+++ /dev/null
-
-/*
- * CINELERRA
- * Copyright (C) 2008 Adam Williams <broadcast at earthling dot net>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
- */
-
-#include "audiodevice.h"
-#include "audioalsa.h"
-#include "bcsignals.h"
-#include "format.inc"
-#include "mutex.h"
-#include "playbackconfig.h"
-#include "preferences.h"
-#include "recordconfig.h"
-
-#include <errno.h>
-
-#ifdef HAVE_ALSA
-
-AudioALSA::AudioALSA(AudioDevice *device)
- : AudioLowLevel(device)
-{
- buffer_position = 0;
- samples_written = 0;
- timer = new Timer;
- delay = 0;
- period_size = 0;
- timer_lock = new Mutex("AudioALSA::timer_lock");
- interrupted = 0;
- dsp_in = 0;
- dsp_out = 0;
-}
-
-AudioALSA::~AudioALSA()
-{
- delete timer_lock;
- delete timer;
-}
-
-// leak checking
-static class alsa_leaks
-{
-public:
-// This is required in the top thread for Alsa to work
- alsa_leaks() {
- ArrayList<char*> *alsa_titles = new ArrayList<char*>;
- AudioALSA::list_devices(alsa_titles, 0, MODEPLAY);
- alsa_titles->remove_all_objects();
- delete alsa_titles;
- }
- ~alsa_leaks() { snd_config_update_free_global(); }
-} alsa_leak;
-
-void AudioALSA::list_devices(ArrayList<char*> *devices, int pcm_title, int mode)
-{
- snd_ctl_t *handle;
- int card, err, dev;
- snd_ctl_card_info_t *info;
- snd_pcm_info_t *pcminfo;
- char string[BCTEXTLEN];
- snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
-
- devices->set_array_delete();
-
-
- switch(mode)
- {
- case MODERECORD:
- stream = SND_PCM_STREAM_CAPTURE;
- break;
- case MODEPLAY:
- stream = SND_PCM_STREAM_PLAYBACK;
- break;
- }
-
-
- snd_ctl_card_info_alloca(&info);
- snd_pcm_info_alloca(&pcminfo);
-
- card = -1;
-#define DEFAULT_DEVICE "default"
- char *result = new char[strlen(DEFAULT_DEVICE) + 1];
- devices->append(result);
- strcpy(result, DEFAULT_DEVICE);
-
- while(snd_card_next(&card) >= 0)
- {
- char name[BCTEXTLEN];
- if(card < 0) break;
- sprintf(name, "hw:%i", card);
-
- if((err = snd_ctl_open(&handle, name, 0)) < 0)
- {
- printf("AudioALSA::list_devices card=%i: %s\n", card, snd_strerror(err));
- continue;
- }
-
- if((err = snd_ctl_card_info(handle, info)) < 0)
- {
- printf("AudioALSA::list_devices card=%i: %s\n", card, snd_strerror(err));
- snd_ctl_close(handle);
- continue;
- }
-
- dev = -1;
-
- while(1)
- {
- if(snd_ctl_pcm_next_device(handle, &dev) < 0)
- printf("AudioALSA::list_devices: snd_ctl_pcm_next_device\n");
-
- if (dev < 0)
- break;
-
- snd_pcm_info_set_device(pcminfo, dev);
- snd_pcm_info_set_subdevice(pcminfo, 0);
- snd_pcm_info_set_stream(pcminfo, stream);
-
- if((err = snd_ctl_pcm_info(handle, pcminfo)) < 0)
- {
- if(err != -ENOENT)
- printf("AudioALSA::list_devices card=%i: %s\n", card, snd_strerror(err));
- continue;
- }
-
- if(pcm_title)
- {
- sprintf(string, "plughw:%d,%d", card, dev);
-// strcpy(string, "cards.pcm.front");
- }
- else
- {
- sprintf(string, "%s #%d",
- snd_ctl_card_info_get_name(info),
- dev);
- }
-
