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-rw-r--r--libao2/ao_alsa.c138
-rw-r--r--libao2/ao_alsa5.c47
-rw-r--r--libao2/ao_arts.c13
-rw-r--r--libao2/ao_dxr2.c5
-rw-r--r--libao2/ao_esd.c7
-rw-r--r--libao2/ao_ivtv.c5
-rw-r--r--libao2/ao_jack.c1
-rw-r--r--libao2/ao_mpegpes.c7
-rw-r--r--libao2/ao_openal.c1
-rw-r--r--libao2/ao_oss.c56
-rw-r--r--libao2/ao_pcm.c7
-rw-r--r--libao2/ao_sdl.c13
-rw-r--r--libao2/ao_sgi.c29
-rw-r--r--libao2/ao_sun.c9
-rw-r--r--libao2/ao_v4l2.c3
-rw-r--r--libao2/audio_out.c16
-rw-r--r--libao2/audio_out.h3
17 files changed, 192 insertions, 168 deletions
diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c
index 57f0bd07cb..03afc5d729 100644
--- a/libao2/ao_alsa.c
+++ b/libao2/ao_alsa.c
@@ -39,7 +39,6 @@
#include "subopt-helper.h"
#include "mixer.h"
#include "mp_msg.h"
-#include "help_mp.h"
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
@@ -72,9 +71,13 @@ static snd_pcm_format_t alsa_format;
static snd_pcm_hw_params_t *alsa_hwparams;
static snd_pcm_sw_params_t *alsa_swparams;
+#define BUFFER_TIME 500000 // 0.5 s
+#define FRAGCOUNT 16
+
static size_t bytes_per_sample;
static int alsa_can_pause;
+static snd_pcm_sframes_t prepause_frames;
#define ALSA_DEVICE_SIZE 256
@@ -134,8 +137,8 @@ static int control(int cmd, void *arg)
mix_index = strtol(test_mix_index, &test_mix_index, 0);
if (*test_mix_index){
- mp_msg(MSGT_AO,MSGL_ERR,
- MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero);
+ mp_tmsg(MSGT_AO,MSGL_ERR,
+ "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
mix_index = 0 ;
}
}
@@ -155,32 +158,32 @@ static int control(int cmd, void *arg)
}
if ((err = snd_mixer_open(&handle, 0)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
return CONTROL_ERROR;
}
if ((err = snd_mixer_attach(handle, card)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
card, snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
err = snd_mixer_load(handle);
if (err < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
}
elem = snd_mixer_find_selem(handle, sid);
if (!elem) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToFindSimpleControl,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
snd_mixer_close(handle);
return CONTROL_ERROR;
@@ -195,7 +198,7 @@ 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
@@ -205,7 +208,7 @@ 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
snd_strerror(err));
snd_mixer_close(handle);
return CONTROL_ERROR;
@@ -253,8 +256,15 @@ static void parse_device (char *dest, const char *src, int len)
static void print_help (void)
{
- mp_msg (MSGT_AO, MSGL_FATAL,
- MSGTR_AO_ALSA_CommandlineHelp);
+ mp_tmsg (MSGT_AO, MSGL_FATAL,
+ "\n[AO_ALSA] -ao alsa commandline help:\n"\
+ "[AO_ALSA] Example: mplayer -ao alsa:device=hw=0.3\n"\
+ "[AO_ALSA] Sets first card fourth hardware device.\n\n"\
+ "[AO_ALSA] Options:\n"\
+ "[AO_ALSA] noblock\n"\
+ "[AO_ALSA] Opens device in non-blocking mode.\n"\
+ "[AO_ALSA] device=<device-name>\n"\
+ "[AO_ALSA] Sets device (change , to . and : to =)\n");
}
static int str_maxlen(void *strp) {
@@ -313,8 +323,6 @@ static int try_open_device(const char *device, int open_mode, int try_ac3)
*/
static int init(int rate_hz, int channels, int format, int flags)
{
- unsigned int alsa_buffer_time = 500000; /* 0.5 s */
- unsigned int alsa_fragcount = 16;
int err;
int block;
strarg_t device;
@@ -340,6 +348,8 @@ static int init(int rate_hz, int channels, int format, int flags)
mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
#endif
+ prepause_frames = 0;
+
snd_lib_error_set_handler(alsa_error_handler);
ao_data.samplerate = rate_hz;
@@ -457,7 +467,7 @@ static int init(int rate_hz, int channels, int format, int flags)
break;
default:
device.str = "default";
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
}
device.len = strlen(device.str);
if (subopt_parse(ao_subdevice, subopts) != 0) {
@@ -475,19 +485,19 @@ static int init(int rate_hz, int channels, int format, int flags)
if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
{
if (err != -EBUSY && !block) {
- mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
+ mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
return 0;
}
} else {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
return 0;
}
}
if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
}
@@ -498,7 +508,7 @@ static int init(int rate_hz, int channels, int format, int flags)
// setting hw-parameters
if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get initial parameters: %s\n",
snd_strerror(err));
return 0;
}
@@ -506,7 +516,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n",
snd_strerror(err));
return 0;
}
@@ -516,8 +526,8 @@ static int init(int rate_hz, int channels, int format, int flags)
if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
- mp_msg(MSGT_AO,MSGL_INFO,
- MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
+ mp_tmsg(MSGT_AO,MSGL_INFO,
+ "[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao_data.format))
ao_data.format = AF_FORMAT_AC3_LE;
@@ -528,7 +538,7 @@ static int init(int rate_hz, int channels, int format, int flags)
if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set format: %s\n",
snd_strerror(err));
return 0;
}
@@ -536,7 +546,7 @@ static int init(int rate_hz, int channels, int format, int flags)
if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
&ao_data.channels)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
snd_strerror(err));
return 0;
}
@@ -548,7 +558,7 @@ static int init(int rate_hz, int channels, int format, int flags)
if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
