C++ snd_pcm_hw_params_set_buffer_size_near函数代码示例

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本文整理汇总了C++中snd_pcm_hw_params_set_buffer_size_near函数的典型用法代码示例。如果您正苦于以下问题：C++ snd_pcm_hw_params_set_buffer_size_near函数的具体用法？C++ snd_pcm_hw_params_set_buffer_size_near怎么用？C++ snd_pcm_hw_params_set_buffer_size_near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_hw_params_set_buffer_size_near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: rdpsnd_alsa_set_params

static void rdpsnd_alsa_set_params(rdpsndAlsaPlugin* alsa)
{
    snd_pcm_hw_params_t* hw_params;
    snd_pcm_sw_params_t* sw_params;
    int error;
    snd_pcm_uframes_t frames;
    snd_pcm_uframes_t start_threshold;

    snd_pcm_drop(alsa->out_handle);

    error = snd_pcm_hw_params_malloc(&hw_params);
    if (error < 0)
    {
        DEBUG_WARN("snd_pcm_hw_params_malloc failed");
        return;
    }
    snd_pcm_hw_params_any(alsa->out_handle, hw_params);
    snd_pcm_hw_params_set_access(alsa->out_handle, hw_params,
                                 SND_PCM_ACCESS_RW_INTERLEAVED);
    snd_pcm_hw_params_set_format(alsa->out_handle, hw_params,
                                 alsa->format);
    snd_pcm_hw_params_set_rate_near(alsa->out_handle, hw_params,
                                    &alsa->actual_rate, NULL);
    snd_pcm_hw_params_set_channels_near(alsa->out_handle, hw_params,
                                        &alsa->actual_channels);
    if (alsa->latency < 0)
        frames = alsa->actual_rate * 4; /* Default to 4-second buffer */
    else
        frames = alsa->latency * alsa->actual_rate * 2 / 1000; /* Double of the latency */
    if (frames < alsa->actual_rate / 2)
        frames = alsa->actual_rate / 2; /* Minimum 0.5-second buffer */
    snd_pcm_hw_params_set_buffer_size_near(alsa->out_handle, hw_params,
                                           &frames);
    snd_pcm_hw_params(alsa->out_handle, hw_params);
    snd_pcm_hw_params_free(hw_params);

    error = snd_pcm_sw_params_malloc(&sw_params);
    if (error < 0)
    {
        DEBUG_WARN("snd_pcm_sw_params_malloc failed");
        return;
    }
    snd_pcm_sw_params_current(alsa->out_handle, sw_params);
    if (alsa->latency == 0)
        start_threshold = 0;
    else
        start_threshold = frames / 2;
    snd_pcm_sw_params_set_start_threshold(alsa->out_handle, sw_params, start_threshold);
    snd_pcm_sw_params(alsa->out_handle, sw_params);
    snd_pcm_sw_params_free(sw_params);

    snd_pcm_prepare(alsa->out_handle);

    DEBUG_SVC("hardware buffer %d frames, playback buffer %.2g seconds",
              (int)frames, (double)frames / 2.0 / (double)alsa->actual_rate);
    if ((alsa->actual_rate != alsa->source_rate) ||
            (alsa->actual_channels != alsa->source_channels))
    {
        DEBUG_SVC("actual rate %d / channel %d is different from source rate %d / channel %d, resampling required.",
                  alsa->actual_rate, alsa->actual_channels, alsa->source_rate, alsa->source_channels);
    }
}

开发者ID:ngraziano，项目名称:FreeRDP，代码行数:62，代码来源:rdpsnd_alsa.c

示例2: drvHostALSAAudioOpen


//.........这里部分代码省略.........
                        rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
                        break;
                    }
                }

                /* Calculate default buffer size here since it might have been changed
                 * in the _near functions */
                buffer_size_f = 4 * period_size_f;

                minval = buffer_size_f;
                err = snd_pcm_hw_params_get_buffer_size_min(pHWParms, &minval);
                if (err < 0)
                {
                    LogRel(("ALSA: Could not retrieve minimal buffer size\n"));
                    rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
                    break;
                }
                else
                {
                    LogFunc(("Minimal buffer size is: %RU32\n", minval));
                    if (buffer_size_f < minval)
                    {
                        if (   ( fIn && s_ALSAConf.buffer_size_in_overriden)
                            || (!fIn && s_ALSAConf.buffer_size_out_overriden))
                        {
                            LogFunc(("Buffer size %RU32 is less than minimal buffer size %RU32\n",
                                     buffer_size_f, minval));
                        }

                        buffer_size_f = minval;
                    }
                }

                err = snd_pcm_hw_params_set_buffer_size_near(phPCM,
                                                             pHWParms, &buffer_size_f);
                LogFunc(("Buffer size is: %RU32\n", buffer_size_f));
                if (err < 0)
                {
                    LogRel(("ALSA: Failed to set buffer size %d: %s\n",
                            buffer_size_f, snd_strerror(err)));
                    rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
                    break;
                }
            }
        }
        else
            LogFunc(("Warning: Buffer size is not set\n"));

        err = snd_pcm_hw_params(phPCM, pHWParms);
        if (err < 0)
        {
            LogRel(("ALSA: Failed to apply audio parameters\n"));
            rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
            break;
        }

        err = snd_pcm_hw_params_get_buffer_size(pHWParms, &obt_buffer_size);
        if (err < 0)
        {
            LogRel(("ALSA: Failed to get buffer size\n"));
            rc = VERR_GENERAL_FAILURE; /** @todo Find a better rc. */
            break;
        }

        snd_pcm_uframes_t obt_period_size;
        int dir = 0;

