alsaaudio: port to the new audio backend api
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: ab9768e73dfe7b7305bd6a51629846e0d77622a5.1568927990.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This commit is contained in:
Kővágó, Zoltán
Gerd Hoffmann
parent
ff095e5231
commit
286a5d201e
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@ -44,9 +44,6 @@ struct pollhlp {
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typedef struct ALSAVoiceOut {
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HWVoiceOut hw;
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int wpos;
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int pending;
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void *pcm_buf;
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snd_pcm_t *handle;
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struct pollhlp pollhlp;
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Audiodev *dev;
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@ -55,7 +52,6 @@ typedef struct ALSAVoiceOut {
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typedef struct ALSAVoiceIn {
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HWVoiceIn hw;
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snd_pcm_t *handle;
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void *pcm_buf;
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struct pollhlp pollhlp;
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Audiodev *dev;
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} ALSAVoiceIn;
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@ -602,102 +598,64 @@ static int alsa_open(bool in, struct alsa_params_req *req,
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return -1;
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}
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static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
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{
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snd_pcm_sframes_t avail;
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avail = snd_pcm_avail_update (handle);
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if (avail < 0) {
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if (avail == -EPIPE) {
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if (!alsa_recover (handle)) {
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avail = snd_pcm_avail_update (handle);
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}
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}
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if (avail < 0) {
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alsa_logerr (avail,
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"Could not obtain number of available frames\n");
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return -1;
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}
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}
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return avail;
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}
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static void alsa_write_pending (ALSAVoiceOut *alsa)
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{
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HWVoiceOut *hw = &alsa->hw;
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while (alsa->pending) {
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int left_till_end_samples = hw->samples - alsa->wpos;
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int len = MIN (alsa->pending, left_till_end_samples);
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char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
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while (len) {
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snd_pcm_sframes_t written;
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written = snd_pcm_writei (alsa->handle, src, len);
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if (written <= 0) {
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switch (written) {
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case 0:
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trace_alsa_wrote_zero(len);
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return;
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case -EPIPE:
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if (alsa_recover (alsa->handle)) {
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alsa_logerr (written, "Failed to write %d frames\n",
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len);
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return;
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}
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trace_alsa_xrun_out();
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continue;
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case -ESTRPIPE:
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/* stream is suspended and waiting for an
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application recovery */
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if (alsa_resume (alsa->handle)) {
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alsa_logerr (written, "Failed to write %d frames\n",
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len);
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return;
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}
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trace_alsa_resume_out();
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continue;
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case -EAGAIN:
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return;
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default:
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alsa_logerr (written, "Failed to write %d frames from %p\n",
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len, src);
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return;
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}
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}
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alsa->wpos = (alsa->wpos + written) % hw->samples;
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alsa->pending -= written;
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len -= written;
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}
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}
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}
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static size_t alsa_run_out(HWVoiceOut *hw, size_t live)
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static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
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{
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ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
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size_t decr;
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snd_pcm_sframes_t avail;
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size_t pos = 0;
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size_t len_frames = len >> hw->info.shift;
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avail = alsa_get_avail (alsa->handle);
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if (avail < 0) {
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dolog ("Could not get number of available playback frames\n");
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return 0;
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while (len_frames) {
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char *src = advance(buf, pos);
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snd_pcm_sframes_t written;
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written = snd_pcm_writei(alsa->handle, src, len_frames);
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if (written <= 0) {
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switch (written) {
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case 0:
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trace_alsa_wrote_zero(len_frames);
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return pos;
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case -EPIPE:
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if (alsa_recover(alsa->handle)) {
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alsa_logerr(written, "Failed to write %zu frames\n",
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len_frames);
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return pos;
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}
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trace_alsa_xrun_out();
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continue;
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case -ESTRPIPE:
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/*
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* stream is suspended and waiting for an application
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* recovery
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*/
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if (alsa_resume(alsa->handle)) {
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alsa_logerr(written, "Failed to write %zu frames\n",
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len_frames);
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return pos;
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}
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trace_alsa_resume_out();
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continue;
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case -EAGAIN:
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return pos;
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default:
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alsa_logerr(written, "Failed to write %zu frames from %p\n",
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len, src);
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return pos;
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}
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}
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pos += written << hw->info.shift;
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if (written < len_frames) {
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break;
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}
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len_frames -= written;
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}
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decr = MIN (live, avail);
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decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
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alsa->pending += decr;
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alsa_write_pending (alsa);
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return decr;
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return pos;
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}
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static void alsa_fini_out (HWVoiceOut *hw)
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@ -706,9 +664,6 @@ static void alsa_fini_out (HWVoiceOut *hw)
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ldebug ("alsa_fini\n");
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alsa_anal_close (&alsa->handle, &alsa->pollhlp);
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g_free(alsa->pcm_buf);
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alsa->pcm_buf = NULL;
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}
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static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
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@ -737,14 +692,6 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
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audio_pcm_init_info (&hw->info, &obt_as);
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hw->samples = obt.samples;
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alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
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if (!alsa->pcm_buf) {
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dolog("Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
