void* RunAudio()
{
while(true)
{
if (!paused)
{
memset(mixbuf, 0, mixlen);
NativeMix((short *)mixbuf, mixlen / 4);
fd_set rfds, wfds;
int nflds;
if (!paused && (FD_ISSET(snd_pcm_file_descriptor(pcm_handle, SND_PCM_CHANNEL_PLAYBACK), &wfds)))
{
snd_pcm_plugin_write(pcm_handle, mixbuf, mixlen);
}
FD_ZERO(&rfds);
FD_ZERO(&wfds);
FD_SET(snd_mixer_file_descriptor(mixer_handle), &rfds);
FD_SET(snd_pcm_file_descriptor(pcm_handle, SND_PCM_CHANNEL_PLAYBACK), &wfds);
nflds = std::max(snd_mixer_file_descriptor(mixer_handle), snd_pcm_file_descriptor(pcm_handle, SND_PCM_CHANNEL_PLAYBACK));
select (nflds+1, &rfds, &wfds, NULL, NULL);
}
else
delay((AUDIO_SAMPLES*1000)/AUDIO_FREQ);
}
}
static int audio_open(audiodevice_t *dev, chanfmt_t fmt) {
audio_alsa05_t *alsa = (audio_alsa05_t *)dev->data_pcm;
snd_pcm_channel_params_t p;
snd_pcm_channel_setup_t s;
if (0 > snd_pcm_open(&alsa->pcm_handle, alsa->card, alsa->pcm_dev, SND_PCM_OPEN_PLAYBACK)) {
WARNING("Opening audio device %d failed\n", alsa->pcm_dev);
goto _err_exit;
}
memset(&alsa->info, 0, sizeof(snd_pcm_channel_info_t));
if (0 > snd_pcm_channel_info(alsa->pcm_handle, &alsa->info)) {
WARNING("param get failed\n");
goto _err_exit;
}
memset(&p, 0, sizeof(p));
p.mode = SND_PCM_MODE_BLOCK;
p.start_mode = SND_PCM_START_DATA;
p.channel = SND_PCM_CHANNEL_PLAYBACK;
p.stop_mode = SND_PCM_STOP_STOP;
p.buf.block.frag_size = 1536;
p.buf.block.frags_max = 6;
p.buf.block.frags_min = 1;
p.format.rate = fmt.rate;
p.format.format = fmt.bit == 8 ? SND_PCM_SFMT_U8 : SND_PCM_SFMT_S16;
p.format.voices = fmt.ch;
p.format.interleave = 1;
alsa->silence = fmt.bit == 8 ? 0x80 : 0;
if (0 > snd_pcm_channel_params(alsa->pcm_handle, &p)) {
WARNING("Unable to set channel params\n");
goto _err_exit;
}
if (0 > snd_pcm_channel_prepare(alsa->pcm_handle, SND_PCM_CHANNEL_PLAYBACK)) {
WARNING("Unable to prepare channel\n");
goto _err_exit;
}
memset(&s, 0, sizeof(s));
s.mode = SND_PCM_MODE_BLOCK;
s.channel = SND_PCM_CHANNEL_PLAYBACK;
if (0 > snd_pcm_channel_setup(alsa->pcm_handle, &s)) {
WARNING("Unable to obtain setup\n");
goto _err_exit;
}
dev->buf.len = s.buf.block.frag_size;
dev->fd = snd_pcm_file_descriptor(alsa->pcm_handle, SND_PCM_CHANNEL_PLAYBACK);
return OK;
_err_exit:
dev->fd = -1;
return NG;
}
static ALCenum qsa_open_playback(ALCdevice* device, const ALCchar* deviceName)
{
qsa_data *data;
int card, dev;
int status;
data = (qsa_data*)calloc(1, sizeof(qsa_data));
if(data == NULL)
return ALC_OUT_OF_MEMORY;
if(!deviceName)
deviceName = qsaDevice;
if(strcmp(deviceName, qsaDevice) == 0)
status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_PLAYBACK);
else
{
const DevMap *iter;
if(VECTOR_SIZE(DeviceNameMap) == 0)
deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap);
#define MATCH_DEVNAME(iter) ((iter)->name && strcmp(deviceName, (iter)->name)==0)
VECTOR_FIND_IF(iter, const DevMap, DeviceNameMap, MATCH_DEVNAME);
#undef MATCH_DEVNAME
if(iter == VECTOR_ITER_END(DeviceNameMap))
{
free(data);
return ALC_INVALID_DEVICE;
}
status = snd_pcm_open(&data->pcmHandle, iter->card, iter->dev, SND_PCM_OPEN_PLAYBACK);
}
if(status < 0)
{
free(data);
return ALC_INVALID_DEVICE;
}
data->audio_fd = snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK);
if(data->audio_fd < 0)
{
snd_pcm_close(data->pcmHandle);
free(data);
return ALC_INVALID_DEVICE;
}
al_string_copy_cstr(&device->DeviceName, deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
}
static int voice_test()
{
int rtn = 0;
char* name;
fd_set rfds, wfds;
Audio_t hPcm_Mic;
Audio_t hPcm_Spk;
snd_pcm_channel_params_t params;
bool bQuit = false;
/************************ CAPTURE **************************/
// configuring capture (mic)
name = "voice";
// get audioman handle
rtn = audio_manager_get_handle(AUDIO_TYPE_VIDEO_CHAT, getpid(), (bool)false, &hPcm_Mic.hAudioman);
if(rtn < 0) {
cli_print("audio_manager_get_handle (mic) failed %s", strerror(-rtn));
return -1;
}
cli_print("Opening %s - for capture", name);
rtn = snd_pcm_open_name(&hPcm_Mic.pDs, name, SND_PCM_OPEN_CAPTURE);
if (rtn < 0) {
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
cli_print("snd_pcm_open_name (mic) failed %s", snd_strerror(rtn));
return -1; // snd_pcm calls return negative values; make positive
}
rtn = snd_pcm_set_audioman_handle(hPcm_Mic.pDs, hPcm_Mic.hAudioman);
if (rtn < 0) {
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
cli_print("snd_pcm_set_audioman_handle (mic) failed %s", snd_strerror(rtn));
(void)snd_pcm_close(hPcm_Mic.pDs);
return -1;
}
// disable mmap
(void)snd_pcm_plugin_set_disable(hPcm_Mic.pDs, (unsigned int)PLUGIN_DISABLE_MMAP);
// set parameters
memset(¶ms, 0, sizeof(params));
params.mode = SND_PCM_MODE_BLOCK;
params.channel = SND_PCM_CHANNEL_CAPTURE;
params.start_mode = SND_PCM_START_GO;
params.stop_mode = SND_PCM_STOP_ROLLOVER;
params.buf.block.frag_size = fragsize;
params.buf.block.frags_max = 1;
params.buf.block.frags_min = 1;
params.format.rate = 8000; //samplerate;
params.format.interleave = 1;
params.format.voices = channels;
params.format.format = SND_PCM_SFMT_S16_LE;
rtn = snd_pcm_plugin_params(hPcm_Mic.pDs, ¶ms);
if (rtn < 0) {
cli_print("snd_pcm_plugin_params (mic) failed %s", snd_strerror(rtn));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
return -1;
}
// get file descriptor for use with the select() call
hPcm_Mic.hFD = snd_pcm_file_descriptor(hPcm_Mic.pDs, SND_PCM_CHANNEL_CAPTURE);
if (hPcm_Mic.hFD < 0) {
cli_print("snd_pcm_file_descriptor (mic) failed %s", snd_strerror(hPcm_Mic.hFD));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
return -1;
}
// Signal the driver to ready the capture channel
rtn = snd_pcm_plugin_prepare(hPcm_Mic.pDs, SND_PCM_CHANNEL_CAPTURE);
if (rtn < 0) {
cli_print("snd_pcm_plugin_prepare (mic) failed %s", snd_strerror(errno));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
return -1;
}
fragsize = params.buf.block.frag_size;
/************************ PLAYBACK **************************/
name = "voice";
// get and set audioman handle
rtn = audio_manager_get_handle(AUDIO_TYPE_VIDEO_CHAT, getpid(), (bool)false, &hPcm_Spk.hAudioman);
if (rtn < 0) {
cli_print("audioman audio_manager_get_handle (spk) failed %s", strerror(-rtn) );
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
return -1;
}
#ifdef HANDSET
// set audio manager handle type
rtn = audio_manager_set_handle_type(hPcm_Spk.hAudioman, AUDIO_TYPE_VIDEO_CHAT, device_type, device_type);
if (rtn < 0) {
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
(void)audio_manager_free_handle(hPcm_Spk.hAudioman);
cli_print("audio_manager_set_handle_type (spk) failed %s", strerror(-rtn));
return -1;
}
#endif
// Create a handle and open a connection to an audio interface specified by name
cli_print("Opening %s - for playback", name);
rtn = snd_pcm_open_name(&hPcm_Spk.pDs, name, SND_PCM_OPEN_PLAYBACK);
if (rtn < 0) {
cli_print("snd_pcm_open_name (spk) failed %s", snd_strerror(rtn));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
(void)audio_manager_free_handle(hPcm_Spk.hAudioman);
return -1;
}
rtn = snd_pcm_set_audioman_handle(hPcm_Spk.pDs, hPcm_Spk.hAudioman);
if (rtn < 0) {
