diff options
Diffstat (limited to 'audio/out/ao.c')
| -rw-r--r-- | audio/out/ao.c | 141 |
1 files changed, 32 insertions, 109 deletions
diff --git a/audio/out/ao.c b/audio/out/ao.c index e7af02276f..4eb115194b 100644 --- a/audio/out/ao.c +++ b/audio/out/ao.c @@ -29,17 +29,19 @@ #include "audio/format.h" #include "options/options.h" -#include "options/m_config.h" +#include "options/m_config_frontend.h" #include "osdep/endian.h" #include "common/msg.h" #include "common/common.h" #include "common/global.h" extern const struct ao_driver audio_out_oss; +extern const struct ao_driver audio_out_audiotrack; extern const struct ao_driver audio_out_audiounit; extern const struct ao_driver audio_out_coreaudio; extern const struct ao_driver audio_out_coreaudio_exclusive; extern const struct ao_driver audio_out_rsound; +extern const struct ao_driver audio_out_pipewire; extern const struct ao_driver audio_out_sndio; extern const struct ao_driver audio_out_pulse; extern const struct ao_driver audio_out_jack; @@ -54,6 +56,9 @@ extern const struct ao_driver audio_out_sdl; static const struct ao_driver * const audio_out_drivers[] = { // native: +#if HAVE_ANDROID + &audio_out_audiotrack, +#endif #if HAVE_AUDIOUNIT &audio_out_audiounit, #endif @@ -82,9 +87,12 @@ static const struct ao_driver * const audio_out_drivers[] = { #if HAVE_OPENSLES &audio_out_opensles, #endif -#if HAVE_SDL2 +#if HAVE_SDL2_AUDIO &audio_out_sdl, #endif +#if HAVE_PIPEWIRE + &audio_out_pipewire, +#endif #if HAVE_SNDIO &audio_out_sndio, #endif @@ -94,9 +102,6 @@ static const struct ao_driver * const audio_out_drivers[] = { #endif &audio_out_pcm, &audio_out_lavc, -#if HAVE_RSOUND - &audio_out_rsound, -#endif NULL }; @@ -123,7 +128,6 @@ static bool get_desc(struct m_obj_desc *dst, int index) static const struct m_obj_list ao_obj_list = { .get_desc = get_desc, .description = "audio outputs", - .allow_unknown_entries = true, .allow_trailer = true, .disallow_positional_parameters = true, .use_global_options = true, @@ -132,11 +136,12 @@ static const struct m_obj_list ao_obj_list = { #define OPT_BASE_STRUCT struct ao_opts const struct m_sub_options ao_conf = { .opts = (const struct m_option[]) { - OPT_SETTINGSLIST("ao", audio_driver_list, 0, &ao_obj_list, ), - OPT_STRING("audio-device", audio_device, UPDATE_AUDIO), - OPT_STRING("audio-client-name", audio_client_name, UPDATE_AUDIO), - OPT_DOUBLE("audio-buffer", audio_buffer, M_OPT_MIN | M_OPT_MAX, - .min = 0, .max = 10), + {"ao", OPT_SETTINGSLIST(audio_driver_list, &ao_obj_list), + .flags = UPDATE_AUDIO}, + {"audio-device", OPT_STRING(audio_device), .flags = UPDATE_AUDIO}, + {"audio-client-name", OPT_STRING(audio_client_name), .flags = UPDATE_AUDIO}, + {"audio-buffer", OPT_DOUBLE(audio_buffer), + .flags = UPDATE_AUDIO, M_RANGE(0, 10)}, {0} }, .size = sizeof(OPT_BASE_STRUCT), @@ -206,14 +211,9 @@ static struct ao *ao_init(bool probing, struct mpv_global *global, af_fmt_to_str(ao->format)); ao->device = talloc_strdup(ao, dev); - - ao->api = ao->driver->play ? &ao_api_push : &ao_api_pull; - ao->api_priv = talloc_zero_size(ao, ao->api->priv_size); - assert(!ao->api->priv_defaults && !ao->api->options); - ao->stream_silence = flags & AO_INIT_STREAM_SILENCE; - ao->period_size = 1; + init_buffer_pre(ao); int r = ao->driver->init(ao); if (r < 0) { @@ -222,18 +222,14 @@ static struct ao *ao_init(bool probing, struct mpv_global *global, char redirect[80], rdevice[80]; snprintf(redirect, sizeof(redirect), "%s", ao->redirect); snprintf(rdevice, sizeof(rdevice), "%s", ao->device ? ao->device : ""); - talloc_free(ao); + ao_uninit(ao); return ao_init(probing, global, wakeup_cb, wakeup_ctx, encode_lavc_ctx, flags, samplerate, format, channels, rdevice, redirect); } goto fail; } - - if (ao->period_size < 1) { - MP_ERR(ao, "Invalid period size set.