native/libjsound/PLATFORM_API_LinuxOS_ALSA_PCM.c

/*
 * Copyright (c) 2002, 2011, Oracle and/or its affiliates. All rights reserved.
 * DO NOT ALTER OR REMOVE COPYRIGHT NOTICES OR THIS FILE HEADER.
 *
 * This code is free software; you can redistribute it and/or modify it
 * under the terms of the GNU General Public License version 2 only, as
 * published by the Free Software Foundation.  Oracle designates this
 * particular file as subject to the "Classpath" exception as provided
 * by Oracle in the LICENSE file that accompanied this code.
 *
 * This code is distributed in the hope that it will be useful, but WITHOUT
 * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
 * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License
 * version 2 for more details (a copy is included in the LICENSE file that
 * accompanied this code).
 *
 * You should have received a copy of the GNU General Public License version
 * 2 along with this work; if not, write to the Free Software Foundation,
 * Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA.
 *
 * Please contact Oracle, 500 Oracle Parkway, Redwood Shores, CA 94065 USA
 * or visit www.oracle.com if you need additional information or have any
 * questions.
 */

#define USE_ERROR
#define USE_TRACE

#include "PLATFORM_API_LinuxOS_ALSA_PCMUtils.h"
#include "PLATFORM_API_LinuxOS_ALSA_CommonUtils.h"
#include "DirectAudio.h"

#if USE_DAUDIO == TRUE

// GetPosition method 1: based on how many bytes are passed to the kernel driver
//                       + does not need much processor resources
//                       - not very exact, "jumps"
// GetPosition method 2: ask kernel about actual position of playback.
//                       - very exact
//                       - switch to kernel layer for each call
// GetPosition method 3: use snd_pcm_avail() call - not yet in official ALSA
// quick tests on a Pentium 200MMX showed max. 1.5% processor usage
// for playing back a CD-quality file and printing 20x per second a line
// on the console with the current time. So I guess performance is not such a
// factor here.
//#define GET_POSITION_METHOD1
#define GET_POSITION_METHOD2


// The default time for a period in microseconds.
// For very small buffers, only 2 periods are used.
#define DEFAULT_PERIOD_TIME 20000 /* 20ms */

///// implemented functions of DirectAudio.h

INT32 DAUDIO_GetDirectAudioDeviceCount() {
    return (INT32) getAudioDeviceCount();
}


INT32 DAUDIO_GetDirectAudioDeviceDescription(INT32 mixerIndex, DirectAudioDeviceDescription* description) {
    ALSA_AudioDeviceDescription adesc;

    adesc.index = (int) mixerIndex;
    adesc.strLen = DAUDIO_STRING_LENGTH;

    adesc.maxSimultaneousLines = (int*) (&(description->maxSimulLines));
    adesc.deviceID = &(description->deviceID);
    adesc.name = description->name;
    adesc.vendor = description->vendor;
    adesc.description = description->description;
    adesc.version = description->version;

    return getAudioDeviceDescriptionByIndex(&adesc);
}

#define MAX_BIT_INDEX 6
// returns
// 6: for anything above 24-bit
// 5: for 4 bytes sample size, 24-bit
// 4: for 3 bytes sample size, 24-bit
// 3: for 3 bytes sample size, 20-bit
// 2: for 2 bytes sample size, 16-bit
// 1: for 1 byte sample size, 8-bit
// 0: for anything else
int getBitIndex(int sampleSizeInBytes, int significantBits) {
    if (significantBits > 24) return 6;
    if (sampleSizeInBytes == 4 && significantBits == 24) return 5;
    if (sampleSizeInBytes == 3) {
        if (significantBits == 24) return 4;
        if (significantBits == 20) return 3;
    }
    if (sampleSizeInBytes == 2 && significantBits == 16) return 2;
    if (sampleSizeInBytes == 1 && significantBits == 8) return 1;
    return 0;
}

int getSampleSizeInBytes(int bitIndex, int sampleSizeInBytes) {
    switch(bitIndex) {
    case 1: return 1;
    case 2: return 2;
    case 3: /* fall through */
    case 4: return 3;
    case 5: return 4;
    }
    return sampleSizeInBytes;
}

int getSignificantBits(int bitIndex, int significantBits) {
    switch(bitIndex) {
    case 1: return 8;
    case 2: return 16;
    case 3: return 20;
    case 4: /* fall through */
    case 5: return 24;
    }
    return significantBits;
}