cli_print("snd_pcm_set_audioman_handle (spk) failed %s", snd_strerror(rtn));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)snd_pcm_close(hPcm_Spk.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
(void)audio_manager_free_handle(hPcm_Spk.hAudioman);
return -1;
}
// disable mmap
(void)snd_pcm_plugin_set_disable(hPcm_Spk.pDs, (unsigned int)PLUGIN_DISABLE_MMAP);
// set parameters
memset(¶ms, 0, sizeof(params));
params.mode = SND_PCM_MODE_BLOCK;
params.channel = SND_PCM_CHANNEL_PLAYBACK;
params.start_mode = SND_PCM_START_GO;
params.stop_mode = SND_PCM_STOP_ROLLOVER;
params.buf.block.frag_size = fragsize;
params.buf.block.frags_max = 1;
params.buf.block.frags_min = 1;
params.format.rate = samplerate;
params.format.interleave = 1;
params.format.voices = channels;
params.format.format = SND_PCM_SFMT_S16_LE;
// Set the configurable parameters for a PCM channel
rtn = snd_pcm_plugin_params(hPcm_Spk.pDs, ¶ms);
if (rtn < 0) {
cli_print("snd_pcm_plugin_params (spk) failed %s", snd_strerror(rtn));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)snd_pcm_close(hPcm_Spk.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
(void)audio_manager_free_handle(hPcm_Spk.hAudioman);
return -1;
}
// get file descriptor for use with the select() call
hPcm_Spk.hFD = snd_pcm_file_descriptor(hPcm_Spk.pDs, SND_PCM_CHANNEL_PLAYBACK);
if (hPcm_Spk.hFD < 0) {
cli_print("snd_pcm_file_descriptor (spk) failed %s", snd_strerror(hPcm_Spk.hFD));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)snd_pcm_close(hPcm_Spk.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
(void)audio_manager_free_handle(hPcm_Spk.hAudioman);
return -1;
}
// Signal the driver to ready the playback channel
rtn = snd_pcm_plugin_prepare(hPcm_Spk.pDs, SND_PCM_CHANNEL_PLAYBACK);
if (rtn < 0) {
cli_print("snd_pcm_plugin_prepare (spk) failed %s", snd_strerror(rtn));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)snd_pcm_close(hPcm_Spk.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
(void)audio_manager_free_handle(hPcm_Spk.hAudioman);
return -1;
}
fragsize = params.buf.block.frag_size;
rtn = snd_pcm_capture_go(hPcm_Mic.pDs);
if( rtn < 0) {
cli_print("snd_pcm_capture_go (mic) failed %s", snd_strerror(rtn));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)snd_pcm_close(hPcm_Spk.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
(void)audio_manager_free_handle(hPcm_Spk.hAudioman);
return -1;
}
rtn = snd_pcm_playback_go(hPcm_Spk.pDs);
if (rtn < 0) {
cli_print("snd_pcm_playback_go (spk) failed %s", snd_strerror(rtn));
(void)snd_pcm_close(hPcm_Mic.pDs);
(void)snd_pcm_close(hPcm_Spk.pDs);
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
(void)audio_manager_free_handle(hPcm_Spk.hAudioman);
return -1;
}
/******************* PLAYBACK/RECORD LOOP **************************/
while(!bQuit)
{
int width;
struct timeval timeout;
timeout.tv_sec = 0;
timeout.tv_usec = 350000; // 350 ms
FD_ZERO(&rfds);
FD_ZERO(&wfds);
FD_SET(hPcm_Mic.hFD, &rfds);
FD_SET(hPcm_Spk.hFD, &wfds);
width = ((hPcm_Spk.hFD > hPcm_Mic.hFD) ? hPcm_Spk.hFD : hPcm_Mic.hFD) + 1;
rtn = select(width, &rfds, &wfds, NULL, &timeout);
if (rtn > 0) {
if (FD_ISSET(hPcm_Spk.hFD, &wfds)) {
bQuit = process_write(hPcm_Spk.pDs);
}
if (FD_ISSET(hPcm_Mic.hFD, &rfds)) {
bQuit = process_read(hPcm_Mic.pDs);
}
}
else if (rtn == 0){
cli_print("select: timed out");
bQuit = true;
}
else { // (rtn < 0)
cli_print("select: %s", strerror(errno));
bQuit = true;
}
}
// Ensure audio processing is stopped
if ((rtn = snd_pcm_plugin_playback_drain(hPcm_Spk.pDs)) < 0) {
cli_print("snd_pcm_plugin_playback_drain (spk) failed %s", snd_strerror(rtn));
}
if ((rtn = snd_pcm_plugin_flush(hPcm_Mic.pDs, SND_PCM_CHANNEL_CAPTURE)) < 0) {
cli_print("snd_pcm_plugin_flush (mic) failed %s", snd_strerror(rtn));
}
if ((rtn = snd_pcm_close(hPcm_Spk.pDs)) < 0) {
cli_print("snd_pcm_close (spk) failed %s", snd_strerror(rtn));
}
if ((rtn = snd_pcm_close(hPcm_Mic.pDs)) < 0) {
cli_print("snd_pcm_close (mic) failed %s", snd_strerror(rtn));
}
if (hPcm_Spk.hAudioman) {
(void)audio_manager_free_handle(hPcm_Spk.hAudioman);
}
if (hPcm_Mic.hAudioman) {
(void)audio_manager_free_handle(hPcm_Mic.hAudioman);
}
return 0;
}
int
main (int argc, char **argv)
{
int card = -1;
int dev = 0;
snd_pcm_t *pcm_handle;
FILE *file;
wave_hdr wav_header;
int samples;
int sample_rate;
int sample_channels;
int sample_bits;
char *sample_buffer;
int fragsize = -1;
int verbose = 0;
int rtn;
int final_return_code = -1;
snd_pcm_channel_info_t pi;
snd_mixer_t *mixer_handle;
snd_mixer_group_t group;
snd_pcm_channel_params_t pp;
snd_pcm_channel_setup_t setup;
int bsize, bytes_read, total_written = 0;
fd_set rfds, wfds;
uint32_t voice_mask[] = { 0, 0, 0, 0 };
snd_pcm_voice_conversion_t voice_conversion;
int voice_override = 0;
int num_frags = -1;
char input_file[PATH_MAX];
char cwd[PATH_MAX];
screen_context_t screen_cxt;
screen_window_t screen_win;
screen_buffer_t screen_buf;
int screen_fill_attribs[] = { SCREEN_BLIT_COLOR, COLOR_PURPLE, SCREEN_BLIT_END };
int screen_dirty[4] = { 0, 0, 1024, 600 }; //start with sane default values
int idle_mode = SCREEN_IDLE_MODE_KEEP_AWAKE;
int usage = SCREEN_USAGE_NATIVE;
if (screen_create_context(&screen_cxt, 0) != 0)
{
return err("failed to create context");
}
if (screen_create_window(&screen_win, screen_cxt) != 0)
{
err("failed to create window");
goto fail1;
}
if (screen_set_window_property_iv(screen_win, SCREEN_PROPERTY_USAGE, &usage) != 0)
{
err("failed to set native usage mode");
goto fail2;
}
if (screen_create_window_buffers(screen_win, 1) != 0)
{
err("failed to set native usage mode");
goto fail2;
}
if(screen_get_window_property_pv(screen_win, SCREEN_PROPERTY_RENDER_BUFFERS, (void **)&screen_buf) != 0)
{
err("failed to get screen buffer");
goto fail2;
}
if (screen_fill(screen_cxt, screen_buf, screen_fill_attribs) != 0) {
err("failed to fill the screen");
goto fail3;
}
if (screen_get_window_property_iv(screen_win, SCREEN_PROPERTY_BUFFER_SIZE, screen_dirty+2) != 0) {
err("failed to get window size");
goto fail3;
}
if (screen_set_window_property_iv(screen_win, SCREEN_PROPERTY_IDLE_MODE, &idle_mode) != 0)
{
err("failed to set idle mode");
goto fail3;
}
if (screen_post_window(screen_win, screen_buf, 1, screen_dirty, 0) != 0) {
err("failed to post the window");
goto fail3;
}
if ((rtn = snd_pcm_open_preferred (&pcm_handle, &card, &dev, SND_PCM_OPEN_PLAYBACK)) < 0)
{
err ("device open");
goto fail3;
}
getcwd(cwd, PATH_MAX);
rtn = snprintf(input_file, PATH_MAX, "%s%s", cwd, WAV_RELATIVE_PATH);
if (rtn > PATH_MAX - 1)
{
err ("File name and path too long");
goto fail4;
}
if ((file = fopen (input_file, "r")) == 0)
{
err ("File open failed");
goto fail4;
}
if (check_hdr (file) == -1) {
err ("check_hdr failed");
goto fail5;
}
samples = find_tag (file, "fmt ");
fread (&wav_header, sizeof (wav_header), 1, file);
fseek (file, (samples - sizeof (wave_hdr)), SEEK_CUR);
sample_rate = ENDIAN_LE32 (wav_header.samples_per_sec);
sample_channels = ENDIAN_LE16 (wav_header.channels);
sample_bits = ENDIAN_LE16 (wav_header.bits_per_sample);
printf ("SampleRate = %d, channels = %d, SampleBits = %d\n", sample_rate, sample_channels,
sample_bits);
/* disabling mmap is not actually required in this example but it is included to
* demonstrate how it is used when it is required.
