From 8004ae341f59dfe8c43bfa53d2b961bc98d1c673 Mon Sep 17 00:00:00 2001 From: Warren Dukes Date: Sat, 5 Mar 2005 20:51:36 +0000 Subject: more alsa work git-svn-id: https://svn.musicpd.org/mpd/trunk@3019 09075e82-0dd4-0310-85a5-a0d7c8717e4f --- src/audioOutputs/audioOutput_alsa.c | 98 +++++++++++++++++++++---------------- 1 file changed, 57 insertions(+), 41 deletions(-) (limited to 'src/audioOutputs') diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c index 6b50c05fc..2c3328d83 100644 --- a/src/audioOutputs/audioOutput_alsa.c +++ b/src/audioOutputs/audioOutput_alsa.c @@ -43,19 +43,21 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer, typedef struct _AlsaData { char * device; - snd_pcm_t * pcm_handle; - int mmap; + snd_pcm_t * pcmHandle; alsa_writei_t * writei; int sampleSize; + int useMmap; + int canPause; + int canResume; } AlsaData; static AlsaData * newAlsaData() { AlsaData * ret = malloc(sizeof(AlsaData)); ret->device = NULL; - ret->pcm_handle = NULL; + ret->pcmHandle = NULL; ret->writei = snd_pcm_writei; - ret->mmap = 0; + ret->useMmap = 0; return ret; } @@ -116,43 +118,43 @@ static int alsa_openDevice(AudioOutput * audioOutput) return -1; } - err = snd_pcm_open(&ad->pcm_handle, ad->device, + err = snd_pcm_open(&ad->pcmHandle, ad->device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); if(err < 0) { - ad->pcm_handle = NULL; + ad->pcmHandle = NULL; goto error; } - err = snd_pcm_nonblock(ad->pcm_handle, 0); + err = snd_pcm_nonblock(ad->pcmHandle, 0); if(err < 0) goto error; /* configure HW params */ snd_pcm_hw_params_alloca(&hwparams); - err = snd_pcm_hw_params_any(ad->pcm_handle, hwparams); + err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams); if(err < 0) goto error; - if(ad->mmap) { - err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams, + if(ad->useMmap) { + err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams, 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)); ERROR("Falling back to direct write mode\n"); - ad->mmap = 0; + ad->useMmap = 0; } else ad->writei = snd_pcm_mmap_writei; } - if(!ad->mmap) { - err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams, + if(!ad->useMmap) { + err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); if(err < 0) goto error; ad->writei = snd_pcm_writei; } - err = snd_pcm_hw_params_set_format(ad->pcm_handle, hwparams, bitformat); + err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat); if(err < 0) { ERROR("Alsa device \"%s\" does not support %i bit audio: " "%s\n", ad->device, (int)bitformat, @@ -160,7 +162,7 @@ static int alsa_openDevice(AudioOutput * audioOutput) goto fail; } - err = snd_pcm_hw_params_set_channels(ad->pcm_handle, hwparams, + err = snd_pcm_hw_params_set_channels(ad->pcmHandle, hwparams, audioFormat->channels); if(err < 0) { ERROR("Alsa device \"%s\" does not support %i channels: " @@ -169,7 +171,7 @@ static int alsa_openDevice(AudioOutput * audioOutput) goto fail; } - err = snd_pcm_hw_params_set_rate_near(ad->pcm_handle, hwparams, + err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams, &sampleRate, 0); if(err < 0 || sampleRate == 0) { ERROR("Alsa device \"%s\" does not support %i Hz audio\n", @@ -177,15 +179,15 @@ static int alsa_openDevice(AudioOutput * audioOutput) goto fail; } - err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm_handle, hwparams, + err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams, &alsa_buffer_time, 