开发者ID:stefano-garzarella，项目名称:virtualbox-org-svn-vbox-trunk，代码行数:67，代码来源:DrvHostALSAAudio.cpp

示例3: printf

eBool tfAlsa::openDevice(const eChar *device, eU32 freq, eU32 bits, eU32 channels)
{
	eInt err = 0;

	if (device == eNULL)
		return eFALSE;

	printf ("opening device '%s' for playback\n", device);

	if ((err = snd_pcm_open (&m_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) 
	{
		fprintf (stderr, "cannot open audio device '%s' for playback(%s)\n", device, snd_strerror (err));
		return eFALSE;
	}

	snd_pcm_hw_params_t *hw_params;
	snd_pcm_format_t format;
	snd_pcm_uframes_t buffer_size = BUFFERSIZE;
	snd_pcm_uframes_t period_size = PERIODSIZE;

	printf("setting device to %i hz %i bits %i channels\n", freq, bits, channels);

	if (bits == 16)
		format = SND_PCM_FORMAT_S16_LE;
	else
		format = SND_PCM_FORMAT_S8;

    	if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) 
	{
		fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err));
		return false;
	}

	if ((err = snd_pcm_hw_params_any (m_handle, hw_params)) < 0) 
	{
		fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err));
		return false;
	}

	if ((err = snd_pcm_hw_params_set_access (m_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) 
	{
		fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err));
		return false;
	}

	if ((err = snd_pcm_hw_params_set_format (m_handle, hw_params, format)) < 0) 
	{
		fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err));
		return false;
	}

	if ((err = snd_pcm_hw_params_set_rate_near (m_handle, hw_params, (unsigned int *)&freq, 0)) < 0) 
	{
		fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err));
		return false;
	}

	if ((err = snd_pcm_hw_params_set_channels (m_handle, hw_params, channels)) < 0) 
	{
		fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err));
		return false;
	}

	if ((err = snd_pcm_hw_params_set_buffer_size_near (m_handle, hw_params, &buffer_size)) < 0) 
	{
		fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err));
		return false;
	}

	if ((err = snd_pcm_hw_params_set_period_size_near (m_handle, hw_params, &period_size, NULL)) < 0) 
	{
		fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err));
		return false;
	}

	if ((err = snd_pcm_hw_params (m_handle, hw_params)) < 0) 
	{
		fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err));
		return false;
	}

	snd_pcm_hw_params_free (hw_params);

	return eTRUE;
}

开发者ID:DX94，项目名称:Enigma-Studio-3，代码行数:85，代码来源:alsa.cpp

示例4: sound_init_alsa

int sound_init_alsa() {
	
	int err;
	snd_pcm_hw_params_t *hw_params;
		
	unsigned int resample,samplerate;
	snd_pcm_uframes_t bufferSize;
	
	err = snd_pcm_open( &_soundDevice, "plughw:0,0", SND_PCM_STREAM_PLAYBACK, 0 );
	if (err<0) {
		return -1;
	}
	
	if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
		snd_pcm_close (_soundDevice);
		return -2;
	}

	if ((err = snd_pcm_hw_params_any (_soundDevice, hw_params)) < 0) {
		snd_pcm_close (_soundDevice);
		return -2;
	}

		
	if ((err = snd_pcm_hw_params_set_access (_soundDevice, hw_params,SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
		snd_pcm_close (_soundDevice);
		return -3;
	}
	
	if (snd_pcm_hw_params_set_format (_soundDevice, hw_params,SND_PCM_FORMAT_U8) >= 0) {
		ordenador.sign=0;
	    ordenador.format=0;
	} else if (snd_pcm_hw_params_set_format (_soundDevice, hw_params,SND_PCM_FORMAT_S8) >= 0) {
		ordenador.sign=-128;
	    ordenador.format=0;
	} else if (snd_pcm_hw_params_set_format (_soundDevice, hw_params,SND_PCM_FORMAT_U16_LE) >= 0) {
		ordenador.sign=0;
	    ordenador.format=1;
	} else if (snd_pcm_hw_params_set_format (_soundDevice, hw_params,SND_PCM_FORMAT_S16_LE) >= 0) {
		ordenador.sign=-128;
	    ordenador.format=1;
	} else if (snd_pcm_hw_params_set_format (_soundDevice, hw_params,SND_PCM_FORMAT_U16_BE) >= 0) {
		ordenador.sign=0;
	    ordenador.format=2;
	} else if (snd_pcm_hw_params_set_format (_soundDevice, hw_params,SND_PCM_FORMAT_S16_BE) >= 0) {
		ordenador.sign=-128;
	    ordenador.format=2;
	} else {
		snd_pcm_close (_soundDevice);
		return -3;
	}
	
	// Disable resampling.
	resample = 0;
	err = snd_pcm_hw_params_set_rate_resample(_soundDevice, hw_params, resample);
	if (err < 0) {
		snd_pcm_close (_soundDevice);
		return -3;
	}
	
	if ((err = snd_pcm_hw_params_set_channels (_soundDevice, hw_params, 1)) >= 0) {
		ordenador.channels=1;
	} else if ((err = snd_pcm_hw_params_set_channels (_soundDevice, hw_params, 2)) >= 0) {
		ordenador.channels=2;
	} else {
		snd_pcm_close (_soundDevice);
		return -3;
	}
	
	samplerate=48000;
	if ((err = snd_pcm_hw_params_set_rate_near (_soundDevice, hw_params, &samplerate, 0)) < 0) {
		snd_pcm_close (_soundDevice);
		return -3;
	}