*/
if ((rtn = snd_pcm_plugin_set_disable (pcm_handle, PLUGIN_DISABLE_MMAP)) < 0)
{
fprintf (stderr, "snd_pcm_plugin_set_disable failed: %s\n", snd_strerror (rtn));
goto fail5;
}
memset (&pi, 0, sizeof (pi));
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((rtn = snd_pcm_plugin_info (pcm_handle, &pi)) < 0)
{
fprintf (stderr, "snd_pcm_plugin_info failed: %s\n", snd_strerror (rtn));
goto fail5;
}
memset (&pp, 0, sizeof (pp));
pp.mode = SND_PCM_MODE_BLOCK;
pp.channel = SND_PCM_CHANNEL_PLAYBACK;
pp.start_mode = SND_PCM_START_FULL;
pp.stop_mode = SND_PCM_STOP_STOP;
pp.buf.block.frag_size = pi.max_fragment_size;
if (fragsize != -1)
{
pp.buf.block.frag_size = fragsize;
}
pp.buf.block.frags_max = num_frags;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.rate = sample_rate;
pp.format.voices = sample_channels;
if (ENDIAN_LE16 (wav_header.format_tag) == 6)
pp.format.format = SND_PCM_SFMT_A_LAW;
else if (ENDIAN_LE16 (wav_header.format_tag) == 7)
pp.format.format = SND_PCM_SFMT_MU_LAW;
else if (sample_bits == 8)
pp.format.format = SND_PCM_SFMT_U8;
else if (sample_bits == 24)
pp.format.format = SND_PCM_SFMT_S24;
else
pp.format.format = SND_PCM_SFMT_S16_LE;
strcpy (pp.sw_mixer_subchn_name, "Wave playback channel");
if ((rtn = snd_pcm_plugin_params (pcm_handle, &pp)) < 0)
{
fprintf (stderr, "snd_pcm_plugin_params failed: %s\n", snd_strerror (rtn));
goto fail5;
}
if ((rtn = snd_pcm_plugin_prepare (pcm_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0) {
fprintf (stderr, "snd_pcm_plugin_prepare failed: %s\n", snd_strerror (rtn));
goto fail5;
}
if (voice_override)
{
snd_pcm_plugin_get_voice_conversion (pcm_handle, SND_PCM_CHANNEL_PLAYBACK,
&voice_conversion);
voice_conversion.matrix[0] = voice_mask[0];
voice_conversion.matrix[1] = voice_mask[1];
voice_conversion.matrix[2] = voice_mask[2];
voice_conversion.matrix[3] = voice_mask[3];
snd_pcm_plugin_set_voice_conversion (pcm_handle, SND_PCM_CHANNEL_PLAYBACK,
&voice_conversion);
}
memset (&setup, 0, sizeof (setup));
memset (&group, 0, sizeof (group));
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
setup.mixer_gid = &group.gid;
if ((rtn = snd_pcm_plugin_setup (pcm_handle, &setup)) < 0)
{
fprintf (stderr, "snd_pcm_plugin_setup failed: %s\n", snd_strerror (rtn));
goto fail5;
}
printf ("Format %s \n", snd_pcm_get_format_name (setup.format.format));
printf ("Frag Size %d \n", setup.buf.block.frag_size);
printf ("Total Frags %d \n", setup.buf.block.frags);
printf ("Rate %d \n", setup.format.rate);
printf ("Voices %d \n", setup.format.voices);
bsize = setup.buf.block.frag_size;
if (group.gid.name[0] == 0)
{
fprintf (stderr, "Mixer Pcm Group [%s] Not Set \n", group.gid.name);
goto fail5;
}
printf ("Mixer Pcm Group [%s]\n", group.gid.name);
if ((rtn = snd_mixer_open (&mixer_handle, card, setup.mixer_device)) < 0)
{
fprintf (stderr, "snd_mixer_open failed: %s\n", snd_strerror (rtn));
goto fail5;
}
samples = find_tag (file, "data");
sample_buffer = malloc (bsize);
FD_ZERO (&rfds);
FD_ZERO (&wfds);
bytes_read = 1;
while (total_written < samples && bytes_read > 0)
{
if (tcgetpgrp (0) == getpid ())
FD_SET (STDIN_FILENO, &rfds);
FD_SET (snd_mixer_file_descriptor (mixer_handle), &rfds);
FD_SET (snd_pcm_file_descriptor (pcm_handle, SND_PCM_CHANNEL_PLAYBACK), &wfds);
rtn = max (snd_mixer_file_descriptor (mixer_handle),
snd_pcm_file_descriptor (pcm_handle, SND_PCM_CHANNEL_PLAYBACK));
if (select (rtn + 1, &rfds, &wfds, NULL, NULL) == -1)
{
err ("select");
goto fail6;
}
if (FD_ISSET (snd_pcm_file_descriptor (pcm_handle, SND_PCM_CHANNEL_PLAYBACK), &wfds))
{
snd_pcm_channel_status_t status;
int written = 0;
if ((bytes_read = fread (sample_buffer, 1, min (samples - total_written, bsize), file)) <= 0)
continue;
written = snd_pcm_plugin_write (pcm_handle, sample_buffer, bytes_read);
if (verbose)
printf ("bytes written = %d \n", written);
if (written < bytes_read)
{
memset (&status, 0, sizeof (status));
status.channel = SND_PCM_CHANNEL_PLAYBACK;
if (snd_pcm_plugin_status (pcm_handle, &status) < 0)
{
fprintf (stderr, "underrun: playback channel status error\n");
goto fail6;
}
if (status.status == SND_PCM_STATUS_READY ||
status.status == SND_PCM_STATUS_UNDERRUN)
{
if (snd_pcm_plugin_prepare (pcm_handle, SND_PCM_CHANNEL_PLAYBACK) < 0)
{
fprintf (stderr, "underrun: playback channel prepare error\n");
goto fail6;
}
}
if (written < 0)
written = 0;
written += snd_pcm_plugin_write (pcm_handle, sample_buffer + written, bytes_read - written);
}
total_written += written;
}
}
bytes_read = snd_pcm_plugin_flush (pcm_handle, SND_PCM_CHANNEL_PLAYBACK);
final_return_code = 0;
fail6:
rtn = snd_mixer_close (mixer_handle);
fail5:
fclose (file);
fail4:
rtn = snd_pcm_close (pcm_handle);
fail3:
screen_destroy_buffer(screen_buf);
fail2:
screen_destroy_window(screen_win);
fail1:
screen_destroy_context(screen_cxt);
return final_return_code;
}
static int NTO_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int rval;
int format;
Uint16 test_format;
int twidth;
int found;
#ifdef DEBUG_AUDIO
fprintf(stderr, "NTO_OpenAudio\n");
#endif
audio_handle = NULL;
this->enabled = 0;
if ( pcm_buf != NULL ) {
free((Uint8 *)pcm_buf);
pcm_buf = NULL;
}
/* initialize channel transfer parameters to default */
init_pcm_cparams(&cparams);
/* Open the audio device */
rval = snd_pcm_open_preferred(&audio_handle, &card_no, &device_no, OPEN_FLAGS);
if ( rval < 0 ) {
SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
return(-1);
}
/* set to nonblocking mode */
if ((rval = snd_pcm_nonblock_mode(audio_handle, 1))<0) //I assume 1 means on
{
SDL_SetError("snd_pcm_nonblock_mode failed: %s\n", snd_strerror(rval));
return(-1);
}
/* enable count status parameter */
if ((rval = snd_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP))<0)
{
SDL_SetError("snd_plugin_set_disable failed: %s\n", snd_strerror(rval));
return(-1);
}
/* Try for a closest match on audio format */
format = 0;
found = 0; // can't use format as SND_PCM_SFMT_U8 = 0 in nto
for ( test_format = SDL_FirstAudioFormat(spec->format); !found ; )
{
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x spec->samples %d\n", test_format,spec->samples);
#endif
/* if match found set format to equivalent ALSA format */
switch ( test_format ) {
case AUDIO_U8:
format = SND_PCM_SFMT_U8;
cparams.buf.block.frag_size = spec->samples * spec->channels;
found = 1;
break;
case AUDIO_S8:
format = SND_PCM_SFMT_S8;
cparams.buf.block.frag_size = spec->samples * spec->channels;
found = 1;
break;
case AUDIO_S16LSB:
format = SND_PCM_SFMT_S16_LE;
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
found = 1;
break;
case AUDIO_S16MSB:
format = SND_PCM_SFMT_S16_BE;
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
found = 1;
break;
case AUDIO_U16LSB:
format = SND_PCM_SFMT_U16_LE;
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
found = 1;
break;
case AUDIO_U16MSB:
format = SND_PCM_SFMT_U16_BE;
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
found = 1;
break;
default:
break;
}
if ( ! found ) {
test_format = SDL_NextAudioFormat();
}
}
/* assumes test_format not 0 on success */
if ( test_format == 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
return(-1);
}
spec->format = test_format;
/* Set the audio format */
cparams.format.format = format;
/* Set mono or stereo audio (currently only two channels supported) */
cparams.format.voices = spec->channels;
#ifdef DEBUG_AUDIO
