diff options
Diffstat (limited to '')
-rw-r--r-- | src/output/alsa_plugin.c | 277 |
1 files changed, 226 insertions, 51 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c index 818c83ca2..9177fabe4 100644 --- a/src/output/alsa_plugin.c +++ b/src/output/alsa_plugin.c @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2009 The Music Player Daemon Project + * Copyright (C) 2003-2010 The Music Player Daemon Project * http://www.musicpd.org * * This program is free software; you can redistribute it and/or modify @@ -17,7 +17,8 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../output_api.h" +#include "config.h" +#include "output_api.h" #include "mixer_list.h" #include <glib.h> @@ -69,6 +70,16 @@ struct alsa_data { /** the size of one audio frame */ size_t frame_size; + + /** + * The size of one period, in number of frames. + */ + snd_pcm_uframes_t period_frames; + + /** + * The number of frames written in the current period. + */ + snd_pcm_uframes_t period_position; }; /** @@ -172,15 +183,148 @@ alsa_test_default_device(void) } static snd_pcm_format_t -get_bitformat(const struct audio_format *af) +get_bitformat(enum sample_format sample_format) +{ + switch (sample_format) { + case SAMPLE_FORMAT_S8: + return SND_PCM_FORMAT_S8; + + case SAMPLE_FORMAT_S16: + return SND_PCM_FORMAT_S16; + + case SAMPLE_FORMAT_S24_P32: + return SND_PCM_FORMAT_S24; + + case SAMPLE_FORMAT_S24: + return G_BYTE_ORDER == G_BIG_ENDIAN + ? SND_PCM_FORMAT_S24_3BE + : SND_PCM_FORMAT_S24_3LE; + + case SAMPLE_FORMAT_S32: + return SND_PCM_FORMAT_S32; + + default: + return SND_PCM_FORMAT_UNKNOWN; + } +} + +static snd_pcm_format_t +byteswap_bitformat(snd_pcm_format_t fmt) +{ + switch(fmt) { + case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE; + case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE; + case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE; + case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE; + case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE; + + case SND_PCM_FORMAT_S24_3BE: + return SND_PCM_FORMAT_S24_3LE; + + case SND_PCM_FORMAT_S24_3LE: + return SND_PCM_FORMAT_S24_3BE; + + case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE; + default: return SND_PCM_FORMAT_UNKNOWN; + } +} + +/** + * Attempts to configure the specified sample format. + */ +static int +alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, + struct audio_format *audio_format, + enum sample_format sample_format) +{ + snd_pcm_format_t alsa_format = get_bitformat(sample_format); + if (alsa_format == SND_PCM_FORMAT_UNKNOWN) + return -EINVAL; + + int err = snd_pcm_hw_params_set_format(pcm, hwparams, alsa_format); + if (err == 0) + audio_format->format = sample_format; + + return err; +} + +/** + * Attempts to configure the specified sample format with reversed + * host byte order. + */ +static int +alsa_output_try_reverse(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, + struct audio_format *audio_format, + enum sample_format sample_format) +{ + snd_pcm_format_t alsa_format = + byteswap_bitformat(get_bitformat(sample_format)); + if (alsa_format == SND_PCM_FORMAT_UNKNOWN) + return -EINVAL; + + int err = snd_pcm_hw_params_set_format(pcm, hwparams, alsa_format); + if (err == 0) { + audio_format->format = sample_format; + audio_format->reverse_endian = true; + } + + return err; +} + +/** + * Attempts to configure the specified sample format, and tries the + * reversed host byte order if was not supported. + */ +static int +alsa_output_try_format_both(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, + struct audio_format *audio_format, + enum sample_format sample_format) +{ + int err = alsa_output_try_format(pcm, hwparams, audio_format, + sample_format); + if (err == -EINVAL) + err = alsa_output_try_reverse(pcm, hwparams, audio_format, + sample_format); + + return err; +} + +/** + * Configure a sample format, and probe other formats if that fails. + */ +static int +alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams, + struct audio_format *audio_format) { - switch (af->bits) { - case 8: return SND_PCM_FORMAT_S8; - case 16: return SND_PCM_FORMAT_S16; - case 24: return SND_PCM_FORMAT_S24; - case 32: return SND_PCM_FORMAT_S32; + /* try the input format first */ + + int err = alsa_output_try_format_both(pcm, hwparams, audio_format, + audio_format->format); + if (err != -EINVAL) + return err; + + /* if unsupported by the hardware, try other formats */ + + static const enum sample_format probe_formats[] = { + SAMPLE_FORMAT_S24_P32, + SAMPLE_FORMAT_S32, + SAMPLE_FORMAT_S24, + SAMPLE_FORMAT_S16, + SAMPLE_FORMAT_S8, + SAMPLE_FORMAT_UNDEFINED, + }; + + for (unsigned i = 0; probe_formats[i] != SAMPLE_FORMAT_UNDEFINED; ++i) { + if (probe_formats[i] == audio_format->format) + continue; + + err = alsa_output_try_format_both(pcm, hwparams, audio_format, + probe_formats[i]); + if (err != -EINVAL) + return err; } - return SND_PCM_FORMAT_UNKNOWN; + + return -EINVAL; } /** @@ -189,7 +333,6 @@ get_bitformat(const struct audio_format *af) */ static bool alsa_setup(struct alsa_data *ad, struct audio_format *audio_format, - snd_pcm_format_t bitformat, GError **error) { snd_pcm_hw_params_t *hwparams; @@ -208,7 +351,6 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format, configure_hw: /* configure HW params */ snd_pcm_hw_params_alloca(&hwparams); - cmd = "snd_pcm_hw_params_any"; err = snd_pcm_hw_params_any(ad->pcm, hwparams); if (err < 0) @@ -235,31 +377,12 @@ configure_hw: ad->writei = snd_pcm_writei; } - err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat); - if (err == -EINVAL && (audio_format->bits == 24 || - audio_format->bits == 16)) { - /* fall back to 32 bit, let pcm_convert.c do the conversion */ - err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, - SND_PCM_FORMAT_S32); - if (err == 0) - audio_format->bits = 32; - } - - if (err == -EINVAL && audio_format->bits != 16) { - /* fall back to 16 bit, let pcm_convert.c do the conversion */ - err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, - SND_PCM_FORMAT_S16); - if (err == 0) { - g_debug("ALSA device \"%s\": converting %u bit to 16 bit\n", - alsa_device(ad), audio_format->bits); - audio_format->bits = 16; - } - } - + err = alsa_output_setup_format(ad->pcm, hwparams, audio_format); if (err < 0) { g_set_error(error, alsa_output_quark(), err, - "ALSA device \"%s\" does not support %u bit audio: %s", - alsa_device(ad), audio_format->bits, + "ALSA device \"%s\" does not support format %s: %s", + alsa_device(ad), + sample_format_to_string(audio_format->format), snd_strerror(-err)); return false; } @@ -285,6 +408,26 @@ configure_hw: } audio_format->sample_rate = sample_rate; + snd_pcm_uframes_t buffer_size_min, buffer_size_max; + snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min); + snd_pcm_hw_params_get_buffer_size_max(hwparams, &buffer_size_max); + unsigned buffer_time_min, buffer_time_max; + snd_pcm_hw_params_get_buffer_time_min(hwparams, &buffer_time_min, 0); + snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time_max, 0); + g_debug("buffer: size=%u..%u time=%u..%u", + (unsigned)buffer_size_min, (unsigned)buffer_size_max, + buffer_time_min, buffer_time_max); + + snd_pcm_uframes_t period_size_min, period_size_max; + snd_pcm_hw_params_get_period_size_min(hwparams, &period_size_min, 0); + snd_pcm_hw_params_get_period_size_max(hwparams, &period_size_max, 0); + unsigned period_time_min, period_time_max; + snd_pcm_hw_params_get_period_time_min(hwparams, &period_time_min, 0); + snd_pcm_hw_params_get_period_time_max(hwparams, &period_time_max, 0); + g_debug("period: size=%u..