	bufferSize=4096;
	if (snd_pcm_hw_params_set_buffer_size_near(_soundDevice, hw_params, &bufferSize) < 0) {
		fprintf(stderr, "Error setting buffersize.\n");
		return -3;
	}

	ordenador.freq=samplerate;
	
	err = snd_pcm_hw_params (_soundDevice, hw_params);
	if (err<0) {
		return -3;
	}
	//snd_pcm_hw_params_get_buffer_size( hw_params, &bufferSize );

	ordenador.buffer_len=bufferSize;
	
	started_sound=0;
	return 0;

	err = snd_pcm_prepare (_soundDevice);
	if (err<0) {
		return -5;
	}
	return 0;
}

开发者ID:Oibaf66，项目名称:fbzx-wii，代码行数:100，代码来源:sound.c

示例5: set_params

static void set_params(void)
{
	snd_pcm_hw_params_t *params;
	snd_pcm_sw_params_t *swparams;
	snd_pcm_uframes_t buffer_size;
	int err;
	size_t n;
	snd_pcm_uframes_t xfer_align;
	unsigned int rate;
	snd_pcm_uframes_t start_threshold, stop_threshold;
	snd_pcm_hw_params_alloca(&params);
	snd_pcm_sw_params_alloca(&swparams);
	err = snd_pcm_hw_params_any(handle, params);
	if (err < 0) {
		error(_("Broken configuration for this PCM: no configurations available"));
		exit(EXIT_FAILURE);
	}
	err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
	if (err < 0) {
		error(_("Access type not available"));
		exit(EXIT_FAILURE);
	}
	err = snd_pcm_hw_params_set_format(handle, params, hwparams.format);
	if (err < 0) {
		error(_("Sample format non available"));
		exit(EXIT_FAILURE);
	}
	err = snd_pcm_hw_params_set_channels(handle, params, hwparams.channels);
	if (err < 0) {
		error(_("Channels count non available"));
		exit(EXIT_FAILURE);
	}

#if 0
	err = snd_pcm_hw_params_set_periods_min(handle, params, 2);
	assert(err >= 0);
#endif
	rate = hwparams.rate;
	err = snd_pcm_hw_params_set_rate_near(handle, params, &hwparams.rate, 0);
	assert(err >= 0);
	if ((float)rate * 1.05 < hwparams.rate || (float)rate * 0.95 > hwparams.rate) {
		if (!quiet_mode) {
			char plugex[64];
			const char *pcmname = snd_pcm_name(handle);
			fprintf(stderr, _("Warning: rate is not accurate (requested = %iHz, got = %iHz)\n"), rate, hwparams.rate);
			if (! pcmname || strchr(snd_pcm_name(handle), ':'))
				*plugex = 0;
			else
				snprintf(plugex, sizeof(plugex), "(-Dplug:%s)",
					 snd_pcm_name(handle));
			fprintf(stderr, _("         please, try the plug plugin %s\n"),
				plugex);
		}
	}
	rate = hwparams.rate;
	if (buffer_time == 0 && buffer_frames == 0) {
		err = snd_pcm_hw_params_get_buffer_time_max(params,
							    &buffer_time, 0);
		assert(err >= 0);
		if (buffer_time > 500000)
			buffer_time = 500000;
	}
	if (period_time == 0 && period_frames == 0) {
		if (buffer_time > 0)
			period_time = buffer_time / 4;
		else
			period_frames = buffer_frames / 4;
	}
	if (period_time > 0)
		err = snd_pcm_hw_params_set_period_time_near(handle, params,
							     &period_time, 0);
	else
		err = snd_pcm_hw_params_set_period_size_near(handle, params,
							     &period_frames, 0);
	assert(err >= 0);
	if (buffer_time > 0) {
		err = snd_pcm_hw_params_set_buffer_time_near(handle, params,
							     &buffer_time, 0);
	} else {
		err = snd_pcm_hw_params_set_buffer_size_near(handle, params,
							     &buffer_frames);
	}
	assert(err >= 0);
	err = snd_pcm_hw_params(handle, params);
	if (err < 0) {
		error(_("Unable to install hw params:"));
		snd_pcm_hw_params_dump(params, log);
		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) {
		error(_("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_get_xfer_align(swparams, &xfer_align);
	if (err < 0) {
		error(_("Unable to obtain xfer align\n"));
//.........这里部分代码省略.........

开发者ID:hishamhm，项目名称:protosampler，代码行数:101，代码来源:play.c

示例6: GLOBAL_DEF

Error AudioDriverALSA::init_device() {
	mix_rate = GLOBAL_DEF("audio/mix_rate", DEFAULT_MIX_RATE);
	speaker_mode = SPEAKER_MODE_STEREO;
	channels = 2;

	// If there is a specified device check that it is really present
	if (device_name != "Default") {
		Array list = get_device_list();
		if (list.find(device_name) == -1) {
			device_name = "Default";
			new_device = "Default";
		}
	}

	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)); \
		if (pcm_handle) {                                        \
			snd_pcm_close(pcm_handle);                           \
			pcm_handle = NULL;                                   \
		}                                                        \
		ERR_FAIL_COND_V(m_cond, ERR_CANT_OPEN);                  \
	}

	//todo, add
	//6 chans - "plug:surround51"
	//4 chans - "plug:surround40";

	if (device_name == "Default") {
		status = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
	} else {
		String device = device_name;
		int pos = device.find(";");
		if (pos != -1) {
			device = device.substr(0, pos);
		}
		status = snd_pcm_open(&pcm_handle, device.utf8().get_data(), 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);

	// In ALSA the period size seems to be the one that will determine the actual latency
	// Ref: https://www.alsa-project.org/main/index.php/FramesPeriods
	unsigned int periods = 2;
	int latency = GLOBAL_DEF("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
	buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
	buffer_size = buffer_frames * periods;
	period_size = buffer_frames;

	// set buffer size from project settings
	status = snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &buffer_size);
	CHECK_FAIL(status < 0);

	status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &period_size, NULL);
	CHECK_FAIL(status < 0);

	if (OS::get_singleton()->is_stdout_verbose()) {
		print_line("audio buffer frames: " + itos(period_size) + " calculated latency: " + itos(period_size * 1000 / mix_rate) + "ms");
	}

	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);
//.........这里部分代码省略.........

