void OpenAudio()
{
int card = -1, dev = 0;
snd_pcm_channel_info_t pi;
snd_mixer_group_t group;
snd_pcm_channel_params_t pp;
snd_pcm_channel_setup_t setup;
mixlen = 2*AUDIO_CHANNELS*AUDIO_SAMPLES;
mixbuf = (uint8_t*)malloc(mixlen);
if (mixbuf == NULL)
return;
memset(mixbuf, 0, mixlen);
if ((snd_pcm_open_preferred(&pcm_handle, &card, &dev, SND_PCM_OPEN_PLAYBACK)) < 0)
return;
memset(&pi, 0, sizeof (pi));
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((snd_pcm_plugin_info (pcm_handle, &pi)) < 0)
return;
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 = -1;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.rate = AUDIO_FREQ;
pp.format.voices = AUDIO_CHANNELS;
pp.format.format = SND_PCM_SFMT_S16_LE;
if ((snd_pcm_plugin_params (pcm_handle, &pp)) < 0)
return;
snd_pcm_plugin_prepare(pcm_handle, SND_PCM_CHANNEL_PLAYBACK);
memset (&setup, 0, sizeof(setup));
memset (&group, 0, sizeof(group));
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
setup.mixer_gid = &group.gid;
if ((snd_pcm_plugin_setup (pcm_handle, &setup)) < 0)
return;
setup.buf.block.frag_size;
if ((snd_mixer_open(&mixer_handle, card, setup.mixer_device)) < 0)
return;
pthread_attr_t attr;
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
pthread_create(&thread_handle, &attr, &BlackberryAudio::staticThreadProc, this);
}
ad_rec_t *ad_open_sps (int32 sps)
{
ad_rec_t *handle;
struct snd_pcm_channel_info info;
snd_pcm_t *dspH;
int err;
int card;
int dev;
if (sps != DEFAULT_SAMPLES_PER_SEC) {
if(abs(sps - DEFAULT_SAMPLES_PER_SEC) <= SPS_EPSILON) {
fprintf(stderr, "Audio sampling rate %d is within %d of %d samples/sec\n",
sps, SPS_EPSILON, DEFAULT_SAMPLES_PER_SEC);
} else {
fprintf(stderr, "Audio sampling rate %d not supported; must be %d samples/sec\n",
sps, DEFAULT_SAMPLES_PER_SEC);
return NULL;
}
}
card = 0;
dev = 0;
err = snd_pcm_open(&dspH, card, dev, SND_PCM_OPEN_DUPLEX); /* XXX */
if (err < 0) {
fprintf(stderr, "Error opening audio device %d,%din full-duplex: %s\n",
card, dev, snd_strerror(err));
return NULL;
}
err = snd_pcm_plugin_info(dspH, &info);
if (err < 0) {
fprintf(stderr, "Error getting PCM plugin info: %s\n", snd_strerror(err));
return NULL;
}
if (setparams(sps, dspH) < 0) {
return NULL;
}
if (setlevels(0, 0) < 0) {
return NULL;
}
if ((handle = (ad_rec_t *) calloc (1, sizeof(ad_rec_t))) == NULL) {
fprintf(stderr, "calloc(%d) failed\n", sizeof(ad_rec_t));
abort();
}
handle->dspH = dspH;
handle->recording = 0;
handle->sps = sps;
handle->bps = sizeof(int16);
return(handle);
}
cst_audiodev *audio_open_alsa(int sps, int channels, cst_audiofmt fmt)
{
snd_pcm_channel_info_t pinfo;
snd_pcm_channel_params_t params;
snd_pcm_channel_setup_t setup;
snd_pcm_t *pcm;
cst_audiodev *ad;
int err;
#ifdef __QNXNTO__
if (snd_pcm_open_preferred(&pcm,&alsa_card,&alsa_device,SND_PCM_OPEN_PLAYBACK) < 0)
{
cst_errmsg("alsa_audio: failed to open audio device\n");
cst_error();
}
if (snd_pcm_plugin_set_disable(pcm,PLUGIN_DISABLE_MMAP) < 0)
{
cst_errmsg("alsa_audio: failed to disable mmap\n");
snd_pcm_close(pcm);
cst_error();
}
#else
if (snd_pcm_open(&pcm,alsa_card,alsa_device,SND_PCM_OPEN_PLAYBACK) < 0)
{
cst_errmsg("alsa_audio: failed to open audio device\n");
cst_error();
}
#endif
memset(&pinfo, 0, sizeof(pinfo));
memset(¶ms, 0, sizeof(params));
memset(&setup, 0, sizeof(setup));
pinfo.channel = SND_PCM_CHANNEL_PLAYBACK;
snd_pcm_plugin_info(pcm,&pinfo);
params.mode = SND_PCM_MODE_BLOCK;
params.channel = SND_PCM_CHANNEL_PLAYBACK;
params.start_mode = SND_PCM_START_DATA;
params.stop_mode = SND_PCM_STOP_STOP;
params.buf.block.frag_size = pinfo.max_fragment_size;
params.buf.block.frags_max = 1;
params.buf.block.frags_min = 1;
params.format.interleave = 1;
params.format.rate = sps;
params.format.voices = channels;
switch (fmt)
{
case CST_AUDIO_LINEAR16:
if (CST_LITTLE_ENDIAN)
params.format.format = SND_PCM_SFMT_S16_LE;
else
params.format.format = SND_PCM_SFMT_S16_BE;
break;
case CST_AUDIO_LINEAR8:
params.format.format = SND_PCM_SFMT_U8;
break;
case CST_AUDIO_MULAW:
params.format.format = SND_PCM_SFMT_MU_LAW;
break;
}
if((err = snd_pcm_plugin_params(pcm,¶ms)) < 0)
{
cst_errmsg("alsa_audio params setting failed: %s\n",snd_strerror(err));
snd_pcm_close(pcm);
cst_error();
}
if((err = snd_pcm_plugin_setup(pcm,SND_PCM_CHANNEL_PLAYBACK)) > 0) {
cst_errmsg("alsa_audio sound prepare setting failed: %s\n",snd_strerror(err));
snd_pcm_close(pcm);
cst_error();
}
if((err = snd_pcm_plugin_prepare(pcm,SND_PCM_CHANNEL_PLAYBACK)) > 0) {
cst_errmsg("alsa_audio sound prepare setting failed: %s\n",snd_strerror(err));
snd_pcm_close(pcm);
cst_error();
}
pinfo.channel = SND_PCM_CHANNEL_PLAYBACK;
snd_pcm_plugin_info(pcm,&pinfo);
ad = cst_alloc(cst_audiodev, 1);
ad->platform_data = pcm;
ad->sps = ad->real_sps = sps;
ad->channels = ad->real_channels = channels;
ad->fmt = ad->real_fmt = fmt;
return ad;
}
static pj_status_t bb10_open_capture (struct bb10_stream *stream,
const pjmedia_aud_param *param)
{
int ret = 0;
unsigned int rate;
unsigned long tmp_buf_size;
int frame_size;
snd_pcm_channel_info_t pi;
snd_mixer_group_t group;
snd_pcm_channel_params_t pp;
snd_pcm_channel_setup_t setup;
if (param->rec_id < 0 || param->rec_id >= stream->af->dev_cnt)
return PJMEDIA_EAUD_INVDEV;
PJ_ASSERT_RETURN(param->bits_per_sample == 16, PJMEDIA_EAUD_SAMPFORMAT);
if ((ret=audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT,
&stream->ca_pcm,
&stream->ca_audio_manager_handle,
(char*)"voice",
SND_PCM_OPEN_CAPTURE)) < 0)
{
TRACE_((THIS_FILE, "audio_manager_snd_pcm_open_name ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
/* Required call from January 2013 gold OS release */
snd_pcm_plugin_set_disable (stream->ca_pcm, PLUGIN_DISABLE_MMAP);
