From 29a25b9933b32800f58dd73d5d1fc21993071c92 Mon Sep 17 00:00:00 2001 From: Avuton Olrich Date: Thu, 20 Jul 2006 16:02:40 +0000 Subject: Add mpd-indent.sh Indent the entire tree, hopefully we can keep it indented. git-svn-id: https://svn.musicpd.org/mpd/trunk@4410 09075e82-0dd4-0310-85a5-a0d7c8717e4f --- src/audioOutputs/audioOutput_alsa.c | 251 +++++++++++++++++++----------------- 1 file changed, 134 insertions(+), 117 deletions(-) (limited to 'src/audioOutputs/audioOutput_alsa.c') diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c index 5150dd502..c98d8b537 100644 --- a/src/audioOutputs/audioOutput_alsa.c +++ b/src/audioOutputs/audioOutput_alsa.c @@ -39,13 +39,13 @@ #include -typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer, - snd_pcm_uframes_t size); +typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer, + snd_pcm_uframes_t size); typedef struct _AlsaData { - char * device; - snd_pcm_t * pcmHandle; - alsa_writei_t * writei; + char *device; + snd_pcm_t *pcmHandle; + alsa_writei_t *writei; unsigned int buffer_time; unsigned int period_time; int sampleSize; @@ -54,8 +54,9 @@ typedef struct _AlsaData { int canResume; } AlsaData; -static AlsaData * newAlsaData(void) { - AlsaData * ret = malloc(sizeof(AlsaData)); +static AlsaData *newAlsaData(void) +{ + AlsaData *ret = malloc(sizeof(AlsaData)); ret->device = NULL; ret->pcmHandle = NULL; @@ -67,22 +68,25 @@ static AlsaData * newAlsaData(void) { return ret; } -static void freeAlsaData(AlsaData * ad) { - if(ad->device) free(ad->device); +static void freeAlsaData(AlsaData * ad) +{ + if (ad->device) + free(ad->device); free(ad); } -static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param) { - AlsaData * ad = newAlsaData(); +static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param) +{ + AlsaData *ad = newAlsaData(); if (param) { - BlockParam * bp = getBlockParam(param, "device"); + BlockParam *bp = getBlockParam(param, "device"); ad->device = bp ? strdup(bp->value) : strdup("default"); if ((bp = getBlockParam(param, "use_mmap")) && - (!strcasecmp(bp->value, "yes") || - !strcasecmp(bp->value, "true"))) + (!strcasecmp(bp->value, "yes") || + !strcasecmp(bp->value, "true"))) ad->useMmap = 1; if ((bp = getBlockParam(param, "buffer_time"))) ad->buffer_time = atoi(bp->value); @@ -94,45 +98,46 @@ static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param) { return 0; } -static void alsa_finishDriver(AudioOutput * audioOutput) { - AlsaData * ad = audioOutput->data; +static void alsa_finishDriver(AudioOutput * audioOutput) +{ + AlsaData *ad = audioOutput->data; freeAlsaData(ad); } static int alsa_testDefault(void) { - snd_pcm_t * handle; + snd_pcm_t *handle; - int ret = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, - SND_PCM_NONBLOCK); + int ret = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, + SND_PCM_NONBLOCK); snd_config_update_free_global(); - - if(ret) { + + if (ret) { WARNING("Error opening default alsa device: %s\n", - snd_strerror(-ret)); + snd_strerror(-ret)); return -1; - } - else snd_pcm_close(handle); + } else + snd_pcm_close(handle); return 0; } -static int alsa_openDevice(AudioOutput * audioOutput) +static int alsa_openDevice(AudioOutput * audioOutput) { - AlsaData * ad = audioOutput->data; - AudioFormat * audioFormat = &audioOutput->outAudioFormat; + AlsaData *ad = audioOutput->data; + AudioFormat *audioFormat = &audioOutput->outAudioFormat; snd_pcm_format_t bitformat; - snd_pcm_hw_params_t * hwparams; - snd_pcm_sw_params_t * swparams; + snd_pcm_hw_params_t *hwparams; + snd_pcm_sw_params_t *swparams; unsigned int sampleRate = audioFormat->sampleRate; unsigned int channels = audioFormat->channels; snd_pcm_uframes_t alsa_buffer_size; snd_pcm_uframes_t alsa_period_size; int err; - char * cmd = NULL; + char *cmd = NULL; - switch(audioFormat->bits) { + switch (audioFormat->bits) { case 8: bitformat = SND_PCM_FORMAT_S8; break; @@ -147,101 +152,107 @@ static int alsa_openDevice(AudioOutput * audioOutput) break; default: ERROR("Alsa device \"%s\" doesn't support %i bit audio\n", - ad->device, audioFormat->bits); + ad->device, audioFormat->bits); return -1; } - err = snd_pcm_open(&ad->pcmHandle, ad->device, - SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); + err = snd_pcm_open(&ad->pcmHandle, ad->device, + SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); snd_config_update_free_global(); - if(err < 0) { + if (err < 0) { ad->pcmHandle = NULL; goto error; } cmd = "snd_pcm_nonblock"; err = snd_pcm_nonblock(ad->pcmHandle, 0); - if(err < 0) goto error; + if (err < 0) + goto error; /* configure HW params */ snd_pcm_hw_params_alloca(&hwparams); cmd = "snd_pcm_hw_params_any"; err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams); - if(err < 0) goto error; + if (err < 0) + goto error; - if(ad->useMmap) { + if (ad->useMmap) { err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams, - SND_PCM_ACCESS_MMAP_INTERLEAVED); - if(err < 0) { + SND_PCM_ACCESS_MMAP_INTERLEAVED); + if (err < 0) { ERROR("Cannot set mmap'ed mode on alsa device \"%s\": " - " %s\n", ad->device, - snd_strerror(-err)); + " %s\n", ad->device, snd_strerror(-err)); ERROR("Falling back to direct write mode\n"); ad->useMmap = 0; - } - else ad->writei = snd_pcm_mmap_writei; + } else + ad->writei = snd_pcm_mmap_writei; } - if(!ad->useMmap) { + if (!ad->useMmap) { cmd = "snd_pcm_hw_params_set_access"; err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams, - SND_PCM_ACCESS_RW_INTERLEAVED); - if(err < 0) goto error; + SND_PCM_ACCESS_RW_INTERLEAVED); + if (err < 0) + goto error; ad->writei = snd_pcm_writei; } err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat); - if(err < 0) { + if (err < 0) { ERROR("Alsa device \"%s\" does not support %i bit audio: " - "%s\n", ad->device, (int)bitformat, - snd_strerror(-err)); + "%s\n", ad->device, (int)bitformat, snd_strerror(-err)); goto fail; } - err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams, - &channels); - if(err < 0) { + err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams, + &channels); + if (err < 0) { ERROR("Alsa device \"%s\" does not support %i channels: " - "%s\n", ad->device, (int)audioFormat->channels, - snd_strerror(-err)); + "%s\n", ad->device, (int)audioFormat->channels, + snd_strerror(-err)); goto fail; } audioFormat->channels = channels; - err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams, - &sampleRate, NULL); - if(err < 0 || sampleRate == 0) { + err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams, + &sampleRate, NULL); + if (err < 0 || sampleRate == 0) { ERROR("Alsa device \"%s\" does not support %i Hz audio\n", - ad->device, (int)audioFormat->sampleRate); + ad->device, (int)audioFormat->sampleRate); goto fail; } audioFormat->sampleRate = sampleRate; cmd = "snd_pcm_hw_params_set_buffer_time_near"; err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams, - &ad->buffer_time, NULL); - if(err < 0) goto error; + &ad->buffer_time, NULL); + if (err < 0) + goto error; if (!ad->period_time && sampleRate > 0) ad->period_time = 1000000 * MPD_ALSA_SAMPLE_XFER / sampleRate; cmd = "snd_pcm_hw_params_set_period_time_near"; err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams, - &ad->period_time, NULL); - if(err < 0) goto error; + &ad->period_time, NULL); + if (err < 0) + goto error; cmd = "snd_pcm_hw_params"; err = snd_pcm_hw_params(ad->pcmHandle, hwparams); - if(err < 0) goto error; + if (err < 0) + goto error; cmd = "snd_pcm_hw_params_get_buffer_size"; err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size); - if(err < 0) goto error; + if (err < 0) + goto error; cmd = "snd_pcm_hw_params_get_period_size"; err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, - NULL); - if(err < 0) goto error; + NULL); + if (err < 0) + goto error; ad->canPause = snd_pcm_hw_params_can_pause(hwparams); ad->canResume = snd_pcm_hw_params_can_resume(hwparams); @@ -251,68 +262,74 @@ static int alsa_openDevice(AudioOutput * audioOutput) cmd = "snd_pcm_sw_params_current"; err = snd_pcm_sw_params_current(ad->pcmHandle, swparams); - if(err < 0) goto error; + if (err < 0) + goto error; cmd = "snd_pcm_sw_params_set_start_threshold"; err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams, - alsa_buffer_size - alsa_period_size); - if(err < 0) goto error; + alsa_buffer_size - + alsa_period_size); + if (err < 0) + goto error; cmd = "snd_pcm_sw_params_set_avail_min"; - err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams, - alsa_period_size); - if(err < 0) goto error; + err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams, + alsa_period_size); + if (err < 