- char *result = devices->append(new char[strlen(string) + 1]);
- strcpy(result, string);
- }
-
-
-
- snd_ctl_close(handle);
- }
-
-// attempt to add pulseaudio "monitor" devices
-// run: pactl list <sources>|<sinks>
-// scan output for <Source/Sink> #n, Name: <device>
-// build alsa device config and add to alsa snd_config
-
- const char *arg = 0;
- switch( mode ) {
- case MODERECORD:
- arg = "source";
- break;
- case MODEPLAY:
- arg = "sink";
- break;
- }
- FILE *pactl = 0;
- char line[BCTEXTLEN];
- if( arg ) {
- sprintf(line, "pactl list %ss", arg);
- pactl = popen(line,"r");
- }
- if( pactl ) {
- if( pcm_title ) snd_config_update();
- char name[BCTEXTLEN], pa_name[BCTEXTLEN], device[BCTEXTLEN];
- name[0] = pa_name[0] = device[0] = 0;
- int arg_len = strlen(arg);
- while( fgets(line, sizeof(line), pactl) ) {
- if( !strncasecmp(line, arg, arg_len) ) {
- char *sp = name, *id = pa_name;
- for( char *cp=line; *cp && *cp!='\n'; *sp++=*cp++ )
- *id++ = (*cp>='A' && *cp<='Z') ||
- (*cp>='a' && *cp<='z') ||
- (*cp>='0' && *cp<='9') ? *cp : '_';
- *sp++ = 0; *id = 0;
- if( !pcm_title )
- devices->append(strcpy(new char[sp-name], name));
- continue;
- }
- if( !pcm_title ) continue;
- if( sscanf(line, " Name: %s", device) != 1 ) continue;
- int len = strlen(pa_name);
- devices->append(strcpy(new char[len+1], pa_name));
- char alsa_config[BCTEXTLEN];
- len = snprintf(alsa_config, sizeof(alsa_config),
- "pcm.!%s {\n type pulse\n device %s\n}\n"
- "ctl.!%s {\n type pulse\n device %s\n}\n",
- pa_name, device, pa_name, device);
- FILE *fp = fmemopen(alsa_config,len,"r");
- snd_input_t *inp;
- snd_input_stdio_attach(&inp, fp, 1);
- snd_config_load(snd_config, inp);
- name[0] = pa_name[0] = device[0] = 0;
- snd_input_close(inp);
- }
- pclose(pactl);
- }
-}
-
-void AudioALSA::translate_name(char *output, char *input, int mode)
-{
- ArrayList<char*> titles;
- ArrayList<char*> pcm_titles;
-
- list_devices(&titles, 0, mode);
- list_devices(&pcm_titles, 1, mode);
-
- sprintf(output, "default");
- for(int i = 0; i < titles.total; i++)
- {
-//printf("AudioALSA::translate_name %s %s\n", titles.values[i], pcm_titles.values[i]);
- if(!strcasecmp(titles.values[i], input))
- {
- strcpy(output, pcm_titles.values[i]);
- break;
- }
- }
-
- titles.remove_all_objects();
- pcm_titles.remove_all_objects();
-}
-
-snd_pcm_format_t AudioALSA::translate_format(int format)
-{
- switch(format)
- {
- case 8:
- return SND_PCM_FORMAT_S8;
- break;
- case 16:
- return SND_PCM_FORMAT_S16_LE;
- break;
- case 24:
- return SND_PCM_FORMAT_S24_LE;
- break;
- case 32:
- return SND_PCM_FORMAT_S32_LE;
- break;
- }
- return SND_PCM_FORMAT_UNKNOWN;
-}
-
-void AudioALSA::set_params(snd_pcm_t *dsp, int mode,
- int channels, int bits, int samplerate, int samples)
-{
- snd_pcm_hw_params_t *params;
- snd_pcm_sw_params_t *swparams;
- int err;
-
- snd_pcm_hw_params_alloca(¶ms);
- snd_pcm_sw_params_alloca(&swparams);
