--- a/src/java.desktop/unix/native/libjsound/PLATFORM_API_BsdOS_ALSA_PCM.c Thu Mar 22 14:10:30 2018 -0700
+++ /dev/null Thu Jan 01 00:00:00 1970 +0000
@@ -1,941 +0,0 @@
-/*
- * Copyright (c) 2002, 2012, 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_BsdOS_ALSA_PCMUtils.h"
-#include "PLATFORM_API_BsdOS_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 Bsd
- return FALSE;
-}
-
-void DAUDIO_Service(void* id, int isSource) {
- // never need servicing on Bsd
-}
-
-
-#endif // USE_DAUDIO