void DAUDIO_GetFormats(INT32 mixerIndex, INT32 deviceID, int isSource, void* creator) {
    snd_pcm_t* handle;
    snd_pcm_format_mask_t* formatMask;
    snd_pcm_format_t format;
    snd_pcm_hw_params_t* hwParams;
    int handledBits[MAX_BIT_INDEX+1];

    int ret;
    int sampleSizeInBytes, significantBits, isSigned, isBigEndian, enc;
    int origSampleSizeInBytes, origSignificantBits;
    unsigned int channels, minChannels, maxChannels;
    int rate, bitIndex;

    for (bitIndex = 0; bitIndex <= MAX_BIT_INDEX; bitIndex++) handledBits[bitIndex] = FALSE;
    if (openPCMfromDeviceID(deviceID, &handle, isSource, TRUE /*query hardware*/) < 0) {
        return;
    }
    ret = snd_pcm_format_mask_malloc(&formatMask);
    if (ret != 0) {
        ERROR1("snd_pcm_format_mask_malloc returned error %d\n", ret);
    } else {
        ret = snd_pcm_hw_params_malloc(&hwParams);
        if (ret != 0) {
            ERROR1("snd_pcm_hw_params_malloc returned error %d\n", ret);
        } else {
            ret = snd_pcm_hw_params_any(handle, hwParams);
            /* snd_pcm_hw_params_any can return a positive value on success too */
            if (ret < 0) {
                 ERROR1("snd_pcm_hw_params_any returned error %d\n", ret);
            } else {
                /* for the logic following this code, set ret to 0 to indicate success */
                ret = 0;
            }
        }
        snd_pcm_hw_params_get_format_mask(hwParams, formatMask);
        if (ret == 0) {
            ret = snd_pcm_hw_params_get_channels_min(hwParams, &minChannels);
            if (ret != 0) {
                ERROR1("snd_pcm_hw_params_get_channels_min returned error %d\n", ret);
            }
        }
        if (ret == 0) {
            ret = snd_pcm_hw_params_get_channels_max(hwParams, &maxChannels);
            if (ret != 0) {
                ERROR1("snd_pcm_hw_params_get_channels_max returned error %d\n", ret);
            }
        }

        // since we queried the hw: device, for many soundcards, it will only
        // report the maximum number of channels (which is the only way to talk
        // to the hw: device). Since we will, however, open the plughw: device
        // when opening the Source/TargetDataLine, we can safely assume that
        // also the channels 1..maxChannels are available.
#ifdef ALSA_PCM_USE_PLUGHW
        minChannels = 1;
#endif
        if (ret == 0) {
            // plughw: supports any sample rate
            rate = -1;
            for (format = 0; format <= SND_PCM_FORMAT_LAST; format++) {
                if (snd_pcm_format_mask_test(formatMask, format)) {
                    // format exists
                    if (getFormatFromAlsaFormat(format, &origSampleSizeInBytes,
                                                &origSignificantBits,
                                                &isSigned, &isBigEndian, &enc)) {
                        // now if we use plughw:, we can use any bit size below the
                        // natively supported ones. Some ALSA drivers only support the maximum
                        // bit size, so we add any sample rates below the reported one.
                        // E.g. this iteration reports support for 16-bit.
                        // getBitIndex will return 2, so it will add entries for
                        // 16-bit (bitIndex=2) and in the next do-while loop iteration,
                        // it will decrease bitIndex and will therefore add 8-bit support.
                        bitIndex = getBitIndex(origSampleSizeInBytes, origSignificantBits);
                        do {
                            if (bitIndex == 0
                                || bitIndex == MAX_BIT_INDEX
                                || !handledBits[bitIndex]) {
                                handledBits[bitIndex] = TRUE;
                                sampleSizeInBytes = getSampleSizeInBytes(bitIndex, origSampleSizeInBytes);
                                significantBits = getSignificantBits(bitIndex, origSignificantBits);
                                if (maxChannels - minChannels > MAXIMUM_LISTED_CHANNELS) {
                                    // avoid too many channels explicitly listed
                                    // just add -1, min, and max
                                    DAUDIO_AddAudioFormat(creator, significantBits,
                                                          -1, -1, rate,
                                                          enc, isSigned, isBigEndian);
                                    DAUDIO_AddAudioFormat(creator, significantBits,
                                                          sampleSizeInBytes * minChannels,
                                                          minChannels, rate,
                                                          enc, isSigned, isBigEndian);
                                    DAUDIO_AddAudioFormat(creator, significantBits,
                                                          sampleSizeInBytes * maxChannels,
                                                          maxChannels, rate,
                                                          enc, isSigned, isBigEndian);
                                } else {
                                    for (channels = minChannels; channels <= maxChannels; channels++) {
                                        DAUDIO_AddAudioFormat(creator, significantBits,
                                                              sampleSizeInBytes * channels,
                                                              channels, rate,
                                                              enc, isSigned, isBigEndian);
                                    }
                                }
                            }
#ifndef ALSA_PCM_USE_PLUGHW
                            // without plugin, do not add fake formats
                            break;
#endif
                        } while (--bitIndex > 0);
                    } else {
                        TRACE1("could not get format from alsa for format %d\n", format);
                    }
                } else {
                    //TRACE1("Format %d not supported\n", format);
                }
            } // for loop
            snd_pcm_hw_params_free(hwParams);
        }
        snd_pcm_format_mask_free(formatMask);
    }
    snd_pcm_close(handle);
}