fprintf(stderr,"intializing channels %d\n", cparams.format.voices);
#endif
/* Set rate */
cparams.format.rate = spec->freq ;
/* Setup the transfer parameters according to cparams */
rval = snd_pcm_plugin_params(audio_handle, &cparams);
if (rval < 0) {
SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
return(-1);
}
/* Make sure channel is setup right one last time */
memset( &csetup, 0, sizeof( csetup ) );
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
if ( snd_pcm_plugin_setup( audio_handle, &csetup ) < 0 )
{
SDL_SetError("Unable to setup playback channel\n" );
return(-1);
}
else
{
#ifdef DEBUG_AUDIO
fprintf(stderr,"requested format: %d\n",cparams.format.format);
fprintf(stderr,"requested frag size: %d\n",cparams.buf.block.frag_size);
fprintf(stderr,"requested max frags: %d\n\n",cparams.buf.block.frags_max);
fprintf(stderr,"real format: %d\n", csetup.format.format );
fprintf(stderr,"real frag size : %d\n", csetup.buf.block.frag_size );
fprintf(stderr,"real max frags : %d\n", csetup.buf.block.frags_max );
#endif // DEBUG_AUDIO
}
/* Allocate memory to the audio buffer and initialize with silence
(Note that buffer size must be a multiple of fragment size, so find closest multiple)
*/
twidth = snd_pcm_format_width(format);
if (twidth < 0) {
printf("snd_pcm_format_width failed\n");
twidth = 0;
}
#ifdef DEBUG_AUDIO
fprintf(stderr,"format is %d bits wide\n",twidth);
#endif
pcm_len = spec->size ;
#ifdef DEBUG_AUDIO
fprintf(stderr,"pcm_len set to %d\n", pcm_len);
#endif
if (pcm_len == 0)
{
pcm_len = csetup.buf.block.frag_size;
}
pcm_buf = (Uint8*)malloc(pcm_len);
if (pcm_buf == NULL) {
SDL_SetError("pcm_buf malloc failed\n");
return(-1);
}
memset(pcm_buf,spec->silence,pcm_len);
#ifdef DEBUG_AUDIO
fprintf(stderr,"pcm_buf malloced and silenced.\n");
#endif
/* get the file descriptor */
if( (audio_fd = snd_pcm_file_descriptor(audio_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
fprintf(stderr, "snd_pcm_file_descriptor failed with error code: %d\n", audio_fd);
}
/* Trigger audio playback */
rval = snd_pcm_plugin_prepare( audio_handle, SND_PCM_CHANNEL_PLAYBACK);
if (rval < 0) {
SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror (rval));
return(-1);
}
this->enabled = 1;
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* We're ready to rock and roll. :-) */
return(0);
}
int
writePcmData (PcmDevice *pcm, const unsigned char *buffer, int count) {
return writeFile(snd_pcm_file_descriptor(pcm->handle, pcm->parameters.channel), buffer, count);
}
static int
QSA_OpenDevice(_THIS, const char *devname, int iscapture)
{
int status = 0;
int format = 0;
SDL_AudioFormat test_format = 0;
int found = 0;
snd_pcm_channel_setup_t csetup;
snd_pcm_channel_params_t cparams;
/* Initialize all variables that we clean on shutdown */
this->hidden =
(struct SDL_PrivateAudioData *) SDL_calloc(1,
(sizeof
(struct
SDL_PrivateAudioData)));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, sizeof(struct SDL_PrivateAudioData));
/* Initialize channel transfer parameters to default */
QSA_InitAudioParams(&cparams);
/* Initialize channel direction: capture or playback */
this->hidden->iscapture = iscapture;
/* Find deviceid and cardid by device name for playback */
if ((!this->hidden->iscapture) && (devname != NULL)) {
uint32_t device;
int32_t status;
/* Search in the playback devices */
device = 0;
do {
status = SDL_strcmp(qsa_playback_device[device].name, devname);
if (status == 0) {
/* Found requested device */
this->hidden->deviceno = qsa_playback_device[device].deviceno;
this->hidden->cardno = qsa_playback_device[device].cardno;
break;
}
device++;
if (device >= qsa_playback_devices) {
QSA_CloseDevice(this);
return SDL_SetError("No such playback device");
}
} while (1);
}
/* Find deviceid and cardid by device name for capture */
if ((this->hidden->iscapture) && (devname != NULL)) {
/* Search in the capture devices */
uint32_t device;
int32_t status;
/* Searching in the playback devices */
device = 0;
do {
status = SDL_strcmp(qsa_capture_device[device].name, devname);
if (status == 0) {
/* Found requested device */
this->hidden->deviceno = qsa_capture_device[device].deviceno;
this->hidden->cardno = qsa_capture_device[device].cardno;
break;
}
device++;
if (device >= qsa_capture_devices) {
QSA_CloseDevice(this);
return SDL_SetError("No such capture device");
}
} while (1);
}
/* Check if SDL requested default audio device */
if (devname == NULL) {
/* Open system default audio device */
if (!this->hidden->iscapture) {
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
SND_PCM_OPEN_PLAYBACK);
} else {
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
SND_PCM_OPEN_CAPTURE);
}
} else {
/* Open requested audio device */
if (!this->hidden->iscapture) {
status =
snd_pcm_open(&this->hidden->audio_handle,
this->hidden->cardno, this->hidden->deviceno,
SND_PCM_OPEN_PLAYBACK);
} else {
status =
snd_pcm_open(&this->hidden->audio_handle,
this->hidden->cardno, this->hidden->deviceno,
SND_PCM_OPEN_CAPTURE);
}
}
/* Check if requested device is opened */
if (status < 0) {
this->hidden->audio_handle = NULL;
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_open", status);
}
if (!QSA_CheckBuggyCards(this, QSA_MMAP_WORKAROUND)) {
/* Disable QSA MMAP plugin for buggy audio drivers */
status =
snd_pcm_plugin_set_disable(this->hidden->audio_handle,
PLUGIN_DISABLE_MMAP);
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_plugin_set_disable", status);
}
}
/* Try for a closest match on audio format */
format = 0;
/* can't use format as SND_PCM_SFMT_U8 = 0 in qsa */
found = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format); !found;) {
/* if match found set format to equivalent QSA format */
switch (test_format) {
case AUDIO_U8:
{
format = SND_PCM_SFMT_U8;
found = 1;
}
break;
case AUDIO_S8:
{
format = SND_PCM_SFMT_S8;
found = 1;
}
break;
case AUDIO_S16LSB:
{
format = SND_PCM_SFMT_S16_LE;
found = 1;
}
break;
case AUDIO_S16MSB:
{
format = SND_PCM_SFMT_S16_BE;
found = 1;
}
break;
case AUDIO_U16LSB:
{
format = SND_PCM_SFMT_U16_LE;
found = 1;
}
break;
case AUDIO_U16MSB:
{
format = SND_PCM_SFMT_U16_BE;
found = 1;
}
break;
case AUDIO_S32LSB:
{
format = SND_PCM_SFMT_S32_LE;
found = 1;
}
break;
case AUDIO_S32MSB:
{
format = SND_PCM_SFMT_S32_BE;
found = 1;
}
break;
case AUDIO_F32LSB:
{
format = SND_PCM_SFMT_FLOAT_LE;
found = 1;
}
break;
case AUDIO_F32MSB:
{
format = SND_PCM_SFMT_FLOAT_BE;
found = 1;
}
break;
default:
{
break;
}
}
if (!found) {
test_format = SDL_NextAudioFormat();
}
}
/* assumes test_format not 0 on success */
if (test_format == 0) {
QSA_CloseDevice(this);
return SDL_SetError("QSA: Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/* Set the audio format */
cparams.format.format = format;
/* Set mono/stereo/4ch/6ch/8ch audio */
cparams.format.voices = this->spec.channels;
/* Set rate */
cparams.format.rate = this->spec.freq;
/* Setup the transfer parameters according to cparams */
status = snd_pcm_plugin_params(this->hidden->audio_handle, &cparams);
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_channel_params", status);
}
/* Make sure channel is setup right one last time */
SDL_memset(&csetup, 0, sizeof(csetup));
if (!this->hidden->iscapture) {
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
} else {
csetup.channel = SND_PCM_CHANNEL_CAPTURE;
}
/* Setup an audio channel */
if (snd_pcm_plugin_setup(this->hidden->audio_handle, &csetup) < 0) {
