mirror of https://github.com/mpv-player/mpv
ao_alsa: cleanups
No functional changes. ALSA_PCM_NEW_HW_PARAMS_API was a pre-ALSA 1.0.0 thing and does nothing with modern ALSA. It stopped being necessary about 10 years ago. 3 functions are moved to avoid forward references.
This commit is contained in:
wm4
parent
26190dbe57
commit
8a7b686597
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@ -40,9 +40,6 @@
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#include "common/msg.h"
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#include "osdep/endian.h"
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#define ALSA_PCM_NEW_HW_PARAMS_API
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#define ALSA_PCM_NEW_SW_PARAMS_API
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#include <alsa/asoundlib.h>
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#define HAVE_CHMAP_API \
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@ -87,10 +84,6 @@ struct priv {
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MP_WARN(ao, "%s: %s\n", (message), snd_strerror(err)); \
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} while (0)
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static double get_delay(struct ao *ao);
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static void uninit(struct ao *ao);
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/* to set/get/query special features/parameters */
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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struct priv *p = ao->priv;
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@ -112,10 +105,8 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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if (AF_FORMAT_IS_SPECIAL(ao->format))
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return CONTROL_FALSE;
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//allocate simple id
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snd_mixer_selem_id_alloca(&sid);
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//sets simple-mixer index and name
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snd_mixer_selem_id_set_index(sid, p->cfg_mixer_index);
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snd_mixer_selem_id_set_name(sid, p->cfg_mixer_name);
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@ -147,7 +138,6 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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ao_control_vol_t *vol = arg;
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set_vol = vol->left / f_multi + pmin + 0.5;
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//setting channels
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err = snd_mixer_selem_set_playback_volume
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(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol);
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CHECK_ALSA_ERROR("Error setting left channel");
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@ -159,8 +149,7 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol);
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CHECK_ALSA_ERROR("Error setting right channel");
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MP_DBG(ao, "right=%li, pmin=%li, pmax=%li, mult=%f\n",
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set_vol, pmin, pmax,
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f_multi);
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set_vol, pmin, pmax, f_multi);
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break;
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}
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case AOCONTROL_GET_VOLUME: {
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@ -296,9 +285,7 @@ static const char *const device_channel_layouts[][2] = {
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{"surround71", "fl-fr-bl-br-fc-lfe-sl-sr"},
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};
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#define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0]))
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#define NUM_ALSA_CHMAPS ARRAY_LEN(device_channel_layouts)
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#define NUM_ALSA_CHMAPS MP_ARRAY_SIZE(device_channel_layouts)
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static const char *select_chmap(struct ao *ao)
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{
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@ -371,7 +358,6 @@ static int try_open_device(struct ao *ao, const char *device, int open_mode)
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if (AF_FORMAT_IS_IEC61937(ao->format)) {
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void *tmp = talloc_new(NULL);
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/* to set the non-audio bit, use AES0=6 */
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char *params = talloc_asprintf(tmp,
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"AES0=%d,AES1=%d,AES2=0,AES3=%d",
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IEC958_AES0_NONAUDIO | IEC958_AES0_PRO_EMPHASIS_NONE,
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@ -388,28 +374,28 @@ static int try_open_device(struct ao *ao, const char *device, int open_mode)
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return snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, open_mode);
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}
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/*
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open & setup audio device
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return: 0=success -1=fail
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*/
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static void uninit(struct ao *ao)
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{
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struct priv *p = ao->priv;
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if (p->alsa) {
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int err;
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err = snd_pcm_close(p->alsa);
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CHECK_ALSA_ERROR("pcm close error");
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}
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alsa_error: ;
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}
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static int init(struct ao *ao)
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{
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int err;
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snd_pcm_uframes_t chunk_size;
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snd_pcm_uframes_t bufsize;
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snd_pcm_uframes_t boundary;
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struct priv *p = ao->priv;
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int err;
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if (!p->cfg_ni)
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ao->format = af_fmt_from_planar(ao->format);
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/* switch for spdif
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* sets opening sequence for SPDIF
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* sets also the playback and other switches 'on the fly'
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* while opening the abstract alias for the spdif subdevice
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* 'iec958'
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*/
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const char *device;
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if (AF_FORMAT_IS_IEC61937(ao->format)) {
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device = "iec958";
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@ -430,9 +416,7 @@ static int init(struct ao *ao)
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MP_VERBOSE(ao, "using device: %s\n", device);
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MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());
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int open_mode = p->cfg_block ? 0 : SND_PCM_NONBLOCK;
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//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
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err = try_open_device(ao, device, open_mode);
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err = try_open_device(ao, device, p->cfg_block ? 0 : SND_PCM_NONBLOCK);
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if (err < 0) {
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if (err != -EBUSY && !p->cfg_block) {
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MP_WARN(ao, "Open in nonblock-mode "
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@ -455,7 +439,6 @@ static int init(struct ao *ao)
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snd_pcm_hw_params_alloca(&alsa_hwparams);
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snd_pcm_sw_params_alloca(&alsa_swparams);
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// setting hw-parameters
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err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
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CHECK_ALSA_ERROR("Unable to get initial parameters");
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@ -530,15 +513,16 @@ static int init(struct ao *ao)
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err = snd_pcm_hw_params(p->alsa, alsa_hwparams);
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CHECK_ALSA_ERROR("Unable to set hw-parameters");
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// end setting hw-params
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/* end setting hw-params */
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// gets buffersize for control
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snd_pcm_uframes_t bufsize;
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err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize);
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CHECK_ALSA_ERROR("Unable to get buffersize");
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p->buffersize = bufsize;
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MP_VERBOSE(ao, "got buffersize=%i samples\n", p->buffersize);