- err = snd_pcm_hw_params_any(dsp, params);
-
- if (err < 0)
- {
- printf("AudioALSA::set_params: no PCM configurations available\n");
- return;
- }
-
- snd_pcm_hw_params_set_access(dsp,
- params,
- SND_PCM_ACCESS_RW_INTERLEAVED);
- snd_pcm_hw_params_set_format(dsp,
- params,
- translate_format(bits));
- snd_pcm_hw_params_set_channels(dsp,
- params,
- channels);
- snd_pcm_hw_params_set_rate_near(dsp,
- params,
- (unsigned int*)&samplerate,
- (int*)0);
-
-// Buffers written must be equal to period_time
- int buffer_time = 0;
- int period_time = (int)(1000000 * (double)samples / samplerate);
- switch( mode ) {
- case MODERECORD:
- buffer_time = 10000000;
- break;
- case MODEPLAY:
- buffer_time = 2 * period_time;
- break;
- }
-
-//printf("AudioALSA::set_params 1 %d %d %d\n", samples, buffer_time, period_time);
- snd_pcm_hw_params_set_buffer_time_near(dsp,
- params,
- (unsigned int*)&buffer_time,
- (int*)0);
- snd_pcm_hw_params_set_period_time_near(dsp,
- params,
- (unsigned int*)&period_time,
- (int*)0);
-//printf("AudioALSA::set_params 5 %d %d\n", buffer_time, period_time);
- err = snd_pcm_hw_params(dsp, params);
- if(err < 0) {
- printf("AudioALSA::set_params: hw_params failed\n");
- return;
- }
-
- snd_pcm_uframes_t chunk_size = 1024;
- snd_pcm_uframes_t buffer_size = 262144;
- snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
- snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
-//printf("AudioALSA::set_params 10 %d %d\n", chunk_size, buffer_size);
-
- snd_pcm_sw_params_current(dsp, swparams);
- //snd_pcm_uframes_t xfer_align = 1;
- //snd_pcm_sw_params_get_xfer_align(swparams, &xfer_align);
- //unsigned int sleep_min = 0;
- //err = snd_pcm_sw_params_set_sleep_min(dsp, swparams, sleep_min);
- period_size = chunk_size;
- err = snd_pcm_sw_params_set_avail_min(dsp, swparams, period_size);
- //err = snd_pcm_sw_params_set_xfer_align(dsp, swparams, xfer_align);
- if(snd_pcm_sw_params(dsp, swparams) < 0) {
- printf("AudioALSA::set_params: snd_pcm_sw_params failed\n");
- }
-
- device->device_buffer = samples * bits / 8 * channels;
- period_size /= 2;
-//printf("AudioALSA::set_params 100 %d %d\n", samples, device->device_buffer);
-
-// snd_pcm_hw_params_free(params);
-// snd_pcm_sw_params_free(swparams);
-}
-
-int AudioALSA::open_input()
-{
- char pcm_name[BCTEXTLEN];
- snd_pcm_stream_t stream = SND_PCM_STREAM_CAPTURE;
- int open_mode = 0;
- int err;
-
- device->in_channels = device->get_ichannels();
- device->in_bits = device->in_config->alsa_in_bits;
-
- 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, pcm_name, stream, open_mode);
-
- if(err < 0)
- {
- printf("AudioALSA::open_input: %s\n", snd_strerror(err));
- return 1;
- }
-
- set_params(dsp_in, MODERECORD,
- device->get_ichannels(),
- device->in_config->alsa_in_bits,
- device->in_samplerate,
- device->in_samples);
-
- return 0;
-}
-
-int AudioALSA::open_output()
-{
- char pcm_name[BCTEXTLEN];
- snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
- int open_mode = SND_PCM_NONBLOCK;
- int err;
-
- device->out_channels = device->get_ochannels();
- device->out_bits = device->out_config->alsa_out_bits;
-
-//printf("AudioALSA::open_output out_device %s\n", device->out_config->alsa_out_device);
- translate_name(pcm_name, device->out_config->alsa_out_device,MODEPLAY);
-//printf("AudioALSA::open_output pcm_name %s\n", pcm_name);
-
- err = snd_pcm_open(&dsp_out, pcm_name, stream, open_mode);
-
- if(err < 0)
- {
- dsp_out = 0;
- printf("AudioALSA::open_output %s: %s\n", pcm_name, snd_strerror(err));
- return 1;
- }
-
- set_params(dsp_out, MODEPLAY,
- device->get_ochannels(),
- device->out_config->alsa_out_bits,
- device->out_samplerate,
- device->out_samples);
- timer->update();
- return 0;
-}
-
-int AudioALSA::stop_output()
-{
-//printf("AudioALSA::stop_output\n");
- if(!device->out_config->interrupt_workaround)
- {
- if( get_output() )
- snd_pcm_drop(get_output());
- }
- else
- flush_device();
- return 0;
-}
-
-int AudioALSA::close_output()
-{
-//printf("AudioALSA::close_output\n");
- if(device->w && dsp_out) {
- stop_output();
- snd_pcm_close(dsp_out);
- dsp_out = 0;
- }
- return 0;
-}
-
-int AudioALSA::close_input()
-{
-//printf("AudioALSA::close_input\n");
- if(device->r && dsp_in) {
-// snd_pcm_reset(dsp_in);
- snd_pcm_drop(dsp_in);
- snd_pcm_drain(dsp_in);
- snd_pcm_close(dsp_in);
- dsp_in = 0;
- }
- return 0;
-}
-
-int AudioALSA::close_all()
-{
-//printf("AudioALSA::close_all\n");
- close_input();
- close_output();
- buffer_position = 0;
- samples_written = 0;
- delay = 0;
- interrupted = 0;
- return 0;
-}
-
-// Undocumented
-int64_t AudioALSA::device_position()
-{
- timer_lock->lock("AudioALSA::device_position");
- int64_t delta = timer->get_scaled_difference(device->out_samplerate);
- int64_t result = buffer_position - delay + delta;
-//printf("AudioALSA::device_position 1 w=" _LD " dt=" _LD " dly=%d pos=" _LD "\n",
-// buffer_position, delta, delay, result);
- timer_lock->unlock();
- return result;
-}
-
-int AudioALSA::read_buffer(char *buffer, int size)
-{
-//printf("AudioALSA::read_buffer 1\n");
- int attempts = 0;
- int done = 0;
- int frame_size = (device->in_bits / 8) * device->get_ichannels();
- int result = 0;
-
- if(!get_input())
- {
- sleep(1);
- return 0;
- }
-
- while(attempts < 1 && !done)
- {
- snd_pcm_uframes_t frames = size / frame_size;
- result = snd_pcm_readi(get_input(), buffer, frames);
- if( result < 0)
- {
- printf("AudioALSA::read_buffer overrun at sample " _LD "\n",
- device->total_samples_read);
-// snd_pcm_resume(get_input());
- close_input(); open_input();
- attempts++;
- }
- else
- done = 1;
-//printf("AudioALSA::read_buffer %d result=%d done=%d\n", __LINE__, result, done);
- }
- return 0;
-}
-
-int AudioALSA::write_buffer(char *buffer, int size)
-{
-//printf("AudioALSA::write_buffer %d\n",size);
-// Don't give up and drop the buffer on the first error.