开发者ID:ippan，项目名称:godot，代码行数:101，代码来源:audio_driver_alsa.cpp

示例7: alloca

static snd_pcm_t *alsa_open(char *dev, int rate, int channels)
{
	snd_pcm_hw_params_t *hwp;
	snd_pcm_sw_params_t *swp;
	snd_pcm_t *h;
	int r;
	int dir;
	snd_pcm_uframes_t period_size_min;
	snd_pcm_uframes_t period_size_max;
	snd_pcm_uframes_t buffer_size_min;
	snd_pcm_uframes_t buffer_size_max;
	snd_pcm_uframes_t period_size;
	snd_pcm_uframes_t buffer_size;

	if ((r = snd_pcm_open(&h, dev, SND_PCM_STREAM_PLAYBACK, 0) < 0))
		return NULL;

	hwp = alloca(snd_pcm_hw_params_sizeof());
	memset(hwp, 0, snd_pcm_hw_params_sizeof());
	snd_pcm_hw_params_any(h, hwp);

	snd_pcm_hw_params_set_access(h, hwp, SND_PCM_ACCESS_RW_INTERLEAVED);
	snd_pcm_hw_params_set_format(h, hwp, SND_PCM_FORMAT_S16_LE);
	snd_pcm_hw_params_set_rate(h, hwp, rate, 0);
	snd_pcm_hw_params_set_channels(h, hwp, channels);

	/* Configurue period */

	dir = 0;
	snd_pcm_hw_params_get_period_size_min(hwp, &period_size_min, &dir);
	dir = 0;
	snd_pcm_hw_params_get_period_size_max(hwp, &period_size_max, &dir);

	period_size = 1024;

	dir = 0;
	r = snd_pcm_hw_params_set_period_size_near(h, hwp, &period_size, &dir);

	if (r < 0) {
		fprintf(stderr, "audio: Unable to set period size %lu (%s)\n",
		        period_size, snd_strerror(r));
		snd_pcm_close(h);
		return NULL;
	}

	dir = 0;
	r = snd_pcm_hw_params_get_period_size(hwp, &period_size, &dir);

	if (r < 0) {
		fprintf(stderr, "audio: Unable to get period size (%s)\n",
		        snd_strerror(r));
		snd_pcm_close(h);
		return NULL;
	}

	/* Configurue buffer size */

	snd_pcm_hw_params_get_buffer_size_min(hwp, &buffer_size_min);
	snd_pcm_hw_params_get_buffer_size_max(hwp, &buffer_size_max);
	buffer_size = period_size * 4;

	dir = 0;
	r = snd_pcm_hw_params_set_buffer_size_near(h, hwp, &buffer_size);

	if (r < 0) {
		fprintf(stderr, "audio: Unable to set buffer size %lu (%s)\n",
		        buffer_size, snd_strerror(r));
		snd_pcm_close(h);
		return NULL;
	}

	r = snd_pcm_hw_params_get_buffer_size(hwp, &buffer_size);

	if (r < 0) {
		fprintf(stderr, "audio: Unable to get buffer size (%s)\n",
		        snd_strerror(r));
		snd_pcm_close(h);
		return NULL;
	}

	/* write the hw params */
	r = snd_pcm_hw_params(h, hwp);

	if (r < 0) {
		fprintf(stderr, "audio: Unable to configure hardware parameters (%s)\n",
		        snd_strerror(r));
		snd_pcm_close(h);
		return NULL;
	}

	/*
	 * Software parameters
	 */

	swp = alloca(snd_pcm_sw_params_sizeof());
	memset(hwp, 0, snd_pcm_sw_params_sizeof());
	snd_pcm_sw_params_current(h, swp);

	r = snd_pcm_sw_params_set_avail_min(h, swp, period_size);

//.........这里部分代码省略.........

开发者ID:truthbk，项目名称:spotifyd，代码行数:101，代码来源:alsa-audio.c

示例8: alsaio_setup

static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate,
                        int nfrags, int frag_size)
{
    int bufsizeforthis, err;
    snd_pcm_hw_params_t* hw_params;
    unsigned int tmp_uint;
    snd_pcm_uframes_t tmp_snd_pcm_uframes;

    if (sys_verbose)
    {
        if (out)
            post("configuring sound output...");
        else post("configuring sound input...");
    }

    /* set hardware parameters... */
    snd_pcm_hw_params_alloca(&hw_params);

    /* get the default params */
    err = snd_pcm_hw_params_any(dev->a_handle, hw_params);
    check_error(err, "snd_pcm_hw_params_any");