0)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to disable resampling: %s\n",
snd_strerror(err));
return 0;
}
@@ -557,7 +567,7 @@ static int init(int rate_hz, int channels, int format, int flags)
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,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
snd_strerror(err));
return 0;
}
@@ -567,16 +577,16 @@ static int init(int rate_hz, int channels, int format, int flags)
ao_data.bps = ao_data.samplerate * bytes_per_sample;
if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
- &alsa_buffer_time, NULL)) < 0)
+ &(unsigned int){BUFFER_TIME}, NULL)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
snd_strerror(err));
return 0;
}
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,
+ &(unsigned int){FRAGCOUNT}, NULL)) < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
snd_strerror(err));
return 0;
}
@@ -584,7 +594,7 @@ static int init(int rate_hz, int channels, int format, int flags)
/* finally install hardware parameters */
if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set hw-parameters: %s\n",
snd_strerror(err));
return 0;
}
@@ -594,7 +604,7 @@ static int init(int rate_hz, int channels, int format, int flags)
// gets buffersize for control
if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err));
return 0;
}
else {
@@ -603,7 +613,7 @@ static int init(int rate_hz, int channels, int format, int flags)
}
if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err));
return 0;
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
@@ -612,13 +622,13 @@ static int init(int rate_hz, int channels, int format, int flags)
/* setting software parameters */
if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get boundary: %s\n",
snd_strerror(err));
return 0;
}
@@ -627,26 +637,26 @@ static int init(int rate_hz, int channels, int format, int flags)
#endif
/* start playing when one period has been written */
if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set start threshold: %s\n",
snd_strerror(err));
return 0;
}
/* disable underrun reporting */
if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set stop threshold: %s\n",
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
/* play silence when there is an underrun */
if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set silence size: %s\n",
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
snd_strerror(err));
return 0;
}
@@ -674,7 +684,7 @@ static void uninit(int immed)
if ((err = snd_pcm_close(alsa_handler)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err));
return;
}
else {
@@ -683,7 +693,7 @@ static void uninit(int immed)
}
}
else {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n");
}
}
@@ -694,14 +704,18 @@ static void audio_pause(void)
if (alsa_can_pause) {
if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPauseError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm pause error: %s\n", snd_strerror(err));
return;
}
mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
} else {
+ if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
+ || prepause_frames < 0)
+ prepause_frames = 0;
+
if ((err = snd_pcm_drop(alsa_handler)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmDropError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err));
return;
}
}
@@ -712,22 +726,27 @@ static void audio_resume(void)
int err;
if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
- mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
+ mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
}
if (alsa_can_pause) {
if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmResumeError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm resume error: %s\n", snd_strerror(err));
return;
}
mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
} else {
if ((err = snd_pcm_prepare(alsa_handler)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
return;
}
+ if (prepause_frames) {
+ void *silence = calloc(prepause_frames, bytes_per_sample);
+ play(silence, prepause_frames * bytes_per_sample, 0);
+ free(silence);
+ }
}
}
@@ -736,14 +755,15 @@ static void reset(void)
{
int err;
+ prepause_frames = 0;
if ((err = snd_pcm_drop(alsa_handler)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
return;
}
if ((err = snd_pcm_prepare(alsa_handler)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
return;
}
return;
@@ -767,7 +787,7 @@ static int play(void* data, int len, int flags)
//mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
if (!alsa_handler) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_DeviceConfigurationError);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Device configuration error.");
return 0;
}
@@ -782,15 +802,15 @@ static int play(void* data, int len, int flags)
res = 0;
}
else if (res == -ESTRPIPE) { /* suspend */
- mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume);
+ mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
sleep(1);
}
if (res < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res));
- mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Write error: %s\n", snd_strerror(res));
+ mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Trying to reset soundcard.\n");
if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(res));
return 0;
break;
}
@@ -810,14 +830,14 @@ static int get_space(void)
if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
{
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Cannot get pcm status: %s\n", 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;
- return ret;
+ unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
+ if (space > ao_data.buffersize) // Buffer underrun?