/** Workaround for cr 7033899, 7030629:
 * dmix plugin doesn't like flush (snd_pcm_drop) when the buffer is empty
 * (just opened, underruned or already flushed).
 * Sometimes it causes PCM falls to -EBADFD error,
 * sometimes causes bufferSize change.
 * To prevent unnecessary flushes AlsaPcmInfo::isRunning & isFlushed are used.
 */
/* ******* ALSA PCM INFO ******************** */
typedef struct tag_AlsaPcmInfo {
    snd_pcm_t* handle;
    snd_pcm_hw_params_t* hwParams;
    snd_pcm_sw_params_t* swParams;
    int bufferSizeInBytes;
    int frameSize; // storage size in Bytes
    unsigned int periods;
    snd_pcm_uframes_t periodSize;
    short int isRunning;    // see comment above
    short int isFlushed;    // see comment above
#ifdef GET_POSITION_METHOD2
    // to be used exclusively by getBytePosition!
    snd_pcm_status_t* positionStatus;
#endif
} AlsaPcmInfo;


int setStartThresholdNoCommit(AlsaPcmInfo* info, int useThreshold) {
    int ret;
    int threshold;

    if (useThreshold) {
        // start device whenever anything is written to the buffer
        threshold = 1;
    } else {
        // never start the device automatically
        threshold = 2000000000; /* near UINT_MAX */
    }
    ret = snd_pcm_sw_params_set_start_threshold(info->handle, info->swParams, threshold);
    if (ret < 0) {
        ERROR1("Unable to set start threshold mode: %s\n", snd_strerror(ret));
        return FALSE;
    }
    return TRUE;
}

int setStartThreshold(AlsaPcmInfo* info, int useThreshold) {
    int ret = 0;

    if (!setStartThresholdNoCommit(info, useThreshold)) {
        ret = -1;
    }
    if (ret == 0) {
        // commit it
        ret = snd_pcm_sw_params(info->handle, info->swParams);
        if (ret < 0) {
            ERROR1("Unable to set sw params: %s\n", snd_strerror(ret));
        }
    }
    return (ret == 0)?TRUE:FALSE;
}


// returns TRUE if successful
int setHWParams(AlsaPcmInfo* info,
                float sampleRate,
                int channels,
                int bufferSizeInFrames,
                snd_pcm_format_t format) {
    unsigned int rrate, periodTime, periods;
    int ret, dir;
    snd_pcm_uframes_t alsaBufferSizeInFrames = (snd_pcm_uframes_t) bufferSizeInFrames;

