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pthreads-based audio and miscellaneous audio clean-up (malc).
ESD support (malc, Frederick Reeve). git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@3917 c046a42c-6fe2-441c-8c8c-71466251a162
This commit is contained in:
balrog
parent
b34d259a81
commit
ca9cc28c62
15 changed files with 883 additions and 54 deletions
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@ -86,9 +86,9 @@ static struct {
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};
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struct alsa_params_req {
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unsigned int freq;
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audfmt_e fmt;
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unsigned int nchannels;
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int freq;
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snd_pcm_format_t fmt;
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int nchannels;
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unsigned int buffer_size;
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unsigned int period_size;
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};
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@ -96,6 +96,7 @@ struct alsa_params_req {
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struct alsa_params_obt {
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int freq;
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audfmt_e fmt;
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int endianness;
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int nchannels;
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snd_pcm_uframes_t samples;
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};
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@ -143,7 +144,7 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len)
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return audio_pcm_sw_write (sw, buf, len);
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}
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static int aud_to_alsafmt (audfmt_e fmt)
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static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
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{
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switch (fmt) {
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case AUD_FMT_S8:
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@ -173,7 +174,8 @@ static int aud_to_alsafmt (audfmt_e fmt)
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}
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}
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static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
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static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
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int *endianness)
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{
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switch (alsafmt) {
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case SND_PCM_FORMAT_S8:
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@ -234,7 +236,6 @@ static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
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return 0;
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}
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#if defined DEBUG_MISMATCHES || defined DEBUG
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static void alsa_dump_info (struct alsa_params_req *req,
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struct alsa_params_obt *obt)
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{
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@ -248,7 +249,6 @@ static void alsa_dump_info (struct alsa_params_req *req,
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req->buffer_size, req->period_size);
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dolog ("obtained: samples %ld\n", obt->samples);
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}
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#endif
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static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
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{
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@ -291,6 +291,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
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unsigned int period_size, buffer_size;
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snd_pcm_uframes_t obt_buffer_size;
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const char *typ = in ? "ADC" : "DAC";
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snd_pcm_format_t obtfmt;
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freq = req->freq;
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period_size = req->period_size;
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@ -327,9 +328,8 @@ static int alsa_open (int in, struct alsa_params_req *req,
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}
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err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
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if (err < 0) {
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if (err < 0 && conf.verbose) {
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alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
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goto err;
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}
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err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
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@ -494,6 +494,17 @@ static int alsa_open (int in, struct alsa_params_req *req,
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goto err;
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}
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err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
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if (err < 0) {
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alsa_logerr2 (err, typ, "Failed to get format\n");
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goto err;
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}
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if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
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dolog ("Invalid format was returned %d\n", obtfmt);
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goto err;
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}
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err = snd_pcm_prepare (handle);
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if (err < 0) {
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alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
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@ -504,28 +515,41 @@ static int alsa_open (int in, struct alsa_params_req *req,
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snd_pcm_uframes_t threshold;
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int bytes_per_sec;
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bytes_per_sec = freq
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<< (nchannels == 2)
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<< (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
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bytes_per_sec = freq << (nchannels == 2);
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switch (obt->fmt) {
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case AUD_FMT_S8:
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case AUD_FMT_U8:
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break;
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case AUD_FMT_S16:
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case AUD_FMT_U16:
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bytes_per_sec <<= 1;
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break;
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case AUD_FMT_S32:
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case AUD_FMT_U32:
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bytes_per_sec <<= 2;
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break;
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}
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threshold = (conf.threshold * bytes_per_sec) / 1000;
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alsa_set_threshold (handle, threshold);
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}
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obt->fmt = req->fmt;
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obt->nchannels = nchannels;
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obt->freq = freq;
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obt->samples = obt_buffer_size;
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*handlep = handle;
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#if defined DEBUG_MISMATCHES || defined DEBUG
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if (obt->fmt != req->fmt ||
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obt->nchannels != req->nchannels ||
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obt->freq != req->freq) {
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dolog ("Audio paramters mismatch for %s\n", typ);
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if (conf.verbose &&
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(obt->fmt != req->fmt ||
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obt->nchannels != req->nchannels ||
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obt->freq != req->freq)) {
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dolog ("Audio paramters for %s\n", typ);
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alsa_dump_info (req, obt);
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}
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#endif
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#ifdef DEBUG
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alsa_dump_info (req, obt);
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@ -665,9 +689,6 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
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ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
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struct alsa_params_req req;
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struct alsa_params_obt obt;
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audfmt_e effective_fmt;
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int endianness;
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int err;
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snd_pcm_t *handle;
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audsettings_t obt_as;
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@ -681,16 +702,10 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
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return -1;
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}
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err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
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if (err) {
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alsa_anal_close (&handle);
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return -1;
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}
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obt_as.freq = obt.freq;
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obt_as.nchannels = obt.nchannels;
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obt_as.fmt = effective_fmt;
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obt_as.endianness = endianness;
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obt_as.fmt = obt.fmt;
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obt_as.endianness = obt.endianness;
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audio_pcm_init_info (&hw->info, &obt_as);
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hw->samples = obt.samples;
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@ -751,9 +766,6 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
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ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
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struct alsa_params_req req;
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struct alsa_params_obt obt;
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int endianness;
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int err;
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audfmt_e effective_fmt;
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snd_pcm_t *handle;
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audsettings_t obt_as;
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@ -767,16 +779,10 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
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return -1;
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}
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err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
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if (err) {
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alsa_anal_close (&handle);
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return -1;
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}
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obt_as.freq = obt.freq;
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obt_as.nchannels = obt.nchannels;
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obt_as.fmt = effective_fmt;
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obt_as.endianness = endianness;
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obt_as.fmt = obt.fmt;
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obt_as.endianness = obt.endianness;
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audio_pcm_init_info (&hw->info, &obt_as);
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hw->samples = obt.samples;
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