/* Required call from January 2013 gold OS release */
snd_pcm_plugin_set_enable(stream->ca_pcm, PLUGIN_ROUTING);
/* sample reads the capabilities of the capture */
memset (&pi, 0, sizeof (pi));
pi.channel = SND_PCM_CHANNEL_CAPTURE;
if ((ret = snd_pcm_plugin_info (stream->ca_pcm, &pi)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_info ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
/* Request the VoIP parameters
* These parameters are different to waverec sample
*/
memset (&pp, 0, sizeof (pp));
/* Blocking read */
pp.mode = SND_PCM_MODE_BLOCK;
pp.channel = SND_PCM_CHANNEL_CAPTURE;
pp.start_mode = SND_PCM_START_FULL;
/* Auto-recover from errors */
pp.stop_mode = SND_PCM_STOP_ROLLOVER;
pp.buf.block.frag_size = param->samples_per_frame * param->bits_per_sample / 8;
/* From January 2013 gold OS release. RIM recommend these for capture */
pp.buf.block.frags_max = 3;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.rate = param->clock_rate;
pp.format.voices = param->channel_count;
pp.format.format = get_alsa_pcm_fmt(param);
/* make the request */
if ((ret = snd_pcm_plugin_params (stream->ca_pcm, &pp)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_params ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
/* Again based on the sample */
memset (&setup, 0, sizeof (setup));
memset (&group, 0, sizeof (group));
setup.channel = SND_PCM_CHANNEL_CAPTURE;
setup.mixer_gid = &group.gid;
if ((ret = snd_pcm_plugin_setup (stream->ca_pcm, &setup)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_setup ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
frame_size = setup.buf.block.frag_size;
if (group.gid.name[0] == 0) {
} else {
}
frame_size = setup.buf.block.frag_size;
PJ_UNUSED_ARG(frame_size); /* Warning about unused var */
/* Set clock rate */
rate = param->clock_rate;
stream->ca_frames = (unsigned long) param->samples_per_frame /
param->channel_count;
/* Set the sound device buffer size and latency */
if (param->flags & PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY) {
tmp_buf_size = rate * param->input_latency_ms / 1000;
} else {
tmp_buf_size = rate * PJMEDIA_SND_DEFAULT_REC_LATENCY / 1000;
}
stream->param.input_latency_ms = tmp_buf_size * 1000 / rate;
/* Set our buffer */
stream->ca_buf_size = stream->ca_frames * param->channel_count *
param->bits_per_sample / 8;
stream->ca_buf = (char *)pj_pool_alloc (stream->pool, stream->ca_buf_size);
TRACE_((THIS_FILE, "bb10_open_capture: ca_frames = %d clock = %d",
stream->ca_frames, param->clock_rate));
return PJ_SUCCESS;
}
static pj_status_t bb10_open_playback (struct bb10_stream *stream,
const pjmedia_aud_param *param)
{
int ret = 0;
snd_pcm_channel_info_t pi;
snd_pcm_channel_setup_t setup;
snd_mixer_group_t group;
snd_pcm_channel_params_t pp;
unsigned int rate;
unsigned long tmp_buf_size;
if (param->play_id < 0 || param->play_id >= stream->af->dev_cnt) {
return PJMEDIA_EAUD_INVDEV;
}
PJ_ASSERT_RETURN(param->bits_per_sample == 16, PJMEDIA_EAUD_SAMPFORMAT);
/* Use the bb10 audio manager API to open as opposed to QNX core audio
* Echo cancellation built in
*/
if ((ret = audio_manager_snd_pcm_open_name(
AUDIO_TYPE_VIDEO_CHAT,
&stream->pb_pcm, &stream->pb_audio_manager_handle,
(char*)"voice",
SND_PCM_OPEN_PLAYBACK)) < 0)
{
TRACE_((THIS_FILE, "audio_manager_snd_pcm_open_name ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
/* Required call from January 2013 gold OS release */
snd_pcm_plugin_set_disable(stream->pb_pcm, PLUGIN_DISABLE_MMAP);
/* Required call from January 2013 gold OS release */
snd_pcm_plugin_set_enable(stream->pb_pcm, PLUGIN_ROUTING);
memset (&pi, 0, sizeof (pi));
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((ret = snd_pcm_plugin_info (stream->pb_pcm, &pi)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_info ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
memset (&pp, 0, sizeof (pp));
/* Request VoIP compatible capabilities */
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;
pp.buf.block.frag_size = param->samples_per_frame * param->bits_per_sample / 8;
/* RIM recommends maximum of 5 */
pp.buf.block.frags_max = 5;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.rate = param->clock_rate;
pp.format.voices = param->channel_count;
pp.format.format = get_alsa_pcm_fmt(param);
/* Make the calls as per the wave sample */
if ((ret = snd_pcm_plugin_params (stream->pb_pcm, &pp)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_params ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
memset (&setup, 0, sizeof (setup));
memset (&group, 0, sizeof (group));
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
setup.mixer_gid = &group.gid;
if ((ret = snd_pcm_plugin_setup (stream->pb_pcm, &setup)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_setup ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
if (group.gid.name[0] == 0) {
return PJMEDIA_EAUD_SYSERR;
}
rate = param->clock_rate;
/* Set the sound device buffer size and latency */
if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY) {
tmp_buf_size = rate * param->output_latency_ms / 1000;
} else {
tmp_buf_size = rate * PJMEDIA_SND_DEFAULT_PLAY_LATENCY / 1000;
}
/* Set period size to samples_per_frame frames. */
stream->pb_frames = param->samples_per_frame / param->channel_count;
stream->param.output_latency_ms = tmp_buf_size * 1000 / rate;
/* Set our buffer */
stream->pb_buf_size = stream->pb_frames * param->channel_count *
param->bits_per_sample / 8;
stream->pb_buf = (char *) pj_pool_alloc(stream->pool, stream->pb_buf_size);
TRACE_((THIS_FILE, "bb10_open_playback: pb_frames = %d clock = %d",
stream->pb_frames, param->clock_rate));
return PJ_SUCCESS;
}
