diff options
Diffstat (limited to 'stream/audio_in.c')
| -rw-r--r-- | stream/audio_in.c | 219 |
1 files changed, 219 insertions, 0 deletions
diff --git a/stream/audio_in.c b/stream/audio_in.c new file mode 100644 index 0000000000..03259614a4 --- /dev/null +++ b/stream/audio_in.c @@ -0,0 +1,219 @@ +#include <stdio.h> +#include <stdlib.h> +#include <unistd.h> + +#include "config.h" + +#include "audio_in.h" +#include "mp_msg.h" +#include "help_mp.h" +#include <string.h> +#include <errno.h> + +// sanitizes ai structure before calling other functions +int audio_in_init(audio_in_t *ai, int type) +{ + ai->type = type; + ai->setup = 0; + + ai->channels = -1; + ai->samplerate = -1; + ai->blocksize = -1; + ai->bytes_per_sample = -1; + ai->samplesize = -1; + + switch (ai->type) { +#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) + case AUDIO_IN_ALSA: + ai->alsa.handle = NULL; + ai->alsa.log = NULL; + ai->alsa.device = strdup("default"); + return 0; +#endif +#ifdef USE_OSS_AUDIO + case AUDIO_IN_OSS: + ai->oss.audio_fd = -1; + ai->oss.device = strdup("/dev/dsp"); + return 0; +#endif + default: + return -1; + } +} + +int audio_in_setup(audio_in_t *ai) +{ + + switch (ai->type) { +#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) + case AUDIO_IN_ALSA: + if (ai_alsa_init(ai) < 0) return -1; + ai->setup = 1; + return 0; +#endif +#ifdef USE_OSS_AUDIO + case AUDIO_IN_OSS: + if (ai_oss_init(ai) < 0) return -1; + ai->setup = 1; + return 0; +#endif + default: + return -1; + } +} + +int audio_in_set_samplerate(audio_in_t *ai, int rate) +{ + switch (ai->type) { +#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) + case AUDIO_IN_ALSA: + ai->req_samplerate = rate; + if (!ai->setup) return 0; + if (ai_alsa_setup(ai) < 0) return -1; + return ai->samplerate; +#endif +#ifdef USE_OSS_AUDIO + case AUDIO_IN_OSS: + ai->req_samplerate = rate; + if (!ai->setup) return 0; + if (ai_oss_set_samplerate(ai) < 0) return -1; + return ai->samplerate; +#endif + default: + return -1; + } +} + +int audio_in_set_channels(audio_in_t *ai, int channels) +{ + switch (ai->type) { +#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) + case AUDIO_IN_ALSA: + ai->req_channels = channels; + if (!ai->setup) return 0; + if (ai_alsa_setup(ai) < 0) return -1; + return ai->channels; +#endif +#ifdef USE_OSS_AUDIO + case AUDIO_IN_OSS: + ai->req_channels = channels; + if (!ai->setup) return 0; + if (ai_oss_set_channels(ai) < 0) return -1; + return ai->channels; +#endif + default: + return -1; + } +} + +int audio_in_set_device(audio_in_t *ai, char *device) +{ +#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) + int i; +#endif + if (ai->setup) return -1; + switch (ai->type) { +#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) + case AUDIO_IN_ALSA: + if (ai->alsa.device) free(ai->alsa.device); + ai->alsa.device = strdup(device); + /* mplayer cannot handle colons in arguments */ + for (i = 0; i < (int)strlen(ai->alsa.device); i++) { + if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; + } + return 0; +#endif +#ifdef USE_OSS_AUDIO + case AUDIO_IN_OSS: + if (ai->oss.device) free(ai->oss.device); + ai->oss.device = strdup(device); + return 0; +#endif + default: + return -1; + } +} + +int audio_in_uninit(audio_in_t *ai) +{ + if (ai->setup) { + switch (ai->type) { +#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) + case AUDIO_IN_ALSA: + if (ai->alsa.log) + snd_output_close(ai->alsa.log); + if (ai->alsa.handle) { + snd_pcm_close(ai->alsa.handle); + } + ai->setup = 0; + return 0; +#endif +#ifdef USE_OSS_AUDIO + case AUDIO_IN_OSS: + close(ai->oss.audio_fd); + ai->setup = 0; + return 0; +#endif + } + } + return -1; +} + +int audio_in_start_capture(audio_in_t *ai) +{ + switch (ai->type) { +#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) + case AUDIO_IN_ALSA: + return snd_pcm_start(ai->alsa.handle); +#endif +#ifdef USE_OSS_AUDIO + case AUDIO_IN_OSS: + return 0; +#endif + default: + return -1; + } +} + +int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) +{ + int ret; + + switch (ai->type) { +#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) + case AUDIO_IN_ALSA: + ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); + if (ret != ai->alsa.chunk_size) { + if (ret < 0) { + mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, snd_strerror(ret)); + if (ret == -EPIPE) { + if (ai_alsa_xrun(ai) == 0) { + mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_XRUNSomeFramesMayBeLeftOut); + } else { + mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrFatalCannotRecover); + } + } + } else { + mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); + } + return -1; + } + return ret; +#endif +#ifdef USE_OSS_AUDIO + case AUDIO_IN_OSS: + ret = read(ai->oss.audio_fd, buffer, ai->blocksize); + if (ret != ai->blocksize) { + if (ret < 0) { + mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno)); + } else { + mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); + } + return -1; + } + return ret; +#endif + default: + return -1; + } +} |