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-rw-r--r--libao2/ao_alsa.c48
-rw-r--r--libao2/ao_alsa5.c16
-rw-r--r--libao2/ao_coreaudio.c28
-rw-r--r--libao2/ao_dsound.c74
-rw-r--r--libao2/ao_dxr2.c8
-rw-r--r--libao2/ao_esd.c24
-rw-r--r--libao2/ao_ivtv.c10
-rw-r--r--libao2/ao_jack.c2
-rw-r--r--libao2/ao_mpegpes.c4
-rw-r--r--libao2/ao_nas.c12
-rw-r--r--libao2/ao_null.c6
-rw-r--r--libao2/ao_openal.c2
-rw-r--r--libao2/ao_oss.c18
-rw-r--r--libao2/ao_pcm.c16
-rw-r--r--libao2/ao_sdl.c22
-rw-r--r--libao2/ao_sgi.c62
-rw-r--r--libao2/ao_sun.c24
-rw-r--r--libao2/ao_v4l2.c10
-rw-r--r--libao2/ao_win32.c2
-rw-r--r--libao2/audio_out.h4
20 files changed, 196 insertions, 196 deletions
diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c
index 032d1d132a..5e848446f2 100644
--- a/libao2/ao_alsa.c
+++ b/libao2/ao_alsa.c
@@ -57,7 +57,7 @@
#include "audio_out_internal.h"
#include "libaf/af_format.h"
-static const ao_info_t info =
+static const ao_info_t info =
{
"ALSA-0.9.x-1.x audio output",
"alsa",
@@ -73,7 +73,7 @@ static snd_pcm_hw_params_t *alsa_hwparams;
static snd_pcm_sw_params_t *alsa_swparams;
/* 16 sets buffersize to 16 * chunksize is as default 1024
- * which seems to be good avarge for most situations
+ * which seems to be good avarge for most situations
* so buffersize is 16384 frames by default */
static int alsa_fragcount = 16;
static snd_pcm_uframes_t chunk_size = 1024;
@@ -156,7 +156,7 @@ static int control(int cmd, void *arg)
//allocate simple id
snd_mixer_selem_id_alloca(&sid);
-
+
//sets simple-mixer index and name
snd_mixer_selem_id_set_index(sid, mix_index);
snd_mixer_selem_id_set_name(sid, mix_name);
@@ -172,7 +172,7 @@ static int control(int cmd, void *arg)
}
if ((err = snd_mixer_attach(handle, card)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
+ mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
card, snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
@@ -207,7 +207,7 @@ static int control(int cmd, void *arg)
//setting channels
if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
+ mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel,
snd_strerror(err));
return CONTROL_ERROR;
}
@@ -216,11 +216,11 @@ static int control(int cmd, void *arg)
set_vol = vol->right / f_multi + pmin + 0.5;
if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel,
+ mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel,
snd_strerror(err));
return CONTROL_ERROR;
}
- mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
+ mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
set_vol, pmin, pmax, f_multi);
if (snd_mixer_selem_has_playback_switch(elem)) {
@@ -245,7 +245,7 @@ static int control(int cmd, void *arg)
snd_mixer_close(handle);
return CONTROL_OK;
}
-
+
} //end switch
return CONTROL_UNKNOWN;
}
@@ -349,7 +349,7 @@ static int init(int rate_hz, int channels, int format, int flags)
#endif
snd_lib_error_set_handler(alsa_error_handler);
-
+
ao_data.samplerate = rate_hz;
ao_data.format = format;
ao_data.channels = channels;
@@ -409,7 +409,7 @@ static int init(int rate_hz, int channels, int format, int flags)
alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
break;
}
-
+
//subdevice parsing
// set defaults
block = 1;
@@ -537,11 +537,11 @@ static int init(int rate_hz, int channels, int format, int flags)
snd_strerror(err));
return 0;
}
-
+
err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
+ mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
snd_strerror(err));
return 0;
}
@@ -585,8 +585,8 @@ static int init(int rate_hz, int channels, int format, int flags)
}
#endif
- if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
- &ao_data.samplerate, NULL)) < 0)
+ if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
+ &ao_data.samplerate, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
snd_strerror(err));
@@ -602,7 +602,7 @@ static int init(int rate_hz, int channels, int format, int flags)
int alsa_buffer_time = 500000; /* original 60 */
int alsa_period_time;
alsa_period_time = alsa_buffer_time/4;
- if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
+ if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
&alsa_buffer_time, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
@@ -611,7 +611,7 @@ static int init(int rate_hz, int channels, int format, int flags)
