本文整理汇总了C++中snd_pcm_sw_params_set_start_threshold函数的典型用法代码示例。如果您正苦于以下问题：C++ snd_pcm_sw_params_set_start_threshold函数的具体用法？C++ snd_pcm_sw_params_set_start_threshold怎么用？C++ snd_pcm_sw_params_set_start_threshold使用的例子？那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。
在下文中一共展示了snd_pcm_sw_params_set_start_threshold函数的20个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于我们的系统推荐出更棒的C++代码示例。
示例1: sizeof
static void *alsa_init(const char *device, unsigned rate, unsigned latency)
{
alsa_t *alsa = (alsa_t*)calloc(1, sizeof(alsa_t));
if (!alsa)
return NULL;
snd_pcm_hw_params_t *params = NULL;
snd_pcm_sw_params_t *sw_params = NULL;
unsigned latency_usec = latency * 1000;
unsigned channels = 2;
unsigned periods = 4;
snd_pcm_format_t format;
const char *alsa_dev = "default";
if (device)
alsa_dev = device;
snd_pcm_uframes_t buffer_size;
TRY_ALSA(snd_pcm_open(&alsa->pcm, alsa_dev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK));
TRY_ALSA(snd_pcm_hw_params_malloc(¶ms));
alsa->has_float = find_float_format(alsa->pcm, params);
format = alsa->has_float ? SND_PCM_FORMAT_FLOAT : SND_PCM_FORMAT_S16;
TRY_ALSA(snd_pcm_hw_params_any(alsa->pcm, params));
TRY_ALSA(snd_pcm_hw_params_set_access(alsa->pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED));
TRY_ALSA(snd_pcm_hw_params_set_format(alsa->pcm, params, format));
TRY_ALSA(snd_pcm_hw_params_set_channels(alsa->pcm, params, channels));
TRY_ALSA(snd_pcm_hw_params_set_rate(alsa->pcm, params, rate, 0));
TRY_ALSA(snd_pcm_hw_params_set_buffer_time_near(alsa->pcm, params, &latency_usec, NULL));
TRY_ALSA(snd_pcm_hw_params_set_periods_near(alsa->pcm, params, &periods, NULL));
TRY_ALSA(snd_pcm_hw_params(alsa->pcm, params));
snd_pcm_hw_params_get_period_size(params, &buffer_size, NULL);
RARCH_LOG("ALSA: Period size: %d frames\n", (int)buffer_size);
snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
RARCH_LOG("ALSA: Buffer size: %d frames\n", (int)buffer_size);
alsa->buffer_size = snd_pcm_frames_to_bytes(alsa->pcm, buffer_size);
alsa->can_pause = snd_pcm_hw_params_can_pause(params);
RARCH_LOG("ALSA: Can pause: %s.\n", alsa->can_pause ? "yes" : "no");
TRY_ALSA(snd_pcm_sw_params_malloc(&sw_params));
TRY_ALSA(snd_pcm_sw_params_current(alsa->pcm, sw_params));
TRY_ALSA(snd_pcm_sw_params_set_start_threshold(alsa->pcm, sw_params, buffer_size / 2));
TRY_ALSA(snd_pcm_sw_params(alsa->pcm, sw_params));
snd_pcm_hw_params_free(params);
snd_pcm_sw_params_free(sw_params);
return alsa;
error:
RARCH_ERR("ALSA: Failed to initialize...\n");
if (params)
snd_pcm_hw_params_free(params);
if (sw_params)
snd_pcm_sw_params_free(sw_params);
if (alsa)
{
if (alsa->pcm)
snd_pcm_close(alsa->pcm);
free(alsa);
}
return NULL;
}
开发者ID:DonelBueno，项目名称:iOS，代码行数:71，代码来源:alsa.c
示例2: alsa_configure
static void
alsa_configure (struct sound_device *sd)
{
int val, err, dir;
unsigned uval;
struct alsa_params *p = (struct alsa_params *) sd->data;
snd_pcm_uframes_t buffer_size;
xassert (p->handle != 0);
err = snd_pcm_hw_params_malloc (&p->hwparams);
if (err < 0)
alsa_sound_perror ("Could not allocate hardware parameter structure", err);
err = snd_pcm_sw_params_malloc (&p->swparams);
if (err < 0)
alsa_sound_perror ("Could not allocate software parameter structure", err);
err = snd_pcm_hw_params_any (p->handle, p->hwparams);
if (err < 0)
alsa_sound_perror ("Could not initialize hardware parameter structure", err);
err = snd_pcm_hw_params_set_access (p->handle, p->hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
alsa_sound_perror ("Could not set access type", err);
val = sd->format;
err = snd_pcm_hw_params_set_format (p->handle, p->hwparams, val);
if (err < 0)
alsa_sound_perror ("Could not set sound format", err);
uval = sd->sample_rate;
err = snd_pcm_hw_params_set_rate_near (p->handle, p->hwparams, &uval, 0);
if (err < 0)
alsa_sound_perror ("Could not set sample rate", err);
val = sd->channels;
err = snd_pcm_hw_params_set_channels (p->handle, p->hwparams, val);
if (err < 0)
alsa_sound_perror ("Could not set channel count", err);
err = snd_pcm_hw_params (p->handle, p->hwparams);
if (err < 0)
alsa_sound_perror ("Could not set parameters", err);
err = snd_pcm_hw_params_get_period_size (p->hwparams, &p->period_size, &dir);
if (err < 0)
alsa_sound_perror ("Unable to get period size for playback", err);
err = snd_pcm_hw_params_get_buffer_size (p->hwparams, &buffer_size);
if (err < 0)
alsa_sound_perror ("Unable to get buffer size for playback", err);
err = snd_pcm_sw_params_current (p->handle, p->swparams);
if (err < 0)
alsa_sound_perror ("Unable to determine current swparams for playback",
err);
/* Start the transfer when the buffer is almost full */
err = snd_pcm_sw_params_set_start_threshold (p->handle, p->swparams,