static void playsetup() {
snd_pcm_channel_setup_t setup;
//Enabling Echo canceller
int ret;
snd_pcm_channel_info_t pi;
snd_mixer_group_t group;
snd_pcm_channel_params_t pp;
//audio routing enabled with AEC
if ((ret = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT,
&g_pcm_handle_p, &g_audio_manager_handle_p, (char*) "voice",
SND_PCM_OPEN_PLAYBACK)) < 0) {
return;
}
snd_pcm_plugin_set_disable(g_pcm_handle_p, PLUGIN_DISABLE_MMAP);
snd_pcm_plugin_set_enable(g_pcm_handle_p, PLUGIN_ROUTING);
memset(&pi, 0, sizeof(pi));
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((ret = snd_pcm_plugin_info(g_pcm_handle_p, &pi)) < 0) {
fprintf(stderr, "snd_pcm_plugin_info failed: %s\n", snd_strerror (ret));
return;
}
fprintf(stderr,"PLAY Minimum Rate = %d\n",pi.min_rate);
// Interestingly on the simulator this returns 4096 but in reality always 170 is the result
fprintf(stderr,"PLAY Minimum fragment size = %d\n",pi.min_fragment_size);
memset(&pp, 0, sizeof(pp));
// Request VoIP compatible capabilities
// On simulator frag_size is always negotiated to 170
pp.mode = SND_PCM_MODE_BLOCK;
pp.channel = SND_PCM_CHANNEL_PLAYBACK;
//pp.start_mode = SND_PCM_START_DATA;
pp.stop_mode = SND_PCM_STOP_ROLLOVER;
pp.start_mode = SND_PCM_START_FULL;
pp.buf.block.frag_size = PREFERRED_FRAME_SIZE;
// Increasing this internal buffer count delays write failure in the loop
pp.buf.block.frags_max = 5;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.rate = VOIP_SAMPLE_RATE;
pp.format.voices = 1;
pp.format.format = SND_PCM_SFMT_S16_LE;
// Make the calls as per the wave sample
if ((ret = snd_pcm_plugin_params(g_pcm_handle_p, &pp)) < 0) {
fprintf(stderr, "pb snd_pcm_plugin_params failed: %s\n", snd_strerror (ret));
return;
}
memset(&setup, 0, sizeof(setup));
memset(&group, 0, sizeof(group));
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
setup.mixer_gid = &group.gid;
if ((ret = snd_pcm_plugin_setup(g_pcm_handle_p, &setup)) < 0) {
fprintf(stderr, "snd_pcm_plugin_setup failed: %s\n", snd_strerror (ret));
return;
}
fprintf(stderr,"PLAY frame_size %d \n", setup.buf.block.frag_size);
fprintf(stderr,"PLAY Rate %d \n", setup.format.rate);
g_frame_size_p = setup.buf.block.frag_size;
if (group.gid.name[0] == 0) {
fprintf(stderr,"FATAL Mixer Pcm Group [%s] Not Set \n", group.gid.name);
exit (-1);
}
printf("Mixer Pcm Group [%s]\n", group.gid.name);
if ((ret = snd_pcm_plugin_prepare(g_pcm_handle_p, SND_PCM_CHANNEL_PLAYBACK))
< 0) {
fprintf(stderr, "snd_pcm_plugin_prepare failed: %s\n", snd_strerror (ret));
}
}
static void capturesetup() {
snd_pcm_channel_setup_t setup;
int ret;
snd_pcm_channel_info_t pi;
snd_mixer_group_t group;
snd_pcm_channel_params_t pp;
int card = setup.mixer_card;
if ((ret = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT, &g_pcm_handle_c,
&g_audio_manager_handle_c, (char*) "voice", SND_PCM_OPEN_CAPTURE)) < 0) {
return;
}
snd_pcm_plugin_set_disable(g_pcm_handle_c, PLUGIN_DISABLE_MMAP);
snd_pcm_plugin_set_enable(g_pcm_handle_c, PLUGIN_ROUTING);
// sample reads the capabilities of the capture
memset(&pi, 0, sizeof(pi));
pi.channel = SND_PCM_CHANNEL_CAPTURE;
if ((ret = snd_pcm_plugin_info(g_pcm_handle_c, &pi)) < 0) {
fprintf(stderr, "snd_pcm_plugin_info failed: %s\n", snd_strerror (ret));
return;
}
fprintf(stderr,"CAPTURE Minimum Rate = %d\n",pi.min_rate);
// Request the VoIP parameters
// These parameters are different to waverec sample
memset(&pp, 0, sizeof(pp));
fprintf(stderr,"CAPTURE Minimum fragment size = %d\n",pi.min_fragment_size);
// Blocking read
pp.mode = SND_PCM_MODE_BLOCK;
pp.channel = SND_PCM_CHANNEL_CAPTURE;
//pp.start_mode = SND_PCM_START_DATA;
// Auto-recover from errors
pp.stop_mode = SND_PCM_STOP_ROLLOVER;
pp.start_mode = SND_PCM_START_DATA;
pp.buf.block.frag_size = PREFERRED_FRAME_SIZE;
pp.buf.block.frags_max = 3;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.rate = VOIP_SAMPLE_RATE;
pp.format.voices = 1;
pp.format.format = SND_PCM_SFMT_S16_LE;
// make the request
if ((ret = snd_pcm_plugin_params(g_pcm_handle_c, &pp)) < 0) {
fprintf(stderr, "ca snd_pcm_plugin_params failed: %s\n", snd_strerror (ret));
return;
}
// Again based on the sample
memset(&setup, 0, sizeof(setup));
memset(&group, 0, sizeof(group));
setup.channel = SND_PCM_CHANNEL_CAPTURE;
setup.mixer_gid = &group.gid;
if ((ret = snd_pcm_plugin_setup(g_pcm_handle_c, &setup)) < 0) {
fprintf(stderr, "snd_pcm_plugin_setup failed: %s\n", snd_strerror (ret));
return;
}
// On the simulator at least, our requested capabilities are accepted.
fprintf(stderr,"CAPTURE Format %s card = %d\n", snd_pcm_get_format_name (setup.format.format),card);
fprintf(stderr,"CAPTURE Rate %d \n", setup.format.rate);
g_frame_size_c = setup.buf.block.frag_size;
if (group.gid.name[0] == 0) {
printf("Mixer Pcm Group [%s] Not Set \n", group.gid.name);
printf("***>>>> Input Gain Controls Disabled <<<<*** \n");
} else {
printf("Mixer Pcm Group [%s]\n", group.gid.name);
}
g_frame_size_c = setup.buf.block.frag_size;
fprintf(stderr, "CAPTURE frame_size = %d\n", g_frame_size_c);
// Sample calls prepare()
if ((ret = snd_pcm_plugin_prepare(g_pcm_handle_c, SND_PCM_CHANNEL_CAPTURE))
< 0) {
fprintf(stderr, "snd_pcm_plugin_prepare failed: %s\n", snd_strerror (ret));
}
}
void *Audio_ALSA::open (AudioConfig &cfg, const char *)
{
AudioConfig tmpCfg;
if (_audioHandle != NULL)
{
_errorString = "ERROR: Device already in use";
return NULL;
}
#ifdef HAVE_ALSA_ASOUNDLIB_H
snd_pcm_uframes_t buffer_frames;
snd_pcm_hw_params_t *hw_params = 0;
if (snd_pcm_open (&_audioHandle, "default", SND_PCM_STREAM_PLAYBACK, 0))
{
_errorString = "ERROR: Could not open audio device.";
goto open_error;
}
// May later be replaced with driver defaults.
