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https://github.com/mpv-player/mpv
synced 2025-03-11 08:37:59 +00:00
ao_alsa: switch to new AO API
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wm4
parent
74487b8430
commit
e1207f2ceb
@ -36,6 +36,7 @@
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#include <alloca.h>
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#include "config.h"
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#include "core/options.h"
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#include "core/subopt-helper.h"
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#include "audio/mixer.h"
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#include "core/mp_msg.h"
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@ -46,19 +47,11 @@
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#include <alsa/asoundlib.h>
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#include "ao.h"
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#include "audio_out_internal.h"
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#include "audio/format.h"
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#include "audio/reorder_ch.h"
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static const ao_info_t info =
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{
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"ALSA-0.9.x-1.x audio output",
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"alsa",
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"Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
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"under development"
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};
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LIBAO_EXTERN(alsa)
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extern struct ao *global_ao;
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#define ao_data (*global_ao)
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static snd_pcm_t *alsa_handler;
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static snd_pcm_format_t alsa_format;
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@ -83,6 +76,9 @@ static float delay_before_pause;
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} \
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} while (0)
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static float get_delay(struct ao *ao);
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static int play(struct ao *ao, void *data, int len, int flags);
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static void alsa_error_handler(const char *file, int line, const char *function,
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int err, const char *format, ...)
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{
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@ -103,7 +99,7 @@ static void alsa_error_handler(const char *file, int line, const char *function,
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}
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/* to set/get/query special features/parameters */
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static int control(int cmd, void *arg)
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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snd_mixer_t *handle = NULL;
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switch (cmd) {
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@ -127,10 +123,10 @@ static int control(int cmd, void *arg)
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if (AF_FORMAT_IS_IEC61937(ao_data.format))
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return CONTROL_TRUE;
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if (mixer_channel) {
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if (global_ao->opts->mixer_channel) {
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char *test_mix_index;
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mix_name = strdup(mixer_channel);
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mix_name = strdup(global_ao->opts->mixer_channel);
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if ((test_mix_index = strchr(mix_name, ','))) {
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*test_mix_index = 0;
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test_mix_index++;
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@ -143,8 +139,8 @@ static int control(int cmd, void *arg)
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}
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}
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}
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if (mixer_device)
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card = mixer_device;
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if (global_ao->opts->mixer_device)
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card = global_ao->opts->mixer_device;
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//allocate simple id
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snd_mixer_selem_id_alloca(&sid);
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@ -153,7 +149,7 @@ static int control(int cmd, void *arg)
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snd_mixer_selem_id_set_index(sid, mix_index);
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snd_mixer_selem_id_set_name(sid, mix_name);
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if (mixer_channel) {
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if (global_ao->opts->mixer_channel) {
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free(mix_name);
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mix_name = NULL;
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}
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@ -402,10 +398,9 @@ static int try_open_device(const char *device, int open_mode, int try_ac3)
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/*
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open & setup audio device
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return: 1=success 0=fail
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return: 0=success -1=fail
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*/
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static int init(int rate_hz, const struct mp_chmap *channels, int format,
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int flags)
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static int init(struct ao *ao, char *params)
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{
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int err;
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int block;
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@ -419,14 +414,15 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
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{NULL}
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};
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global_ao = ao;
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char alsa_device[ALSA_DEVICE_SIZE + 1];
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// make sure alsa_device is null-terminated even when using strncpy etc.
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memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
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mp_msg(MSGT_AO, MSGL_V,
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"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
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ao_data.channels.num,
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format);
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"alsa-init: requested format: %d Hz, %d channels, %x\n",
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ao->samplerate, ao->channels.num, ao->format);
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alsa_handler = NULL;
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mp_msg(MSGT_AO, MSGL_V, "alsa-init: using ALSA %s\n", snd_asoundlib_version());
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@ -435,7 +431,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
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snd_lib_error_set_handler(alsa_error_handler);
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alsa_format = find_alsa_format(format);
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alsa_format = find_alsa_format(ao->format);
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//subdevice parsing
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// set defaults
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@ -447,7 +443,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
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* 'iec958'
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*/
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device.str = NULL;
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if (AF_FORMAT_IS_IEC61937(format)) {
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if (AF_FORMAT_IS_IEC61937(ao->format)) {
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device.str = "iec958";
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mp_msg(MSGT_AO, MSGL_V,
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"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n",
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@ -463,14 +459,14 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
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name, ao_data.channels.num);
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talloc_free(name);
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}
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if (strcmp(device.str, "default") != 0 && format == AF_FORMAT_FLOAT_NE)
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if (strcmp(device.str, "default") != 0 && ao->format == AF_FORMAT_FLOAT_NE)
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{
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// hack - use the converter plugin (why the heck?)