(buffer_size / p->period_size)
* p->period_size);
if (err < 0)
alsa_sound_perror ("Unable to set start threshold mode for playback", err);
/* Allow the transfer when at least period_size samples can be processed */
err = snd_pcm_sw_params_set_avail_min (p->handle, p->swparams, p->period_size);
if (err < 0)
alsa_sound_perror ("Unable to set avail min for playback", err);
err = snd_pcm_sw_params (p->handle, p->swparams);
if (err < 0)
alsa_sound_perror ("Unable to set sw params for playback\n", err);
snd_pcm_hw_params_free (p->hwparams);
p->hwparams = NULL;
snd_pcm_sw_params_free (p->swparams);
p->swparams = NULL;
err = snd_pcm_prepare (p->handle);
if (err < 0)
alsa_sound_perror ("Could not prepare audio interface for use", err);
if (sd->volume > 0)
{
int chn;
snd_mixer_t *handle;
snd_mixer_elem_t *e;
const char *file = sd->file ? sd->file : DEFAULT_ALSA_SOUND_DEVICE;
if (snd_mixer_open (&handle, 0) >= 0)
{
if (snd_mixer_attach (handle, file) >= 0
&& snd_mixer_load (handle) >= 0
&& snd_mixer_selem_register (handle, NULL, NULL) >= 0)
for (e = snd_mixer_first_elem (handle);
e;
e = snd_mixer_elem_next (e))
//.........这里部分代码省略.........
开发者ID:azuk，项目名称:emacs，代码行数:101，代码来源:sound.c
示例3: GLOBAL_DEF
Error AudioDriverALSA::init() {
active=false;
thread_exited=false;
exit_thread=false;
pcm_open = false;
samples_in = NULL;
samples_out = NULL;
mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
output_format = OUTPUT_STEREO;
channels = 2;
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
#define CHECK_FAIL(m_cond)\
if (m_cond) {\
fprintf(stderr,"ALSA ERR: %s\n",snd_strerror(status));\
snd_pcm_close(pcm_handle);\
ERR_FAIL_COND_V(m_cond,ERR_CANT_OPEN);\
}
//todo, add
//6 chans - "plug:surround51"
//4 chans - "plug:surround40";
status = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
ERR_FAIL_COND_V( status<0, ERR_CANT_OPEN );
snd_pcm_hw_params_alloca(&hwparams);
status = snd_pcm_hw_params_any(pcm_handle, hwparams);
CHECK_FAIL( status<0 );
status = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
CHECK_FAIL( status<0 );
//not interested in anything else
status = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE);
CHECK_FAIL( status<0 );
//todo: support 4 and 6
status = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2);
CHECK_FAIL( status<0 );
status = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &mix_rate, NULL);
CHECK_FAIL( status<0 );
int latency = GLOBAL_DEF("audio/output_latency",25);
buffer_size = nearest_power_of_2( latency * mix_rate / 1000 );
// set buffer size from project settings
status = snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &buffer_size);
CHECK_FAIL( status<0 );
// make period size 1/8
period_size = buffer_size >> 3;
status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &period_size, NULL);
CHECK_FAIL( status<0 );
unsigned int periods=2;
status = snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, &periods, NULL);
CHECK_FAIL( status<0 );
status = snd_pcm_hw_params(pcm_handle,hwparams);
CHECK_FAIL( status<0 );
//snd_pcm_hw_params_free(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
status = snd_pcm_sw_params_current(pcm_handle, swparams);
CHECK_FAIL( status<0 );
status = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, period_size);
CHECK_FAIL( status<0 );
status = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
CHECK_FAIL( status<0 );
status = snd_pcm_sw_params(pcm_handle, swparams);
CHECK_FAIL( status<0 );
samples_in = memnew_arr(int32_t, period_size*channels);
samples_out = memnew_arr(int16_t, period_size*channels);
snd_pcm_nonblock(pcm_handle, 0);
mutex=Mutex::create();
thread = Thread::create(AudioDriverALSA::thread_func, this);
return OK;
};
开发者ID:lonesurvivor，项目名称:godot，代码行数:98，代码来源:audio_driver_alsa.cpp
示例4: alsa_set_swparams
/* setup alsa data transfer behavior */
static inline int alsa_set_swparams(ao_device *device)
{
ao_alsa_internal *internal = (ao_alsa_internal *) device->internal;
snd_pcm_sw_params_t *params;
snd_pcm_uframes_t boundary;
int err;
/* allocate the software parameter structure */
snd_pcm_sw_params_alloca(¶ms);
/* fetch the current software parameters */
err = snd_pcm_sw_params_current(internal->pcm_handle, params);
if (err < 0){
adebug("snd_pcm_sw_params_current() failed.\n");
return err;
}
#if 0 /* the below causes more trouble than it cures */
/* allow transfers to start when there is one period */
err = snd_pcm_sw_params_set_start_threshold(internal->pcm_handle,
params, internal->period_size);
if (err < 0){
adebug("snd_pcm_sw_params_set_start_threshold() failed.\n");
//return err;
}
/* require a minimum of one full transfer in the buffer */
err = snd_pcm_sw_params_set_avail_min(internal->pcm_handle, params,