tmpCfg = cfg;
if (snd_pcm_hw_params_malloc (&hw_params))
{
_errorString = "ERROR: could not malloc hwparams.";
goto open_error;
}
if (snd_pcm_hw_params_any (_audioHandle, hw_params))
{
_errorString = "ERROR: could not initialize hw params";
goto open_error;
}
if (snd_pcm_hw_params_set_access (_audioHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED))
{
_errorString = "ERROR: could not set access type";
goto open_error;
}
snd_pcm_format_t alsamode;
switch (tmpCfg.precision)
{
case 8:
tmpCfg.encoding = AUDIO_UNSIGNED_PCM;
alsamode = SND_PCM_FORMAT_U8;
break;
case 16:
tmpCfg.encoding = AUDIO_SIGNED_PCM;
alsamode = SND_PCM_FORMAT_S16;
break;
default:
_errorString = "ERROR: set desired number of bits for audio device.";
goto open_error;
}
if (snd_pcm_hw_params_set_format (_audioHandle, hw_params, alsamode))
{
_errorString = "ERROR: could not set sample format";
goto open_error;
}
if (snd_pcm_hw_params_set_channels (_audioHandle, hw_params, tmpCfg.channels))
{
_errorString = "ERROR: could not set channel count";
goto open_error;
}
{ // Gentoo bug #98769, comment 4
unsigned int rate = tmpCfg.frequency;
if (snd_pcm_hw_params_set_rate_near (_audioHandle, hw_params, &rate, 0))
{
_errorString = "ERROR: could not set sample rate";
goto open_error;
}
}
_alsa_to_frames_divisor = tmpCfg.channels * tmpCfg.precision / 8;
buffer_frames = 4096;
snd_pcm_hw_params_set_period_size_near(_audioHandle, hw_params, &buffer_frames, 0);
if (snd_pcm_hw_params (_audioHandle, hw_params))
{
_errorString = "ERROR: could not set hw parameters";
goto open_error;
}
snd_pcm_hw_params_free (hw_params);
hw_params = 0;
if (snd_pcm_prepare (_audioHandle))
{
_errorString = "ERROR: could not prepare audio interface for use";
goto open_error;
}
tmpCfg.bufSize = buffer_frames * _alsa_to_frames_divisor;
#else // HAVE_ALSA_ASOUNDLIB_H
// Obsolete interface
int mask, wantedFormat, format;
int rtn;
int card = -1, dev = 0;
if ((rtn = snd_pcm_open_preferred (&_audioHandle, &card, &dev, SND_PCM_OPEN_PLAYBACK)))
{
_errorString = "ERROR: Could not open audio device.";
goto open_error;
}
// Transfer input parameters to this object.
// May later be replaced with driver defaults.
tmpCfg = cfg;
snd_pcm_channel_params_t pp;
snd_pcm_channel_setup_t setup;
snd_pcm_channel_info_t pi;
memset (&pi, 0, sizeof (pi));
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((rtn = snd_pcm_plugin_info (_audioHandle, &pi)))
{
_errorString = "ALSA: snd_pcm_plugin_info failed.";
goto open_error;
}
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 = 1;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.rate = tmpCfg.frequency;
pp.format.voices = tmpCfg.channels;
// Set sample precision and type of encoding.
if ( tmpCfg.precision == 8 )
{
tmpCfg.encoding = AUDIO_UNSIGNED_PCM;
pp.format.format = SND_PCM_SFMT_U8;
}
if ( tmpCfg.precision == 16 )
{
tmpCfg.encoding = AUDIO_SIGNED_PCM;
pp.format.format = SND_PCM_SFMT_S16_LE;
}
if ((rtn = snd_pcm_plugin_params (_audioHandle, &pp)) < 0)
{
_errorString = "ALSA: snd_pcm_plugin_params failed.";
goto open_error;
}
if ((rtn = snd_pcm_plugin_prepare (_audioHandle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
_errorString = "ALSA: snd_pcm_plugin_prepare failed.";
goto open_error;
}
memset (&setup, 0, sizeof (setup));
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((rtn = snd_pcm_plugin_setup (_audioHandle, &setup)) < 0)
{
_errorString = "ALSA: snd_pcm_plugin_setup failed.";
goto open_error;
}
tmpCfg.bufSize = setup.buf.block.frag_size;
#endif
#ifdef HAVE_EXCEPTIONS
_sampleBuffer = new(std::nothrow) int_least8_t[tmpCfg.bufSize];
#else
_sampleBuffer = new int_least8_t[tmpCfg.bufSize];
#endif
if (!_sampleBuffer)
{
_errorString = "AUDIO: Unable to allocate memory for sample buffers.";
goto open_error;
}
// Setup internal Config
_settings = tmpCfg;
// Update the users settings
getConfig (cfg);
return _sampleBuffer;
open_error:
#ifdef HAVE_ALSA_ASOUNDLIB_H
if (hw_params)
snd_pcm_hw_params_free (hw_params);
#endif
if (_audioHandle != NULL)
close ();
perror ("ALSA");
return NULL;
}
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 pj_status_t bb10_open_capture (struct bb10_stream *stream,
const pjmedia_aud_param *param)
{
int ret = 0;
unsigned int rate;
unsigned long tmp_buf_size;
int card = -1;
int dev = 0;
int frame_size;
snd_pcm_channel_info_t pi;
snd_mixer_group_t group;
snd_pcm_channel_params_t pp;
snd_pcm_channel_setup_t setup;
unsigned int handle;
if (param->rec_id < 0 || param->rec_id >= stream->af->dev_cnt)
return PJMEDIA_EAUD_INVDEV;
#if PJ_BBSDK_VER >= 0x100006
if ((ret = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VOICE,
&stream->ca_pcm,
&handle,
"/dev/snd/voicec",
SND_PCM_OPEN_CAPTURE)) < 0)
{
TRACE_((THIS_FILE, "audio_manager_snd_pcm_open_name ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
#else
/* BB10 Audio init here (not prepare) */
PJ_UNUSED_ARG(handle);
if ((ret = snd_pcm_open_preferred (&stream->ca_pcm, &card, &dev,
SND_PCM_OPEN_CAPTURE)) < 0)
{
TRACE_((THIS_FILE, "snd_pcm_open_preferred ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
#endif
/* sample reads the capabilities of the capture */
memset (&pi, 0, sizeof (pi));
pi.channel = SND_PCM_CHANNEL_CAPTURE;
if ((ret = snd_pcm_plugin_info (stream->ca_pcm, &pi)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_info ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
/* Request the VoIP parameters
* These parameters are different to waverec sample
*/
memset (&pp, 0, sizeof (pp));
/* Blocking read */
pp.mode = SND_PCM_MODE_BLOCK;
pp.channel = SND_PCM_CHANNEL_CAPTURE;