+ space = ao_data.buffersize;
+ return space;
}
/* delay in seconds between first and last sample in buffer */
diff --git a/libao2/ao_alsa5.c b/libao2/ao_alsa5.c
index a9b470e9d5..d6ab7eeb22 100644
--- a/libao2/ao_alsa5.c
+++ b/libao2/ao_alsa5.c
@@ -30,7 +30,6 @@
#include "libaf/af_format.h"
#include "mp_msg.h"
-#include "help_mp.h"
static const ao_info_t info =
{
@@ -65,7 +64,7 @@ static int init(int rate_hz, int channels, int format, int flags)
snd_pcm_info_t info;
snd_pcm_channel_info_t chninfo;
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_InitInfo, rate_hz,
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ALSA5] alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz,
channels, af_fmt2str_short(format));
alsa_handler = NULL;
@@ -75,7 +74,7 @@ static int init(int rate_hz, int channels, int format, int flags)
if ((cards = snd_cards()) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_SoundCardNotFound);
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: no soundcards found.\n");
return 0;
}
@@ -121,7 +120,7 @@ static int init(int rate_hz, int channels, int format, int flags)
ao_data.bps *= 2;
break;
case -1:
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,af_fmt2str_short(format));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: invalid format (%s) requested - output disabled.\n",af_fmt2str_short(format));
return 0;
default:
break;
@@ -173,17 +172,17 @@ static int init(int rate_hz, int channels, int format, int flags)
if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PlayBackError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: playback open error: %s\n", snd_strerror(err));
return 0;
}
if ((err = snd_pcm_info(alsa_handler, &info)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmInfoError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: PCM info error: %s\n", snd_strerror(err));
return 0;
}
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_SoundcardsFound,
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ALSA5] alsa-init: %d soundcard(s) found, using: %s\n",
cards, info.name);
if (info.flags & SND_PCM_INFO_PLAYBACK)
@@ -192,7 +191,7 @@ static int init(int rate_hz, int channels, int format, int flags)
chninfo.channel = SND_PCM_CHANNEL_PLAYBACK;
if ((err = snd_pcm_channel_info(alsa_handler, &chninfo)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmChanInfoError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: PCM channel info error: %s\n", snd_strerror(err));
return 0;
}
@@ -216,7 +215,7 @@ static int init(int rate_hz, int channels, int format, int flags)
if ((err = snd_pcm_channel_params(alsa_handler, &params)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_CantSetParms, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: error setting parameters: %s\n", snd_strerror(err));
return 0;
}
@@ -229,13 +228,13 @@ static int init(int rate_hz, int channels, int format, int flags)
if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_CantSetChan, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: error setting up channel: %s\n", snd_strerror(err));
return 0;
}
if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ChanPrepareError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: channel prepare error: %s\n", snd_strerror(err));
return 0;
}
@@ -252,19 +251,19 @@ static void uninit(int immed)
if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_DrainError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-uninit: playback drain error: %s\n", snd_strerror(err));
return;
}
if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_FlushError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-uninit: playback flush error: %s\n", snd_strerror(err));
return;
}
if ((err = snd_pcm_close(alsa_handler)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmCloseError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-uninit: PCM close error: %s\n", snd_strerror(err));
return;
}
}
@@ -276,19 +275,19 @@ static void reset(void)
if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetDrainError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-reset: playback drain error: %s\n", snd_strerror(err));
return;
}
if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetFlushError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-reset: playback flush error: %s\n", snd_strerror(err));
return;
}
if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetChanPrepareError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-reset: channel prepare error: %s\n", snd_strerror(err));
return;
}
}
@@ -300,13 +299,13 @@ static void audio_pause(void)
if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseDrainError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-pause: playback drain error: %s\n", snd_strerror(err));
return;
}
if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseFlushError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-pause: playback flush error: %s\n", snd_strerror(err));
return;
}
}
@@ -317,7 +316,7 @@ static void audio_resume(void)
int err;
if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResumePrepareError, snd_strerror(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-resume: channel prepare error: %s\n", snd_strerror(err));