internal->period_size);
if (err < 0){
adebug("snd_pcm_sw_params_set_avail_min() failed.\n");
//return err;
}
#endif
/* do not align transfers; this is obsolete/deprecated in ALSA
1.x where the transfer alignemnt is always 1 (except for
buggy drivers like VIA 82xx which still demand aligned
transfers regardless of setting, in violation of the ALSA
API docs) */
err = snd_pcm_sw_params_set_xfer_align(internal->pcm_handle, params, 1);
if (err < 0){
adebug("snd_pcm_sw_params_set_xfer_align() failed.\n");
//return err;
}
/* get the boundary size */
err = snd_pcm_sw_params_get_boundary(params,&boundary);
if (err < 0){
adebug("snd_pcm_sw_params_get_boundary() failed.\n");
}else{
/* force a work-ahead silence buffer; this is a fix, again for
VIA 82xx, where non-MMIO transfers will buffer into
period-size transfers, but the last transfer is usually
undersized and playback falls off the end of the submitted
data. */
err = snd_pcm_sw_params_set_silence_size(internal->pcm_handle, params, boundary);
if (err < 0){
adebug("snd_pcm_sw_params_set_silence_size() failed.\n");
//return err;
}
}
/* commit the params structure to ALSA */
err = snd_pcm_sw_params(internal->pcm_handle, params);
if (err < 0){
adebug("snd_pcm_sw_params() failed.\n");
return err;
}
return 1;
}
开发者ID:BenWBrown，项目名称:cs453_final_project，代码行数:75，代码来源:ao_alsa.c
示例5: AlsaOutput_Configure
//.........这里部分代码省略.........
case 24: pcm_format_id = SND_PCM_FORMAT_U24_3BE; break;
case 32: pcm_format_id = SND_PCM_FORMAT_U32_BE; break;
}
break;
case BLT_PCM_SAMPLE_FORMAT_FLOAT_BE:
ATX_LOG_FINE("sample format is BLT_PCM_SAMPLE_FORMAT_FLOAT_LE");
switch (format->bits_per_sample) {
case 32: pcm_format_id = SND_PCM_FORMAT_FLOAT_BE; break;
}
break;
}
if (pcm_format_id == SND_PCM_FORMAT_UNKNOWN) {
return BLT_ERROR_INVALID_MEDIA_TYPE;
}
ior = snd_pcm_hw_params_set_format(self->device_handle,
hw_params,
pcm_format_id);
if (ior != 0) {
ATX_LOG_WARNING_2("snd_pcm_hw_params_set_format() failed (%d:%s)", ior, snd_strerror(ior));
return BLT_FAILURE;
}
/* set the period size */
ior = snd_pcm_hw_params_set_period_size_near(self->device_handle,
hw_params,
&period_size,
NULL);
if (ior != 0) {
ATX_LOG_WARNING_2("snd_pcm_hw_params_set_period_size_near() failed (%d:%s)", ior, snd_strerror(ior));
return BLT_FAILURE;
}
/* set the buffer time (duration) */
ior = snd_pcm_hw_params_set_buffer_time_near(self->device_handle,
hw_params,
&buffer_time,
NULL);
if (ior != 0) {
ATX_LOG_WARNING_2("snd_pcm_hw_params_set_buffer_time_near() failed (%d:%s)", ior, snd_strerror(ior));
return BLT_FAILURE;
}
/* get the actual buffer size */
snd_pcm_hw_params_get_buffer_size(hw_params, &buffer_size);
/* activate this configuration */
ior = snd_pcm_hw_params(self->device_handle, hw_params);
if (ior != 0) {
ATX_LOG_WARNING_2("snd_pcm_hw_params() failed (%d:%s)", ior, snd_strerror(ior));
return BLT_FAILURE;
}
/* configure the software parameters */
snd_pcm_sw_params_alloca_no_assert(&sw_params);
snd_pcm_sw_params_current(self->device_handle, sw_params);
/* set the start threshold to 1/2 the buffer size */
snd_pcm_sw_params_set_start_threshold(self->device_handle,
sw_params,
buffer_size/2);
/* set the buffer alignment */
/* NOTE: this call is now obsolete */
/* snd_pcm_sw_params_set_xfer_align(self->device_handle, sw_params, 1); */
/* activate the sofware parameters */
ior = snd_pcm_sw_params(self->device_handle, sw_params);
if (ior != 0) {
ATX_LOG_SEVERE_2("snd_pcm_sw_params() failed (%d:%s)", ior, snd_strerror(ior));
return BLT_FAILURE;
}
/* print status info */
{
snd_pcm_uframes_t val;
ATX_LOG_FINER_1("sample type = %x", pcm_format_id);
if (rate != format->sample_rate) {
ATX_LOG_FINER_1("actual sample = %d", rate);
}
ATX_LOG_FINER_1("actual buffer time = %d", (int)buffer_time);
ATX_LOG_FINER_1("buffer size = %d", (int)buffer_size);
snd_pcm_sw_params_get_start_threshold(sw_params, &val);
ATX_LOG_FINER_1("start threshold = %d", (int)val);
snd_pcm_sw_params_get_stop_threshold(sw_params, &val);
ATX_LOG_FINER_1("stop threshold = %d", (int)val);
snd_pcm_hw_params_get_period_size(hw_params, &val, NULL);
ATX_LOG_FINER_1("period size = %d", (int)val);
}
break;
}
/* update the state */
AlsaOutput_SetState(self, BLT_ALSA_OUTPUT_STATE_CONFIGURED);
return BLT_SUCCESS;
}
开发者ID:axiomatic-systems，项目名称:BlueTune，代码行数:101，代码来源:BltAlsaOutput.c
示例6: set_swparams
static int
set_swparams (GstAlsaSrc * alsa)
{
int err;
snd_pcm_sw_params_t *params;
snd_pcm_sw_params_malloc (¶ms);
/* get the current swparams */
CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
/* allow the transfer when at least period_size samples can be processed */
CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
alsa->period_size), set_avail);