QSA_CloseDevice(this);
return SDL_SetError("QSA: Unable to setup channel");
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this->hidden->pcm_len = this->spec.size;
if (this->hidden->pcm_len == 0) {
this->hidden->pcm_len =
csetup.buf.block.frag_size * this->spec.channels *
(snd_pcm_format_width(format) / 8);
}
/*
* Allocate memory to the audio buffer and initialize with silence
* (Note that buffer size must be a multiple of fragment size, so find
* closest multiple)
*/
this->hidden->pcm_buf =
(Uint8 *) SDL_AllocAudioMem(this->hidden->pcm_len);
if (this->hidden->pcm_buf == NULL) {
QSA_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->pcm_buf, this->spec.silence,
this->hidden->pcm_len);
/* get the file descriptor */
if (!this->hidden->iscapture) {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (this->hidden->audio_fd < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_file_descriptor", status);
}
/* Prepare an audio channel */
if (!this->hidden->iscapture) {
/* Prepare audio playback */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
/* Prepare audio capture */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_plugin_prepare", status);
}
/* We're really ready to rock and roll. :-) */
return 0;
}
static ALCenum qsa_open_capture(ALCdevice* device, const ALCchar* deviceName)
{
qsa_data *data;
int card, dev;
int format=-1;
int status;
data=(qsa_data*)calloc(1, sizeof(qsa_data));
if (data==NULL)
{
return ALC_OUT_OF_MEMORY;
}
if(!deviceName)
deviceName = qsaDevice;
if(strcmp(deviceName, qsaDevice) == 0)
status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_CAPTURE);
else
{
const DevMap *iter;
if(VECTOR_SIZE(CaptureNameMap) == 0)
deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap);
#define MATCH_DEVNAME(iter) ((iter)->name && strcmp(deviceName, (iter)->name)==0)
VECTOR_FIND_IF(iter, const DevMap, CaptureNameMap, MATCH_DEVNAME);
#undef MATCH_DEVNAME
if(iter == VECTOR_ITER_END(CaptureNameMap))
{
free(data);
return ALC_INVALID_DEVICE;
}
status = snd_pcm_open(&data->pcmHandle, iter->card, iter->dev, SND_PCM_OPEN_CAPTURE);
}
if(status < 0)
{
free(data);
return ALC_INVALID_DEVICE;
}
data->audio_fd = snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE);
if(data->audio_fd < 0)
{
snd_pcm_close(data->pcmHandle);
free(data);
return ALC_INVALID_DEVICE;
}
al_string_copy_cstr(&device->DeviceName, deviceName);
device->ExtraData = data;
switch (device->FmtType)
{
case DevFmtByte:
format=SND_PCM_SFMT_S8;
break;
case DevFmtUByte:
format=SND_PCM_SFMT_U8;
break;
case DevFmtShort:
format=SND_PCM_SFMT_S16_LE;
break;
case DevFmtUShort:
format=SND_PCM_SFMT_U16_LE;
break;
case DevFmtInt:
format=SND_PCM_SFMT_S32_LE;
break;
case DevFmtUInt:
format=SND_PCM_SFMT_U32_LE;
break;
case DevFmtFloat:
format=SND_PCM_SFMT_FLOAT_LE;
break;
}
/* we actually don't want to block on reads */
snd_pcm_nonblock_mode(data->pcmHandle, 1);
/* Disable mmap to control data transfer to the audio device */
snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP);
/* configure a sound channel */
memset(&data->cparams, 0, sizeof(data->cparams));
data->cparams.mode=SND_PCM_MODE_BLOCK;
data->cparams.channel=SND_PCM_CHANNEL_CAPTURE;
data->cparams.start_mode=SND_PCM_START_GO;
data->cparams.stop_mode=SND_PCM_STOP_STOP;
data->cparams.buf.block.frag_size=device->UpdateSize*
ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType);
data->cparams.buf.block.frags_max=device->NumUpdates;
data->cparams.buf.block.frags_min=device->NumUpdates;
data->cparams.format.interleave=1;
data->cparams.format.rate=device->Frequency;
data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans);
data->cparams.format.format=format;
if(snd_pcm_plugin_params(data->pcmHandle, &data->cparams) < 0)
{
snd_pcm_close(data->pcmHandle);
free(data);
device->ExtraData=NULL;
return ALC_INVALID_VALUE;
}
return ALC_NO_ERROR;
}
static gboolean
alsa_open (void *dp)
{
alsa_driver * const d = dp;
int mf, err;
snd_pcm_format_t pf;
// snd_pcm_channel_info_t pcm_info;
snd_pcm_channel_params_t pp;
snd_pcm_channel_setup_t setup;
snd_pcm_channel_flush(d->soundfd, SND_PCM_CHANNEL_PLAYBACK);
memset(&pp, 0, sizeof(pp));
err = snd_pcm_open(&(d->soundfd), d->card_number, d->device_number, SND_PCM_OPEN_PLAYBACK);
if (err != 0) {
char buf[256];
g_sprintf(buf, _("Couldn't open ALSA device for sound output (card:%d, device:%d):\n%s"),
d->card_number, d->device_number, snd_strerror(err));
error_error(buf);
goto out;
}
// ---
// Set non-blocking mode.
// ---
// snd_pcm_nonblock_mode(d->soundfd, 1);
d->outtime = 0;
d->bits = 0;
mf = 0;
// --
// Set channel parameters
// --
pp.mode = SND_PCM_MODE_BLOCK;
pp.channel = SND_PCM_CHANNEL_PLAYBACK;
pp.start_mode = SND_PCM_START_FULL;
pp.stop_mode = SND_PCM_STOP_ROLLOVER;
// ---
// Select audio format
// ---
memset(&pf, 0, sizeof(pf));
pf.interleave = 1;
if (d->p_resolution == 16) {
pf.format = SND_PCM_SFMT_S16_LE;
d->bits = 16;
mf = ST_MIXER_FORMAT_S16_LE;
}
else {
pf.format = SND_PCM_SFMT_U8;
d->bits = 8;
mf = ST_MIXER_FORMAT_S8;
}
pf.rate = d->p_mixfreq;
d->playrate = d->p_mixfreq;
if(d->p_channels == 2) {
d->stereo = 1;
pf.voices = d->p_channels;
mf |= ST_MIXER_FORMAT_STEREO;
}
else {
pf.voices = d->p_channels;
d->stereo = 0;
}
d->mf = mf;
memcpy(&pp.format, &pf, sizeof(pf));
// pp.buf.block.frag_size = d->p_fragsize * pf.voices * (d->bits / 8);
pp.buf.block.frag_size = d->p_fragsize;
pp.buf.block.frags_max = -1;
pp.buf.block.frags_min = 1;
err = snd_pcm_channel_params(d->soundfd, &pp);
if (err < 0) {
error_error(_("Required output-channel parameters not supported.\n"));
goto out;
}
if (snd_pcm_channel_prepare(d->soundfd, SND_PCM_CHANNEL_PLAYBACK) < 0) {
error_error(_("Unable to prepare ALSA channel.\n"));
goto out;
}
// ---
// Get buffering parameters
// ---
memset(&setup, 0, sizeof(setup));
setup.mode = SND_PCM_MODE_BLOCK;
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
err = snd_pcm_channel_setup(d->soundfd, &setup);
if (err < 0) {
error_error(_("Alsa setup error.\n"));
goto out;
}
// snd_pcm_channel_status(d->soundfd, &pbstat);
// d->fragsize = pbstat.fragment_size;
d->numfrags = setup.buf.block.frags;
d->fragsize = setup.buf.block.frag_size;
/* fprintf(stderr, "Numfrags: %d\n", d->numfrags);
fprintf(stderr, "Fragsize: %d\n", d->fragsize); */
d->sndbuf = calloc(1, d->fragsize);
if(d->stereo == 1) {
d->fragsize /= 2;
}
if(d->bits == 16) {
d->fragsize /= 2;
}
d->polltag = audio_poll_add(snd_pcm_file_descriptor(d->soundfd, SND_PCM_CHANNEL_PLAYBACK), GDK_INPUT_WRITE, alsa_poll_ready_playing, d);
/* d->firstpoll = TRUE; */
d->firstpoll = TRUE;
d->playtime = 0;
return TRUE;
out:
alsa_release(dp);
return FALSE;
}
int
main(int argc, char **argv)
{
FILE *file;
int samples;
char *sample_buffer;
int rtn, final_return_code = -1, exit_application = 0;
int bsize, bytes_read, total_written = 0;
fd_set rfds, wfds;
char input_file[PATH_MAX];
char cwd[PATH_MAX];
/*
* Before we can listen for events from the BlackBerry Tablet OS platform
* services, we need to initialize the BPS infrastructure
*/
bps_initialize();
if (setup_screen() != EXIT_SUCCESS) {
printf("Unable to set up the screen. Exiting.");
exit(-1);
}
/*
* Once the BPS infrastructure has been initialized we can register for
* events from the various BlackBerry Tablet OS platform services. The
* Navigator service manages and delivers application life cycle and
* visibility events.