\n"); - goto fail; - } + ao->driver_initialized = true; ao->sstride = af_fmt_to_bytes(ao->format); ao->num_planes = 1; @@ -244,8 +240,10 @@ static struct ao *ao_init(bool probing, struct mpv_global *global, } ao->bps = ao->samplerate * ao->sstride; - if (!ao->device_buffer && ao->driver->get_space) - ao->device_buffer = ao->driver->get_space(ao); + if (ao->device_buffer <= 0 && ao->driver->write) { + MP_ERR(ao, "Device buffer size not set.\n"); + goto fail; + } if (ao->device_buffer) MP_VERBOSE(ao, "device buffer: %d samples.\n", ao->device_buffer); ao->buffer = MPMAX(ao->device_buffer, ao->def_buffer * ao->samplerate); @@ -255,12 +253,12 @@ static struct ao *ao_init(bool probing, struct mpv_global *global, ao->buffer = (ao->buffer + align - 1) / align * align; MP_VERBOSE(ao, "using soft-buffer of %d samples.\n", ao->buffer); - if (ao->api->init(ao) < 0) + if (!init_buffer_post(ao)) goto fail; return ao; fail: - talloc_free(ao); + ao_uninit(ao); return NULL; } @@ -354,95 +352,20 @@ struct ao *ao_init_best(struct mpv_global *global, return ao; } -// Uninitialize and destroy the AO. Remaining audio must be dropped. -void ao_uninit(struct ao *ao) -{ - if (ao) - ao->api->uninit(ao); - talloc_free(ao); -} - -// Queue the given audio data. Start playback if it hasn't started yet. Return -// the number of samples that was accepted (the core will try to queue the rest -// again later). Should never block. -// data: start pointer for each plane. If the audio data is packed, only -// data[0] is valid, otherwise there is a plane for each channel. -// samples: size of the audio data (see ao->sstride) -// flags: currently AOPLAY_FINAL_CHUNK can be set -int ao_play(struct ao *ao, void **data, int samples, int flags) -{ - return ao->api->play(ao, data, samples, flags); -} - -int ao_control(struct ao *ao, enum aocontrol cmd, void *arg) -{ - return ao->api->control ? ao->api->control(ao, cmd, arg) : CONTROL_UNKNOWN; -} - -// Return size of the buffered data in seconds. Can include the device latency. -// Basically, this returns how much data there is still to play, and how long -// it takes until the last sample in the buffer reaches the speakers. This is -// used for audio/video synchronization, so it's very important to implement -// this correctly. -double ao_get_delay(struct ao *ao) -{ - return ao->api->get_delay(ao); -} - -// Return free size of the internal audio buffer. This controls how much audio -// the core should decode and try to queue with ao_play(). -int ao_get_space(struct ao *ao) -{ - return ao->api->get_space(ao); -} - -// Stop playback and empty buffers. Essentially go back to the state after -// ao->init(). -void ao_reset(struct ao *ao) -{ - if (ao->api->reset) - ao->api->reset(ao); -} - -// Pause playback. Keep the current buffer. ao_get_delay() must return the -// same value as before pausing. -void ao_pause(struct ao *ao) -{ - if (ao->api->pause) - ao->api->pause(ao); -} - -// Resume playback. Play the remaining buffer. If the driver doesn't support -// pausing, it has to work around this and e.g. use ao_play_silence() to fill -// the lost audio. -void ao_resume(struct ao *ao) -{ - if (ao->api->resume) - ao->api->resume(ao); -} - -// Block until the current audio buffer has played completely. -void ao_drain(struct ao *ao) -{ - if (ao->api->drain) - ao->api->drain(ao); -} - -bool ao_eof_reached(struct ao *ao) -{ - return ao->api->get_eof ? ao->api->get_eof(ao) : true; -} - // Query the AO_EVENT_*s as requested by the events parameter, and return them. int ao_query_and_reset_events(struct ao *ao, int events) { return atomic_fetch_and(&ao->events_, ~(unsigned)events) & events; } -void ao_add_events(struct ao *ao, int events) +// Returns events that were set by this calls. +int ao_add_events(struct ao *ao, int events) { - atomic_fetch_or(&ao->events_, events); - ao->wakeup_cb(ao->wakeup_ctx); + unsigned prev_events = atomic_fetch_or(&ao->events_, events); + unsigned new = events & ~prev_events; + if (new) + ao->wakeup_cb(ao->wakeup_ctx); + return new; } // Request that the player core destroys and recreates the AO. Fully thread-safe. |