pp.start_mode = SND_PCM_START_DATA;
/* Auto-recover from errors */
pp.stop_mode = SND_PCM_STOP_ROLLOVER;
/* HARD CODE for the time being PJMEDIA expects 640 for 16khz */
pp.buf.block.frag_size = PREFERRED_FRAME_SIZE*2;
/* Not applicable for capture hence -1 */
pp.buf.block.frags_max = -1;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
/* HARD CODE for the time being PJMEDIA expects 16khz */
PJ_TODO(REMOVE_SAMPLE_RATE_HARD_CODE);
pj_assert(param->clock_rate == VOIP_SAMPLE_RATE * 2);
pp.format.rate = VOIP_SAMPLE_RATE*2;
pp.format.voices = 1;
pp.format.format = SND_PCM_SFMT_S16_LE;
/* make the request */
if ((ret = snd_pcm_plugin_params (stream->ca_pcm, &pp)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_params ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
/* Again based on the sample */
memset (&setup, 0, sizeof (setup));
memset (&group, 0, sizeof (group));
setup.channel = SND_PCM_CHANNEL_CAPTURE;
setup.mixer_gid = &group.gid;
if ((ret = snd_pcm_plugin_setup (stream->ca_pcm, &setup)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_setup ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
frame_size = setup.buf.block.frag_size;
if (group.gid.name[0] == 0) {
} else {
}
/* frag_size should be 160 */
frame_size = setup.buf.block.frag_size;
/* END BB10 init */
/* Set clock rate */
rate = param->clock_rate;
stream->ca_frames = (unsigned long) param->samples_per_frame /
param->channel_count;
/* Set the sound device buffer size and latency */
if (param->flags & PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY) {
tmp_buf_size = (rate / 1000) * param->input_latency_ms;
} else {
tmp_buf_size = (rate / 1000) * PJMEDIA_SND_DEFAULT_REC_LATENCY;
}
stream->param.input_latency_ms = tmp_buf_size / (rate / 1000);
/* Set our buffer */
stream->ca_buf_size = stream->ca_frames * param->channel_count *
(param->bits_per_sample/8);
stream->ca_buf = (char *)pj_pool_alloc (stream->pool, stream->ca_buf_size);
TRACE_((THIS_FILE, "bb10_open_capture: ca_frames = %d clock = %d",
stream->ca_frames, param->clock_rate));
return PJ_SUCCESS;
}
static pj_status_t bb10_open_playback (struct bb10_stream *stream,
const pjmedia_aud_param *param)
{
int card = -1;
int dev = 0;
int ret = 0;
snd_pcm_channel_info_t pi;
snd_pcm_channel_setup_t setup;
snd_mixer_group_t group;
snd_pcm_channel_params_t pp;
unsigned int rate;
unsigned long tmp_buf_size;
unsigned int handle;
if (param->play_id < 0 || param->play_id >= stream->af->dev_cnt) {
return PJMEDIA_EAUD_INVDEV;
}
#if PJ_BBSDK_VER >= 0x100006
if ((ret = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VOICE,
&stream->pb_pcm, &handle,
"/dev/snd/voicep",
SND_PCM_OPEN_PLAYBACK)) < 0)
{
TRACE_((THIS_FILE, "audio_manager_snd_pcm_open_name ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
#else
if ((ret = snd_pcm_open_preferred (&stream->pb_pcm, &card, &dev,
SND_PCM_OPEN_PLAYBACK)) < 0)
{
TRACE_((THIS_FILE, "snd_pcm_open_preferred ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
#endif
/* TODO PJ_ZERO */
memset (&pi, 0, sizeof (pi));
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((ret = snd_pcm_plugin_info (stream->pb_pcm, &pi)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_info ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
memset (&pp, 0, sizeof (pp));
/* Request VoIP compatible capabilities
* On simulator frag_size is always negotiated to 170
*/
pp.mode = SND_PCM_MODE_BLOCK;
pp.channel = SND_PCM_CHANNEL_PLAYBACK;
pp.start_mode = SND_PCM_START_DATA;
pp.stop_mode = SND_PCM_STOP_ROLLOVER;
/* HARD CODE for the time being PJMEDIA expects 640 for 16khz */
pp.buf.block.frag_size = PREFERRED_FRAME_SIZE*2;
/* Increasing this internal buffer count delays write failure in the loop */
pp.buf.block.frags_max = 4;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
/* HARD CODE for the time being PJMEDIA expects 16khz */
PJ_TODO(REMOVE_SAMPLE_RATE_HARD_CODE);
pj_assert(param->clock_rate == VOIP_SAMPLE_RATE * 2);
pp.format.rate = VOIP_SAMPLE_RATE*2;
pp.format.voices = 1;
pp.format.format = SND_PCM_SFMT_S16_LE;
/* Make the calls as per the wave sample */
if ((ret = snd_pcm_plugin_params (stream->pb_pcm, &pp)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_params ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
memset (&setup, 0, sizeof (setup));
memset (&group, 0, sizeof (group));
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
setup.mixer_gid = &group.gid;
if ((ret = snd_pcm_plugin_setup (stream->pb_pcm, &setup)) < 0) {
TRACE_((THIS_FILE, "snd_pcm_plugin_setup ret = %d", ret));
return PJMEDIA_EAUD_SYSERR;
}
if (group.gid.name[0] == 0) {
return PJMEDIA_EAUD_SYSERR;
}
rate = param->clock_rate;
/* Set the sound device buffer size and latency */
if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY) {
tmp_buf_size = (rate / 1000) * param->output_latency_ms;
} else {
tmp_buf_size = (rate / 1000) * PJMEDIA_SND_DEFAULT_PLAY_LATENCY;
}
/* Set period size to samples_per_frame frames. */
stream->pb_frames = param->samples_per_frame;
stream->param.output_latency_ms = tmp_buf_size / (rate / 1000);
/* Set our buffer */
stream->pb_buf_size = stream->pb_frames * param->channel_count *
(param->bits_per_sample/8);
stream->pb_buf = (char *) pj_pool_alloc(stream->pool, stream->pb_buf_size);
TRACE_((THIS_FILE, "bb10_open_playback: pb_frames = %d clock = %d",
stream->pb_frames, param->clock_rate));
return PJ_SUCCESS;
}
Error AudioDriverBB10::init(const char *p_name) {
active = false;
thread_exited = false;
exit_thread = false;
pcm_open = false;
samples_in = NULL;
