/*

 * 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) {