} else
alsa_buffer_time = err;
- if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
+ if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
&alsa_period_time, NULL)) < 0)
/* original: alsa_buffer_time/ao_data.bps */
{
@@ -621,13 +621,13 @@ static int init(int rate_hz, int channels, int format, int flags)
}
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_BufferTimePeriodTime,
alsa_buffer_time, err);
- }
+ }
#endif//end SET_BUFFERTIME
#ifdef SET_CHUNKSIZE
{
//set chunksize
- if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
+ if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
&chunk_size, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodSize,
@@ -639,7 +639,7 @@ static int init(int rate_hz, int channels, int format, int flags)
}
if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
&alsa_fragcount, NULL)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
+ mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
snd_strerror(err));
return 0;
}
@@ -870,15 +870,15 @@ static int get_space(void)
{
snd_pcm_status_t *status;
int ret;
-
+
snd_pcm_status_alloca(&status);
-
+
if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret));
return 0;
}
-
+
ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
if (ret > ao_data.buffersize) // Buffer underrun?
ret = ao_data.buffersize;
@@ -890,10 +890,10 @@ static float get_delay(void)
{
if (alsa_handler) {
snd_pcm_sframes_t delay;
-
+
if (snd_pcm_delay(alsa_handler, &delay) < 0)
return 0;
-
+
if (delay < 0) {
/* underrun - move the application pointer forward to catch up */
#if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
diff --git a/libao2/ao_alsa5.c b/libao2/ao_alsa5.c
index 007a5f1b4f..feb2919f1d 100644
--- a/libao2/ao_alsa5.c
+++ b/libao2/ao_alsa5.c
@@ -32,7 +32,7 @@
#include "mp_msg.h"
#include "help_mp.h"
-static const ao_info_t info =
+static const ao_info_t info =
{
"ALSA-0.5.x audio output",
"alsa5",
@@ -117,7 +117,7 @@ static int init(int rate_hz, int channels, int format, int flags)
alsa_format.format = SND_PCM_SFMT_MPEG;
break;
}
-
+
switch(alsa_format.format)
{
case SND_PCM_SFMT_S16_LE:
@@ -230,7 +230,7 @@ static int init(int rate_hz, int channels, int format, int flags)
setup.format = alsa_format;
setup.buf.stream.queue_size = ao_data.buffersize;
setup.msbits_per_sample = ao_data.bps;
-
+
if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
{
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_CantSetChan, snd_strerror(err));
@@ -333,10 +333,10 @@ static void audio_resume(void)
static int play(void* data, int len, int flags)
{
int got_len;
-
+
if (!len)
return 0;
-
+
if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
{
if (got_len == -EPIPE) /* underrun? */
@@ -365,7 +365,7 @@ static int play(void* data, int len, int flags)
static int get_space(void)
{
snd_pcm_channel_status_t ch_stat;
-
+
ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
@@ -378,9 +378,9 @@ static int get_space(void)
static float get_delay(void)
{
snd_pcm_channel_status_t ch_stat;
-
+
ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
-
+
if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
return (float)ao_data.buffersize/(float)ao_data.bps; /* error occurred */
else
diff --git a/libao2/ao_coreaudio.c b/libao2/ao_coreaudio.c
index 3ef6d3367f..76cb9174be 100644
--- a/libao2/ao_coreaudio.c
+++ b/libao2/ao_coreaudio.c
@@ -174,7 +174,7 @@ Float32 vol;
ao->b_muted = 0;
return CONTROL_TRUE;
}
-
+
vol=(control_vol->left+control_vol->right)*4.0/200.0;
err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0);
if(err==0) {
@@ -189,7 +189,7 @@ Float32 vol;
default:
return CONTROL_FALSE;
}
-
+
}
@@ -237,8 +237,8 @@ static OSStatus DeviceListener( AudioDeviceID inDevice,
static int init(int rate,int channels,int format,int flags)
{
AudioStreamBasicDescription inDesc;
-ComponentDescription desc;
-Component comp;
+ComponentDescription desc;
+Component comp;
AURenderCallbackStruct renderCallback;
OSStatus err;
UInt32 size, maxFrames, i_param_size;
@@ -396,20 +396,20 @@ int b_alive;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
-
+
comp = FindNextComponent(NULL, &desc); //Finds an component that meets the desc spec's
if (comp == NULL) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