/* start the transfer on first read */
CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
0), start_threshold);
/* use monotonic timestamping */
CHECK (snd_pcm_sw_params_set_tstamp_mode (alsa->handle, params,
SND_PCM_TSTAMP_MMAP), tstamp_mode);
#if GST_CHECK_ALSA_VERSION(1,0,16)
/* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
#else
/* align all transfers to 1 sample */
CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
#endif
/* write the parameters to the recording device */
CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
snd_pcm_sw_params_free (params);
return 0;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to determine current swparams for playback: %s",
snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
start_threshold:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set start threshold mode for playback: %s",
snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
set_avail:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set avail min for playback: %s", snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
tstamp_mode:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set tstamp mode for playback: %s", snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
#if !GST_CHECK_ALSA_VERSION(1,0,16)
set_align:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set transfer align for playback: %s", snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
#endif
set_sw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set sw params for playback: %s", snd_strerror (err)));
snd_pcm_sw_params_free (params);
return err;
}
}
开发者ID:Distrotech，项目名称:gst-plugins-base，代码行数:81，代码来源:gstalsasrc.c
示例7: snd_pcm_open
//.........这里部分代码省略.........
if (rc < 0) {
printf("Failed to set PCM device to mono: %s\n",
snd_strerror(rc));
exit(1);
}
/* 44100 bits/second sampling rate (CD quality) */
val = rate;
rc = snd_pcm_hw_params_set_rate_near(this->m_pHandle, this->m_pParams,
&val, 0);
if (rc < 0) {
printf("Failed to set PCM device to sample rate =%d: %s\n",
rate,snd_strerror(rc));
exit(1);
}
if(val != rate){
printf("Rate doesn't match (request %iHz, get %iHz\n",rate,val);
exit(1);
}
rc = snd_pcm_hw_params_set_buffer_time_near(this->m_pHandle, this->m_pParams, &buffer_time, &dir);
if (rc < 0) {
fprintf(stderr, "Failed to set PCM device to buffer time=%d: %s\n",
val,snd_strerror(rc));
exit(1);
}
rc = snd_pcm_hw_params_set_period_time_near(this->m_pHandle, this->m_pParams, &period_time, &dir);
if (rc < 0) {
fprintf(stderr, "Failed to set PCM device to period time=%d: %s\n",
val,snd_strerror(rc));
exit(1);
}
//chunk size
rc = snd_pcm_hw_params_get_period_size(this->m_pParams, &period_size, &dir);
if(rc < 0){
fprintf(stderr, "Failed to get period size for capture: %s\n", snd_strerror(rc));
exit(1);
}
this->m_frames = period_size;
rc = snd_pcm_hw_params_get_buffer_size(this->m_pParams, &buffer_size);
if(rc < 0){
fprintf(stderr, "Failed to get buffer size for capture: %s\n", snd_strerror(rc));
exit(1);
}
#ifdef DEBUG
printf("Recorder alsa driver hw params setting\n");
printf("period size =%d frames\n", (int)period_size);
printf("buffer size =%d frames\n", (int)buffer_size);
snd_pcm_hw_params_get_period_time(this->m_pParams,&val, &dir);
printf("period time =%d us\n",val);
snd_pcm_hw_params_get_buffer_time(this->m_pParams,&val, &dir);
printf("buffer time =%d us\n", val);
snd_pcm_hw_params_get_periods(this->m_pParams, &val, &dir);
printf("period per buffer =%d frames\n", val);
#endif
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(this->m_pHandle, this->m_pParams);
if (rc < 0) {
printf("unable to set hw parameters: %s\n",
snd_strerror(rc));
exit(1);
}
#ifdef ALSA_OPT
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
/* get the current swparams */
rc = snd_pcm_sw_params_current(this->m_pHandle, swparams);
if (rc < 0) {
printf("Unable to determine current swparams for playback: %s\n", snd_strerror(rc));
exit(1);
}
/* start the transfer when the buffer is almost full: */
/* (buffer_size / avail_min) * avail_min */
rc = snd_pcm_sw_params_set_start_threshold(this->m_pHandle, swparams, (buffer_size / period_size) * period_size);
if (rc < 0) {
printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(rc));
exit(1);
}
rc = snd_pcm_sw_params_set_avail_min(this->m_pHandle, swparams, period_size);
if (rc < 0) {
printf("Unable to set avail min for playback: %s\n", snd_strerror(rc));
exit(1);
}
/* write the parameters to the playback device */
rc = snd_pcm_sw_params(this->m_pHandle, swparams);
if (rc < 0) {
printf("Unable to set sw params for playback: %s\n", snd_strerror(rc));
exit(1);
}
#endif
}
开发者ID:JangolVoIP，项目名称:Codes，代码行数:101，代码来源:Player.cpp
示例8: set_params
//.........这里部分代码省略.........