return;
}
}
@@ -337,21 +336,21 @@ static int play(void* data, int len, int flags)
{
if (got_len == -EPIPE) /* underrun? */
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_Underrun);
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-play: alsa underrun, resetting stream.\n");
if ((got_len = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PlaybackPrepareError, snd_strerror(got_len));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-play: playback prepare error: %s\n", snd_strerror(got_len));
return 0;
}
if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
{
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_WriteErrorAfterReset,
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-play: write error after reset: %s - giving up.\n",
snd_strerror(got_len));
return 0;
}
return got_len; /* 2nd write was ok */
}
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_OutPutError, snd_strerror(got_len));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-play: output error: %s\n", snd_strerror(got_len));
return 0;
}
return got_len;
diff --git a/libao2/ao_arts.c b/libao2/ao_arts.c
index 4a0f446c65..d828e7953e 100644
--- a/libao2/ao_arts.c
+++ b/libao2/ao_arts.c
@@ -28,7 +28,6 @@
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "mp_msg.h"
-#include "help_mp.h"
#define OBTAIN_BITRATE(a) (((a != AF_FORMAT_U8) && (a != AF_FORMAT_S8)) ? 16 : 8)
@@ -59,10 +58,10 @@ static int init(int rate_hz, int channels, int format, int flags)
int frag_spec;
if( (err=arts_init()) ) {
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantInit, arts_error_text(err));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ARTS] %s\n", arts_error_text(err));
return 0;
}
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_ServerConnect);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ARTS] Connected to sound server.\n");
/*
* arts supports 8bit unsigned and 16bit signed sample formats
@@ -93,7 +92,7 @@ static int init(int rate_hz, int channels, int format, int flags)
stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "MPlayer");
if(stream == NULL) {
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantOpenStream);
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ARTS] Unable to open a stream.\n");
arts_free();
return 0;
}
@@ -104,11 +103,11 @@ static int init(int rate_hz, int channels, int format, int flags)
frag_spec = ARTS_PACKET_SIZE_LOG2 | ARTS_PACKETS << 16;
arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec);
ao_data.buffersize = arts_stream_get(stream, ARTS_P_BUFFER_SIZE);
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_StreamOpen);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ARTS] Stream opened.\n");
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize,
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ARTS] buffer size: %d\n",
ao_data.buffersize);
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize,
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ARTS] buffer size: %d\n",
arts_stream_get(stream, ARTS_P_PACKET_SIZE));
return 1;
diff --git a/libao2/ao_dxr2.c b/libao2/ao_dxr2.c
index 37bf6f636d..cac669db58 100644
--- a/libao2/ao_dxr2.c
+++ b/libao2/ao_dxr2.c
@@ -26,7 +26,6 @@
#include <dxr2ioctl.h>
#include "config.h"
#include "mp_msg.h"
-#include "help_mp.h"
#include "libavutil/common.h"
#include "mpbswap.h"
@@ -73,7 +72,7 @@ static int control(int cmd,void *arg){
if(v.arg != volume) {
volume = v.arg;
if( ioctl(dxr2_fd,DXR2_IOC_SET_AUDIO_VOLUME,&v) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_SetVolFailed,volume);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO DXR2] Setting volume to %d failed.\n",volume);
return CONTROL_ERROR;
}
}
@@ -135,7 +134,7 @@ static int init(int rate,int channels,int format,int flags){
break;
#endif
default:
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_UnsupSamplerate,rate);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO DXR2] %d Hz not supported, try to resample.\n",rate);
return 0;
}
diff --git a/libao2/ao_esd.c b/libao2/ao_esd.c
index 593d5a934a..1097d4dfa6 100644
--- a/libao2/ao_esd.c
+++ b/libao2/ao_esd.c
@@ -51,7 +51,6 @@
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "mp_msg.h"
-#include "help_mp.h"
#define ESD_RESAMPLES 0
@@ -168,7 +167,7 @@ static int init(int rate_hz, int channels, int format, int flags)
if (esd_fd < 0) {
esd_fd = esd_open_sound(server);
if (esd_fd < 0) {
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ESD_CantOpenSound,
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ESD] esd_open_sound failed: %s\n",
strerror(errno));
return 0;
}
@@ -245,14 +244,14 @@ static int init(int rate_hz, int channels, int format, int flags)
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_tmsg(MSGT_AO, MSGL_INFO,"[AO ESD] latency: [server: %0.2fs, net: %0.2fs] (adjust %0.2fs)\n",
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) {
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ESD_CantOpenPBStream, strerror(errno));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ESD] failed to open ESD playback stream: %s\n", strerror(errno));
return 0;
}
diff --git a/libao2/ao_ivtv.c b/libao2/ao_ivtv.c
index b09e5308bb..1fe1b1edda 100644
--- a/libao2/ao_ivtv.c
+++ b/libao2/ao_ivtv.c