    /* try to set interleaved access */
    err = snd_pcm_hw_params_set_access(dev->a_handle,
                                       hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
    if (err < 0)
        return (-1);
    check_error(err, "snd_pcm_hw_params_set_access");
#if 0       /* enable this to print out which formats are available */
    {
        int i;
        for (i = 0; i <= SND_PCM_FORMAT_LAST; i++)
            fprintf(stderr, "%d -> %d\n",
                    i, snd_pcm_hw_params_test_format(dev->a_handle, hw_params, i));
    }
#endif
    /* Try to set 32 bit format first */
    err = snd_pcm_hw_params_set_format(dev->a_handle,
                                       hw_params, SND_PCM_FORMAT_S32);
    if (err < 0)
    {
        /* fprintf(stderr,
            "PD-ALSA: 32 bit format not available - trying 24\n"); */
        err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,
                                           SND_PCM_FORMAT_S24_3LE);
        if (err < 0)
        {
            /* fprintf(stderr,
                "PD-ALSA: 32/24 bit format not available - using 16\n"); */
            err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,
                                               SND_PCM_FORMAT_S16);
            check_error(err, "snd_pcm_hw_params_set_format");
            dev->a_sampwidth = 2;
        }
        else dev->a_sampwidth = 3;
    }
    else dev->a_sampwidth = 4;

    if (sys_verbose)
        post("Sample width set to %d bytes", dev->a_sampwidth);

    /* set the subformat */
    err = snd_pcm_hw_params_set_subformat(dev->a_handle,
                                          hw_params, SND_PCM_SUBFORMAT_STD);
    check_error(err, "snd_pcm_hw_params_set_subformat");

    /* set the number of channels */
    tmp_uint = *channels;
    err = snd_pcm_hw_params_set_channels_min(dev->a_handle,
            hw_params, &tmp_uint);
    check_error(err, "snd_pcm_hw_params_set_channels");
    if (tmp_uint != (unsigned)*channels)
        post("ALSA: set %s channels to %d", (out?"output":"input"), tmp_uint);
    *channels = tmp_uint;
    dev->a_channels = *channels;

    /* set the sampling rate */
    err = snd_pcm_hw_params_set_rate_min(dev->a_handle, hw_params,
                                         (unsigned int *)rate, 0);
    check_error(err, "snd_pcm_hw_params_set_rate_min (input)");
#if 0
    err = snd_pcm_hw_params_get_rate(hw_params, &subunitdir);
    post("input sample rate %d", err);
#endif

    /* post("frag size %d, nfrags %d", frag_size, nfrags); */
    /* set "period size" */
#ifdef ALSAAPI9
    err = snd_pcm_hw_params_set_period_size_near(dev->a_handle,
            hw_params, (snd_pcm_uframes_t)frag_size, 0);
#else
    tmp_snd_pcm_uframes = frag_size;
    err = snd_pcm_hw_params_set_period_size_near(dev->a_handle,
            hw_params, &tmp_snd_pcm_uframes, 0);
#endif
    check_error(err, "snd_pcm_hw_params_set_period_size_near (input)");

    /* set the buffer size */
#ifdef ALSAAPI9
    err = snd_pcm_hw_params_set_buffer_size_near(dev->a_handle,
//.........这里部分代码省略.........

开发者ID:bryansum，项目名称:pdlib，代码行数:101，代码来源:s_audio_alsa.c

示例9: snd_pcm_hw_params_alloca

bool CAESinkALSA::InitializeHW(AEAudioFormat &format)
{
  snd_pcm_hw_params_t *hw_params;

  snd_pcm_hw_params_alloca(&hw_params);
  memset(hw_params, 0, snd_pcm_hw_params_sizeof());

  snd_pcm_hw_params_any(m_pcm, hw_params);
  snd_pcm_hw_params_set_access(m_pcm, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);

  unsigned int sampleRate   = format.m_sampleRate;
  unsigned int channelCount = format.m_channelLayout.Count();
  snd_pcm_hw_params_set_rate_near    (m_pcm, hw_params, &sampleRate, NULL);
  snd_pcm_hw_params_set_channels_near(m_pcm, hw_params, &channelCount);

  /* ensure we opened X channels or more */
  if (format.m_channelLayout.Count() > channelCount)
  {
    CLog::Log(LOGERROR, "CAESinkALSA::InitializeHW - Unable to open the required number of channels");
    return false;
  }

  /* update the channelLayout to what we managed to open */
  format.m_channelLayout.Reset();
  for (unsigned int i = 0; i < channelCount; ++i)
    format.m_channelLayout += ALSAChannelMap[i];

  snd_pcm_format_t fmt = AEFormatToALSAFormat(format.m_dataFormat);
  if (fmt == SND_PCM_FORMAT_UNKNOWN)
  {
    /* if we dont support the requested format, fallback to float */
    format.m_dataFormat = AE_FMT_FLOAT;
    fmt                 = SND_PCM_FORMAT_FLOAT;
  }

  /* try the data format */
  if (snd_pcm_hw_params_set_format(m_pcm, hw_params, fmt) < 0)
  {
    /* if the chosen format is not supported, try each one in decending order */
    CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Your hardware does not support %s, trying other formats", CAEUtil::DataFormatToStr(format.m_dataFormat));
    for (enum AEDataFormat i = AE_FMT_MAX; i > AE_FMT_INVALID; i = (enum AEDataFormat)((int)i - 1))
    {
      if (AE_IS_RAW(i) || i == AE_FMT_MAX)
        continue;

      if (m_passthrough && i != AE_FMT_S16BE && i != AE_FMT_S16LE)
	continue;

      fmt = AEFormatToALSAFormat(i);

      if (fmt == SND_PCM_FORMAT_UNKNOWN || snd_pcm_hw_params_set_format(m_pcm, hw_params, fmt) < 0)
      {
        fmt = SND_PCM_FORMAT_UNKNOWN;
        continue;
      }

      int fmtBits = CAEUtil::DataFormatToBits(i);
      int bits    = snd_pcm_hw_params_get_sbits(hw_params);
      if (bits != fmtBits)
      {
        /* if we opened in 32bit and only have 24bits, pack into 24 */
        if (fmtBits == 32 && bits == 24)
          i = AE_FMT_S24NE4;
        else
          continue;
      }

