int in, struct alsa_params_req *req, } err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); - if (err < 0) { + if (err < 0 && conf.verbose) { alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); - goto err; } err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); @@ -494,6 +494,17 @@ static int alsa_open (int in, struct alsa_params_req *req, goto err; } + err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to get format\n"); + goto err; + } + + if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { + dolog ("Invalid format was returned %d\n", obtfmt); + goto err; + } + err = snd_pcm_prepare (handle); if (err < 0) { alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); @@ -504,28 +515,41 @@ static int alsa_open (int in, struct alsa_params_req *req, snd_pcm_uframes_t threshold; int bytes_per_sec; - bytes_per_sec = freq - << (nchannels == 2) - << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16); + bytes_per_sec = freq << (nchannels == 2); + + switch (obt->fmt) { + case AUD_FMT_S8: + case AUD_FMT_U8: + break; + + case AUD_FMT_S16: + case AUD_FMT_U16: + bytes_per_sec <<= 1; + break; + + case AUD_FMT_S32: + case AUD_FMT_U32: + bytes_per_sec <<= 2; + break; + } threshold = (conf.threshold * bytes_per_sec) / 1000; alsa_set_threshold (handle, threshold); } - obt->fmt = req->fmt; obt->nchannels = nchannels; obt->freq = freq; obt->samples = obt_buffer_size; + *handlep = handle; -#if defined DEBUG_MISMATCHES || defined DEBUG - if (obt->fmt != req->fmt || - obt->nchannels != req->nchannels || - obt->freq != req->freq) { - dolog ("Audio paramters mismatch for %s\n", typ); + if (conf.verbose && + (obt->fmt != req->fmt || + obt->nchannels != req->nchannels || + obt->freq != req->freq)) { + dolog ("Audio paramters for %s\n", typ); alsa_dump_info (req, obt); } -#endif #ifdef DEBUG alsa_dump_info (req, obt); @@ -665,9 +689,6 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; struct alsa_params_req req; struct alsa_params_obt obt; - audfmt_e effective_fmt; - int endianness; - int err; snd_pcm_t *handle; audsettings_t obt_as; @@ -681,16 +702,10 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) return -1; } - err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); - if (err) { - alsa_anal_close (&handle); - return -1; - } - obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; - obt_as.fmt = effective_fmt; - obt_as.endianness = endianness; + obt_as.fmt = obt.fmt; + obt_as.endianness = obt.endianness; audio_pcm_init_info (&hw->info, &obt_as); hw->samples = obt.samples; @@ -751,9 +766,6 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; struct alsa_params_req req; struct alsa_params_obt obt; - int endianness; - int err; - audfmt_e effective_fmt; snd_pcm_t *handle; audsettings_t obt_as; @@ -767,16 +779,10 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) return -1; } - err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); - if (err) { - alsa_anal_close (&handle); - return -1; - } - obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; - obt_as.fmt = effective_fmt; - obt_as.endianness = endianness; + obt_as.fmt = obt.fmt; + obt_as.endianness = obt.endianness; audio_pcm_init_info (&hw->info, &obt_as); hw->samples = obt.samples; |