@@ -30,7 +30,6 @@
#include "config.h"
#include "mp_msg.h"
-#include "help_mp.h"
#include "audio_out.h"
#include "audio_out_internal.h"
@@ -86,7 +85,7 @@ init (int rate, int channels, int format, int flags)
/* check for supported audio rate */
if (rate != 32000 || rate != 41000 || rate != 48000)
{
- mp_msg (MSGT_AO, MSGL_ERR, MSGTR_AO_MPEGPES_UnsupSamplerate, rate);
+ mp_tmsg (MSGT_AO, MSGL_ERR, "[AO MPEGPES] %d Hz not supported, try to resample.\n", rate);
rate = 48000;
}
@@ -147,7 +146,7 @@ get_space (void)
static int
play (void *data, int len, int flags)
{
- int ivtv_write (unsigned char *data, int len);
+ int ivtv_write (const 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 0d2492c02b..65bd229bf3 100644
--- a/libao2/ao_jack.c
+++ b/libao2/ao_jack.c
@@ -28,7 +28,6 @@
#include "config.h"
#include "mp_msg.h"
-#include "help_mp.h"
#include "audio_out.h"
#include "audio_out_internal.h"
diff --git a/libao2/ao_mpegpes.c b/libao2/ao_mpegpes.c
index 7b2d3a5788..52c2b6cca0 100644
--- a/libao2/ao_mpegpes.c
+++ b/libao2/ao_mpegpes.c
@@ -38,7 +38,6 @@
#include "subopt-helper.h"
#include "mp_msg.h"
-#include "help_mp.h"
#ifdef CONFIG_DVB
#include <poll.h>
@@ -89,7 +88,7 @@ static int control(int cmd,void *arg){
if(dvb_mixer.volume_right>255) dvb_mixer.volume_right=255;
// printf("Setting DVB volume: %d ; %d \n",dvb_mixer.volume_left,dvb_mixer.volume_right);
if ( (ioctl(vo_mpegpes_fd2,AUDIO_SET_MIXER, &dvb_mixer) < 0)){
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_MPEGPES_CantSetMixer,
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO MPEGPES] DVB audio set mixer failed: %s.\n",
strerror(errno));
return CONTROL_ERROR;
}
@@ -253,7 +252,7 @@ static int init(int rate,int channels,int format,int flags){
case 44100: freq_id=2;break;
case 32000: freq_id=3;break;
default:
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_MPEGPES_UnsupSamplerate, rate);
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO MPEGPES] %d Hz not supported, try to resample.\n", rate);
#if 0
if(rate>48000) rate=96000; else
if(rate>44100) rate=48000; else
@@ -293,8 +292,6 @@ static void audio_resume(void)
{
}
-void send_pes_packet(unsigned char* data,int len,int id,int timestamp);
-void send_lpcm_packet(unsigned char* data,int len,int id,int timestamp,int freq_id);
extern int vo_pts;
// return: how many bytes can be played without blocking
diff --git a/libao2/ao_openal.c b/libao2/ao_openal.c
index 030304007c..e425b5769c 100644
--- a/libao2/ao_openal.c
+++ b/libao2/ao_openal.c
@@ -34,7 +34,6 @@
#endif
#include "mp_msg.h"
-#include "help_mp.h"
#include "audio_out.h"
#include "audio_out_internal.h"
diff --git a/libao2/ao_oss.c b/libao2/ao_oss.c
index 6c23026e87..82a0dd51e5 100644
--- a/libao2/ao_oss.c
+++ b/libao2/ao_oss.c
@@ -33,7 +33,6 @@
#include "config.h"
#include "mp_msg.h"
#include "mixer.h"
-#include "help_mp.h"
#ifdef HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
@@ -142,7 +141,7 @@ static int oss2format(int format)
case AFMT_AC3: return AF_FORMAT_AC3_NE;
#endif
}
- mp_msg(MSGT_GLOBAL,MSGL_ERR,MSGTR_AO_OSS_UnknownUnsupportedFormat, format);
+ mp_tmsg(MSGT_GLOBAL,MSGL_ERR,"[AO OSS] Unknown/Unsupported OSS format: %x.\n", format);
return -1;
}
@@ -154,6 +153,29 @@ static int prepause_space;
static const char *oss_mixer_device = PATH_DEV_MIXER;
static int oss_mixer_channel = SOUND_MIXER_PCM;
+#ifdef SNDCTL_DSP_GETPLAYVOL
+static int volume_oss4(ao_control_vol_t *vol, int cmd) {
+ int v;
+
+ if (audio_fd < 0)
+ return CONTROL_ERROR;
+
+ if (cmd == AOCONTROL_GET_VOLUME) {
+ if (ioctl(audio_fd, SNDCTL_DSP_GETPLAYVOL, &v) == -1)
+ return CONTROL_ERROR;
+ vol->right = (v & 0xff00) >> 8;
+ vol->left = v & 0x00ff;
+ return CONTROL_OK;
+ } else if (cmd == AOCONTROL_SET_VOLUME) {
+ v = ((int) vol->right << 8) | (int) vol->left;
+ if (ioctl(audio_fd, SNDCTL_DSP_SETPLAYVOL, &v) == -1)
+ return CONTROL_ERROR;
+ return CONTROL_OK;
+ } else
+ return CONTROL_UNKNOWN;
+}
+#endif
+
// to set/get/query special features/parameters
static int control(int cmd,void *arg){
switch(cmd){
@@ -179,6 +201,12 @@ static int control(int cmd,void *arg){
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
int fd, v, devs;
+#ifdef SNDCTL_DSP_GETPLAYVOL
+ // Try OSS4 first
+ if (volume_oss4(vol, cmd) == CONTROL_OK)
+ return CONTROL_OK;
+#endif
+
if(AF_FORMAT_IS_AC3(ao_data.format))
return CONTROL_TRUE;
@@ -247,7 +275,7 @@ static int init(int rate,int channels,int format,int flags){
int fd, devs, i;
if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenMixer,
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open mixer device %s: %s\n",
oss_mixer_device, strerror(errno));
}else{
ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
@@ -256,7 +284,7 @@ static int init(int rate,int channels,int format,int flags){
for (i=0; i<SOUND_MIXER_NRDEVICES; i++){
if(!strcasecmp(mixer_channels[i], mchan)){
if(!(devs & (1 << i))){
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,mchan);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Audio card mixer does not have channel '%s', using default.\n",mchan);
i = SOUND_MIXER_NRDEVICES+1;
break;
}
@@ -265,7 +293,7 @@ static int init(int rate,int channels,int format,int flags){
}
}
if(i==SOUND_MIXER_NRDEVICES){