      /* record that the format fell back to X */
      format.m_dataFormat = i;
      CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Using data format %s", CAEUtil::DataFormatToStr(format.m_dataFormat));
      break;
    }

    /* if we failed to find a valid output format */
    if (fmt == SND_PCM_FORMAT_UNKNOWN)
    {
      CLog::Log(LOGERROR, "CAESinkALSA::InitializeHW - Unable to find a suitable output format");
      return false;
    }
  }

  unsigned int periods;

  snd_pcm_uframes_t periodSize, bufferSize;
  snd_pcm_hw_params_get_buffer_size_max(hw_params, &bufferSize);

  bufferSize  = std::min(bufferSize, (snd_pcm_uframes_t)8192);
  periodSize  = bufferSize / ALSA_PERIODS;
  periods     = ALSA_PERIODS;

  CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Request: periodSize %lu, periods %u, bufferSize %lu", periodSize, periods, bufferSize);

  /* work on a copy of the hw params */
  snd_pcm_hw_params_t *hw_params_copy;
  snd_pcm_hw_params_alloca(&hw_params_copy);

  /* try to set the buffer size then the period size */
  snd_pcm_hw_params_copy(hw_params_copy, hw_params);
  snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize);
  snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL);
//.........这里部分代码省略.........

开发者ID:JohnsonAugustine，项目名称:xbmc-rbp，代码行数:101，代码来源:AESinkALSA.cpp

示例10: sndout_alsa_start

int sndout_alsa_start(int rate_, int stereo)
{
	snd_pcm_hw_params_t *hwparams = NULL;
	unsigned int rate = rate_;
	int samples, shift;
	int ret;

	samples = rate * 40 / 1000;
	for (shift = 8; (1 << shift) < samples; shift++)
		;
	period_size = 1 << shift;
	buffer_size = 8 * period_size;

	snd_pcm_hw_params_alloca(&hwparams);

	ret  = snd_pcm_hw_params_any(handle, hwparams);
	ret |= snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
	ret |= snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
	ret |= snd_pcm_hw_params_set_channels(handle, hwparams, stereo ? 2 : 1);
	ret |= snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, 0);
	ret |= snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
	ret |= snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, NULL);

	if (ret != 0) {
		fprintf(stderr, PFX "failed to set hwparams\n");
		goto fail;
	}

	ret = snd_pcm_hw_params(handle, hwparams);
	if (ret != 0) {
		fprintf(stderr, PFX "failed to apply hwparams: %d\n", ret);
		goto fail;
	}

	snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
	snd_pcm_hw_params_get_period_size(hwparams, &period_size, NULL);
	snd_pcm_hw_params_get_channels(hwparams, &channels);

	silent_period = realloc(silent_period, period_size * 2 * channels);
	if (silent_period != NULL)
		memset(silent_period, 0, period_size * 2 * channels);

	ret = snd_pcm_prepare(handle);
	if (ret != 0) {
		fprintf(stderr, PFX "snd_pcm_prepare failed: %d\n", ret);
		goto fail;
	}

	ret = snd_pcm_start(handle);
	if (ret != 0) {
		fprintf(stderr, PFX "snd_pcm_start failed: %d\n", ret);
		goto fail;
	}

	failure_counter = 0;

	return 0;

fail:
	// to flush out redirected logs
	fflush(stdout);
	fflush(stderr);
	return -1;
}

开发者ID:CraigularB，项目名称:Provenance，代码行数:64，代码来源:sndout_alsa.c

示例11: alsa_open

static snd_pcm_t *
alsa_open (int channels, unsigned samplerate, int realtime)
{	const char * device = "default" ;
	snd_pcm_t *alsa_dev = NULL ;
	snd_pcm_hw_params_t *hw_params ;
	snd_pcm_uframes_t buffer_size ;
	snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ;
	snd_pcm_sw_params_t *sw_params ;

	int err ;

	if (realtime)
	{	alsa_period_size = 256 ;
		alsa_buffer_frames = 3 * alsa_period_size ;
		}
	else
	{	alsa_period_size = 1024 ;
		alsa_buffer_frames = 4 * alsa_period_size ;
		} ;

	if ((err = snd_pcm_open (&alsa_dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
	{	fprintf (stderr, "cannot open audio device \"%s\" (%s)\n", device, snd_strerror (err)) ;
		goto catch_error ;
		} ;

	snd_pcm_nonblock (alsa_dev, 0) ;

	if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
	{	fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_any (alsa_dev, hw_params)) < 0)
	{	fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_access (alsa_dev, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
	{	fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_format (alsa_dev, hw_params, SND_PCM_FORMAT_FLOAT)) < 0)
	{	fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_rate_near (alsa_dev, hw_params, &samplerate, 0)) < 0)
	{	fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_channels (alsa_dev, hw_params, channels)) < 0)
	{	fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_buffer_size_near (alsa_dev, hw_params, &alsa_buffer_frames)) < 0)
	{	fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_period_size_near (alsa_dev, hw_params, &alsa_period_size, 0)) < 0)
	{	fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params (alsa_dev, hw_params)) < 0)
	{	fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	/* extra check: if we have only one period, this code won't work */
	snd_pcm_hw_params_get_period_size (hw_params, &alsa_period_size, 0) ;
	snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_size) ;
	if (alsa_period_size == buffer_size)
	{	fprintf (stderr, "Can't use period equal to buffer size (%lu == %lu)", alsa_period_size, buffer_size) ;
		goto catch_error ;
		} ;

	snd_pcm_hw_params_free (hw_params) ;

	if ((err = snd_pcm_sw_params_malloc (&sw_params)) != 0)
	{	fprintf (stderr, "%s: snd_pcm_sw_params_malloc: %s", __func__, snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_sw_params_current (alsa_dev, sw_params)) != 0)
	{	fprintf (stderr, "%s: snd_pcm_sw_params_current: %s", __func__, snd_strerror (err)) ;
		goto catch_error ;
		} ;