if (err < 0) {
printf("Access type not available");
exit(EXIT_FAILURE);
}
printf("format = %s, channels = %d, rate = %d\n",
snd_pcm_format_name(hwparams.format),hwparams.channels,hwparams.rate);
//err = snd_pcm_hw_params_set_format(handle, params, hwparams.format);
err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
if (err < 0) {
printf("Sample format non available");
exit(EXIT_FAILURE);
}
//err = snd_pcm_hw_params_set_channels(handle, params, hwparams.channels);
err = snd_pcm_hw_params_set_channels(handle, params, 2);
if (err < 0) {
printf("Channels count non available");
exit(EXIT_FAILURE);
}
#if 0
//add by yjc 2012/08/21
err = set_audio_clk_freq(hwparams.rate);
if (err < 0){
printf("set_audio_clk_freq fail..........\n");
exit(EXIT_FAILURE);
}
#endif
err = snd_pcm_hw_params_set_rate(handle, params, hwparams.rate, 0);
if (err < 0) {
printf("Rate non available");
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_get_buffer_time_max(params,
&buffer_time, 0);
assert(err >= 0);
if (buffer_time > 500000)
buffer_time = 500000;
period_time = buffer_time / 4;
err = snd_pcm_hw_params_set_period_time_near(handle, params,
&period_time, 0);
assert(err >= 0);
err = snd_pcm_hw_params_set_buffer_time_near(handle, params,
&buffer_time, 0);
assert(err >= 0);
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
printf("Unable to install hw params:");
exit(EXIT_FAILURE);
}
snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
if (chunk_size == buffer_size) {
printf("Can't use period equal to buffer size (%lu == %lu)",
chunk_size, buffer_size);
exit(EXIT_FAILURE);
}
snd_pcm_sw_params_current(handle, swparams);
err = snd_pcm_sw_params_set_avail_min(handle, swparams, chunk_size);
if(stream == SND_PCM_STREAM_PLAYBACK)
start_threshold = (buffer_size / chunk_size) * chunk_size;
else
start_threshold = 1;
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
assert(err >= 0);
err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, buffer_size);
assert(err >= 0);
if (snd_pcm_sw_params(handle, swparams) < 0) {
printf("unable to install sw params:");
exit(EXIT_FAILURE);
}
//bits_per_sample = snd_pcm_format_physical_width(hwparams.format);
bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
//bits_per_frame = bits_per_sample * hwparams.channels;
bits_per_frame = bits_per_sample * 2;
chunk_bytes = chunk_size * bits_per_frame / 8;
printf("chunk_size = %d,chunk_bytes = %d,buffer_size = %d\n\n",
(int)chunk_size,chunk_bytes,(int)buffer_size);
return chunk_size;
}
开发者ID:houwentaoff，项目名称:rtsp_audio，代码行数:101，代码来源:test_audio.cpp
示例9: sound_lowlevel_init
//.........这里部分代码省略.........
ui_error( UI_ERROR_ERROR, "couldn't set period size %d on '%s'.",
(int)sound_periodsize, pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( first_init && ( exact_periodsize < sound_periodsize / 1.5 ||
exact_periodsize > sound_periodsize * 1.5 ) ) {
fprintf( stderr,
"The period size %d is not supported by your hardware. "
"Using %d instead.\n", (int)sound_periodsize,
(int)exact_periodsize );
}
periods = nperiods;
/* Set number of periods. Periods used to be called fragments. */
if( snd_pcm_hw_params_set_periods_near( pcm_handle, hw_params, &periods,
NULL ) < 0 ) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR, "couldn't set periods on '%s'.", pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( first_init && periods != nperiods ) {
fprintf( stderr, "%d periods is not supported by your hardware. "
"Using %d instead.\n", nperiods, periods );
}
snd_pcm_hw_params_get_buffer_size( hw_params, &exact_bsize );
/* Apply HW parameter settings to */
/* PCM device and prepare device */
if( snd_pcm_hw_params( pcm_handle, hw_params ) < 0 ) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR,"couldn't set hw_params on %s", pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
snd_pcm_sw_params_alloca( &sw_params );
if( ( err = snd_pcm_sw_params_current( pcm_handle, sw_params ) ) < 0 ) {
ui_error( UI_ERROR_ERROR,"couldn't get sw_params from %s: %s", pcm_name,
snd_strerror ( err ) );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( ( err = snd_pcm_sw_params_set_start_threshold( pcm_handle,
sw_params, exact_periodsize * ( nperiods - 1 ) ) ) < 0 ) {
ui_error( UI_ERROR_ERROR,"couldn't set start_treshold on %s: %s", pcm_name,
snd_strerror ( err ) );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( !avail_min )
avail_min = exact_periodsize >> 1;
if( snd_pcm_sw_params_set_avail_min( pcm_handle,
sw_params, avail_min ) < 0 ) {
#if SND_LIB_VERSION < 0x10010
if( ( err = snd_pcm_sw_params_set_sleep_min( pcm_handle,
sw_params, 1 ) ) < 0 ) {
fprintf( stderr, "Unable to set minimal sleep 1 for %s: %s\n", pcm_name,
snd_strerror ( err ) );
}
#else
fprintf( stderr, "Unable to set avail min %s: %s\n", pcm_name,
snd_strerror( err ) );
#endif
}
#if SND_LIB_VERSION < 0x10010
if( ( err = snd_pcm_sw_params_set_xfer_align( pcm_handle, sw_params, 1 ) ) < 0 ) {
ui_error( UI_ERROR_ERROR,"couldn't set xfer_allign on %s: %s", pcm_name,
snd_strerror ( err ) );
init_running = 0;
return 1;
}
#endif
if( ( err = snd_pcm_sw_params( pcm_handle, sw_params ) ) < 0 ) {
ui_error( UI_ERROR_ERROR,"couldn't set sw_params on %s: %s", pcm_name,
snd_strerror ( err ) );
init_running = 0;
return 1;
}
if( first_init ) snd_output_stdio_attach(&output, stdout, 0);
first_init = 0;
init_running = 0;
return 0; /* success */
}
开发者ID:CiaranG，项目名称:ZXdroid，代码行数:101，代码来源:alsasound.c
示例10: qWarning
//.........这里部分代码省略.........