goto err_out;
}
-
+
err = OpenAComponent(comp, &(ao->theOutputUnit)); //gains access to the services provided by the component
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
goto err_out;
}
- // Initialize AudioUnit
+ // Initialize AudioUnit
err = AudioUnitInitialize(ao->theOutputUnit);
if (err) {
ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
@@ -426,7 +426,7 @@ int b_alive;
size = sizeof(UInt32);
err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
-
+
if (err)
{
ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
@@ -434,7 +434,7 @@ int b_alive;
}
ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
-
+
ao_data.samplerate = inDesc.mSampleRate;
ao_data.channels = inDesc.mChannelsPerFrame;
ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
@@ -444,7 +444,7 @@ int b_alive;
ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
ao->buffer_len = ao->num_chunks * ao->chunk_size;
ao->buffer = av_fifo_alloc(ao->buffer_len);
-
+
ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
renderCallback.inputProc = theRenderProc;
@@ -456,7 +456,7 @@ int b_alive;
}
reset();
-
+
return CONTROL_OK;
err_out2:
@@ -467,7 +467,7 @@ err_out:
av_fifo_free(ao->buffer);
free(ao);
ao = NULL;
- return CONTROL_FALSE;
+ return CONTROL_FALSE;
}
/*****************************************************************************
@@ -734,7 +734,7 @@ err_out:
av_fifo_free(ao->buffer);
free(ao);
ao = NULL;
- return CONTROL_FALSE;
+ return CONTROL_FALSE;
}
/*****************************************************************************
@@ -943,7 +943,7 @@ static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
static int play(void* output_samples,int num_bytes,int flags)
-{
+{
int wrote, b_digital;
// Check whether we need to reset the digital output stream.
diff --git a/libao2/ao_dsound.c b/libao2/ao_dsound.c
index 729084b145..4197c30e27 100644
--- a/libao2/ao_dsound.c
+++ b/libao2/ao_dsound.c
@@ -113,17 +113,17 @@ static const int channel_mask[] = {
};
static HINSTANCE hdsound_dll = NULL; ///handle to the dll
-static LPDIRECTSOUND hds = NULL; ///direct sound object
+static LPDIRECTSOUND hds = NULL; ///direct sound object
static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer
static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///secondary direct sound buffer (stream buffer)
-static int buffer_size = 0; ///size in bytes of the direct sound buffer
+static int buffer_size = 0; ///size in bytes of the direct sound buffer
static int write_offset = 0; ///offset of the write cursor in the direct sound buffer
static int min_free_space = 0; ///if the free space is below this value get_space() will return 0
///there will always be at least this amout of free space to prevent
///get_space() from returning wrong values when buffer is 100% full.
///will be replaced with nBlockAlign in init()
static int device_num = 0; ///wanted device number
-static GUID device; ///guid of the device
+static GUID device; ///guid of the device
/***************************************************************************************/
@@ -221,17 +221,17 @@ static int InitDirectSound(void)
// initialize directsound
HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
- HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
+ HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
int device_index=0;
opt_t subopts[] = {
{"device", OPT_ARG_INT, &device_num,NULL},
{NULL}
- };
+ };
if (subopt_parse(ao_subdevice, subopts) != 0) {
print_help();
return 0;
}
-
+
hdsound_dll = LoadLibrary("DSOUND.DLL");
if (hdsound_dll == NULL) {
mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n");
@@ -245,7 +245,7 @@ static int InitDirectSound(void)
FreeLibrary(hdsound_dll);
return 0;
}
-
+
// Enumerate all directsound devices
mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n");
OurDirectSoundEnumerate(DirectSoundEnum,&device_index);
@@ -310,22 +310,22 @@ static void DestroyBuffer(void)
static int write_buffer(unsigned char *data, int len)
{
HRESULT res;
- LPVOID lpvPtr1;
- DWORD dwBytes1;
- LPVOID lpvPtr2;
- DWORD dwBytes2;
-
+ LPVOID lpvPtr1;
+ DWORD dwBytes1;
+ LPVOID lpvPtr2;
+ DWORD dwBytes2;
+
// Lock the buffer
- res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
- // If the buffer was lost, restore and retry lock.