samples_out = NULL;
mix_rate = 44100;
speaker_mode = SPEAKER_MODE_STEREO;
char *dev_name;
if (p_name == NULL) {
dev_name = "pcmPreferred";
} else {
dev_name = (char *)p_name;
}
printf("******** reconnecting to device %s\n", dev_name);
int ret = snd_pcm_open_name(&pcm_handle, dev_name, SND_PCM_OPEN_PLAYBACK);
ERR_FAIL_COND_V(ret < 0, FAILED);
pcm_open = true;
snd_pcm_channel_info_t cinfo;
zeromem(&cinfo, sizeof(cinfo));
cinfo.channel = SND_PCM_CHANNEL_PLAYBACK;
snd_pcm_plugin_info(pcm_handle, &cinfo);
printf("rates %i, %i, %i, %i, %i\n", cinfo.rates, cinfo.rates & SND_PCM_RATE_44100, cinfo.rates & SND_PCM_RATE_32000, cinfo.rates & SND_PCM_RATE_22050, cinfo.max_rate);
mix_rate = cinfo.max_rate;
printf("formats %i, %i, %i\n", cinfo.formats, cinfo.formats & SND_PCM_FMT_S16_BE, cinfo.formats & SND_PCM_FMT_S16_LE);
ERR_FAIL_COND_V(!(cinfo.formats & SND_PCM_FMT_S16_LE), FAILED);
printf("voices %i\n", cinfo.max_voices);
speaker_mode = SPEAKER_MODE_STEREO;
snd_pcm_channel_params_t cp;
zeromem(&cp, sizeof(cp));
cp.mode = SND_PCM_MODE_BLOCK;
cp.channel = SND_PCM_CHANNEL_PLAYBACK;
cp.start_mode = SND_PCM_START_DATA;
cp.stop_mode = SND_PCM_STOP_STOP;
//cp.buf.block.frag_size = cinfo.max_fragment_size;
cp.buf.block.frag_size = 512;
cp.buf.block.frags_max = 1;
cp.buf.block.frags_min = 1;
cp.format.interleave = 1;
cp.format.rate = mix_rate;
cp.format.voices = speaker_mode;
cp.format.format = SND_PCM_SFMT_S16_LE;
ret = snd_pcm_plugin_params(pcm_handle, &cp);
printf("ret is %i, %i\n", ret, cp.why_failed);
ERR_FAIL_COND_V(ret < 0, FAILED);
ret = snd_pcm_plugin_prepare(pcm_handle, SND_PCM_CHANNEL_PLAYBACK);
ERR_FAIL_COND_V(ret < 0, FAILED);
snd_mixer_group_t group;
zeromem(&group, sizeof(group));
snd_pcm_channel_setup_t setup;
zeromem(&setup, sizeof(setup));
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
setup.mode = SND_PCM_MODE_BLOCK;
setup.mixer_gid = &group.gid;
ret = snd_pcm_plugin_setup(pcm_handle, &setup);
ERR_FAIL_COND_V(ret < 0, FAILED);
pcm_frag_size = setup.buf.block.frag_size;
pcm_max_frags = 1;
sample_buf_count = pcm_frag_size * pcm_max_frags / 2;
printf("sample count %i, %i, %i\n", sample_buf_count, pcm_frag_size, pcm_max_frags);
samples_in = memnew_arr(int32_t, sample_buf_count);
samples_out = memnew_arr(int16_t, sample_buf_count);
thread = Thread::create(AudioDriverBB10::thread_func, this);
return OK;
};
/*
* General setup of the libasound audio mixer and pcm components.
* Some of the settings are based on the format of the wav data.
*/
int
setup_snd(const char * name)
{
int fragsize = -1;
int num_frags = -1;
int rtn;
char *dev_name;
if (NULL == name) {
dev_name = "pcmPreferred";
} else {
dev_name = (char *) name;
}
if ((rtn = snd_pcm_open_name(&pcm_handle, dev_name, SND_PCM_OPEN_PLAYBACK)) < 0) {
snprintf(msg, MSG_SIZE, "snd_pcm_open_name failed: %s\n", snd_strerror(rtn));
show_dialog_message(msg);
return FAILURE;
}
if ((rtn = snd_pcm_info(pcm_handle, &info)) < 0) {
snprintf(msg, MSG_SIZE, "snd_pcm_info failed: %s\n", snd_strerror(rtn));
goto setup_failure;
}
card = info.card;
/* 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) {
snprintf(msg, MSG_SIZE, "snd_pcm_plugin_set_disable failed: %s\n", snd_strerror(rtn));
goto setup_failure;
}
memset(&pi, 0, sizeof(pi));
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((rtn = snd_pcm_plugin_info(pcm_handle, &pi)) < 0) {
snprintf(msg, MSG_SIZE, "snd_pcm_plugin_info failed: %s\n", snd_strerror(rtn));
goto setup_failure;
}
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) {
snprintf(msg, MSG_SIZE, "snd_pcm_plugin_params failed: %s\n", snd_strerror(rtn));
goto setup_failure;
}
if ((rtn = snd_pcm_plugin_prepare(pcm_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0) {
snprintf(msg, MSG_SIZE, "snd_pcm_plugin_prepare failed: %s\n", snd_strerror(rtn));
goto setup_failure;
}
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) {
snprintf(msg, MSG_SIZE, "snd_pcm_plugin_setup failed: %s\n", snd_strerror(rtn));
goto setup_failure;
}
if (group.gid.name[0] == 0) {
snprintf(msg, MSG_SIZE, "Mixer Pcm Group [%s] Not Set \n", group.gid.name);
goto setup_failure;
}
if ((rtn = snd_mixer_open(&mixer_handle, card, setup.mixer_device)) < 0) {
snprintf(msg, MSG_SIZE, "snd_mixer_open failed: %s\n", snd_strerror(rtn));
goto setup_failure;
}
char tmp[MSG_SIZE];
snprintf(msg, MSG_SIZE, "Format %s \n", snd_pcm_get_format_name(setup.format.format));
snprintf(tmp, MSG_SIZE, "Frag Size %d \n", setup.buf.block.frag_size);
strlcat(msg, tmp, MSG_SIZE);
snprintf(tmp, MSG_SIZE, "Total Frags %d \n", setup.buf.block.frags);
strlcat(msg, tmp, MSG_SIZE);
snprintf(tmp, MSG_SIZE, "Rate %d \n", setup.format.rate);
strlcat(msg, tmp, MSG_SIZE);
snprintf(tmp, MSG_SIZE, "Voices %d \n", setup.format.voices);
strlcat(msg, tmp, MSG_SIZE);
snprintf(tmp, MSG_SIZE, "Mixer Pcm Group [%s]\n", group.gid.name);
strlcat(msg, tmp, MSG_SIZE);
show_dialog_message(msg);
return SUCCESS;
setup_failure:
show_dialog_message(msg);
snd_pcm_close(pcm_handle);
return FAILURE;
}
/* public methods (static but exported through the sysdep_dsp or plugin
struct) */
static void *alsa_dsp_create(const void *flags)
{
int i, j;
// audio_buf_info info;
struct alsa_dsp_priv_data *priv = NULL;
struct sysdep_dsp_struct *dsp = NULL;
const struct sysdep_dsp_create_params *params = flags;
const char *device = params->device;
int err;
int bytespersample;
fprintf(stderr,"info: dsp_create called\n");
/* allocate the dsp struct */
if (!(dsp = calloc(1, sizeof(struct sysdep_dsp_struct))))
{
fprintf(stderr,
"error: malloc failed for struct sysdep_dsp_struct\n");
