merge: repair break in audio capture

diff --git a/cinelerra-5.1/cinelerra/audioalsa.C b/cinelerra-5.1/cinelerra/audioalsa.C
index adb6c8bb0b9d80def3eaf073477d9ea9e714f13e..ee5b65303fbb9a4e1e68013a933d444d7a9695d9 100644 (file)
--- a/cinelerra-5.1/cinelerra/audioalsa.C
+++ b/cinelerra-5.1/cinelerra/audioalsa.C
@@ -58,6 +58,7 @@ 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 +75,6 @@ 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 +93,6 @@ 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)
@@ -219,7 +216,10 @@ void AudioALSA::list_devices(ArrayList<char*> *devices, int pcm_title, int mode)
 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);
@@ -382,7 +382,7 @@ 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 +427,7 @@ 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,