0) + goto error; cmd = "snd_pcm_sw_params_set_xfer_align"; err = snd_pcm_sw_params_set_xfer_align(ad->pcmHandle, swparams, 1); - if(err < 0) goto error; + if (err < 0) + goto error; cmd = "snd_pcm_sw_params"; err = snd_pcm_sw_params(ad->pcmHandle, swparams); - if(err < 0) goto error; - - ad->sampleSize = (audioFormat->bits/8)*audioFormat->channels; + if (err < 0) + goto error; + + ad->sampleSize = (audioFormat->bits / 8) * audioFormat->channels; audioOutput->open = 1; DEBUG("alsa device \"%s\" will be playing %i bit, %i channel audio at " - "%i Hz\n", ad->device, (int)audioFormat->bits, - channels, sampleRate); + "%i Hz\n", ad->device, (int)audioFormat->bits, + channels, sampleRate); return 0; -error: - if(cmd) { - ERROR("Error opening alsa device \"%s\" (%s): %s\n", - ad->device, cmd, snd_strerror(-err)); - } - else { - ERROR("Error opening alsa device \"%s\": %s\n", ad->device, - snd_strerror(-err)); + error: + if (cmd) { + ERROR("Error opening alsa device \"%s\" (%s): %s\n", + ad->device, cmd, snd_strerror(-err)); + } else { + ERROR("Error opening alsa device \"%s\": %s\n", ad->device, + snd_strerror(-err)); } -fail: - if(ad->pcmHandle) snd_pcm_close(ad->pcmHandle); + fail: + if (ad->pcmHandle) + snd_pcm_close(ad->pcmHandle); ad->pcmHandle = NULL; audioOutput->open = 0; return -1; } -static int alsa_errorRecovery(AlsaData * ad, int err) { - if(err == -EPIPE) { +static int alsa_errorRecovery(AlsaData * ad, int err) +{ + if (err == -EPIPE) { DEBUG("Underrun on alsa device \"%s\"\n", ad->device); - } - else if(err == -ESTRPIPE) { + } else if (err == -ESTRPIPE) { DEBUG("alsa device \"%s\" was suspended\n", ad->device); } - switch(snd_pcm_state(ad->pcmHandle)) { + switch (snd_pcm_state(ad->pcmHandle)) { case SND_PCM_STATE_PAUSED: err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0); break; case SND_PCM_STATE_SUSPENDED: err = ad->canResume ? - snd_pcm_resume(ad->pcmHandle) : - snd_pcm_prepare(ad->pcmHandle); + snd_pcm_resume(ad->pcmHandle) : + snd_pcm_prepare(ad->pcmHandle); break; case SND_PCM_STATE_SETUP: case SND_PCM_STATE_XRUN: @@ -331,16 +348,18 @@ static int alsa_errorRecovery(AlsaData * ad, int err) { return err; } -static void alsa_dropBufferedAudio(AudioOutput * audioOutput) { - AlsaData * ad = audioOutput->data; +static void alsa_dropBufferedAudio(AudioOutput * audioOutput) +{ + AlsaData *ad = audioOutput->data; - alsa_errorRecovery( ad, snd_pcm_drop(ad->pcmHandle) ); + alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle)); } -static void alsa_closeDevice(AudioOutput * audioOutput) { - AlsaData * ad = audioOutput->data; +static void alsa_closeDevice(AudioOutput * audioOutput) +{ + AlsaData *ad = audioOutput->data; - if(ad->pcmHandle) { + if (ad->pcmHandle) { snd_pcm_drain(ad->pcmHandle); snd_pcm_close(ad->pcmHandle); ad->pcmHandle = NULL; @@ -349,10 +368,9 @@ static void alsa_closeDevice(AudioOutput * audioOutput) { audioOutput->open = 0; } -static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk, - int size) +static int alsa_playAudio(AudioOutput * audioOutput, char *playChunk, int size) { - AlsaData * ad = audioOutput->data; + AlsaData *ad = audioOutput->data; int ret; size /= ad->sampleSize; @@ -360,13 +378,14 @@ static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk, while (size > 0) { ret = ad->writei(ad->pcmHandle, playChunk, size); - if(ret == -EAGAIN || ret == -EINTR) continue; - - if(ret < 0) { - if( alsa_errorRecovery(ad, ret) < 0) { + if (ret == -EAGAIN || ret == -EINTR) + continue; + + if (ret < 0) { + if (alsa_errorRecovery(ad, ret) < 0) { ERROR("closing alsa device \"%s\" due to write " - "error: %s\n", ad->device, - snd_strerror(-errno)); + "error: %s\n", ad->device, + snd_strerror(-errno)); alsa_closeDevice(audioOutput); return -1; } @@ -380,8 +399,7 @@ static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk, return 0; } -AudioOutputPlugin alsaPlugin = -{ +AudioOutputPlugin alsaPlugin = { "alsa", alsa_testDefault, alsa_initDriver, @@ -390,11 +408,10 @@ AudioOutputPlugin alsaPlugin = alsa_playAudio, alsa_dropBufferedAudio, alsa_closeDevice, - NULL, /* sendMetadataFunc */ + NULL, /* sendMetadataFunc */ }; -#else /* HAVE ALSA */ +#else /* HAVE ALSA */ DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin) - -#endif /* HAVE_ALSA */ +#endif /* HAVE_ALSA */ -- cgit v1.2.3