- if (DSERR_BUFFERLOST == res)
- {
+ res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
+ // If the buffer was lost, restore and retry lock.
+ if (DSERR_BUFFERLOST == res)
+ {
IDirectSoundBuffer_Restore(hdsbuf);
res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
}
-
-
- if (SUCCEEDED(res))
+
+
+ if (SUCCEEDED(res))
{
if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) {
// reorder channels while writing to pointers.
@@ -354,27 +354,27 @@ static int write_buffer(unsigned char *data, int len)
write_offset+=dwBytes1+dwBytes2;
if(write_offset>=buffer_size)write_offset=dwBytes2;
} else {
- // Write to pointers without reordering.
+ // Write to pointers without reordering.
fast_memcpy(lpvPtr1,data,dwBytes1);
if (NULL != lpvPtr2 )fast_memcpy(lpvPtr2,data+dwBytes1,dwBytes2);
write_offset+=dwBytes1+dwBytes2;
if(write_offset>=buffer_size)write_offset=dwBytes2;
}
-
- // Release the data back to DirectSound.
+
+ // Release the data back to DirectSound.
res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2);
- if (SUCCEEDED(res))
- {
- // Success.
+ if (SUCCEEDED(res))
+ {
+ // Success.
DWORD status;
IDirectSoundBuffer_GetStatus(hdsbuf, &status);
if (!(status & DSBSTATUS_PLAYING)){
res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
}
- return dwBytes1+dwBytes2;
- }
- }
- // Lock, Unlock, or Restore failed.
+ return dwBytes1+dwBytes2;
+ }
+ }
+ // Lock, Unlock, or Restore failed.
return 0;
}
@@ -408,7 +408,7 @@ static int control(int cmd, void *arg)
return -1;
}
-/**
+/**
\brief setup sound device
\param rate samplerate
\param channels number of channels
@@ -436,7 +436,7 @@ static int init(int rate, int channels, int format, int flags)
default:
mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
format=AF_FORMAT_S16_LE;
- }
+ }
//fill global ao_data
ao_data.channels = channels;
ao_data.samplerate = rate;
@@ -493,7 +493,7 @@ static int init(int rate, int channels, int format, int flags)
ao_data.outburst = wformat.Format.nBlockAlign * 512;
// create primary buffer and set its format
-
+
res = IDirectSound_CreateSoundBuffer( hds, &dsbpridesc, &hdspribuf, NULL );
if ( res != DS_OK ) {
UninitDirectSound();
@@ -553,7 +553,7 @@ static void audio_resume(void)
IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
}
-/**
+/**
\brief close audio device
\param immed stop playback immediately
*/
@@ -579,7 +579,7 @@ static int get_space(void)
int space;
DWORD play_offset;
IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
- space=buffer_size-(write_offset-play_offset);
+ space=buffer_size-(write_offset-play_offset);
// | | <-- const --> | | |
// buffer start play_cursor write_cursor write_offset buffer end
// play_cursor is the actual postion of the play cursor
@@ -601,10 +601,10 @@ static int play(void* data, int len, int flags)
{
DWORD play_offset;
int space;
-
+
// make sure we have enough space to write data
IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
- space=buffer_size-(write_offset-play_offset);
+ space=buffer_size-(write_offset-play_offset);
if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
if(space < len) len = space;
@@ -622,7 +622,7 @@ static float get_delay(void)
DWORD play_offset;
int space;
IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
- space=play_offset-write_offset;
+ space=play_offset-write_offset;
if(space <= 0)space += buffer_size;