%u time=%u..%u", + (unsigned)period_size_min, (unsigned)period_size_max, + period_time_min, period_time_max); + if (ad->buffer_time > 0) { buffer_time = ad->buffer_time; cmd = "snd_pcm_hw_params_set_buffer_time_near"; @@ -365,6 +508,9 @@ configure_hw: g_debug("buffer_size=%u period_size=%u", (unsigned)alsa_buffer_size, (unsigned)alsa_period_size); + ad->period_frames = alsa_period_size; + ad->period_position = 0; + return true; error: @@ -378,19 +524,9 @@ static bool alsa_open(void *data, struct audio_format *audio_format, GError **error) { struct alsa_data *ad = data; - snd_pcm_format_t bitformat; int err; bool success; - bitformat = get_bitformat(audio_format); - if (bitformat == SND_PCM_FORMAT_UNKNOWN) { - /* sample format is not supported by this plugin - - fall back to 16 bit samples */ - - audio_format->bits = 16; - bitformat = SND_PCM_FORMAT_S16; - } - err = snd_pcm_open(&ad->pcm, alsa_device(ad), SND_PCM_STREAM_PLAYBACK, ad->mode); if (err < 0) { @@ -400,7 +536,7 @@ alsa_open(void *data, struct audio_format *audio_format, GError **error) return false; } - success = alsa_setup(ad, audio_format, bitformat, error); + success = alsa_setup(ad, audio_format, error); if (!success) { snd_pcm_close(ad->pcm); return false; @@ -431,6 +567,7 @@ alsa_recover(struct alsa_data *ad, int err) /* fall-through to snd_pcm_prepare: */ case SND_PCM_STATE_SETUP: case SND_PCM_STATE_XRUN: + ad->period_position = 0; err = snd_pcm_prepare(ad->pcm); break; case SND_PCM_STATE_DISCONNECTED: @@ -448,11 +585,47 @@ alsa_recover(struct alsa_data *ad, int err) } static void +alsa_drain(void *data) +{ + struct alsa_data *ad = data; + + if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING) + return; + + if (ad->period_position > 0) { + /* generate some silence to finish the partial + period */ + snd_pcm_uframes_t nframes = + ad->period_frames - ad->period_position; + size_t nbytes = nframes * ad->frame_size; + void *buffer = g_malloc(nbytes); + snd_pcm_hw_params_t *params; + snd_pcm_format_t format; + unsigned channels; + + snd_pcm_hw_params_alloca(¶ms); + snd_pcm_hw_params_current(ad->pcm, params); + snd_pcm_hw_params_get_format(params, &format); + snd_pcm_hw_params_get_channels(params, &channels); + + snd_pcm_format_set_silence(format, buffer, nframes * channels); + ad->writei(ad->pcm, buffer, nframes); + g_free(buffer); + } + + snd_pcm_drain(ad->pcm); + + ad->period_position = 0; +} + +static void alsa_cancel(void *data) { struct alsa_data *ad = data; - alsa_recover(ad, snd_pcm_drop(ad->pcm)); + ad->period_position = 0; + + snd_pcm_drop(ad->pcm); } static void @@ -460,9 +633,6 @@ alsa_close(void *data) { struct alsa_data *ad = data; - if (snd_pcm_state(ad->pcm) == SND_PCM_STATE_RUNNING) - snd_pcm_drain(ad->pcm); - snd_pcm_close(ad->pcm); } @@ -475,8 +645,11 @@ alsa_play(void *data, const void *chunk, size_t size, GError **error) while (true) { snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size); - if (ret > 0) + if (ret > 0) { + ad->period_position = (ad->period_position + ret) + % ad->period_frames; return ret * ad->frame_size; + } if (ret < 0 && ret != -EAGAIN && ret != -EINTR && alsa_recover(ad, ret) < 0) { @@ -494,7 +667,9 @@ const struct audio_output_plugin alsaPlugin = { .finish = alsa_finish, .open = alsa_open, .play = alsa_play, + .drain = alsa_drain, .cancel = alsa_cancel, .close = alsa_close, - .mixer_plugin = &alsa_mixer, + + .mixer_plugin = &alsa_mixer_plugin, }; |