    /* choose all parameters */
    ret = snd_pcm_hw_params_any(info->handle, info->hwParams);
    if (ret < 0) {
        ERROR1("Broken configuration: no configurations available: %s\n", snd_strerror(ret));
        return FALSE;
    }
    /* set the interleaved read/write format */
    ret = snd_pcm_hw_params_set_access(info->handle, info->hwParams, SND_PCM_ACCESS_RW_INTERLEAVED);
    if (ret < 0) {
        ERROR1("SND_PCM_ACCESS_RW_INTERLEAVED access type not available: %s\n", snd_strerror(ret));
        return FALSE;
    }
    /* set the sample format */
    ret = snd_pcm_hw_params_set_format(info->handle, info->hwParams, format);
    if (ret < 0) {
        ERROR1("Sample format not available: %s\n", snd_strerror(ret));
        return FALSE;
    }
    /* set the count of channels */
    ret = snd_pcm_hw_params_set_channels(info->handle, info->hwParams, channels);
    if (ret < 0) {
        ERROR2("Channels count (%d) not available: %s\n", channels, snd_strerror(ret));
        return FALSE;
    }
    /* set the stream rate */
    rrate = (int) (sampleRate + 0.5f);
    dir = 0;
    ret = snd_pcm_hw_params_set_rate_near(info->handle, info->hwParams, &rrate, &dir);
    if (ret < 0) {
        ERROR2("Rate %dHz not available for playback: %s\n", (int) (sampleRate+0.5f), snd_strerror(ret));
        return FALSE;
    }
    if ((rrate-sampleRate > 2) || (rrate-sampleRate < - 2)) {
        ERROR2("Rate doesn't match (requested %2.2fHz, got %dHz)\n", sampleRate, rrate);
        return FALSE;
    }
    /* set the buffer time */
    ret = snd_pcm_hw_params_set_buffer_size_near(info->handle, info->hwParams, &alsaBufferSizeInFrames);
    if (ret < 0) {
        ERROR2("Unable to set buffer size to %d frames: %s\n",
               (int) alsaBufferSizeInFrames, snd_strerror(ret));
        return FALSE;
    }
    bufferSizeInFrames = (int) alsaBufferSizeInFrames;
    /* set the period time */
    if (bufferSizeInFrames > 1024) {
        dir = 0;
        periodTime = DEFAULT_PERIOD_TIME;
        ret = snd_pcm_hw_params_set_period_time_near(info->handle, info->hwParams, &periodTime, &dir);
        if (ret < 0) {
            ERROR2("Unable to set period time to %d: %s\n", DEFAULT_PERIOD_TIME, snd_strerror(ret));
            return FALSE;
        }
    } else {
        /* set the period count for very small buffer sizes to 2 */
        dir = 0;
        periods = 2;
        ret = snd_pcm_hw_params_set_periods_near(info->handle, info->hwParams, &periods, &dir);
        if (ret < 0) {
            ERROR2("Unable to set period count to %d: %s\n", /*periods*/ 2, snd_strerror(ret));
            return FALSE;
        }
    }
    /* write the parameters to device */
    ret = snd_pcm_hw_params(info->handle, info->hwParams);
    if (ret < 0) {
        ERROR1("Unable to set hw params: %s\n", snd_strerror(ret));
        return FALSE;
    }
    return TRUE;
}

// returns 1 if successful
int setSWParams(AlsaPcmInfo* info) {
    int ret;

    /* get the current swparams */
    ret = snd_pcm_sw_params_current(info->handle, info->swParams);
    if (ret < 0) {
        ERROR1("Unable to determine current swparams: %s\n", snd_strerror(ret));
        return FALSE;
    }
    /* never start the transfer automatically */
    if (!setStartThresholdNoCommit(info, FALSE /* don't use threshold */)) {
        return FALSE;
    }

    /* allow the transfer when at least period_size samples can be processed */
    ret = snd_pcm_sw_params_set_avail_min(info->handle, info->swParams, info->periodSize);
    if (ret < 0) {
        ERROR1("Unable to set avail min for playback: %s\n", snd_strerror(ret));
        return FALSE;
    }
    /* write the parameters to the playback device */
    ret = snd_pcm_sw_params(info->handle, info->swParams);
    if (ret < 0) {
        ERROR1("Unable to set sw params: %s\n", snd_strerror(ret));
        return FALSE;
    }
    return TRUE;
}

static snd_output_t* ALSA_OUTPUT = NULL;

void* DAUDIO_Open(INT32 mixerIndex, INT32 deviceID, int isSource,
                  int encoding, float sampleRate, int sampleSizeInBits,
                  int frameSize, int channels,
                  int isSigned, int isBigEndian, int bufferSizeInBytes) {
    snd_pcm_format_mask_t* formatMask;
    snd_pcm_format_t format;
    int dir;
    int ret = 0;
    AlsaPcmInfo* info = NULL;
    /* snd_pcm_uframes_t is 64 bit on 64-bit systems */
    snd_pcm_uframes_t alsaBufferSizeInFrames = 0;