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound,mchan);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Audio card mixer does not have channel '%s', using default.\n",mchan);
}
}
} else
@@ -281,14 +309,14 @@ static int init(int rate,int channels,int format,int flags){
audio_fd=open(dsp, O_WRONLY);
#endif
if(audio_fd<0){
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenDev, dsp, strerror(errno));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open audio device %s: %s\n", dsp, strerror(errno));
return 0;
}
#ifdef __linux__
/* Remove the non-blocking flag */
if(fcntl(audio_fd, F_SETFL, 0) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantMakeFd, strerror(errno));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't make file descriptor blocking: %s\n", strerror(errno));
return 0;
}
#endif
@@ -317,7 +345,7 @@ ac3_retry:
}
if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 ||
oss_format != format2oss(format)) {
- mp_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSet, dsp,
+ mp_tmsg(MSGT_AO,MSGL_WARN, "[AO OSS] Can't set audio device %s to %s output, trying %s...\n", dsp,
af_fmt2str_short(format), af_fmt2str_short(AF_FORMAT_S16_NE) );
format=AF_FORMAT_S16_NE;
goto ac3_retry;
@@ -339,14 +367,14 @@ ac3_retry:
if (ao_data.channels > 2) {
if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
ao_data.channels != channels ) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, channels);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", channels);
return 0;
}
}
else {
int c = ao_data.channels-1;
if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, ao_data.channels);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Failed to set audio device to %d channels.\n", ao_data.channels);
return 0;
}
ao_data.channels=c+1;
@@ -360,7 +388,7 @@ ac3_retry:
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
int r=0;
- mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_OSS_CantUseGetospace);
+ mp_tmsg(MSGT_AO,MSGL_WARN,"[AO OSS] audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
} else {
@@ -391,7 +419,7 @@ ac3_retry:
}
free(data);
if(ao_data.buffersize==0){
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantUseSelect);
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\n *** Your audio driver DOES NOT support select() ***\n Recompile MPlayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
return 0;
}
#endif
@@ -440,7 +468,7 @@ static void reset(void){
uninit(1);
audio_fd=open(dsp, O_WRONLY);
if(audio_fd < 0){
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantReopen, strerror(errno));
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS]\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE *** %s\n", strerror(errno));
return;
}
@@ -492,8 +520,6 @@ static int get_space(void){
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
// calculate exact buffer space:
playsize = zz.fragments*zz.fragsize;
- if (playsize > MAX_OUTBURST)
- playsize = (MAX_OUTBURST / zz.fragsize) * zz.fragsize;
return playsize;
}
#endif
diff --git a/libao2/ao_pcm.c b/libao2/ao_pcm.c
index 10ee82a0e2..3639647284 100644
--- a/libao2/ao_pcm.c
+++ b/libao2/ao_pcm.c
@@ -32,7 +32,6 @@
#include "audio_out.h"
#include "audio_out_internal.h"
#include "mp_msg.h"
-#include "help_mp.h"
#ifdef __MINGW32__
// for GetFileType to detect pipes
@@ -181,10 +180,10 @@ static int init(int rate,int channels,int format,int flags){
ao_data.format=format;
ao_data.bps=channels*rate*(af_fmt2bits(format)/8);
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename,
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\nPCM: Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename,
(ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
(channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] Info: Faster dumping is achieved with -vc null -vo null -ao pcm:fast\n[AO PCM] Info: To write WAVE files use -ao pcm:waveheader (default).\n");
fp = fopen(ao_outputfilename, "wb");
if(fp) {
@@ -193,7 +192,7 @@ static int init(int rate,int channels,int format,int flags){
}
return 1;
}
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile,
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO PCM] Failed to open %s for writing!\n",
ao_outputfilename);
return 0;
}
diff --git a/libao2/ao_sdl.c b/libao2/ao_sdl.c
index c886cd8bb6..e9ae7298d5 100644
--- a/libao2/ao_sdl.c
+++ b/libao2/ao_sdl.c
@@ -26,7 +26,6 @@
#include "config.h"
#include "mp_msg.h"
-#include "help_mp.h"
#include "audio_out.h"
#include "audio_out_internal.h"
@@ -139,11 +138,11 @@ static int init(int rate,int channels,int format,int flags){
/* Allocate ring-buffer memory */
buffer = av_fifo_alloc(BUFFSIZE);
- mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_INFO, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
+ mp_tmsg(MSGT_AO,MSGL_INFO,"[AO SDL] Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
if(ao_subdevice) {
setenv("SDL_AUDIODRIVER", ao_subdevice, 1);
- mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_DriverInfo, ao_subdevice);
+ mp_tmsg(MSGT_AO,MSGL_INFO,"[AO SDL] using %s audio driver.\n", ao_subdevice);
}
ao_data.channels=channels;
@@ -177,7 +176,7 @@ static int init(int rate,int channels,int format,int flags){
default:
aspec.format = AUDIO_S16LSB;