return NULL;
}
/* alloc private data */
if(!(priv = calloc(1, sizeof(struct alsa_dsp_priv_data))))
{
fprintf(stderr,
"error: malloc failed for struct dsp_priv_data\n");
alsa_dsp_destroy(dsp);
return NULL;
}
/* fill in the functions and some data */
dsp->_priv = priv;
dsp->get_freespace = alsa_dsp_get_freespace;
dsp->write = alsa_dsp_write;
dsp->destroy = alsa_dsp_destroy;
dsp->hw_info.type = params->type;
dsp->hw_info.samplerate = params->samplerate;
priv->audio_dev.bMute = 0;
priv->audio_dev.m_AudioHandle = NULL;
priv->audio_dev.m_MixerHandle = NULL;
priv->audio_dev.m_Acard = 0;
priv->audio_dev.m_Adevice = 0;
if (preferred_device)
{
if((err = snd_pcm_open_preferred(&(priv->audio_dev.m_AudioHandle), &priv->audio_dev.m_Acard,
&priv->audio_dev.m_Adevice, SND_PCM_OPEN_PLAYBACK)) < 0)
{
fprintf(stderr,"info: snd_pcm_open_preferred failed: %s \n", snd_strerror(err));
alsa_dsp_destroy(dsp);
return NULL;
}
}
else
{
fprintf(stderr,"info: audio is using primary device\n");
if((err = snd_pcm_open(&(priv->audio_dev.m_AudioHandle), 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
{
fprintf(stderr,"info: snd_pcm_open failed: %s \n", snd_strerror(err));
alsa_dsp_destroy(dsp);
return NULL;
}
}
memset (&(priv->audio_dev.m_Achaninfo), 0, sizeof (priv->audio_dev.m_Achaninfo));
priv->audio_dev.m_Achaninfo.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((err = snd_pcm_plugin_info (priv->audio_dev.m_AudioHandle, &(priv->audio_dev.m_Achaninfo))) < 0)
{
fprintf (stderr, "info: snd_pcm_plugin_info failed: %s\n", snd_strerror (err));
alsa_dsp_destroy(dsp);
return NULL;
}
//needed to enable the count status parameter, mmap plugin disables this
if((err = snd_plugin_set_disable(priv->audio_dev.m_AudioHandle, PLUGIN_DISABLE_MMAP)) < 0)
{
fprintf (stderr, "info: snd_plugin_set_disable failed: %s\n", snd_strerror (err));
alsa_dsp_destroy(dsp);
return NULL;
}
/* calculate and set the fragsize & number of frags */
/* fragsize (as power of 2) */
i = 8;
if (dsp->hw_info.type & SYSDEP_DSP_16BIT) i++;
if (dsp->hw_info.type & SYSDEP_DSP_STEREO) i++;
i += dsp->hw_info.samplerate / 22000;
/* number of frags */
j = ((dsp->hw_info.samplerate * alsa_dsp_bytes_per_sample[dsp->hw_info.type] *
params->bufsize) / (0x01 << i)) + 1;
bytespersample=1;
// dsp->hw_info.type &= ~SYSDEP_DSP_16BIT;
// dsp->hw_info.type &= ~SYSDEP_DSP_STEREO;
if (dsp->hw_info.type & SYSDEP_DSP_16BIT) bytespersample++;
if (dsp->hw_info.type & SYSDEP_DSP_STEREO) bytespersample <<= 1;
memset( &(priv->audio_dev.m_Aparams), 0, sizeof(priv->audio_dev.m_Aparams));
priv->audio_dev.m_Aparams.mode = SND_PCM_MODE_BLOCK;
priv->audio_dev.m_Aparams.channel = SND_PCM_CHANNEL_PLAYBACK;
priv->audio_dev.m_Aparams.start_mode = SND_PCM_START_FULL;
priv->audio_dev.m_Aparams.stop_mode = SND_PCM_STOP_ROLLOVER;
#if 0
priv->audio_dev.m_Aparams.buf.stream.queue_size = 512 * bytespersample;
priv->audio_dev.m_Aparams.buf.stream.fill = SND_PCM_FILL_SILENCE;
priv->audio_dev.m_Aparams.buf.stream.max_fill = 512 * bytespersample;
#endif
priv->audio_dev.m_Aparams.format.interleave = 1;
priv->audio_dev.m_Aparams.format.rate = dsp->hw_info.samplerate;
priv->audio_dev.m_Aparams.format.voices = (dsp->hw_info.type & SYSDEP_DSP_STEREO) ? 2 : 1;
priv->audio_dev.m_Aparams.buf.block.frag_size = 1000;
priv->audio_dev.m_Aparams.buf.block.frags_min = 1;
priv->audio_dev.m_Aparams.buf.block.frags_max = 5;
priv->audio_dev.m_BytesPerSample = bytespersample;
priv->audio_dev.m_Aparams.format.format =
#ifdef LSB_FIRST
(dsp->hw_info.type & SYSDEP_DSP_16BIT) ? SND_PCM_SFMT_S16_LE : SND_PCM_SFMT_U8;
#else
(dsp->hw_info.type & SYSDEP_DSP_16BIT) ? SND_PCM_SFMT_S16_BE : SND_PCM_SFMT_U8;
#endif
if ((err = snd_pcm_plugin_params (priv->audio_dev.m_AudioHandle, &(priv->audio_dev.m_Aparams))) < 0)
{
fprintf (stderr, "info: snd_pcm_plugin_params failed: %s\n", snd_strerror (err));
alsa_dsp_destroy(dsp);
return NULL;
}
if ((err = snd_pcm_plugin_prepare (priv->audio_dev.m_AudioHandle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
fprintf (stderr, "warning: snd_pcm_plugin_prepare failed: %s\n", snd_strerror (err));
}
memset (&(priv->audio_dev.m_Asetup), 0, sizeof (priv->audio_dev.m_Asetup));
priv->audio_dev.m_Asetup.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((err = snd_pcm_plugin_setup (priv->audio_dev.m_AudioHandle, &(priv->audio_dev.m_Asetup))) < 0)
{
fprintf (stderr, "warning: snd_pcm_plugin_setup failed: %s\n", snd_strerror (err));
alsa_dsp_destroy(dsp);
return NULL;
}
memset (&(priv->audio_dev.m_Astatus), 0, sizeof (priv->audio_dev.m_Astatus));
priv->audio_dev.m_Astatus.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((err = snd_pcm_plugin_status (priv->audio_dev.m_AudioHandle, &(priv->audio_dev.m_Astatus))) < 0)
{
fprintf (stderr, "warning: snd_pcm_plugin_status failed: %s\n", snd_strerror (err));
}
dsp->hw_info.bufsize = priv->audio_dev.m_Asetup.buf.stream.queue_size
/ alsa_dsp_bytes_per_sample[dsp->hw_info.type];
#if 0
if ((err=snd_pcm_nonblock_mode(priv->audio_dev.m_AudioHandle, 1))<0)
{
fprintf(stderr, "error: error with non block mode: %s\n", snd_strerror (err));
}
#endif
return dsp;
}
void *Audio_ALSA::open (AudioConfig &cfg, const char *)
{
AudioConfig tmpCfg;
int mask, wantedFormat, format;
int rtn;
int card = -1, dev = 0;
if (_audioHandle != NULL)
{
_errorString = "ERROR: Device already in use";
return NULL;
}
if ((rtn = snd_pcm_open_preferred (&_audioHandle, &card, &dev, SND_PCM_OPEN_PLAYBACK)))
{
_errorString = "ERROR: Could not open audio device.";
goto open_error;
}
// Transfer input parameters to this object.