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snd_pcm_uframes_t chunk_size;
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err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL);
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CHECK_ALSA_ERROR("Unable to get period size");
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@ -549,6 +533,7 @@ static int init(struct ao *ao)
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err = snd_pcm_sw_params_current(p->alsa, alsa_swparams);
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CHECK_ALSA_ERROR("Unable to get sw-parameters");
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snd_pcm_uframes_t boundary;
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err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary);
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CHECK_ALSA_ERROR("Unable to get boundary");
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@ -568,7 +553,7 @@ static int init(struct ao *ao)
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CHECK_ALSA_ERROR("Unable to set silence size");
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err = snd_pcm_sw_params(p->alsa, alsa_swparams);
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CHECK_ALSA_ERROR("Unable to get sw-parameters");
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CHECK_ALSA_ERROR("Unable to set sw-parameters");
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/* end setting sw-params */
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@ -594,34 +579,11 @@ static int init(struct ao *ao)
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}
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#endif
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MP_VERBOSE(ao, "opened: %d Hz/%d channels/%d bps/%d samples buffer/%s\n",
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ao->samplerate, ao->channels.num, af_fmt2bits(ao->format),
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p->buffersize, snd_pcm_format_description(p->alsa_fmt));
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return 0;
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alsa_error:
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uninit(ao);
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return -1;
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} // end init
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/* close audio device */
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static void uninit(struct ao *ao)
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{
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struct priv *p = ao->priv;
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if (p->alsa) {
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int err;
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err = snd_pcm_close(p->alsa);
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CHECK_ALSA_ERROR("pcm close error");
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MP_VERBOSE(ao, "uninit: pcm closed\n");
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}
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alsa_error:
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p->alsa = NULL;
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}
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static void drain(struct ao *ao)
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@ -630,6 +592,46 @@ static void drain(struct ao *ao)
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snd_pcm_drain(p->alsa);
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}
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static int get_space(struct ao *ao)
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{
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struct priv *p = ao->priv;
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snd_pcm_status_t *status;
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int err;
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snd_pcm_status_alloca(&status);
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err = snd_pcm_status(p->alsa, status);
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CHECK_ALSA_ERROR("cannot get pcm status");
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unsigned space = snd_pcm_status_get_avail(status);
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if (space > p->buffersize) // Buffer underrun?
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space = p->buffersize;
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return space / p->outburst * p->outburst;
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alsa_error:
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return 0;
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}
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/* delay in seconds between first and last sample in buffer */
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static double get_delay(struct ao *ao)
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{
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struct priv *p = ao->priv;
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snd_pcm_sframes_t delay;
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if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED)
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return p->delay_before_pause;
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if (snd_pcm_delay(p->alsa, &delay) < 0)
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return 0;
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if (delay < 0) {
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/* underrun - move the application pointer forward to catch up */
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snd_pcm_forward(p->alsa, -delay);
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delay = 0;
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}
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return delay / (double)ao->samplerate;
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}
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static void audio_pause(struct ao *ao)
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{
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struct priv *p = ao->priv;
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@ -683,7 +685,6 @@ static void audio_resume(struct ao *ao)
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alsa_error: ;
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}
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/* stop playing and empty buffers (for seeking/pause) */
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static void reset(struct ao *ao)
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{
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struct priv *p = ao->priv;
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@ -716,8 +717,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
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res = snd_pcm_writei(p->alsa, data[0], samples);
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}
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if (res == -EINTR || res == -EAGAIN) {
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/* nothing to do */
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if (res == -EINTR || res == -EAGAIN) { /* retry */
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res = 0;
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} else if (res == -ESTRPIPE) { /* suspend */
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audio_resume(ao);
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@ -736,46 +736,6 @@ alsa_error:
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return -1;
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}
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static int get_space(struct ao *ao)
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{
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struct priv *p = ao->priv;
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snd_pcm_status_t *status;
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int err;
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snd_pcm_status_alloca(&status);
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err = snd_pcm_status(p->alsa, status);
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CHECK_ALSA_ERROR("cannot get pcm status");
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unsigned space = snd_pcm_status_get_avail(status);
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if (space > p->buffersize) // Buffer underrun?
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space = p->buffersize;
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return space / p->outburst * p->outburst;
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alsa_error:
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return 0;
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}
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/* delay in seconds between first and last sample in buffer */
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static double get_delay(struct ao *ao)
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{
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struct priv *p = ao->priv;
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snd_pcm_sframes_t delay;
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if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED)
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return p->delay_before_pause;
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if (snd_pcm_delay(p->alsa, &delay) < 0)
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return 0;
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if (delay < 0) {
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/* underrun - move the application pointer forward to catch up */
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snd_pcm_forward(p->alsa, -delay);
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delay = 0;
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}
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return delay / (double)ao->samplerate;
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}
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#define MAX_POLL_FDS 20
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static int audio_wait(struct ao *ao, pthread_mutex_t *lock)
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{
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