ao_data.format = AF_FORMAT_S16_LE;
- mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, format);
+ mp_tmsg(MSGT_AO,MSGL_WARN,"[AO SDL] Unsupported audio format: 0x%x.\n", format);
}
/* The desired audio frequency in samples-per-second. */
@@ -198,13 +197,13 @@ void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer s
/* initialize the SDL Audio system */
if (SDL_Init (SDL_INIT_AUDIO/*|SDL_INIT_NOPARACHUTE*/)) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantInit, SDL_GetError());
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO SDL] SDL Audio initialization failed: %s\n", SDL_GetError());
return 0;
}
/* Open the audio device and start playing sound! */
if(SDL_OpenAudio(&aspec, &obtained) < 0) {
- mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_SDL_CantOpenAudio, SDL_GetError());
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO SDL] Unable to open audio: %s\n", SDL_GetError());
return 0;
}
@@ -232,7 +231,7 @@ void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer s
ao_data.format = AF_FORMAT_U16_BE;
break;
default:
- mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, obtained.format);
+ mp_tmsg(MSGT_AO,MSGL_WARN,"[AO SDL] Unsupported audio format: 0x%x.\n", obtained.format);
return 0;
}
diff --git a/libao2/ao_sgi.c b/libao2/ao_sgi.c
index 602ee25a27..40bc6b9177 100644
--- a/libao2/ao_sgi.c
+++ b/libao2/ao_sgi.c
@@ -29,7 +29,6 @@
#include "audio_out.h"
#include "audio_out_internal.h"
#include "mp_msg.h"
-#include "help_mp.h"
#include "libaf/af_format.h"
static const ao_info_t info =
@@ -117,7 +116,7 @@ static int fmt2sgial(int *format, int *width) {
// to set/get/query special features/parameters
static int control(int cmd, void *arg){
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_INFO);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] control.\n");
switch(cmd) {
case AOCONTROL_QUERY_FORMAT:
@@ -138,7 +137,7 @@ static int init(int rate, int channels, int format, int flags) {
smpfmt = fmt2sgial(&format, &smpwidth);
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
{ /* from /usr/share/src/dmedia/audio/setrate.c */
@@ -148,7 +147,7 @@ static int init(int rate, int channels, int format, int flags) {
if(ao_subdevice) {
rv = alGetResourceByName(AL_SYSTEM, ao_subdevice, AL_OUTPUT_DEVICE_TYPE);
if (!rv) {
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InvalidDevice);
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] play: invalid device.\n");
return 0;
}
}
@@ -161,20 +160,20 @@ static int init(int rate, int channels, int format, int flags) {
x[1].value.i = AL_CRYSTAL_MCLK_TYPE;
if (alSetParams(rv,x, 2)<0) {
- mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetParms_Samplerate, alGetErrorString(oserror()));
+ mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: setparams failed: %s\nCould not set desired samplerate.\n", alGetErrorString(oserror()));
}
if (x[0].sizeOut < 0) {
- mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetAlRate);
+ mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: AL_RATE was not accepted on the given resource.\n");
}
if (alGetParams(rv,x, 1)<0) {
- mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantGetParms, alGetErrorString(oserror()));
+ mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: getparams failed: %s\n", alGetErrorString(oserror()));
}
realrate = alFixedToDouble(x[0].value.ll);
if (frate != realrate) {
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_SampleRateInfo, realrate, frate);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: samplerate is now %f (desired rate is %f)\n", realrate, frate);
}
sample_rate = (int)realrate;
}
@@ -191,7 +190,7 @@ static int init(int rate, int channels, int format, int flags) {
ao_config = alNewConfig();
if (!ao_config) {
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
return 0;
}
@@ -200,14 +199,14 @@ static int init(int rate, int channels, int format, int flags) {
alSetSampFmt(ao_config, smpfmt) < 0 ||
alSetQueueSize(ao_config, sample_rate) < 0 ||
alSetDevice(ao_config, rv) < 0) {
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
return 0;
}
ao_port = alOpenPort("mplayer", "w", ao_config);
if (!ao_port) {
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitOpenAudioFailed, alGetErrorString(oserror()));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: Unable to open audio channel: %s\n", alGetErrorString(oserror()));
return 0;
}
@@ -222,7 +221,7 @@ static void uninit(int immed) {
/* TODO: samplerate should be set back to the value before mplayer was started! */
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Uninit);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] uninit: ...\n");
if (ao_config) {
alFreeConfig(ao_config);
@@ -241,7 +240,7 @@ static void uninit(int immed) {
// stop playing and empty buffers (for seeking/pause)
static void reset(void) {
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Reset);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] reset: ...\n");
alDiscardFrames(ao_port, queue_size);
}
@@ -249,14 +248,14 @@ static void reset(void) {
// stop playing, keep buffers (for pause)
static void audio_pause(void) {
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_PauseInfo);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] audio_pause: ...\n");