}
qLog(QAudioInput) << "channels = " << vval;
snd_pcm_hw_params_get_rate(hwparams, &vval, &dir);
if ( (int)vval != (int)frequency ) {
qLog(QAudioInput) << QString("frequency type not set, want %1 got %2").arg(frequency).arg(vval);
frequency = vval;
}
qLog(QAudioInput) << "rate =" << vval << " bps";
snd_pcm_hw_params_get_period_time(hwparams,&period_time, &dir);
qLog(QAudioInput) << "period time =" << period_time << " us";
snd_pcm_hw_params_get_period_size(hwparams,&period_size, &dir);
qLog(QAudioInput) << "period size =" << (int)period_size;
snd_pcm_hw_params_get_buffer_time(hwparams,&buffer_time, &dir);
qLog(QAudioInput) << "buffer time =" << buffer_time;
snd_pcm_hw_params_get_buffer_size(hwparams,(snd_pcm_uframes_t *) &buffer_size);
qLog(QAudioInput) << "buffer size =" << (int)buffer_size;
snd_pcm_hw_params_get_periods(hwparams, &vval, &dir);
qLog(QAudioInput) << "periods per buffer =" << vval;
snd_pcm_hw_params_get_rate_numden(hwparams, &vval, &vval2);
qLog(QAudioInput) << QString("exact rate = %1/%2 bps").arg(vval).arg(vval2).toLatin1().constData();
vval = snd_pcm_hw_params_get_sbits(hwparams);
qLog(QAudioInput) << "significant bits =" << vval;
snd_pcm_hw_params_get_tick_time(hwparams,&vval, &dir);
qLog(QAudioInput) << "tick time =" << vval;
vval = snd_pcm_hw_params_is_batch(hwparams);
qLog(QAudioInput) << "is batch =" << vval;
vval = snd_pcm_hw_params_is_block_transfer(hwparams);
qLog(QAudioInput) << "is block transfer =" << vval;
vval = snd_pcm_hw_params_is_double(hwparams);
qLog(QAudioInput) << "is double =" << vval;
vval = snd_pcm_hw_params_is_half_duplex(hwparams);
qLog(QAudioInput) << "is half duplex =" << vval;
vval = snd_pcm_hw_params_is_joint_duplex(hwparams);
qLog(QAudioInput) << "is joint duplex =" << vval;
vval = snd_pcm_hw_params_can_overrange(hwparams);
qLog(QAudioInput) << "can overrange =" << vval;
vval = snd_pcm_hw_params_can_mmap_sample_resolution(hwparams);
qLog(QAudioInput) << "can mmap =" << vval;
vval = snd_pcm_hw_params_can_pause(hwparams);
qLog(QAudioInput) << "can pause =" << vval;
vval = snd_pcm_hw_params_can_resume(hwparams);
qLog(QAudioInput) << "can resume =" << vval;
vval = snd_pcm_hw_params_can_sync_start(hwparams);
qLog(QAudioInput) << "can sync start =" << vval;
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_sw_params_current(handle, swparams);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params_current: err %d",err);
}
err = snd_pcm_sw_params_set_start_threshold(handle,swparams,period_size);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params_set_start_threshold: err %d",err);
}
err = snd_pcm_sw_params_set_avail_min(handle, swparams,period_size);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params_set_avail_min: err %d",err);
}
err = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params_set_xfer_align: err %d",err);
}
err = snd_pcm_sw_params(handle, swparams);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params: err %d",err);
}
snd_pcm_prepare(handle);
snd_pcm_start(handle);
int count = snd_pcm_poll_descriptors_count(handle);
pollfd *pfds = new pollfd[count];
snd_pcm_poll_descriptors(handle, pfds, count);
for (int i = 0; i < count; ++i)
{
if ((pfds[i].events & POLLIN) != 0) {
notifier = new QSocketNotifier(pfds[i].fd, QSocketNotifier::Read);
QObject::connect(notifier,
SIGNAL(activated(int)),
input,
SIGNAL(readyRead()));
break;
}
}
if (notifier == NULL) {
rc = false;
}
delete pfds;
}
return rc;
}
开发者ID:Camelek，项目名称:qtmoko，代码行数:101，代码来源:qaudioinput_alsa.cpp
示例11: sndo_pcm_initialize
static int sndo_pcm_initialize(sndo_pcm_t *pcm)
{
int err;
snd_pcm_uframes_t boundary;
snd_pcm_uframes_t p_period_size = ~0UL, c_period_size = ~0UL;
snd_pcm_uframes_t p_buffer_size = ~0UL, c_buffer_size = ~0UL;
if (pcm->playback) {
err = snd_pcm_hw_params(pcm->playback, pcm->p_hw_params);
if (err < 0)
return err;
err = snd_pcm_hw_params_get_period_size(pcm->p_hw_params, &p_period_size, NULL);
if (err < 0)
return err;
err = snd_pcm_hw_params_get_buffer_size(pcm->p_hw_params, &p_buffer_size);
if (err < 0)
return err;
}
if (pcm->capture) {
err = snd_pcm_hw_params(pcm->capture, pcm->c_hw_params);
if (err < 0)
return err;
err = snd_pcm_hw_params_get_period_size(pcm->c_hw_params, &c_period_size, NULL);
if (err < 0)
return err;