    TRACE0("> DAUDIO_Open\n");
#ifdef USE_TRACE
    // for using ALSA debug dump methods
    if (ALSA_OUTPUT == NULL) {
        snd_output_stdio_attach(&ALSA_OUTPUT, stdout, 0);
    }
#endif
    if (channels <= 0) {
        ERROR1("ERROR: Invalid number of channels=%d!\n", channels);
        return NULL;
    }
    info = (AlsaPcmInfo*) malloc(sizeof(AlsaPcmInfo));
    if (!info) {
        ERROR0("Out of memory\n");
        return NULL;
    }
    memset(info, 0, sizeof(AlsaPcmInfo));
    // initial values are: stopped, flushed
    info->isRunning = 0;
    info->isFlushed = 1;

    ret = openPCMfromDeviceID(deviceID, &(info->handle), isSource, FALSE /* do open device*/);
    if (ret == 0) {
        // set to blocking mode
        snd_pcm_nonblock(info->handle, 0);
        ret = snd_pcm_hw_params_malloc(&(info->hwParams));
        if (ret != 0) {
            ERROR1("  snd_pcm_hw_params_malloc returned error %d\n", ret);
        } else {
            ret = -1;
            if (getAlsaFormatFromFormat(&format, frameSize / channels, sampleSizeInBits,
                                        isSigned, isBigEndian, encoding)) {
                if (setHWParams(info,
                                sampleRate,
                                channels,
                                bufferSizeInBytes / frameSize,
                                format)) {
                    info->frameSize = frameSize;
                    ret = snd_pcm_hw_params_get_period_size(info->hwParams, &info->periodSize, &dir);
                    if (ret < 0) {
                        ERROR1("ERROR: snd_pcm_hw_params_get_period: %s\n", snd_strerror(ret));
                    }
                    snd_pcm_hw_params_get_periods(info->hwParams, &(info->periods), &dir);
                    snd_pcm_hw_params_get_buffer_size(info->hwParams, &alsaBufferSizeInFrames);
                    info->bufferSizeInBytes = (int) alsaBufferSizeInFrames * frameSize;
                    TRACE3("  DAUDIO_Open: period size = %d frames, periods = %d. Buffer size: %d bytes.\n",
                           (int) info->periodSize, info->periods, info->bufferSizeInBytes);
                }
            }
        }
        if (ret == 0) {
            // set software parameters
            ret = snd_pcm_sw_params_malloc(&(info->swParams));
            if (ret != 0) {
                ERROR1("snd_pcm_hw_params_malloc returned error %d\n", ret);
            } else {
                if (!setSWParams(info)) {
                    ret = -1;
                }
            }
        }
        if (ret == 0) {
            // prepare device
            ret = snd_pcm_prepare(info->handle);
            if (ret < 0) {
                ERROR1("ERROR: snd_pcm_prepare: %s\n", snd_strerror(ret));
            }
        }

#ifdef GET_POSITION_METHOD2
        if (ret == 0) {
            ret = snd_pcm_status_malloc(&(info->positionStatus));
            if (ret != 0) {
                ERROR1("ERROR in snd_pcm_status_malloc: %s\n", snd_strerror(ret));
            }
        }
#endif
    }
    if (ret != 0) {
        DAUDIO_Close((void*) info, isSource);
        info = NULL;
    } else {
        // set to non-blocking mode
        snd_pcm_nonblock(info->handle, 1);
        TRACE1("< DAUDIO_Open: Opened device successfully. Handle=%p\n",
               (void*) info->handle);
    }
    return (void*) info;
}

#ifdef USE_TRACE
void printState(snd_pcm_state_t state) {
    if (state == SND_PCM_STATE_OPEN) {
        TRACE0("State: SND_PCM_STATE_OPEN\n");
    }
    else if (state == SND_PCM_STATE_SETUP) {
        TRACE0("State: SND_PCM_STATE_SETUP\n");
    }
    else if (state == SND_PCM_STATE_PREPARED) {
        TRACE0("State: SND_PCM_STATE_PREPARED\n");
    }
    else if (state == SND_PCM_STATE_RUNNING) {
        TRACE0("State: SND_PCM_STATE_RUNNING\n");
    }
    else if (state == SND_PCM_STATE_XRUN) {
        TRACE0("State: SND_PCM_STATE_XRUN\n");
    }
    else if (state == SND_PCM_STATE_DRAINING) {
        TRACE0("State: SND_PCM_STATE_DRAINING\n");
    }
    else if (state == SND_PCM_STATE_PAUSED) {
        TRACE0("State: SND_PCM_STATE_PAUSED\n");
    }
    else if (state == SND_PCM_STATE_SUSPENDED) {
        TRACE0("State: SND_PCM_STATE_SUSPENDED\n");
    }
}
#endif

int DAUDIO_Start(void* id, int isSource) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;
    int ret;
    snd_pcm_state_t state;