	/* note: set start threshold to delay start until the ring buffer is full */
	snd_pcm_sw_params_current (alsa_dev, sw_params) ;

	if ((err = snd_pcm_sw_params_set_start_threshold (alsa_dev, sw_params, buffer_size)) < 0)
	{	fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

//.........这里部分代码省略.........

开发者ID:5in4，项目名称:libsox.dll，代码行数:101，代码来源:sndfile-play.c

示例12: alsa_init


//.........这里部分代码省略.........
		if (rc < 0)
		{
			device = "pulse";
			rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
		}

		if (rc < 0)
			printf("ALSA: unable to automatically determine audio device.\n");
	}
	else {
		rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
	}

	if (rc < 0)
	{
		fprintf(stderr, "ALSA: unable to open PCM device %s: %s\n", device.c_str(), snd_strerror(rc));
		return;
	}

	printf("ALSA: Successfully initialized \"%s\"\n", device.c_str());

	/* Allocate a hardware parameters object. */
	snd_pcm_hw_params_alloca(&params);

	/* Fill it in with default values. */
	rc=snd_pcm_hw_params_any(handle, params);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_any %s\n", snd_strerror(rc));
		return;
	}

	/* Set the desired hardware parameters. */

	/* Interleaved mode */
	rc=snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_access %s\n", snd_strerror(rc));
		return;
	}

	/* Signed 16-bit little-endian format */
	rc=snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_format %s\n", snd_strerror(rc));
		return;
	}

	/* Two channels (stereo) */
	rc=snd_pcm_hw_params_set_channels(handle, params, 2);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_channels %s\n", snd_strerror(rc));
		return;
	}

	/* 44100 bits/second sampling rate (CD quality) */
	val = 44100;
	rc=snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_rate_near %s\n", snd_strerror(rc));
		return;
	}

	/* Set period size to settings.aica.BufferSize frames. */
	period_size = settings.aica.BufferSize;
	rc=snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, &dir);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_buffer_size_near %s\n", snd_strerror(rc));
		return;
	}
	else
	{
		printf("ALSA: period size set to %ld\n", period_size);
	}

	buffer_size = (44100 * 100 /* settings.omx.Audio_Latency */ / 1000 / period_size + 1) * period_size;
	rc=snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_buffer_size_near %s\n", snd_strerror(rc));
		return;
	}
	else
	{
		printf("ALSA: buffer size set to %ld\n", buffer_size);
	}

	/* Write the parameters to the driver */
	rc = snd_pcm_hw_params(handle, params);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Unable to set hw parameters: %s\n", snd_strerror(rc));
		return;
	}
}

开发者ID:reicast，项目名称:reicast-emulator，代码行数:101，代码来源:audiobackend_alsa.cpp

示例13: alsa_open_capture

static ALCenum alsa_open_capture(ALCdevice *Device, const ALCchar *deviceName)
{
    const char *driver = NULL;
    snd_pcm_hw_params_t *hp;
    snd_pcm_uframes_t bufferSizeInFrames;
    snd_pcm_uframes_t periodSizeInFrames;
    ALboolean needring = AL_FALSE;
    snd_pcm_format_t format;
    const char *funcerr;
    alsa_data *data;
    int err;

    if(deviceName)
    {
        size_t idx;

        if(!allCaptureDevNameMap)
            allCaptureDevNameMap = probe_devices(SND_PCM_STREAM_CAPTURE, &numCaptureDevNames);

        for(idx = 0;idx < numCaptureDevNames;idx++)
        {
            if(strcmp(deviceName, allCaptureDevNameMap[idx].name) == 0)
            {
                driver = allCaptureDevNameMap[idx].device;
                break;
            }
        }
        if(idx == numCaptureDevNames)
            return ALC_INVALID_VALUE;
    }
    else
    {
        deviceName = alsaDevice;
        driver = GetConfigValue("alsa", "capture", "default");
    }

    data = (alsa_data*)calloc(1, sizeof(alsa_data));

    err = snd_pcm_open(&data->pcmHandle, driver, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
    if(err < 0)
    {
        ERR("Could not open capture device '%s': %s\n", driver, snd_strerror(err));
        free(data);
        return ALC_INVALID_VALUE;
    }

    format = -1;
    switch(Device->FmtType)
    {
        case DevFmtByte:
            format = SND_PCM_FORMAT_S8;
            break;
        case DevFmtUByte:
            format = SND_PCM_FORMAT_U8;
            break;
        case DevFmtShort:
            format = SND_PCM_FORMAT_S16;
            break;
        case DevFmtUShort:
            format = SND_PCM_FORMAT_U16;
            break;
        case DevFmtInt:
            format = SND_PCM_FORMAT_S32;
            break;
        case DevFmtUInt:
            format = SND_PCM_FORMAT_U32;
            break;
        case DevFmtFloat:
            format = SND_PCM_FORMAT_FLOAT;
            break;
    }