err = snd_pcm_hw_params_get_buffer_size(pcm->c_hw_params, &c_buffer_size);
if (err < 0)
return err;
}
if (p_period_size < c_period_size)
pcm->transfer_block = p_period_size;
else
pcm->transfer_block = c_period_size;
if (p_buffer_size < c_buffer_size)
pcm->ring_size = p_buffer_size;
else
pcm->ring_size = c_buffer_size;
if (pcm->playback) {
err = snd_pcm_sw_params_get_boundary(pcm->p_sw_params, &boundary);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_start_threshold(pcm->playback, pcm->p_sw_params, boundary);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_stop_threshold(pcm->playback, pcm->p_sw_params, pcm->xrun == SNDO_PCM_XRUN_IGNORE ? boundary : pcm->ring_size);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_xfer_align(pcm->playback, pcm->p_sw_params, 1);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_avail_min(pcm->playback, pcm->p_sw_params, pcm->transfer_block);
if (err < 0)
return err;
err = snd_pcm_sw_params(pcm->playback, pcm->p_sw_params);
if (err < 0)
return err;
}
if (pcm->capture) {
err = snd_pcm_sw_params_get_boundary(pcm->c_sw_params, &boundary);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_start_threshold(pcm->capture, pcm->c_sw_params, boundary);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_stop_threshold(pcm->capture, pcm->c_sw_params, pcm->xrun == SNDO_PCM_XRUN_IGNORE ? boundary : pcm->ring_size);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_xfer_align(pcm->capture, pcm->c_sw_params, 1);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_avail_min(pcm->capture, pcm->c_sw_params, pcm->transfer_block);
if (err < 0)
return err;
err = snd_pcm_sw_params(pcm->capture, pcm->c_sw_params);
if (err < 0)
return err;
}
pcm->initialized = 1;
return 0;
}
开发者ID:xenyinzen，项目名称:lx_toolset，代码行数:80，代码来源:ordinary_pcm.c
示例12: pcm_open
static int pcm_open(pcm_handle_t* pcm, const pcm_desc_t* desc)
{
const snd_pcm_format_t fmt = SND_PCM_FORMAT_S16_LE;
snd_pcm_stream_t stm;
int err;
if (desc->flags & PCM_FLAG_IN) stm = SND_PCM_STREAM_CAPTURE;
else stm = SND_PCM_STREAM_PLAYBACK;
err = snd_pcm_open
(&pcm->pcm, desc->name, stm, SND_PCM_NONBLOCK);
if (err) PERROR_GOTO(snd_strerror(err), on_error_0);
err = snd_pcm_hw_params_malloc(&pcm->hw_params);
if (err) PERROR_GOTO(snd_strerror(err), on_error_1);
err = snd_pcm_hw_params_any(pcm->pcm, pcm->hw_params);
if (err) PERROR_GOTO(snd_strerror(err), on_error_2);
err = snd_pcm_hw_params_set_access
(pcm->pcm, pcm->hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err) PERROR_GOTO(snd_strerror(err), on_error_2);
err = snd_pcm_hw_params_set_format(pcm->pcm, pcm->hw_params, fmt);
if (err) PERROR_GOTO(snd_strerror(err), on_error_2);
err = snd_pcm_hw_params_set_rate
(pcm->pcm, pcm->hw_params, desc->fsampl, 0);
if (err) PERROR_GOTO(snd_strerror(err), on_error_2);
pcm->nchan = desc->nchan;
pcm->wchan = (size_t)snd_pcm_format_physical_width(fmt) / 8;
pcm->scale = pcm->nchan * pcm->wchan;
err = snd_pcm_hw_params_set_channels
(pcm->pcm, pcm->hw_params, desc->nchan);
if (err) PERROR_GOTO(snd_strerror(err), on_error_2);
err = snd_pcm_hw_params(pcm->pcm, pcm->hw_params);
if (err) PERROR_GOTO(snd_strerror(err), on_error_2);
err = snd_pcm_sw_params_malloc(&pcm->sw_params);
if (err) PERROR_GOTO(snd_strerror(err), on_error_2);
err = snd_pcm_sw_params_current(pcm->pcm, pcm->sw_params);
if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
#if 1
err = snd_pcm_sw_params_set_avail_min
(pcm->pcm, pcm->sw_params, 1024);
if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
#endif
#if 1
err = snd_pcm_sw_params_set_start_threshold
(pcm->pcm, pcm->sw_params, 0U);
if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
#endif
err = snd_pcm_sw_params(pcm->pcm, pcm->sw_params);
if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
err = snd_pcm_prepare(pcm->pcm);
if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
pcm->rpos = 0;
pcm->wpos = 0;
pcm->nsampl = (size_t)desc->fsampl * 10;
pcm->buf = malloc(pcm->nsampl * pcm->scale);
if (pcm->buf == NULL) goto on_error_3;
return 0;
on_error_3:
snd_pcm_sw_params_free(pcm->sw_params);
on_error_2:
snd_pcm_hw_params_free(pcm->hw_params);
on_error_1:
snd_pcm_close(pcm->pcm);
on_error_0:
return -1;
}
开发者ID:texane，项目名称:aspect，代码行数:83，代码来源:main.c
示例13: TIesrFA_ALSA_open
//.........这里部分代码省略.........