// May later be replaced with driver defaults.
tmpCfg = cfg;
snd_pcm_channel_params_t pp;
snd_pcm_channel_setup_t setup;
snd_pcm_channel_info_t pi;
memset (&pi, 0, sizeof (pi));
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((rtn = snd_pcm_plugin_info (_audioHandle, &pi)))
{
_errorString = "ALSA: snd_pcm_plugin_info failed.";
goto open_error;
}
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 = 1;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.rate = tmpCfg.frequency;
pp.format.voices = tmpCfg.channels;
// Set sample precision and type of encoding.
if ( tmpCfg.precision == 8 )
{
tmpCfg.encoding = AUDIO_UNSIGNED_PCM;
pp.format.format = SND_PCM_SFMT_U8;
}
if ( tmpCfg.precision == 16 )
{
tmpCfg.encoding = AUDIO_SIGNED_PCM;
pp.format.format = SND_PCM_SFMT_S16_LE;
}
if ((rtn = snd_pcm_plugin_params (_audioHandle, &pp)) < 0)
{
_errorString = "ALSA: snd_pcm_plugin_params failed.";
goto open_error;
}
if ((rtn = snd_pcm_plugin_prepare (_audioHandle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
_errorString = "ALSA: snd_pcm_plugin_prepare failed.";
goto open_error;
}
memset (&setup, 0, sizeof (setup));
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((rtn = snd_pcm_plugin_setup (_audioHandle, &setup)) < 0)
{
_errorString = "ALSA: snd_pcm_plugin_setup failed.";
goto open_error;
}
tmpCfg.bufSize = setup.buf.block.frag_size;
#ifdef HAVE_EXCEPTIONS
_sampleBuffer = new(std::nothrow) int_least8_t[tmpCfg.bufSize];
#else
_sampleBuffer = new int_least8_t[tmpCfg.bufSize];
#endif
if (!_sampleBuffer)
{
_errorString = "AUDIO: Unable to allocate memory for sample buffers.";
goto open_error;
}
// Setup internal Config
_settings = tmpCfg;
// Update the users settings
getConfig (cfg);
return _sampleBuffer;
open_error:
if (_audioHandle != NULL)
{
close ();
}
perror ("ALSA");
return NULL;
}
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;
}
/*****************************************************************************
* Open : creates a handle and opens an alsa device
*****************************************************************************
* This function opens an alsa device, through the alsa API
*****************************************************************************/
int E_(OpenAudio)( vlc_object_t *p_this )
{
aout_instance_t *p_aout = (aout_instance_t *)p_this;
int i_ret;
int i_bytes_per_sample;
int i_nb_channels;
snd_pcm_channel_info_t pi;
snd_pcm_channel_params_t pp;
aout_instance_t *p_aout = (aout_instance_t *)p_this;
/* allocate structure */
p_aout->output.p_sys = malloc( sizeof( aout_sys_t ) );
if( p_aout->output.p_sys == NULL )
{
msg_Err( p_aout, "out of memory" );
return -1;
}
/* open audio device */
if( ( i_ret = snd_pcm_open_preferred( &p_aout->output.p_sys->p_pcm_handle,
&p_aout->output.p_sys->i_card,
&p_aout->output.p_sys->i_device,
SND_PCM_OPEN_PLAYBACK ) ) < 0 )
{
msg_Err( p_aout, "unable to open audio device (%s)",
snd_strerror( i_ret ) );
free( p_aout->output.p_sys );
return -1;
}
/* disable mmap */
if( ( i_ret = snd_pcm_plugin_set_disable( p_aout->output.p_sys->p_pcm_handle,
PLUGIN_DISABLE_MMAP ) ) < 0 )
{
msg_Err( p_aout, "unable to disable mmap (%s)", snd_strerror(i_ret) );
E_(CloseAudio)( p_this );
free( p_aout->output.p_sys );
return -1;
}
p_aout->output.p_sys->p_silent_buffer = malloc( DEFAULT_FRAME_SIZE * 4 );
p_aout->output.pf_play = Play;
aout_VolumeSoftInit( p_aout );
memset( &pi, 0, sizeof(pi) );
memset( &pp, 0, sizeof(pp) );
pi.channel = SND_PCM_CHANNEL_PLAYBACK;
if( ( i_ret = snd_pcm_plugin_info( p_aout->output.p_sys->p_pcm_handle,
&pi ) ) < 0 )
{
msg_Err( p_aout, "unable to get plugin info (%s)",
snd_strerror( i_ret ) );
E_(CloseAudio)( p_this );
free( p_aout->output.p_sys );
return -1;
}
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.frags_max = 3;
pp.buf.block.frags_min = 1;
pp.format.interleave = 1;
pp.format.rate = p_aout->output.output.i_rate;
i_nb_channels = aout_FormatNbChannels( &p_aout->output.output );
if ( i_nb_channels > 2 )
{
/* I don't know if QNX supports more than two channels. */
i_nb_channels = 2;
p_aout->output.output.i_channels = AOUT_CHAN_STEREO;
}
pp.format.voices = i_nb_channels;
p_aout->output.output.i_format = AOUT_FMT_S16_NE;
p_aout->output.i_nb_samples = DEFAULT_FRAME_SIZE;
pp.format.format = SND_PCM_SFMT_S16;
i_bytes_per_sample = 2;
pp.buf.block.frag_size = p_aout->output.i_nb_samples *
p_aout->output.output.i_channels *
i_bytes_per_sample;
/* set parameters */
if( ( i_ret = snd_pcm_plugin_params( p_aout->output.p_sys->p_pcm_handle,
&pp ) ) < 0 )
{
msg_Err( p_aout, "unable to set parameters (%s)", snd_strerror(i_ret) );
E_(CloseAudio)( p_this );
free( p_aout->output.p_sys );
return -1;
}
/* prepare channel */
if( ( i_ret = snd_pcm_plugin_prepare( p_aout->output.p_sys->p_pcm_handle,
SND_PCM_CHANNEL_PLAYBACK ) ) < 0 )
{
msg_Err( p_aout, "unable to prepare channel (%s)",
snd_strerror( i_ret ) );
E_(CloseAudio)( p_this );
free( p_aout->output.p_sys );
return -1;
}
/* Create audio thread and wait for its readiness. */
if( vlc_thread_create( p_aout, "aout", QNXaoutThread,
VLC_THREAD_PRIORITY_OUTPUT, VLC_FALSE ) )
{
msg_Err( p_aout, "cannot create QNX audio thread (%s)", strerror(errno) );
E_(CloseAudio)( p_this );
free( p_aout->output.p_sys );
return -1;
}
return( 0 );
}