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device.str = talloc_asprintf(global_ao, "plug:%s", device.str);
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}
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}
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device.len = strlen(device.str);
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if (subopt_parse(ao_subdevice, subopts) != 0) {
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if (subopt_parse(params, subopts) != 0) {
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print_help();
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return 0;
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}
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@ -481,7 +477,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
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alsa_can_pause = 1;
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int open_mode = block ? 0 : SND_PCM_NONBLOCK;
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int isac3 = AF_FORMAT_IS_IEC61937(format);
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int isac3 = AF_FORMAT_IS_IEC61937(ao->format);
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//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
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err = try_open_device(alsa_device, open_mode, isac3);
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if (err < 0) {
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@ -522,7 +518,7 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
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(alsa_handler, alsa_hwparams, alsa_format);
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if (err < 0) {
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mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Format %s is not supported "
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"by hardware, trying default.\n", af_fmt2str_short(format));
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"by hardware, trying default.\n", af_fmt2str_short(ao->format));
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alsa_format = SND_PCM_FORMAT_S16_LE;
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if (AF_FORMAT_IS_AC3(ao_data.format))
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ao_data.format = AF_FORMAT_AC3_LE;
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@ -619,15 +615,15 @@ static int init(int rate_hz, const struct mp_chmap *channels, int format,
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ao_data.samplerate, ao_data.channels.num, (int)bytes_per_sample,
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ao_data.buffersize, snd_pcm_format_description(alsa_format));
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return 1;
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return 0;
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alsa_error:
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return 0;
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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(int immed)
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static void uninit(struct ao *ao, bool immed)
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{
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if (alsa_handler) {
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@ -648,12 +644,12 @@ static void uninit(int immed)
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alsa_error: ;
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}
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static void audio_pause(void)
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static void audio_pause(struct ao *ao)
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{
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int err;
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if (alsa_can_pause) {
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delay_before_pause = get_delay();
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delay_before_pause = get_delay(ao);
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err = snd_pcm_pause(alsa_handler, 1);
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CHECK_ALSA_ERROR("pcm pause error");
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mp_msg(MSGT_AO, MSGL_V, "alsa-pause: pause supported by hardware\n");
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@ -670,7 +666,7 @@ static void audio_pause(void)
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alsa_error: ;
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}
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static void audio_resume(void)
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static void audio_resume(struct ao *ao)
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{
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int err;
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@ -689,7 +685,7 @@ static void audio_resume(void)
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CHECK_ALSA_ERROR("pcm prepare error");
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if (prepause_frames) {
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void *silence = calloc(prepause_frames, bytes_per_sample);
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play(silence, prepause_frames * bytes_per_sample, 0);
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play(ao, silence, prepause_frames * bytes_per_sample, 0);
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free(silence);
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}
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}
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@ -698,7 +694,7 @@ 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(void)
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static void reset(struct ao *ao)
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{
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int err;
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@ -719,7 +715,7 @@ alsa_error: ;
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thanxs for marius <marius@rospot.com> for giving us the light ;)
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*/
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static int play(void *data, int len, int flags)
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static int play(struct ao *ao, void *data, int len, int flags)
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{
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int num_frames;
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snd_pcm_sframes_t res = 0;
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@ -768,7 +764,7 @@ alsa_error:
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}
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/* how many byes are free in the buffer */
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static int get_space(void)
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static int get_space(struct ao *ao)
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{
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snd_pcm_status_t *status;
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int err;
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@ -788,7 +784,7 @@ alsa_error:
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}
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/* delay in seconds between first and last sample in buffer */
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static float get_delay(void)
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static float get_delay(struct ao *ao)
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{
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if (alsa_handler) {
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snd_pcm_sframes_t delay;
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@ -808,3 +804,22 @@ static float get_delay(void)
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} else
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return 0;
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}
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const struct ao_driver audio_out_alsa = {
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.is_new = true,
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.info = &(const struct ao_info) {
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"ALSA-0.9.x-1.x audio output",
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"alsa",
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"Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
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"under development"
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},
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.init = init,
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.uninit = uninit,
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.control = control,
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.get_space = get_space,
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.play = play,
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.get_delay = get_delay,
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.pause = audio_pause,
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.resume = audio_resume,
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.reset = reset,
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};
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