*
* For this sample, we request Navigator events so we can track when
* the system is terminating the application (NAVIGATOR_EXIT event).
* This allows us to clean up application resources.
*
* We request Audio Device events because we want to make sure that
* we properly handle changes in audio output.
*
* We request dialog events to properly initialize the dialog
* subsystem in order to display status and error messages.
*/
if (BPS_SUCCESS != navigator_request_events(0)) {
fprintf(stderr, "Error requesting navigator events: %s", strerror(errno));
exit(-1);
}
if (BPS_SUCCESS != dialog_request_events(0)) {
fprintf(stderr, "Error requesting dialog events: %s", strerror(errno));
exit(-1);
}
if (BPS_SUCCESS != audiodevice_request_events(0)) {
fprintf(stderr, "Error requesting audio device events: %s", strerror(errno));
exit(-1);
}
/*
* Create and display the dialog.
*/
create_dialog();
/*
* Open and check the input file.
*/
getcwd(cwd, PATH_MAX);
rtn = snprintf(input_file, PATH_MAX, "%s/%s", cwd, WAV_RELATIVE_PATH);
if (rtn > PATH_MAX - 1) {
err("File name and path too long");
goto fail1;
}
if ((file = fopen(input_file, "r")) == 0) {
err("File open failed");
goto fail1;
}
if (check_hdr(file) == -1) {
err("check_hdr failed");
goto fail2;
}
/*
* Parse the headers in the wav file to figure out what kind of data we
* are dealing with in the file.
*/
samples = find_tag(file, "fmt ");
fread(&wav_header, sizeof(wav_header), 1, file);
fseek(file,(samples - sizeof(wave_hdr)), SEEK_CUR);
sample_rate = ENDIAN_LE32(wav_header.samples_per_sec);
sample_channels = ENDIAN_LE16(wav_header.channels);
sample_bits = ENDIAN_LE16(wav_header.bits_per_sample);
snprintf(msg, MSG_SIZE, "SampleRate = %d, channels = %d, SampleBits = %d\n", sample_rate, sample_channels,
sample_bits);
show_dialog_message(msg);
if (setup_snd(NULL)) {
goto fail2;
}
bsize = setup.buf.block.frag_size;
samples = find_tag(file, "data");
sample_buffer = malloc(bsize);
if (!sample_buffer) {
goto fail3;
}
FD_ZERO(&rfds);
FD_ZERO(&wfds);
bytes_read = 1;
while (total_written < samples && bytes_read > 0 ) {
bps_event_t *event = NULL;
while (BPS_SUCCESS == bps_get_event(&event, 0) && event) {
/*
* If it is a NAVIGATOR_EXIT event then we are done so stop
* processing events, clean up and exit
*/
if (bps_event_get_domain(event) == navigator_get_domain()) {
if (NAVIGATOR_EXIT == bps_event_get_code(event)) {
exit_application = 1;
goto success;
}
}
if (bps_event_get_domain(event) == audiodevice_get_domain()) {
/*
* If it is a audio device event then it means a new audio device
* has been enabled and a switch is required. We close the old,
* open the new audio device using the path and get the card number so
* that we can close and re-open the mixer with the new card
* number.
*/
const char * audiodevice_path = audiodevice_event_get_path(event);
if (NULL == audiodevice_path) {
snprintf(msg, MSG_SIZE, "audiodevice_event_get_path failed: %s\n", snd_strerror(rtn));
show_dialog_message(msg);
goto fail5;
}
if ((rtn = snd_mixer_close(mixer_handle)) < 0) {
snprintf(msg, MSG_SIZE, "snd_mixer_close failed: %s\n", snd_strerror(rtn));
show_dialog_message(msg);
goto fail4;
}
if ((rtn = snd_pcm_close(pcm_handle)) < 0) {
snprintf(msg, MSG_SIZE, "snd_pcm_close failed: %s\n", snd_strerror(rtn));
show_dialog_message(msg);
goto fail3;
}
if (setup_snd(audiodevice_path)) {
/*
* setup_snd() closes pcm and mixer handles in the case of error so we
* skip clean up of the handles in the case of failure.
*/
goto fail3;
}
}
}
if (tcgetpgrp(0) == getpid())
FD_SET(STDIN_FILENO, &rfds);
FD_SET(snd_mixer_file_descriptor(mixer_handle), &rfds);
FD_SET(snd_pcm_file_descriptor(pcm_handle, SND_PCM_CHANNEL_PLAYBACK), &wfds);
rtn = max(snd_mixer_file_descriptor(mixer_handle),
snd_pcm_file_descriptor(pcm_handle, SND_PCM_CHANNEL_PLAYBACK));
if (select(rtn + 1, &rfds, &wfds, NULL, NULL) == -1) {
err("select");
goto fail5;
}
if (FD_ISSET(snd_pcm_file_descriptor(pcm_handle, SND_PCM_CHANNEL_PLAYBACK), &wfds)) {
snd_pcm_channel_status_t status;
int written = 0;
if ((bytes_read = fread(sample_buffer, 1, min(samples - total_written, bsize), file)) <= 0)
continue;
written = snd_pcm_plugin_write(pcm_handle, sample_buffer, bytes_read);
if (written < bytes_read) {
memset(&status, 0, sizeof(status));
status.channel = SND_PCM_CHANNEL_PLAYBACK;
if (snd_pcm_plugin_status(pcm_handle, &status) < 0) {
show_dialog_message("underrun: playback channel status error\n");
goto fail5;
}
if (status.status == SND_PCM_STATUS_READY ||
status.status == SND_PCM_STATUS_UNDERRUN) {
if (snd_pcm_plugin_prepare(pcm_handle, SND_PCM_CHANNEL_PLAYBACK) < 0) {
show_dialog_message("underrun: playback channel prepare error\n");
goto fail5;
}
}
if (written < 0)
written = 0;
written += snd_pcm_plugin_write(pcm_handle, sample_buffer + written, bytes_read - written);
}
total_written += written;
}
}
success:
bytes_read = snd_pcm_plugin_flush(pcm_handle, SND_PCM_CHANNEL_PLAYBACK);
final_return_code = 0;
/*
* there are return codes to these close calls, but we would do the same
* thing regardless of error or success so we do not check the return codes.
*/
fail5:
snd_mixer_close(mixer_handle);
fail4:
snd_pcm_close(pcm_handle);
fail3:
free(sample_buffer);
sample_buffer = NULL;
fail2:
fclose(file);
fail1:
while (!exit_application) {
/*
* Something went wrong so there is probably an error message and
* we don't want to exit right away because we want the user to see
* the message in the dialog.
*
* Using a negative timeout (-1) in the call to bps_get_event(...)
* ensures that we don't busy wait by blocking until an event is
* available.
*/
bps_event_t *event = NULL;
bps_get_event(&event, -1);
if (event) {
/*
* If it is a NAVIGATOR_EXIT event then we are done so stop
* processing events, clean up and exit
*/
if (bps_event_get_domain(event) == navigator_get_domain()) {
if (NAVIGATOR_EXIT == bps_event_get_code(event)) {
exit_application = 1;
}
}
}
}
/*
* Destroy the dialog, if it exists and cleanup screen resources.