return (float)(buffer_size - space) / (float)ao_data.bps;
}
diff --git a/libao2/ao_dxr2.c b/libao2/ao_dxr2.c
index 7f9594fb4d..136efd1a9e 100644
--- a/libao2/ao_dxr2.c
+++ b/libao2/ao_dxr2.c
@@ -67,7 +67,7 @@ static int control(int cmd,void *arg){
ao_control_vol_t* vol = (ao_control_vol_t*)arg;
// We need this trick because the volume stepping is often too small
diff = ((vol->left+vol->right) / 2 - (volume*19.0/100.0)) * 19.0 / 100.0;
- v.arg = volume + (diff > 0 ? ceil(diff) : floor(diff));
+ v.arg = volume + (diff > 0 ? ceil(diff) : floor(diff));
if(v.arg > 19) v.arg = 19;
if(v.arg < 0) v.arg = 0;
if(v.arg != volume) {
@@ -95,7 +95,7 @@ static int init(int rate,int channels,int format,int flags){
return 0;
last_freq_id = -1;
-
+
ao_data.outburst=2048;
ao_data.samplerate=rate;
ao_data.channels=channels;
@@ -178,11 +178,11 @@ static int get_space(void){
static void dxr2_send_lpcm_packet(unsigned char* data,int len,int id,unsigned int timestamp,int freq_id)
{
int write_dxr2(const unsigned char *data, int len);
-
+
if(dxr2_fd < 0) {
mp_msg(MSGT_AO,MSGL_ERR,"DXR2 fd is not valid\n");
return;
- }
+ }
if(last_freq_id != freq_id) {
ioctl(dxr2_fd, DXR2_IOC_SET_AUDIO_SAMPLE_FREQUENCY, &freq_id);
diff --git a/libao2/ao_esd.c b/libao2/ao_esd.c
index d54f843c65..593d5a934a 100644
--- a/libao2/ao_esd.c
+++ b/libao2/ao_esd.c
@@ -122,7 +122,7 @@ static int control(int cmd, void *arg)
vol_cache_time = now;
}
esd_free_all_info(esd_i);
-
+
return CONTROL_OK;
case AOCONTROL_SET_VOLUME:
@@ -236,19 +236,19 @@ static int init(int rate_hz, int channels, int format, int flags)
#ifdef CONFIG_ESD_LATENCY
esd_latency = esd_get_latency(esd_fd);
#else
- esd_latency = ((channels == 1 ? 2 : 1) * ESD_DEFAULT_RATE *
+ esd_latency = ((channels == 1 ? 2 : 1) * ESD_DEFAULT_RATE *
(ESD_BUF_SIZE + 64 * (4.0f / bytes_per_sample))
- ) / rate_hz;
- esd_latency += ESD_BUF_SIZE * 2;
+ ) / rate_hz;
+ esd_latency += ESD_BUF_SIZE * 2;
#endif
if(esd_latency > 0) {
lag_serv = (esd_latency * 4.0f) / (bytes_per_sample * rate_hz);
lag_seconds = lag_net + lag_serv;
audio_delay += lag_seconds;
- mp_msg(MSGT_AO, MSGL_INFO,MSGTR_AO_ESD_LatencyInfo,
+ mp_msg(MSGT_AO, MSGL_INFO,MSGTR_AO_ESD_LatencyInfo,
lag_serv, lag_net, lag_seconds);
}
-
+
esd_play_fd = esd_play_stream_fallback(esd_fmt, rate_hz,
server, ESD_CLIENT_NAME);
if (esd_play_fd < 0) {
@@ -333,7 +333,7 @@ static int play(void* data, int len, int flags)
*/
n = write(esd_play_fd, (char*)data + offs, ESD_BUF_SIZE);
if ( n < 0 ) {
- if ( errno != EAGAIN )
+ if ( errno != EAGAIN )
dprintf("esd play: write failed: %s\n", strerror(errno));
break;
} else if ( n != ESD_BUF_SIZE ) {
@@ -343,13 +343,13 @@ static int play(void* data, int len, int flags)
nwritten += n;
}
#endif
-
+
if (nwritten > 0) {
if (!esd_play_start.tv_sec)
gettimeofday(&esd_play_start, NULL);
nsamples = nwritten / esd_bytes_per_sample;
esd_samples_written += nsamples;
-
+
dprintf("esd play: %d %lu\n", nsamples, esd_samples_written);
} else {
dprintf("esd play: blocked / %lu\n", esd_samples_written);
@@ -395,7 +395,7 @@ static void reset(void)
{
#ifdef __svr4__
/* throw away data buffered in the esd connection */
- if (ioctl(esd_play_fd, I_FLUSH, FLUSHW))
+ if (ioctl(esd_play_fd, I_FLUSH, FLUSHW))
perror("I_FLUSH");
#endif
}
@@ -411,7 +411,7 @@ static int get_space(void)
float current_delay;
int space;
- /*
+ /*
* Don't buffer too much data in the esd daemon.
