JDK14/Java14源码在线阅读
/*
* 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);
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