- int attempts = 0;
- int done = 0;
- int sample_size = (device->out_bits / 8) * device->get_ochannels();
- int samples = size / sample_size;
-//printf("AudioALSA::write_buffer %d\n",samples);
- int count = samples;
- snd_pcm_sframes_t delay = 0;
-
-// static FILE *debug_fd = 0;
-// if(!debug_fd)
-// {
-// debug_fd = fopen("/tmp/debug.pcm", "w");
-// }
-// fwrite(buffer, size, 1, debug_fd);
-// fflush(debug_fd);
-
-
- if(!get_output()) return 0;
- if( buffer_position == 0 )
- timer->update();
-
- AudioThread *audio_out = device->audio_out;
- while(attempts < 2 && !done && !device->playback_interrupted)
- {
-// Buffers written must be equal to period_time
- audio_out->Thread::enable_cancel();
- int ret = snd_pcm_avail_update(get_output());
- if( ret >= period_size ) {
- if( ret > count ) ret = count;
-//FILE *alsa_fp = 0;
-//if( !alsa_fp ) alsa_fp = fopen("/tmp/alsa.raw","w");
-//if( alsa_fp ) fwrite(buffer, sample_size, ret, alsa_fp);
-//printf("AudioALSA::snd_pcm_writei start %d\n",count);
- ret = snd_pcm_writei(get_output(),buffer,ret);
-//printf("AudioALSA::snd_pcm_writei done %d\n", ret);
- }
- else if( ret >= 0 || ret == -EAGAIN ) {
- ret = snd_pcm_wait(get_output(),15);
- if( ret > 0 ) ret = 0;
- }
- audio_out->Thread::disable_cancel();
- if( ret == 0 ) continue;
-
- if( ret > 0 ) {
- samples_written += ret;
- if( (count-=ret) > 0 ) {
- buffer += ret * sample_size;
- attempts = 0;
- }
- else
- done = 1;
- }
- else {
- printf("AudioALSA::write_buffer err %d(%s) at sample " _LD "\n",
- ret, snd_strerror(ret), device->current_position());
- Timer::delay(50);
-// snd_pcm_resume(get_output());
- snd_pcm_recover(get_output(), ret, 1);
-// close_output(); open_output();
- attempts++;
- }
- }
-
- if( !interrupted && device->playback_interrupted )
- {
- interrupted = 1;
-//printf("AudioALSA::write_buffer interrupted\n");
- stop_output();
- }
-
- if(done)
- {
- timer_lock->lock("AudioALSA::write_buffer");
- snd_pcm_delay(get_output(), &delay);
- this->delay = delay;
- timer->update();
- buffer_position += samples;
-//printf("AudioALSA::write_buffer ** wrote %d, delay %d\n",samples,(int)delay);
- timer_lock->unlock();
- }
- return 0;
-}
-
-//this delay seems to prevent a problem where the sound system outputs
-//a lot of silence while waiting for the device drain to happen.
-int AudioALSA::output_wait()
-{
- snd_pcm_sframes_t delay = 0;
- snd_pcm_delay(get_output(), &delay);
- if( delay <= 0 ) return 0;
- int64_t udelay = 1e6 * delay / device->out_samplerate;
- // don't allow more than 10 seconds
- if( udelay > 10000000 ) udelay = 10000000;
- while( udelay > 0 && !device->playback_interrupted ) {
- int64_t usecs = udelay;
- if( usecs > 100000 ) usecs = 100000;
- usleep(usecs);
- udelay -= usecs;
- }
- if( device->playback_interrupted &&
- !device->out_config->interrupt_workaround )
- snd_pcm_drop(get_output());
- return 0;
-}
-
-int AudioALSA::flush_device()
-{
-//printf("AudioALSA::flush_device\n");
- if(get_output())
- {
- output_wait();
- //this causes the output to stutter.
- //snd_pcm_nonblock(get_output(), 0);
- snd_pcm_drain(get_output());
- //snd_pcm_nonblock(get_output(), 1);
- }
- return 0;
-}
-
-int AudioALSA::interrupt_playback()
-{
-//printf("AudioALSA::interrupt_playback *********\n");
-// if(get_output())
-// {
-// Interrupts the playback but may not have caused snd_pcm_writei to exit.
-// With some soundcards it causes snd_pcm_writei to freeze for a few seconds.
-// if(!device->out_config->interrupt_workaround)
-// snd_pcm_drop(get_output());
-
-// Makes sure the current buffer finishes before stopping.
-// snd_pcm_drain(get_output());
-
-// The only way to ensure snd_pcm_writei exits, but
-// got a lot of crashes when doing this.
-// device->Thread::cancel();
-// }
- return 0;
-}
-
-
-snd_pcm_t* AudioALSA::get_output()
-{
- return dsp_out;
-}
-
-snd_pcm_t* AudioALSA::get_input()
-{
- return dsp_in;
-}
-
-#endif
projects / goodguy / history.git / blobdiff