diff stream/audio_in.c @ 19271:64d82a45a05d

introduce new 'stream' directory for all stream layer related components and split them from libmpdemux

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/stream/audio_in.c	Mon Jul 31 17:39:17 2006 +0000
@@ -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;
+    }
+}