X-Git-Url: http://git.cinelerra-gg.org/git/?p=goodguy%2Fhistory.git;a=blobdiff_plain;f=cinelerra-5.1%2Fcinelerra%2Faudioalsa.C;h=7a8d712c3b8f2687280dd33a9bccfa0925872c12;hp=adb6c8bb0b9d80def3eaf073477d9ea9e714f13e;hb=7e5a0760f40ff787cc3d93cb7768a901ebe52809;hpb=30bdb85eb33a8ee7ba675038a86c6be59c43d7bd

diff --git a/cinelerra-5.1/cinelerra/audioalsa.C b/cinelerra-5.1/cinelerra/audioalsa.C
index adb6c8bb..7a8d712c 100644
--- a/cinelerra-5.1/cinelerra/audioalsa.C
+++ b/cinelerra-5.1/cinelerra/audioalsa.C
@@ -22,6 +22,7 @@
 #include "audiodevice.h"
 #include "audioalsa.h"
 #include "bcsignals.h"
+#include "language.h"
 #include "mutex.h"
 #include "playbackconfig.h"
 #include "preferences.h"
@@ -58,6 +59,7 @@ public:
 // This is required in the top thread for Alsa to work
 	alsa_leaks() {
 		ArrayList<char*> *alsa_titles = new ArrayList<char*>;
+		alsa_titles->set_array_delete();
 		AudioALSA::list_devices(alsa_titles, 0, MODEPLAY);
 		alsa_titles->remove_all_objects();
 		delete alsa_titles;
@@ -74,9 +76,6 @@ void AudioALSA::list_devices(ArrayList<char*> *devices, int pcm_title, int mode)
 	char string[BCTEXTLEN];
 	snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
 
-	devices->set_array_delete();
-
-
 	switch(mode)
 	{
 		case MODERECORD:
@@ -95,7 +94,6 @@ void AudioALSA::list_devices(ArrayList<char*> *devices, int pcm_title, int mode)
 #define DEFAULT_DEVICE "default"
 	char *result = new char[strlen(DEFAULT_DEVICE) + 1];
 	devices->append(result);
-	devices->set_array_delete();     // since we are allocating by new[]
 	strcpy(result, DEFAULT_DEVICE);
 
 	while(snd_card_next(&card) >= 0)
@@ -159,6 +157,7 @@ void AudioALSA::list_devices(ArrayList<char*> *devices, int pcm_title, int mode)
 		snd_ctl_close(handle);
 	}
 
+#ifdef HAVE_PACTL
 // attempt to add pulseaudio "monitor" devices
 //  run: pactl list <sources>|<sinks>
 //   scan output for <Source/Sink> #n,  Name: <device>
@@ -214,12 +213,16 @@ void AudioALSA::list_devices(ArrayList<char*> *devices, int pcm_title, int mode)
 		}
 		pclose(pactl);
 	}
+#endif
 }
 
 void AudioALSA::translate_name(char *output, char *input, int mode)
 {
 	ArrayList<char*> titles;
+	titles.set_array_delete();
+
 	ArrayList<char*> pcm_titles;
+	pcm_titles.set_array_delete();
 
 	list_devices(&titles, 0, mode);
 	list_devices(&pcm_titles, 1, mode);
@@ -271,7 +274,8 @@ int AudioALSA::set_params(snd_pcm_t *dsp, int mode,
 	err = snd_pcm_hw_params_any(dsp, params);
 
 	if (err < 0) {
-		fprintf(stderr, "AudioALSA::set_params: no PCM configurations available\n");
+		fprintf(stderr, "AudioALSA::set_params: ");
+		fprintf(stderr, _("no PCM configurations available\n"));
 		return 1;
 	}
 
@@ -279,8 +283,8 @@ int AudioALSA::set_params(snd_pcm_t *dsp, int mode,
 		params,
 		SND_PCM_ACCESS_RW_INTERLEAVED);
         if(err) {
-		fprintf(stderr, "AudioALSA::set_params: failed to set up "
-				"interleaved device access.\n");
+		fprintf(stderr, "AudioALSA::set_params: ");
+		fprintf(stderr, _("failed to set up interleaved device access.\n"));
 		return 1;
         }
 
@@ -288,7 +292,8 @@ int AudioALSA::set_params(snd_pcm_t *dsp, int mode,
 		params,
 		translate_format(bits));
         if(err) {
-		fprintf(stderr, "AudioALSA::set_params: failed to set output format.\n");
+		fprintf(stderr, "AudioALSA::set_params: ");
+		fprintf(stderr, _("failed to set output format.\n"));
 		return 1;
         }
 
@@ -296,8 +301,8 @@ int AudioALSA::set_params(snd_pcm_t *dsp, int mode,
 		params,
 		channels);
         if(err) {
-		fprintf(stderr, "AudioALSA::set_params: Configured ALSA device "
-				"does not support %d channel operation.\n",
+		fprintf(stderr, "AudioALSA::set_params: ");
+		fprintf(stderr, _("Configured ALSA device does not support %d channel operation.\n"),
 			channels);
 		return 1;
         }
@@ -307,8 +312,8 @@ int AudioALSA::set_params(snd_pcm_t *dsp, int mode,
 		(unsigned int*)&samplerate,
 		(int*)0);
         if(err) {
-		fprintf(stderr, "AudioALSA::set_params: Configured ALSA device "
-				"does not support %u Hz playback.\n",
+		fprintf(stderr, "AudioALSA::set_params: ");
+		fprintf(stderr, _(" Configured ALSA device does not support %u Hz playback.\n"),
 			(unsigned int)samplerate);
 		return 1;
         }
@@ -382,7 +387,7 @@ int AudioALSA::open_input()
 	translate_name(pcm_name, device->in_config->alsa_in_device,MODERECORD);
 //printf("AudioALSA::open_input %s\n", pcm_name);
 
-	err = snd_pcm_open(&dsp_in, device->in_config->alsa_in_device, stream, open_mode);
+	err = snd_pcm_open(&dsp_in, pcm_name, stream, open_mode);
 
 	if(err < 0) {
 		dsp_in = 0;
@@ -427,7 +432,7 @@ int AudioALSA::open_output()
 		return 1;
 	}
 
-	set_params(dsp_out, MODEPLAY,
+	err = set_params(dsp_out, MODEPLAY,
 		device->get_ochannels(),
 		device->out_config->alsa_out_bits,
 		device->out_samplerate,
@@ -562,8 +567,8 @@ int AudioALSA::write_buffer(char *buffer, int size)
 		timer->update();
 
 	AudioThread *audio_out = device->audio_out;
-	while(attempts < 2 && !done && !device->playback_interrupted)
-	{
+	while( count > 0 && attempts < 2 && !done ) {
+		if( device->playback_interrupted ) break;
 // Buffers written must be equal to period_time
 		audio_out->Thread::enable_cancel();
 		int ret = snd_pcm_avail_update(get_output());
@@ -581,6 +586,7 @@ int AudioALSA::write_buffer(char *buffer, int size)
 			if( ret > 0 ) ret = 0;
 		}
 		audio_out->Thread::disable_cancel();
+		if( device->playback_interrupted ) break;
 		if( ret == 0 ) continue;
 
 		if( ret > 0 ) {