    TRACE0("> DAUDIO_Start\n");
    // set to blocking mode
    snd_pcm_nonblock(info->handle, 0);
    // set start mode so that it always starts as soon as data is there
    setStartThreshold(info, TRUE /* use threshold */);
    state = snd_pcm_state(info->handle);
    if (state == SND_PCM_STATE_PAUSED) {
        // in case it was stopped previously
        TRACE0("  Un-pausing...\n");
        ret = snd_pcm_pause(info->handle, FALSE);
        if (ret != 0) {
            ERROR2("  NOTE: error in snd_pcm_pause:%d: %s\n", ret, snd_strerror(ret));
        }
    }
    if (state == SND_PCM_STATE_SUSPENDED) {
        TRACE0("  Resuming...\n");
        ret = snd_pcm_resume(info->handle);
        if (ret < 0) {
            if ((ret != -EAGAIN) && (ret != -ENOSYS)) {
                ERROR2("  ERROR: error in snd_pcm_resume:%d: %s\n", ret, snd_strerror(ret));
            }
        }
    }
    if (state == SND_PCM_STATE_SETUP) {
        TRACE0("need to call prepare again...\n");
        // prepare device
        ret = snd_pcm_prepare(info->handle);
        if (ret < 0) {
            ERROR1("ERROR: snd_pcm_prepare: %s\n", snd_strerror(ret));
        }
    }
    // in case there is still data in the buffers
    ret = snd_pcm_start(info->handle);
    if (ret != 0) {
        if (ret != -EPIPE) {
            ERROR2("  NOTE: error in snd_pcm_start: %d: %s\n", ret, snd_strerror(ret));
        }
    }
    // set to non-blocking mode
    ret = snd_pcm_nonblock(info->handle, 1);
    if (ret != 0) {
        ERROR1("  ERROR in snd_pcm_nonblock: %s\n", snd_strerror(ret));
    }
    state = snd_pcm_state(info->handle);
#ifdef USE_TRACE
    printState(state);
#endif
    ret = (state == SND_PCM_STATE_PREPARED)
        || (state == SND_PCM_STATE_RUNNING)
        || (state == SND_PCM_STATE_XRUN)
        || (state == SND_PCM_STATE_SUSPENDED);
    if (ret) {
        info->isRunning = 1;
        // source line should keep isFlushed value until Write() is called;
        // for target data line reset it right now.
        if (!isSource) {
            info->isFlushed = 0;
        }
    }
    TRACE1("< DAUDIO_Start %s\n", ret?"success":"error");
    return ret?TRUE:FALSE;
}

int DAUDIO_Stop(void* id, int isSource) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;
    int ret;

    TRACE0("> DAUDIO_Stop\n");
    // set to blocking mode
    snd_pcm_nonblock(info->handle, 0);
    setStartThreshold(info, FALSE /* don't use threshold */); // device will not start after buffer xrun
    ret = snd_pcm_pause(info->handle, 1);
    // set to non-blocking mode
    snd_pcm_nonblock(info->handle, 1);
    if (ret != 0) {
        ERROR1("ERROR in snd_pcm_pause: %s\n", snd_strerror(ret));
        return FALSE;
    }
    info->isRunning = 0;
    TRACE0("< DAUDIO_Stop success\n");
    return TRUE;
}

void DAUDIO_Close(void* id, int isSource) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;

    TRACE0("DAUDIO_Close\n");
    if (info != NULL) {
        if (info->handle != NULL) {
            snd_pcm_close(info->handle);
        }
        if (info->hwParams) {
            snd_pcm_hw_params_free(info->hwParams);
        }
        if (info->swParams) {
            snd_pcm_sw_params_free(info->swParams);
        }
#ifdef GET_POSITION_METHOD2
        if (info->positionStatus) {
            snd_pcm_status_free(info->positionStatus);
        }
#endif
        free(info);
    }
}

/*
 * Underrun and suspend recovery
 * returns
 * 0:  exit native and return 0
 * 1:  try again to write/read
 * -1: error - exit native with return value -1
 */
int xrun_recovery(AlsaPcmInfo* info, int err) {
    int ret;

    if (err == -EPIPE) {    /* underrun / overflow */
        TRACE0("xrun_recovery: underrun/overflow.\n");
        ret = snd_pcm_prepare(info->handle);
        if (ret < 0) {
            ERROR1("Can't recover from underrun/overflow, prepare failed: %s\n", snd_strerror(ret));
            return -1;
        }
        return 1;
    } else if (err == -ESTRPIPE) {
        TRACE0("xrun_recovery: suspended.\n");
        ret = snd_pcm_resume(info->handle);
        if (ret < 0) {
            if (ret == -EAGAIN) {
                return 0; /* wait until the suspend flag is released */
            }
            return -1;
        }
        ret = snd_pcm_prepare(info->handle);
        if (ret < 0) {
            ERROR1("Can't recover from underrun/overflow, prepare failed: %s\n", snd_strerror(ret));
            return -1;
        }
        return 1;
    } else if (err == -EAGAIN) {
        TRACE0("xrun_recovery: EAGAIN try again flag.\n");
        return 0;
    }

