new v4l capture patch by Jindrich Makovicka <makovick@kmlinux.fjfi.cvut.cz>:

- multithreaded audio/video buffering (I know mplayer crew hates threads
  but it seems to me as the only way of doing reliable a/v capture)
- a/v timebase synchronization (sample count vs. gettimeofday)
- "immediate" mode support for mplayer
- fixed colorspace stuff - RGB?? and YUY2 modes now work as expected
- native ALSA audio capture
- separated audio input layer


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7061 b3059339-0415-0410-9bf9-f77b7e298cf2

1 files changed, 192 insertions, 0 deletions

diff --git a/libmpdemux/audio_in.c b/libmpdemux/audio_in.c
new file mode 100644
index 0000000000..c662bfab0f
--- /dev/null
+++ b/libmpdemux/audio_in.c
@@ -0,0 +1,192 @@
+#include "config.h"
+#include "audio_in.h"
+#include "mp_msg.h"
+#include <stdio.h>
+#include <stdlib.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) {
+#ifdef HAVE_ALSA9	  
+    case AUDIO_IN_ALSA:
+	ai->alsa.handle = NULL;
+	ai->alsa.log = NULL;
+	ai->alsa.device = strdup("default");
+	return 0;
+#endif
+    case AUDIO_IN_OSS:
+	ai->oss.audio_fd = -1;
+	ai->oss.device = strdup("/dev/dsp");
+	return 0;
+    default:
+	return -1;
+    }
+}
+
+int audio_in_setup(audio_in_t *ai)
+{
+    int err;
+    
+    switch (ai->type) {
+#ifdef HAVE_ALSA9	  
+    case AUDIO_IN_ALSA:
+	if (ai_alsa_init(ai) < 0) return -1;
+	ai->setup = 1;
+	return 0;
+#endif
+    case AUDIO_IN_OSS:
+	if (ai_oss_init(ai) < 0) return -1;
+	ai->setup = 1;
+	return 0;
+    default:
+	return -1;
+    }
+}
+
+int audio_in_set_samplerate(audio_in_t *ai, int rate)
+{
+    switch (ai->type) {
+#ifdef HAVE_ALSA9	  
+    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
+    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;
+    default:
+	return -1;
+    }
+}
+
+int audio_in_set_channels(audio_in_t *ai, int channels)
+{
+    switch (ai->type) {
+#ifdef HAVE_ALSA9	  
+    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
+    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;
+    default:
+	return -1;
+    }
+}
+
+int audio_in_set_device(audio_in_t *ai, char *device)
+{
+    int i;
+    if (ai->setup) return -1;
+    switch (ai->type) {
+#ifdef HAVE_ALSA9	  
+    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 < strlen(ai->alsa.device); i++) {
+	    if (ai->alsa.device[i] == ',') ai->alsa.device[i] = ':';
+	}
+	return 0;
+#endif
+    case AUDIO_IN_OSS:
+	if (ai->oss.device) free(ai->oss.device);
+	ai->oss.device = strdup(device);
+	return 0;
+    default:
+	return -1;
+    }
+}
+
+int audio_in_uninit(audio_in_t *ai)
+{
+    if (ai->setup) {
+	switch (ai->type) {
+#ifdef HAVE_ALSA9	  
+	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
+	case AUDIO_IN_OSS:
+	    close(ai->oss.audio_fd);
+	    ai->setup = 0;
+	    return 0;
+	default:
+	    return -1;
+	}
+    }
+}
+
+int audio_in_start_capture(audio_in_t *ai)
+{
+    switch (ai->type) {
+#ifdef HAVE_ALSA9	  
+    case AUDIO_IN_ALSA:
+	return snd_pcm_start(ai->alsa.handle);
+#endif
+    case AUDIO_IN_OSS:
+	return 0;
+    default:
+	return -1;
+    }
+}
+
+int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer)
+{
+    int ret;
+    
+    switch (ai->type) {
+#ifdef HAVE_ALSA9	  
+    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, "\nerror reading audio: %s\n", snd_strerror(ret));
+	    } else {
+		mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
+	    }
+	    return -1;
+	}
+	return ret;
+#endif
+    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, "\nerror reading audio: %s\n", strerror(errno));
+	    } else {
+		mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
+	    }
+	    return -1;
+	}
+	return ret;
+    default:
+	return -1;
+    }
+}