/* Set up the information for recording from the audio channelport. This
will include data specified by the user, such as data in
aTIesrFAInstance->encoding, etc. If setup fails, then the audio channel
should be shut down and failure reported. */
/* Setup hardware parameters of the ALSA pcm */
rtnStatus = snd_pcm_hw_params_malloc( &hw_params );
if( rtnStatus < 0 )
goto openExit;
rtnStatus = snd_pcm_hw_params_any( ALSAData->alsa_handle, hw_params );
if( rtnStatus < 0 )
goto openExit;
/* There will only be a single channel */
rtnStatus = snd_pcm_hw_params_set_channels( ALSAData->alsa_handle,
hw_params, 1 );
if( rtnStatus < 0 )
goto openExit;
/* Even though only one channel, must specify type of read. */
rtnStatus = snd_pcm_hw_params_set_access( ALSAData->alsa_handle,
hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED );
if( rtnStatus < 0 )
goto openExit;
/* Format type - only 16 bit linear for now */
rtnStatus = snd_pcm_hw_params_set_format( ALSAData->alsa_handle,
hw_params,
SND_PCM_FORMAT_S16 );
if( rtnStatus < 0 )
goto openExit;
/* Set sample rate, it must be exactly supported */
rtnStatus = snd_pcm_hw_params_set_rate( ALSAData->alsa_handle,
hw_params,
aTIesrFAInstance->sample_rate, 0 );
if( rtnStatus < 0 )
goto openExit;
/* Size of ALSA PCM ring buffer is set to approximately the desired
number of multiples of read_samples in the buffer */
numSamples = aTIesrFAInstance->num_audio_buffer_frames *
ALSAData->read_samples;
rtnStatus = snd_pcm_hw_params_set_buffer_size_near( ALSAData->alsa_handle,
hw_params, &numSamples );
if( rtnStatus < 0 )
goto openExit;
/* Set the hardware parameters in the PCM*/
rtnStatus = snd_pcm_hw_params( ALSAData->alsa_handle, hw_params );
if( rtnStatus < 0 )
goto openExit;
/* Set software parameters to interrupt at the end of an audio buffer
number of samples and to start-up manually */
rtnStatus = snd_pcm_sw_params_malloc( &sw_params );
if( rtnStatus < 0 )
goto openExit;
rtnStatus = snd_pcm_sw_params_current( ALSAData->alsa_handle, sw_params );
if( rtnStatus < 0 )
goto openExit;
/* Number of samples needed in PCM buffer prior to interrupt */
rtnStatus = snd_pcm_sw_params_set_avail_min( ALSAData->alsa_handle,
sw_params, ALSAData->read_samples );
if( rtnStatus < 0 )
goto openExit;
/* Set start threshold so startup is done manually */
rtnStatus = snd_pcm_sw_params_set_start_threshold( ALSAData->alsa_handle,
sw_params, ULONG_MAX );
if( rtnStatus < 0 )
goto openExit;
/* Channel opened successfully */
openSuccess = 1;
openExit:
if( hw_params != NULL )
snd_pcm_hw_params_free( hw_params );
if( sw_params != NULL )
snd_pcm_sw_params_free( sw_params );
if( openSuccess )
return TIesrFA_ALSAErrNone;
return TIesrFA_ALSAErrFail;
}
开发者ID:proegssilb，项目名称:tiesr-dialer，代码行数:101，代码来源:TIesrFA_ALSA.c
示例14: setparams_set
int setparams_set(snd_pcm_t *handle,
snd_pcm_hw_params_t *params,
snd_pcm_sw_params_t *swparams,
const char *id)
{
int err;
snd_pcm_uframes_t val;
unsigned int sleep_min = 0;
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
printf("Unable to set hw params for %s: %s\n", id, snd_strerror(err));
return err;
}
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0) {
printf("Unable to determine current swparams for %s: %s\n", id, snd_strerror(err));
return err;
}
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0x7fffffff);
if (err < 0) {
printf("Unable to set start threshold mode for %s: %s\n", id, snd_strerror(err));
return err;
}
tick_time_ok = 0;
if (tick_time > 0) {
int time, ttime;
snd_pcm_hw_params_get_period_time(params, &time, NULL);
snd_pcm_hw_params_get_tick_time(params, &ttime, NULL);
if (time < ttime) {
printf("Skipping to set minimal sleep: period time < tick time\n");
} else if (ttime <= 0) {
printf("Skipping to set minimal sleep: tick time <= 0 (%i)\n", ttime);
} else {
sleep_min = tick_time / ttime;
if (sleep_min <= 0)
sleep_min = 1;
err = snd_pcm_sw_params_set_sleep_min(handle, swparams, sleep_min);
if (err < 0) {
printf("Unable to set minimal sleep %i for %s: %s\n", sleep_min, id, snd_strerror(err));
r
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