0); if(err < 0) goto error; - err = snd_pcm_hw_params_set_period_time_near(ad->pcm_handle, hwparams, + err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams, &alsa_period_time, 0); if(err < 0) goto error; - err = snd_pcm_hw_params(ad->pcm_handle, hwparams); + err = snd_pcm_hw_params(ad->pcmHandle, hwparams); if(err < 0) goto error; err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size); @@ -194,15 +196,18 @@ static int alsa_openDevice(AudioOutput * audioOutput) err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, 0); if(err < 0) goto error; + ad->canPause = snd_pcm_hw_params_can_pause(hwparams); + ad->canResume = snd_pcm_hw_params_can_resume(hwparams); + /* configure SW params */ snd_pcm_sw_params_alloca(&swparams); - snd_pcm_sw_params_current(ad->pcm_handle, swparams); + snd_pcm_sw_params_current(ad->pcmHandle, swparams); - err = snd_pcm_sw_params_set_start_threshold(ad->pcm_handle, swparams, + err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams, alsa_buffer_size - alsa_period_size); if(err < 0) goto error; - err = snd_pcm_sw_params(ad->pcm_handle, swparams); + err = snd_pcm_sw_params(ad->pcmHandle, swparams); if(err < 0) goto error; ad->sampleSize = (audioFormat->bits/8)*audioFormat->channels; @@ -215,8 +220,8 @@ error: ERROR("Error opening alsa device \"%s\": %s\n", ad->device, snd_strerror(-err)); fail: - if(ad->pcm_handle) snd_pcm_close(ad->pcm_handle); - ad->pcm_handle = NULL; + if(ad->pcmHandle) snd_pcm_close(ad->pcmHandle); + ad->pcmHandle = NULL; audioOutput->open = 0; return -1; } @@ -224,8 +229,8 @@ fail: static void alsa_dropBufferedAudio(AudioOutput * audioOutput) { AlsaData * ad = audioOutput->data; - snd_pcm_drop(ad->pcm_handle); - snd_pcm_prepare(ad->pcm_handle); + snd_pcm_drop(ad->pcmHandle); + snd_pcm_prepare(ad->pcmHandle); } inline static int alsa_errorRecovery(AlsaData * ad, int err) { @@ -236,12 +241,19 @@ inline static int alsa_errorRecovery(AlsaData * ad, int err) { DEBUG("alsa device \"%s\" was suspended\n", ad->device); } - switch(snd_pcm_state(ad->pcm_handle)) { + 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); + break; case SND_PCM_STATE_SETUP: case SND_PCM_STATE_XRUN: - err = snd_pcm_prepare(ad->pcm_handle); - if(err < 0) return -1; - return 0; + err = snd_pcm_prepare(ad->pcmHandle); + break; default: /* unknown state, do nothing */ break; @@ -253,10 +265,10 @@ inline static int alsa_errorRecovery(AlsaData * ad, int err) { static void alsa_closeDevice(AudioOutput * audioOutput) { AlsaData * ad = audioOutput->data; - if(ad->pcm_handle) { - snd_pcm_drain(ad->pcm_handle); - snd_pcm_close(ad->pcm_handle); - ad->pcm_handle = NULL; + if(ad->pcmHandle) { + snd_pcm_drain(ad->pcmHandle); + snd_pcm_close(ad->pcmHandle); + ad->pcmHandle = NULL; } audioOutput->open = 0; @@ -271,17 +283,21 @@ static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk, size /= ad->sampleSize; while (size > 0) { - ret = ad->writei(ad->pcm_handle, playChunk, size); + ret = ad->writei(ad->pcmHandle, playChunk, size); if(ret == -EAGAIN) continue; - if(ret < 0 && alsa_errorRecovery(ad, ret) < 0) { - ERROR("closing alsa device \"%s\" due to write error:" - " %s\n", ad->device, - snd_strerror(-errno)); - alsa_closeDevice(audioOutput); - return -1; + 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)); + alsa_closeDevice(audioOutput); + return -1; + } + continue; } + playChunk += ret * ad->sampleSize; size -= ret; } -- cgit v1.2.3