*/
destroy_dialog();
cleanup_screen();
bps_shutdown();
return final_return_code;
}
bool QnxAudioInput::open()
{
if (!m_format.isValid() || m_format.sampleRate() <= 0) {
if (!m_format.isValid())
qWarning("QnxAudioInput: open error, invalid format.");
else
qWarning("QnxAudioInput: open error, invalid sample rate (%d).", m_format.sampleRate());
return false;
}
int errorCode = 0;
int card = 0;
int device = 0;
if ((errorCode = snd_pcm_open_preferred(&m_pcmHandle, &card, &device, SND_PCM_OPEN_CAPTURE)) < 0) {
qWarning("QnxAudioInput: open error, couldn't open card (0x%x)", -errorCode);
return false;
}
// Necessary so that bytesFree() which uses the "free" member of the status struct works
snd_pcm_plugin_set_disable(m_pcmHandle, PLUGIN_DISABLE_MMAP);
snd_pcm_channel_info_t info;
memset(&info, 0, sizeof(info));
info.channel = SND_PCM_CHANNEL_CAPTURE;
if ((errorCode = snd_pcm_plugin_info(m_pcmHandle, &info)) < 0) {
qWarning("QnxAudioInput: open error, couldn't get channel info (0x%x)", -errorCode);
close();
return false;
}
snd_pcm_channel_params_t params = QnxAudioUtils::formatToChannelParams(m_format, QAudio::AudioInput, info.max_fragment_size);
if ((errorCode = snd_pcm_plugin_params(m_pcmHandle, ¶ms)) < 0) {
qWarning("QnxAudioInput: open error, couldn't set channel params (0x%x)", -errorCode);
close();
return false;
}
if ((errorCode = snd_pcm_plugin_prepare(m_pcmHandle, SND_PCM_CHANNEL_CAPTURE)) < 0) {
qWarning("QnxAudioInput: open error, couldn't prepare channel (0x%x)", -errorCode);
close();
return false;
}
snd_pcm_channel_setup_t setup;
memset(&setup, 0, sizeof(setup));
setup.channel = SND_PCM_CHANNEL_CAPTURE;
if ((errorCode = snd_pcm_plugin_setup(m_pcmHandle, &setup)) < 0) {
qWarning("QnxAudioInput: open error, couldn't get channel setup (0x%x)", -errorCode);
close();
return false;
}
m_periodSize = qMin(2048, setup.buf.block.frag_size);
m_clockStamp.restart();
m_timeStamp.restart();
m_elapsedTimeOffset = 0;
m_totalTimeValue = 0;
m_bytesRead = 0;
m_pcmNotifier = new QSocketNotifier(snd_pcm_file_descriptor(m_pcmHandle, SND_PCM_CHANNEL_CAPTURE),
QSocketNotifier::Read, this);
connect(m_pcmNotifier, SIGNAL(activated(int)), SLOT(userFeed()));
return true;
}
static int NTO_OpenAudio(_THIS, SDL_AudioSpec* spec)
{
int rval;
int format;
Uint16 test_format;
int found;
audio_handle = NULL;
this->enabled = 0;
if (pcm_buf != NULL)
{
SDL_FreeAudioMem(pcm_buf);
pcm_buf = NULL;
}
/* initialize channel transfer parameters to default */
NTO_InitAudioParams(&cparams);
/* Open the audio device */
rval = snd_pcm_open_preferred(&audio_handle, &cardno, &deviceno, OPEN_FLAGS);
if (rval < 0)
{
SDL_SetError("NTO_OpenAudio(): snd_pcm_open failed: %s\n", snd_strerror(rval));
return (-1);
}
if (!NTO_CheckBuggyCards(this, QSA_MMAP_WORKAROUND))
{
/* enable count status parameter */
if ((rval = snd_pcm_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP)) < 0)
{
SDL_SetError("snd_pcm_plugin_set_disable failed: %s\n", snd_strerror(rval));
return (-1);
}
}
/* Try for a closest match on audio format */
format = 0;
/* can't use format as SND_PCM_SFMT_U8 = 0 in nto */
found = 0;
for (test_format=SDL_FirstAudioFormat(spec->format); !found ;)
{
/* if match found set format to equivalent ALSA format */
switch (test_format)
{
case AUDIO_U8:
format = SND_PCM_SFMT_U8;
found = 1;
break;
case AUDIO_S8:
format = SND_PCM_SFMT_S8;
found = 1;
break;
case AUDIO_S16LSB:
format = SND_PCM_SFMT_S16_LE;
found = 1;
break;
case AUDIO_S16MSB:
format = SND_PCM_SFMT_S16_BE;
found = 1;
break;
case AUDIO_U16LSB:
format = SND_PCM_SFMT_U16_LE;
found = 1;
break;
case AUDIO_U16MSB:
format = SND_PCM_SFMT_U16_BE;
found = 1;
break;
default:
break;
}
if (!found)
{
test_format = SDL_NextAudioFormat();
}
}
/* assumes test_format not 0 on success */
if (test_format == 0)
{
SDL_SetError("NTO_OpenAudio(): Couldn't find any hardware audio formats");
return (-1);
}
spec->format = test_format;
/* Set the audio format */
cparams.format.format = format;
/* Set mono or stereo audio (currently only two channels supported) */
cparams.format.voices = spec->channels;
/* Set rate */
cparams.format.rate = spec->freq;
/* Setup the transfer parameters according to cparams */
rval = snd_pcm_plugin_params(audio_handle, &cparams);
if (rval < 0)
{
SDL_SetError("NTO_OpenAudio(): snd_pcm_channel_params failed: %s\n", snd_strerror(rval));
return (-1);
}
/* Make sure channel is setup right one last time */
SDL_memset(&csetup, 0x00, sizeof(csetup));
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
if (snd_pcm_plugin_setup(audio_handle, &csetup) < 0)
{
SDL_SetError("NTO_OpenAudio(): Unable to setup playback channel\n");
return -1;
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
pcm_len = spec->size;
if (pcm_len==0)
{
pcm_len = csetup.buf.block.frag_size * spec->channels * (snd_pcm_format_width(format)/8);
}
/* Allocate memory to the audio buffer and initialize with silence (Note that
buffer size must be a multiple of fragment size, so find closest multiple)
*/
pcm_buf = (Uint8*)SDL_AllocAudioMem(pcm_len);
if (pcm_buf == NULL)
{
SDL_SetError("NTO_OpenAudio(): pcm buffer allocation failed\n");
return (-1);
}
SDL_memset(pcm_buf, spec->silence, pcm_len);
/* get the file descriptor */
if ((audio_fd = snd_pcm_file_descriptor(audio_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
SDL_SetError("NTO_OpenAudio(): snd_pcm_file_descriptor failed with error code: %s\n", snd_strerror(rval));
return (-1);
}
/* Trigger audio playback */
rval = snd_pcm_plugin_prepare(audio_handle, SND_PCM_CHANNEL_PLAYBACK);
if (rval < 0)
{
SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror(rval));
return (-1);
}
this->enabled = 1;
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* We're really ready to rock and roll. :-) */
return (0);
}
static gboolean
alsa_open (void *dp)
{
alsa_driver * const d = dp;
int mf;
snd_pcm_format_t pf;
snd_pcm_capture_info_t pcm_info;
snd_pcm_capture_params_t pp;
snd_pcm_capture_status_t pbstat;
// int err = snd_pcm_open(&(d->soundfd), d->card_number, d->device_number,
// SND_PCM_OPEN_CAPTURE);
int err = snd_pcm_open(&(d->soundfd), 0, 0, SND_PCM_OPEN_CAPTURE);
if (err != 0) {
char buf[256];
g_sprintf(buf, _("Couldn't open ALSA device for sound input (card:%d, device:%d):\n%s"),
d->card_number, d->device_number, snd_strerror(err));
error_error(buf);
goto out;
}
// ---
// Set non-blocking mode.