*
* If we send too much, esd will block in write()s to the sound
@@ -461,7 +461,7 @@ static float get_delay(void)
gettimeofday(&now, NULL);
play_time = now.tv_sec - esd_play_start.tv_sec;
play_time += (now.tv_usec - esd_play_start.tv_usec) / 1000000.;
-
+
/* dprintf("esd delay: %f %f\n", play_time, buffered_samples_time); */
if (play_time > buffered_samples_time) {
diff --git a/libao2/ao_ivtv.c b/libao2/ao_ivtv.c
index c2b367a414..b09e5308bb 100644
--- a/libao2/ao_ivtv.c
+++ b/libao2/ao_ivtv.c
@@ -41,7 +41,7 @@
static int freq = 0;
-static const ao_info_t info =
+static const ao_info_t info =
{
"IVTV MPEG Audio Decoder output",
"ivtv",
@@ -73,7 +73,7 @@ init (int rate, int channels, int format, int flags)
"AO: [ivtv] can only handle MPEG audio streams.\n");
return 0;
}
-
+
ao_data.outburst = 2048;
ao_data.samplerate = rate;
ao_data.channels = channels;
@@ -132,11 +132,11 @@ get_space (void)
x = (float) (vo_pts - ao_data.pts) / 90000.0;
if (x <= 0)
return 0;
-
+
y = freq * 4 * x;
y /= ao_data.outburst;
y *= ao_data.outburst;
-
+
if (y > 32000)
y = 32000;
@@ -148,7 +148,7 @@ static int
play (void *data, int len, int flags)
{
int ivtv_write (unsigned char *data, int len);
-
+
if (ao_data.format != AF_FORMAT_MPEG2)
return 0;
diff --git a/libao2/ao_jack.c b/libao2/ao_jack.c
index 8ee5550602..e92786c28d 100644
--- a/libao2/ao_jack.c
+++ b/libao2/ao_jack.c
@@ -40,7 +40,7 @@
#include <jack/jack.h>
-static const ao_info_t info =
+static const ao_info_t info =
{
"JACK audio output",
"jack",
diff --git a/libao2/ao_mpegpes.c b/libao2/ao_mpegpes.c
index 3746d24b4e..64cb05fa03 100644
--- a/libao2/ao_mpegpes.c
+++ b/libao2/ao_mpegpes.c
@@ -63,7 +63,7 @@ int vo_mpegpes_fd2 = -1;
#include <errno.h>
-static const ao_info_t info =
+static const ao_info_t info =
{
#ifdef CONFIG_DVB
"DVB audio output",
@@ -194,7 +194,7 @@ static int preinit(const char *arg)
#ifdef CONFIG_DVB
if(!ao_file)
return init_device(card);
-#else
+#else
if(!ao_file)
return vo_mpegpes_fd; //video fd
#endif
diff --git a/libao2/ao_nas.c b/libao2/ao_nas.c
index e5e675be3f..fb49c5e60e 100644
--- a/libao2/ao_nas.c
+++ b/libao2/ao_nas.c
@@ -119,7 +119,7 @@ static const char* nas_state(unsigned int state) {
return nas_states[state];
}
-static const ao_info_t info =
+static const ao_info_t info =
{
"NAS audio output",
"nas",
@@ -201,7 +201,7 @@ static int nas_readBuffer(struct ao_nas_data *nas_data, unsigned int num)
* Now write the new buffer to the network.