    TRACE2("xrun_recovery: unexpected error %d: %s\n", err, snd_strerror(err));
    return -1;
}

// returns -1 on error
int DAUDIO_Write(void* id, char* data, int byteSize) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;
    int ret, count;
    snd_pcm_sframes_t frameSize, writtenFrames;

    TRACE1("> DAUDIO_Write %d bytes\n", byteSize);

    /* sanity */
    if (byteSize <= 0 || info->frameSize <= 0) {
        ERROR2(" DAUDIO_Write: byteSize=%d, frameSize=%d!\n",
               (int) byteSize, (int) info->frameSize);
        TRACE0("< DAUDIO_Write returning -1\n");
        return -1;
    }

    count = 2; // maximum number of trials to recover from underrun
    //frameSize = snd_pcm_bytes_to_frames(info->handle, byteSize);
    frameSize = (snd_pcm_sframes_t) (byteSize / info->frameSize);
    do {
        writtenFrames = snd_pcm_writei(info->handle, (const void*) data, (snd_pcm_uframes_t) frameSize);

        if (writtenFrames < 0) {
            ret = xrun_recovery(info, (int) writtenFrames);
            if (ret <= 0) {
                TRACE1("DAUDIO_Write: xrun recovery returned %d -> return.\n", ret);
                return ret;
            }
            if (count-- <= 0) {
                ERROR0("DAUDIO_Write: too many attempts to recover from xrun/suspend\n");
                return -1;
            }
        } else {
            break;
        }
    } while (TRUE);
    //ret =  snd_pcm_frames_to_bytes(info->handle, writtenFrames);

    if (writtenFrames > 0) {
        // reset "flushed" flag
        info->isFlushed = 0;
    }

    ret =  (int) (writtenFrames * info->frameSize);
    TRACE1("< DAUDIO_Write: returning %d bytes.\n", ret);
    return ret;
}

// returns -1 on error
int DAUDIO_Read(void* id, char* data, int byteSize) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;
    int ret, count;
    snd_pcm_sframes_t frameSize, readFrames;

    TRACE1("> DAUDIO_Read %d bytes\n", byteSize);
    /*TRACE3("  info=%p, data=%p, byteSize=%d\n",
      (void*) info, (void*) data, (int) byteSize);
      TRACE2("  info->frameSize=%d, info->handle=%p\n",
      (int) info->frameSize, (void*) info->handle);
    */
    /* sanity */
    if (byteSize <= 0 || info->frameSize <= 0) {
        ERROR2(" DAUDIO_Read: byteSize=%d, frameSize=%d!\n",
               (int) byteSize, (int) info->frameSize);
        TRACE0("< DAUDIO_Read returning -1\n");
        return -1;
    }
    if (!info->isRunning && info->isFlushed) {
        // PCM has nothing to read
        return 0;
    }

    count = 2; // maximum number of trials to recover from error
    //frameSize = snd_pcm_bytes_to_frames(info->handle, byteSize);
    frameSize = (snd_pcm_sframes_t) (byteSize / info->frameSize);
    do {
        readFrames = snd_pcm_readi(info->handle, (void*) data, (snd_pcm_uframes_t) frameSize);
        if (readFrames < 0) {
            ret = xrun_recovery(info, (int) readFrames);
            if (ret <= 0) {
                TRACE1("DAUDIO_Read: xrun recovery returned %d -> return.\n", ret);
                return ret;
            }
            if (count-- <= 0) {
                ERROR0("DAUDIO_Read: too many attempts to recover from xrun/suspend\n");
                return -1;
            }
        } else {
            break;
        }
    } while (TRUE);
    //ret =  snd_pcm_frames_to_bytes(info->handle, readFrames);
    ret =  (int) (readFrames * info->frameSize);
    TRACE1("< DAUDIO_Read: returning %d bytes.\n", ret);
    return ret;
}


int DAUDIO_GetBufferSize(void* id, int isSource) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;

    return info->bufferSizeInBytes;
}

int DAUDIO_StillDraining(void* id, int isSource) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;
    snd_pcm_state_t state;

    state = snd_pcm_state(info->handle);
    //printState(state);
    //TRACE1("Still draining: %s\n", (state != SND_PCM_STATE_XRUN)?"TRUE":"FALSE");
    return (state == SND_PCM_STATE_RUNNING)?TRUE:FALSE;
}


int DAUDIO_Flush(void* id, int isSource) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;
    int ret;