}
// resume playing, after audio_pause()
static void audio_resume(void) {
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_ResumeInfo);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] audio_resume: ...\n");
}
diff --git a/libao2/ao_sun.c b/libao2/ao_sun.c
index d43a6ad3c2..02ffa1edb6 100644
--- a/libao2/ao_sun.c
+++ b/libao2/ao_sun.c
@@ -47,7 +47,6 @@
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "mp_msg.h"
-#include "help_mp.h"
static const ao_info_t info =
{
@@ -154,13 +153,13 @@ static int realtime_samplecounter_available(char *dev)
info.play.samples = 0;
if (ioctl(fd, AUDIO_SETINFO, &info)) {
if ( mp_msg_test(MSGT_AO,MSGL_V) )
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed);
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] rtsc: SETINFO failed.\n");
goto error;
}
if (write(fd, silence, len) != len) {
if ( mp_msg_test(MSGT_AO,MSGL_V) )
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed);
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] rtsc: write failed.\n");
goto error;
}
@@ -494,7 +493,7 @@ static int init(int rate,int channels,int format,int flags){
audio_fd=open(audio_dev, O_WRONLY);
if(audio_fd<0){
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_CantOpenAudioDev, audio_dev, strerror(errno));
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] Can't open audio device %s, %s -> nosound.\n", audio_dev, strerror(errno));
return 0;
}
@@ -572,7 +571,7 @@ static int init(int rate,int channels,int format,int flags){
if (!ok) {
char buf[128];
- mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate,
+ mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate.\n",
channels, af_fmt2str(format, buf, 128), rate);
return 0;
}
diff --git a/libao2/ao_v4l2.c b/libao2/ao_v4l2.c
index 3781fab359..52fba877cb 100644
--- a/libao2/ao_v4l2.c
+++ b/libao2/ao_v4l2.c
@@ -27,7 +27,6 @@
#include "config.h"
#include "mp_msg.h"
-#include "help_mp.h"
#include "audio_out.h"
#include "audio_out_internal.h"
@@ -84,7 +83,7 @@ init (int rate, int channels, int format, int flags)
/* check for supported audio rate */
if (rate != 32000 || rate != 41000 || rate != 48000)
{
- mp_msg (MSGT_AO, MSGL_ERR, MSGTR_AO_MPEGPES_UnsupSamplerate, rate);
+ mp_tmsg (MSGT_AO, MSGL_ERR, "[AO MPEGPES] %d Hz not supported, try to resample.\n", rate);
rate = 48000;
}
diff --git a/libao2/audio_out.c b/libao2/audio_out.c
index 4ac5489d0e..d6cb01fb3f 100644
--- a/libao2/audio_out.c
+++ b/libao2/audio_out.c
@@ -24,8 +24,6 @@
#include "audio_out.h"
#include "mp_msg.h"
-#include "help_mp.h"
-#include "mp_core.h" /* for exit_player() */
// there are some globals:
ao_data_t ao_data={0,0,0,0,OUTBURST,-1,0};
@@ -129,7 +127,7 @@ const ao_functions_t* const audio_out_drivers[] =
void list_audio_out(void){
int i=0;
- mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AvailableAudioOutputDrivers);
+ mp_tmsg(MSGT_AO, MSGL_INFO, "Available audio output drivers:\n");
mp_msg(MSGT_IDENTIFY, MSGL_INFO, "ID_AUDIO_OUTPUTS\n");
while (audio_out_drivers[i]) {
const ao_info_t *info = audio_out_drivers[i++]->info;
@@ -145,10 +143,6 @@ const ao_functions_t* init_best_audio_out(char** ao_list,int use_plugin,int rate
while(ao_list[0][0]){
char* ao=ao_list[0];
int ao_len;
- if (strncmp(ao, "alsa9", 5) == 0 || strncmp(ao, "alsa1x", 6) == 0) {
- mp_msg(MSGT_AO, MSGL_FATAL, MSGTR_AO_ALSA9_1x_Removed);
- exit_player(EXIT_NONE);
- }
if (ao_subdevice) {
free(ao_subdevice);
ao_subdevice = NULL;
@@ -161,7 +155,7 @@ const ao_functions_t* init_best_audio_out(char** ao_list,int use_plugin,int rate
else
ao_len = strlen(ao);
- mp_msg(MSGT_AO, MSGL_V, MSGTR_AO_TryingPreferredAudioDriver,
+ mp_tmsg(MSGT_AO, MSGL_V, "Trying preferred audio driver '%.*s', options '%s'\n",
ao_len, ao, ao_subdevice ? ao_subdevice : "[none]");
for(i=0;audio_out_drivers[i];i++){
@@ -171,12 +165,12 @@ const ao_functions_t* init_best_audio_out(char** ao_list,int use_plugin,int rate
if(audio_out->init(rate,channels,format,flags))
return audio_out; // success!
else
- mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_FailedInit, ao);
+ mp_tmsg(MSGT_AO, MSGL_WARN, "Failed to initialize audio driver '%s'\n", ao);
break;
}
}
if (!audio_out_drivers[i]) // we searched through the entire list
- mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_NoSuchDriver, ao_len, ao);
+ mp_tmsg(MSGT_AO, MSGL_WARN, "No such audio driver '%.*s'\n", ao_len, ao);
// continue...
++ao_list;
if(!(ao_list[0])) return NULL; // do NOT fallback to others
@@ -186,7 +180,7 @@ const ao_functions_t* init_best_audio_out(char** ao_list,int use_plugin,int rate
ao_subdevice = NULL;
}
- mp_msg(MSGT_AO, MSGL_V, MSGTR_AO_TryingEveryKnown);
+ mp_tmsg(MSGT_AO, MSGL_V, "Trying every known audio driver...\n");
// now try the rest...
for(i=0;audio_out_drivers[i];i++){
diff --git a/libao2/audio_out.h b/libao2/audio_out.h
index 049cfc5f21..e483a88422 100644
--- a/libao2/audio_out.h
+++ b/libao2/audio_out.h
@@ -47,8 +47,7 @@ typedef struct ao_functions_s
} ao_functions_t;
/* global data used by mplayer and plugins */
-typedef struct ao_data_s
-{
+typedef struct ao_data {
int samplerate;
int channels;
int format;
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-07 11:29:44 +0000