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hw->samples, 1 << hw->info.shift);
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alsa_anal_close1 (&handle);
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return -1;
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}
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alsa->pollhlp.s = hw->s;
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alsa->handle = handle;
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alsa->dev = dev;
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@ -839,14 +786,6 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
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audio_pcm_init_info (&hw->info, &obt_as);
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hw->samples = obt.samples;
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alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
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if (!alsa->pcm_buf) {
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dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
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hw->samples, 1 << hw->info.shift);
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alsa_anal_close1 (&handle);
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return -1;
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}
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alsa->pollhlp.s = hw->s;
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alsa->handle = handle;
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alsa->dev = dev;
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@ -858,129 +797,48 @@ static void alsa_fini_in (HWVoiceIn *hw)
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ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
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alsa_anal_close (&alsa->handle, &alsa->pollhlp);
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g_free(alsa->pcm_buf);
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alsa->pcm_buf = NULL;
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}
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static size_t alsa_run_in(HWVoiceIn *hw)
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static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
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{
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ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
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int hwshift = hw->info.shift;
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int i;
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size_t live = audio_pcm_hw_get_live_in (hw);
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size_t dead = hw->samples - live;
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size_t decr;
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struct {
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size_t add;
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size_t len;
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} bufs[2] = {
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{ .add = hw->wpos, .len = 0 },
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{ .add = 0, .len = 0 }
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};
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snd_pcm_sframes_t avail;
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snd_pcm_uframes_t read_samples = 0;
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size_t pos = 0;
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if (!dead) {
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return 0;
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}
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avail = alsa_get_avail (alsa->handle);
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if (avail < 0) {
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dolog ("Could not get number of captured frames\n");
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return 0;
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}
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if (!avail) {
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snd_pcm_state_t state;
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state = snd_pcm_state (alsa->handle);
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switch (state) {
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case SND_PCM_STATE_PREPARED:
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avail = hw->samples;
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break;
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case SND_PCM_STATE_SUSPENDED:
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/* stream is suspended and waiting for an application recovery */
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if (alsa_resume (alsa->handle)) {
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dolog ("Failed to resume suspended input stream\n");
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return 0;
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}
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trace_alsa_resume_in();
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break;
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default:
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trace_alsa_no_frames(state);
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return 0;
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}
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}
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decr = MIN(dead, avail);
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if (!decr) {
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return 0;
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}
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if (hw->wpos + decr > hw->samples) {
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bufs[0].len = (hw->samples - hw->wpos);
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bufs[1].len = (decr - (hw->samples - hw->wpos));
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}
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else {
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bufs[0].len = decr;
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}
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for (i = 0; i < 2; ++i) {
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void *src;
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struct st_sample *dst;
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while (len) {
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void *dst = advance(buf, pos);
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snd_pcm_sframes_t nread;
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snd_pcm_uframes_t len;
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len = bufs[i].len;
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nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
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src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
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dst = hw->conv_buf + bufs[i].add;
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if (nread <= 0) {
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switch (nread) {
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case 0:
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trace_alsa_read_zero(len);
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return pos;;
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while (len) {
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nread = snd_pcm_readi (alsa->handle, src, len);
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if (nread <= 0) {
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switch (nread) {
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case 0:
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trace_alsa_read_zero(len);
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goto exit;
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case -EPIPE:
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if (alsa_recover (alsa->handle)) {
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alsa_logerr (nread, "Failed to read %ld frames\n", len);
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goto exit;
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}
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trace_alsa_xrun_in();
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continue;
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case -EAGAIN:
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goto exit;
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default:
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alsa_logerr (
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nread,
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"Failed to read %ld frames from %p\n",
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len,
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src
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);
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goto exit;
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case -EPIPE:
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if (alsa_recover(alsa->handle)) {
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alsa_logerr(nread, "Failed to read %zu frames\n", len);
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return pos;
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}
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trace_alsa_xrun_in();
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continue;
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case -EAGAIN:
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return pos;
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default:
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alsa_logerr(nread, "Failed to read %zu frames to %p\n",
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len, dst);
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return pos;;
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}
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hw->conv (dst, src, nread);
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src = advance (src, nread << hwshift);
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dst += nread;
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read_samples += nread;
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len -= nread;
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}
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pos += nread << hw->info.shift;
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len -= nread << hw->info.shift;
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}
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exit:
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hw->wpos = (hw->wpos + read_samples) % hw->samples;
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return read_samples;
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return pos;
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}
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static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
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@ -1065,12 +923,12 @@ static void alsa_audio_fini (void *opaque)
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static struct audio_pcm_ops alsa_pcm_ops = {
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.init_out = alsa_init_out,
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.fini_out = alsa_fini_out,
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.run_out = alsa_run_out,
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.write = alsa_write,
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.ctl_out = alsa_ctl_out,
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.init_in = alsa_init_in,
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.fini_in = alsa_fini_in,
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.run_in = alsa_run_in,
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.read = alsa_read,
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.ctl_in = alsa_ctl_in,
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};
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