    TRACE0("DAUDIO_Flush\n");

    if (info->isFlushed) {
        // nothing to drop
        return 1;
    }

    ret = snd_pcm_drop(info->handle);
    if (ret != 0) {
        ERROR1("ERROR in snd_pcm_drop: %s\n", snd_strerror(ret));
        return FALSE;
    }

    info->isFlushed = 1;
    if (info->isRunning) {
        ret = DAUDIO_Start(id, isSource);
    }
    return ret;
}

int DAUDIO_GetAvailable(void* id, int isSource) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;
    snd_pcm_sframes_t availableInFrames;
    snd_pcm_state_t state;
    int ret;

    state = snd_pcm_state(info->handle);
    if (info->isFlushed || state == SND_PCM_STATE_XRUN) {
        // if in xrun state then we have the entire buffer available,
        // not 0 as alsa reports
        ret = info->bufferSizeInBytes;
    } else {
        availableInFrames = snd_pcm_avail_update(info->handle);
        if (availableInFrames < 0) {
            ret = 0;
        } else {
            //ret = snd_pcm_frames_to_bytes(info->handle, availableInFrames);
            ret = (int) (availableInFrames * info->frameSize);
        }
    }
    TRACE1("DAUDIO_GetAvailable returns %d bytes\n", ret);
    return ret;
}

INT64 estimatePositionFromAvail(AlsaPcmInfo* info, int isSource, INT64 javaBytePos, int availInBytes) {
    // estimate the current position with the buffer size and
    // the available bytes to read or write in the buffer.
    // not an elegant solution - bytePos will stop on xruns,
    // and in race conditions it may jump backwards
    // Advantage is that it is indeed based on the samples that go through
    // the system (rather than time-based methods)
    if (isSource) {
        // javaBytePos is the position that is reached when the current
        // buffer is played completely
        return (INT64) (javaBytePos - info->bufferSizeInBytes + availInBytes);
    } else {
        // javaBytePos is the position that was when the current buffer was empty
        return (INT64) (javaBytePos + availInBytes);
    }
}

INT64 DAUDIO_GetBytePosition(void* id, int isSource, INT64 javaBytePos) {
    AlsaPcmInfo* info = (AlsaPcmInfo*) id;
    int ret;
    INT64 result = javaBytePos;
    snd_pcm_state_t state;
    state = snd_pcm_state(info->handle);

    if (!info->isFlushed && state != SND_PCM_STATE_XRUN) {
#ifdef GET_POSITION_METHOD2
        snd_timestamp_t* ts;
        snd_pcm_uframes_t framesAvail;

        // note: slight race condition if this is called simultaneously from 2 threads
        ret = snd_pcm_status(info->handle, info->positionStatus);
        if (ret != 0) {
            ERROR1("ERROR in snd_pcm_status: %s\n", snd_strerror(ret));
            result = javaBytePos;
        } else {
            // calculate from time value, or from available bytes
            framesAvail = snd_pcm_status_get_avail(info->positionStatus);
            result = estimatePositionFromAvail(info, isSource, javaBytePos, framesAvail * info->frameSize);
        }
#endif
#ifdef GET_POSITION_METHOD3
        snd_pcm_uframes_t framesAvail;
        ret = snd_pcm_avail(info->handle, &framesAvail);
        if (ret != 0) {
            ERROR1("ERROR in snd_pcm_avail: %s\n", snd_strerror(ret));
            result = javaBytePos;
        } else {
            result = estimatePositionFromAvail(info, isSource, javaBytePos, framesAvail * info->frameSize);
        }
#endif
#ifdef GET_POSITION_METHOD1
        result = estimatePositionFromAvail(info, isSource, javaBytePos, DAUDIO_GetAvailable(id, isSource));
#endif
    }
    //printf("getbyteposition: javaBytePos=%d , return=%d\n", (int) javaBytePos, (int) result);
    return result;
}


void DAUDIO_SetBytePosition(void* id, int isSource, INT64 javaBytePos) {
    /* save to ignore, since GetBytePosition
     * takes the javaBytePos param into account
     */
}

int DAUDIO_RequiresServicing(void* id, int isSource) {
    // never need servicing on Linux
    return FALSE;
}

void DAUDIO_Service(void* id, int isSource) {
    // never need servicing on Linux
}


#endif // USE_DAUDIO