    funcerr = NULL;
    bufferSizeInFrames = maxu(Device->UpdateSize*Device->NumUpdates,
                              100*Device->Frequency/1000);
    periodSizeInFrames = minu(bufferSizeInFrames, 25*Device->Frequency/1000);

    snd_pcm_hw_params_malloc(&hp);
#define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error
    CHECK(snd_pcm_hw_params_any(data->pcmHandle, hp));
    /* set interleaved access */
    CHECK(snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_RW_INTERLEAVED));
    /* set format (implicitly sets sample bits) */
    CHECK(snd_pcm_hw_params_set_format(data->pcmHandle, hp, format));
    /* set channels (implicitly sets frame bits) */
    CHECK(snd_pcm_hw_params_set_channels(data->pcmHandle, hp, ChannelsFromDevFmt(Device->FmtChans)));
    /* set rate (implicitly constrains period/buffer parameters) */
    CHECK(snd_pcm_hw_params_set_rate(data->pcmHandle, hp, Device->Frequency, 0));
    /* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
    if(snd_pcm_hw_params_set_buffer_size_min(data->pcmHandle, hp, &bufferSizeInFrames) < 0)
    {
        TRACE("Buffer too large, using intermediate ring buffer\n");
        needring = AL_TRUE;
        CHECK(snd_pcm_hw_params_set_buffer_size_near(data->pcmHandle, hp, &bufferSizeInFrames));
    }
    /* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
    CHECK(snd_pcm_hw_params_set_period_size_near(data->pcmHandle, hp, &periodSizeInFrames, NULL));
    /* install and prepare hardware configuration */
    CHECK(snd_pcm_hw_params(data->pcmHandle, hp));
    /* retrieve configuration info */
//.........这里部分代码省略.........

开发者ID:LighFusion，项目名称:surreal，代码行数:101，代码来源:alsa.c

示例14: printf

int AudioAlsa::setHWParams( const ch_cnt_t _channels, snd_pcm_access_t _access )
{
	int err, dir;

	// choose all parameters
	if( ( err = snd_pcm_hw_params_any( m_handle, m_hwParams ) ) < 0 )
	{
		printf( "Broken configuration for playback: no configurations "
				"available: %s\n", snd_strerror( err ) );
		return err;
	}

	// set the interleaved read/write format
	if( ( err = snd_pcm_hw_params_set_access( m_handle, m_hwParams,
							_access ) ) < 0 )
	{
		printf( "Access type not available for playback: %s\n",
							snd_strerror( err ) );
		return err;
	}

	// set the sample format
	if( ( snd_pcm_hw_params_set_format( m_handle, m_hwParams,
						SND_PCM_FORMAT_S16_LE ) ) < 0 )
	{
		if( ( snd_pcm_hw_params_set_format( m_handle, m_hwParams,
						SND_PCM_FORMAT_S16_BE ) ) < 0 )
		{
			printf( "Neither little- nor big-endian available for "
					"playback: %s\n", snd_strerror( err ) );
			return err;
		}
		m_convertEndian = isLittleEndian();
	}
	else
	{
		m_convertEndian = !isLittleEndian();
	}

	// set the count of channels
	if( ( err = snd_pcm_hw_params_set_channels( m_handle, m_hwParams,
							_channels ) ) < 0 )
	{
		printf( "Channel count (%i) not available for playbacks: %s\n"
				"(Does your soundcard not support surround?)\n",
					_channels, snd_strerror( err ) );
		return err;
	}

	// set the sample rate
	if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
						sampleRate(), 0 ) ) < 0 )
	{
		if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
				mixer()->baseSampleRate(), 0 ) ) < 0 )
		{
			printf( "Could not set sample rate: %s\n",
							snd_strerror( err ) );
			return err;
		}
	}

	m_periodSize = mixer()->framesPerPeriod();
	m_bufferSize = m_periodSize * 8;
	dir = 0;
	err = snd_pcm_hw_params_set_period_size_near( m_handle, m_hwParams,
							&m_periodSize, &dir );
	if( err < 0 )
	{
		printf( "Unable to set period size %lu for playback: %s\n",
					m_periodSize, snd_strerror( err ) );
		return err;
	}
	dir = 0;
	err = snd_pcm_hw_params_get_period_size( m_hwParams, &m_periodSize,
									&dir );
	if( err < 0 )
	{
		printf( "Unable to get period size for playback: %s\n",
							snd_strerror( err ) );
	}

	dir = 0;
	err = snd_pcm_hw_params_set_buffer_size_near( m_handle, m_hwParams,
								&m_bufferSize );
	if( err < 0 )
	{
		printf( "Unable to set buffer size %lu for playback: %s\n",
					m_bufferSize, snd_strerror( err ) );
		return ( err );
	}
	err = snd_pcm_hw_params_get_buffer_size( m_hwParams, &m_bufferSize );

	if( 2 * m_periodSize > m_bufferSize )
	{
		printf( "buffer to small, could not use\n" );
		return ( err );
	}


//.........这里部分代码省略.........

开发者ID:uro5h，项目名称:lmms，代码行数:101，代码来源:AudioAlsa.cpp

示例15: initialize_device

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C++ snd_pcm_hw_params_set_buffer_size_near函数代码示例

示例1: rdpsnd_alsa_set_params

示例2: drvHostALSAAudioOpen

示例3: printf

示例4: sound_init_alsa

示例5: set_params

示例6: GLOBAL_DEF

示例7: alloca

示例8: alsaio_setup

示例9: snd_pcm_hw_params_alloca

示例10: sndout_alsa_start

示例11: alsa_open

示例12: alsa_init

示例13: alsa_open_capture

示例14: printf

示例15: initialize_device

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