// ---
snd_pcm_block_mode(d->soundfd, 1); // enable block mode
d->bits = 0;
mf = 0;
// ---
// Select audio format
// ---
memset(&pf, 0, sizeof(pf));
if (d->p_resolution == 16) {
pf.format = SND_PCM_SFMT_S16_LE;
d->bits = 16;
mf = ST_MIXER_FORMAT_S16_LE;
}
else {
pf.format = SND_PCM_SFMT_U8;
d->bits = 8;
mf = ST_MIXER_FORMAT_S8;
}
/* if(d->p_channels == 2) { */
/* d->stereo = 1; */
/* pf.channels = d->p_channels; */
/* mf |= ST_MIXER_FORMAT_STEREO; */
/* } */
/* else { */
/* pf.channels = d->p_channels; */
/* d->stereo = 0; */
/* } */
d->stereo = 0;
pf.channels = 1;
pf.rate = d->p_mixfreq;
d->playrate = d->p_mixfreq;
mf = ST_MIXER_FORMAT_S16_LE;
d->mf = mf;
err = snd_pcm_capture_format(d->soundfd, &pf);
if (err < 0) {
error_error(_("Required sound output format not supported.\n"));
goto out;
}
snd_pcm_capture_info(d->soundfd, &pcm_info);
memset(&pp, 0, sizeof(pp));
pp.fragment_size = d->p_fragsize * pf.channels * d->bits / 8;
pp.fragments_min = 1;
err = snd_pcm_capture_params(d->soundfd, &pp);
if (err < 0) {
error_error(_("Required sound output parameters not supported.\n"));
goto out;
}
snd_pcm_capture_status(d->soundfd, &pbstat);
/* d->numfrags = pbstat.fragments; */
d->numfrags = 1;
d->fragsize = pbstat.fragment_size;
d->sndbuf = calloc(1, d->fragsize);
if(d->stereo == 1) {
d->fragsize /= 2;
}
if(d->bits == 16) {
d->fragsize /= 2;
}
d->polltag = gdk_input_add(snd_pcm_file_descriptor(d->soundfd), GDK_INPUT_READ, alsa_poll_ready_sampling, d);
// d->firstpoll = TRUE;
// d->playtime = 0;
return TRUE;
out:
alsa_release(dp);
return FALSE;
}
static gboolean
alsa_open (void *dp)
{
alsa_driver * const d = dp;
int mf;
snd_pcm_format_t pf;
snd_pcm_playback_info_t pcm_info;
snd_pcm_playback_params_t pp;
snd_pcm_playback_status_t pbstat;
int err = snd_pcm_open(&(d->soundfd), d->card_number, d->device_number,
SND_PCM_OPEN_PLAYBACK);
if (err != 0) {
char buf[256];
g_sprintf(buf, _("Couldn't open ALSA device for sound output (card:%d, device:%d):\n%s"),
d->card_number, d->device_number, snd_strerror(err));
error_error(buf);
goto out;
}
// ---
// Set non-blocking mode.
// ---
snd_pcm_block_mode(d->soundfd, 0); // enable block mode
d->outtime = 0;
d->bits = 0;
mf = 0;
// ---
// Select audio format
// ---
memset(&pf, 0, sizeof(pf));
if (d->p_resolution == 16) {
pf.format = SND_PCM_SFMT_S16_LE;
d->bits = 16;
mf = ST_MIXER_FORMAT_S16_LE;
}
else {
pf.format = SND_PCM_SFMT_U8;
d->bits = 8;
mf = ST_MIXER_FORMAT_S8;
}
if(d->p_channels == 2) {
d->stereo = 1;
pf.channels = d->p_channels;
mf |= ST_MIXER_FORMAT_STEREO;
}
else {
pf.channels = d->p_channels;
d->stereo = 0;
}
pf.rate = d->p_mixfreq;
d->playrate = d->p_mixfreq;
d->mf = mf;
err = snd_pcm_playback_format(d->soundfd, &pf);
if (err < 0) {
error_error(_("Required sound output format not supported.\n"));
goto out;
}
snd_pcm_playback_info(d->soundfd, &pcm_info);
memset(&pp, 0, sizeof(pp));
pp.fragment_size = d->p_fragsize * pf.channels * (d->bits / 8);
pp.fragments_max = 1;
// pp.fragments_max = 16pcm_info.buffer_size / pp.fragment_size;
pp.fragments_room = 1;
err = snd_pcm_playback_params(d->soundfd, &pp);
if (err < 0) {
error_error(_("Required sound output parameters not supported.\n"));
goto out;
}
snd_pcm_playback_status(d->soundfd, &pbstat);
// d->numfrags = pbstat.fragments;
// d->numfrags = 1;
d->fragsize = pbstat.fragment_size;
d->numfrags = 1;
/* fprintf(stderr, "Numfrags: %d\n", d->numfrags);
fprintf(stderr, "Fragsize: %d\n", d->fragsize); */
d->sndbuf = calloc(1, d->fragsize);
if(d->stereo == 1) {
d->fragsize /= 2;
}
if(d->bits == 16) {
d->fragsize /= 2;
}
d->polltag = audio_poll_add(snd_pcm_file_descriptor(d->soundfd), GDK_INPUT_WRITE, alsa_poll_ready_playing, d);
/* d->firstpoll = TRUE; */
d->firstpoll = TRUE;
d->playtime = 0;
return TRUE;
out:
alsa_release(dp);
return FALSE;
}
// Initialize audio and report back a few parameters of the channel
int setupAudio( )
{
int error;
snd_pcm_channel_info_t pi;
snd_pcm_channel_params_t pp;
snd_pcm_channel_setup_t ps;
//if ((error = snd_pcm_open_name (&playback_handle, "pcmPreferred", SND_PCM_OPEN_PLAYBACK)) < 0) {
if ((error = snd_pcm_open_name (&playback_handle, "tones", SND_PCM_OPEN_PLAYBACK)) < 0) {
slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to open preferred audio driver\n");
return -error;
}
sample_handle = snd_pcm_file_descriptor( playback_handle, SND_PCM_CHANNEL_PLAYBACK );
if ((error = snd_pcm_plugin_set_disable (playback_handle, PLUGIN_DISABLE_MMAP)) < 0) {
slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to disable mmap\n");
snd_pcm_close(playback_handle);
return -error;
}
if ((error = snd_pcm_plugin_set_enable (playback_handle, PLUGIN_ROUTING)) < 0)
{
slogf ( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "snd_pcm_plugin_set_enable failed: %s\n", snd_strerror (error));
snd_pcm_close(playback_handle);
return -error;
}
// Find what sample rate and format to use
memset( &pi, 0, sizeof(pi) );
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((error = snd_pcm_channel_info (playback_handle, &pi)) < 0) {
slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to get plugin info\n");
snd_pcm_close(playback_handle);
return -error;
}
// Initialize the playback channel
memset(&pp, 0, sizeof(pp) );
pp.mode = SND_PCM_MODE_BLOCK;
pp.channel = SND_PCM_CHANNEL_PLAYBACK;
pp.start_mode = SND_PCM_START_FULL;
pp.stop_mode = SND_PCM_STOP_STOP;
pp.buf.block.frag_size = pi.max_fragment_size;
pp.buf.block.frags_max = 3;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.format = SND_PCM_SFMT_S16_LE;
pp.format.rate = 48000;
pp.format.voices = 2;
memset( &ps, 0, sizeof( ps ) );
memset( &group, 0, sizeof( group ) );
ps.channel = SND_PCM_CHANNEL_PLAYBACK;
ps.mixer_gid = &group.gid;
//strcpy(pp.sw_mixer_subchn_name, "Wave playback channel");
strcpy(pp.sw_mixer_subchn_name, "voice_ringtone");
if ((error = snd_pcm_plugin_params( playback_handle, &pp)) < 0)
{
slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to set plugin params\n");
snd_pcm_close(playback_handle);
return -error;
}
if ((error = snd_pcm_plugin_prepare (playback_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0) {
slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to set prepare plugin\n");
snd_pcm_close(playback_handle);
return -error;
}
if ((error = snd_pcm_plugin_setup (playback_handle, &ps)) < 0) {
slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to setup plugin\n");
snd_pcm_close(playback_handle);
return -error;
}
card = ps.mixer_card;
device = ps.mixer_device;
if( audioman_handle ) {
if ((error = snd_pcm_set_audioman_handle (playback_handle, audioman_handle)) < 0) {
slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to set audioman handle \n");
snd_pcm_close(playback_handle);
return -error;
}
}
sample_frequency = ps.format.rate;
frag_samples = ps.buf.block.frag_size / 4;
frame_size = ps.buf.block.frag_size;
format = pp.format.format;
/*
// Only support 16-bit samples for now
frag_buffer = malloc( ps.buf.block.frag_size * 2 );
if( frag_buffer == NULL ) {
slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to alloc frag buffer\n");
snd_pcm_close(playback_handle);
return ENOMEM;
}
*/
fprintf (stderr,"Playback Format %s card = %d\n", snd_pcm_get_format_name (ps.format.format),card);
fprintf (stderr,"Playback preferred frame_size %d \n", pi.max_fragment_size);
fprintf (stderr,"Playback frame_size %d \n", ps.buf.block.frag_size);
fprintf (stderr,"Playback frame_samples %d \n", frag_samples);
fprintf (stderr,"Playback Rate %d \n", ps.format.rate);
if (group.gid.name[0] == 0)
{
fprintf (stderr,"Playback Mixer Pcm Group [%s] Not Set \n", group.gid.name);
return -1;
}
fprintf (stderr, "Playback Mixer Pcm Group [%s]\n", group.gid.name);
return EOK;
}