*/
AuWriteElement(nas_data->aud, nas_data->flow, 0, num, nas_data->server_buffer, AuFalse, &as);
- if (as != AuSuccess)
+ if (as != AuSuccess)
nas_print_error(nas_data->aud, "nas_readBuffer(): AuWriteElement", as);
return num;
@@ -229,7 +229,7 @@ static int nas_empty_event_queue(struct ao_nas_data *nas_data)
{
AuEvent ev;
int result = 0;
-
+
while (AuScanForTypedEvent(nas_data->aud, AuEventsQueuedAfterFlush,
AuTrue, AuEventTypeElementNotify, &ev)) {
AuDispatchEvent(nas_data->aud, &ev);
@@ -462,7 +462,7 @@ static int init(int rate,int channels,int format,int flags)
mp_msg(MSGT_AO, MSGL_V, "ao_nas: init(): Using audioserver %s\n", server);
nas_data->aud = AuOpenServer(server, 0, NULL, 0, NULL, NULL);
- if (!nas_data->aud) {
+ if (!nas_data->aud) {
mp_msg(MSGT_AO, MSGL_ERR, "ao_nas: init(): Can't open nas audio server -> nosound\n");
return 0;
}
@@ -571,7 +571,7 @@ static void audio_resume(void)
static int get_space(void)
{
int result;
-
+
mp_msg(MSGT_AO, MSGL_DBG3, "ao_nas: get_space()\n");
pthread_mutex_lock(&nas_data->buffer_mutex);
@@ -631,7 +631,7 @@ static int play(void* data,int len,int flags)
static float get_delay(void)
{
float result;
-
+
mp_msg(MSGT_AO, MSGL_DBG3, "ao_nas: get_delay()\n");
pthread_mutex_lock(&nas_data->buffer_mutex);
diff --git a/libao2/ao_null.c b/libao2/ao_null.c
index 20b715fb96..17b3bc7e4e 100644
--- a/libao2/ao_null.c
+++ b/libao2/ao_null.c
@@ -27,7 +27,7 @@
#include "audio_out.h"
#include "audio_out_internal.h"
-static const ao_info_t info =
+static const ao_info_t info =
{
"Null audio output",
"null",
@@ -41,14 +41,14 @@ struct timeval last_tv;
int buffer;
static void drain(void){
-
+
struct timeval now_tv;
int temp, temp2;
gettimeofday(&now_tv, 0);
temp = now_tv.tv_sec - last_tv.tv_sec;
temp *= ao_data.bps;
-
+
temp2 = now_tv.tv_usec - last_tv.tv_usec;
temp2 /= 1000;
temp2 *= ao_data.bps;
diff --git a/libao2/ao_openal.c b/libao2/ao_openal.c
index c80c49b27a..829eb7ae58 100644
--- a/libao2/ao_openal.c
+++ b/libao2/ao_openal.c
@@ -42,7 +42,7 @@
#include "osdep/timer.h"
#include "subopt-helper.h"
-static const ao_info_t info =
+static const ao_info_t info =
{
"OpenAL audio output",
"openal",
diff --git a/libao2/ao_oss.c b/libao2/ao_oss.c
index e4688d1723..4666c400e9 100644
--- a/libao2/ao_oss.c
+++ b/libao2/ao_oss.c
@@ -48,7 +48,7 @@
#include "audio_out.h"
#include "audio_out_internal.h"
-static const ao_info_t info =
+static const ao_info_t info =
{
"OSS/ioctl audio output",
"oss",
@@ -199,7 +199,7 @@ static int control(int cmd,void *arg){
if(ao_data.format == AF_FORMAT_AC3)
return CONTROL_TRUE;
-
+
if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
{
ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
@@ -260,17 +260,17 @@ static int init(int rate,int channels,int format,int flags){
oss_mixer_device=mdev;
else
oss_mixer_device=PATH_DEV_MIXER;
-
+
if(mchan){
int fd, devs, i;
-
+
if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenMixer,
oss_mixer_device, strerror(errno));
}else{
ioctl(fd, SOUND_MIXE