diff options
Diffstat (limited to 'libao2')
| -rw-r--r-- | libao2/ao_alsa.c | 134 | ||||
| -rw-r--r-- | libao2/ao_alsa5.c | 46 | ||||
| -rw-r--r-- | libao2/ao_arts.c | 13 | ||||
| -rw-r--r-- | libao2/ao_dxr2.c | 5 | ||||
| -rw-r--r-- | libao2/ao_esd.c | 6 | ||||
| -rw-r--r-- | libao2/ao_ivtv.c | 4 | ||||
| -rw-r--r-- | libao2/ao_jack.c | 1 | ||||
| -rw-r--r-- | libao2/ao_mpegpes.c | 8 | ||||
| -rw-r--r-- | libao2/ao_openal.c | 1 | ||||
| -rw-r--r-- | libao2/ao_oss.c | 53 | ||||
| -rw-r--r-- | libao2/ao_pcm.c | 12 | ||||
| -rw-r--r-- | libao2/ao_sdl.c | 20 | ||||
| -rw-r--r-- | libao2/ao_sgi.c | 34 | ||||
| -rw-r--r-- | libao2/ao_sun.c | 9 | ||||
| -rw-r--r-- | libao2/ao_v4l2.c | 4 | ||||
| -rw-r--r-- | libao2/audio_out.c | 16 | ||||
| -rw-r--r-- | libao2/audio_out.h | 3 |
17 files changed, 193 insertions, 176 deletions
diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c index 5e848446f2..4c3629b8c9 100644 --- a/libao2/ao_alsa.c +++ b/libao2/ao_alsa.c @@ -84,6 +84,7 @@ static int ao_noblock = 0; static int open_mode; static int alsa_can_pause = 0; +static snd_pcm_sframes_t prepause_frames; #define ALSA_DEVICE_SIZE 256 @@ -146,8 +147,8 @@ 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 ; } } @@ -167,32 +168,32 @@ 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; @@ -207,7 +208,7 @@ 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)); return CONTROL_ERROR; } @@ -216,7 +217,7 @@ 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)); return CONTROL_ERROR; } @@ -263,8 +264,15 @@ 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(strarg_t *str) { @@ -348,6 +356,8 @@ 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; @@ -450,7 +460,7 @@ 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) { @@ -510,19 +520,19 @@ static int init(int rate_hz, int channels, int format, int flags) if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0) { if (err != -EBUSY && ao_noblock) { - 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, format == AF_FORMAT_AC3)) < 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"); } @@ -533,7 +543,7 @@ 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; } @@ -541,7 +551,7 @@ 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; } @@ -551,8 +561,8 @@ 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; ao_data.format = AF_FORMAT_S16_LE; } @@ -560,7 +570,7 @@ 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; } @@ -568,7 +578,7 @@ 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; } @@ -579,7 +589,7 @@ 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; } @@ -588,7 +598,7 @@ 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; } @@ -605,7 +615,7 @@ static int init(int rate_hz, int channels, int format, int flags) 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, + mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n", snd_strerror(err)); return 0; } else @@ -615,11 +625,11 @@ static int init(int rate_hz, int channels, int format, int flags) &alsa_period_time, NULL)) < 0) /* original: alsa_buffer_time/ao_data.bps */ { - mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriodTime, + mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set period time: %s\n", snd_strerror(err)); return 0; } - mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_BufferTimePeriodTime, + mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] buffer_time: %d, period_time :%d\n", alsa_buffer_time, err); } #endif//end SET_BUFFERTIME @@ -630,7 +640,7 @@ static int init(int rate_hz, int channels, int format, int flags) 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, + mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to set period size(%ld): %s\n", chunk_size, snd_strerror(err)); return 0; } @@ -639,7 +649,7 @@ 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n", snd_strerror(err)); return 0; } @@ -652,7 +662,7 @@ 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; } @@ -662,7 +672,7 @@ 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 { @@ -671,7 +681,7 @@ 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); @@ -680,13 +690,13 @@ 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; } @@ -695,26 +705,26 @@ 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; } @@ -742,7 +752,7 @@ 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 { @@ -751,7 +761,7 @@ 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"); } } @@ -762,14 +772,18 @@ 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; } } @@ -780,22 +794,27 @@ 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); + } } } @@ -804,14 +823,15 @@ 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; @@ -832,7 +852,7 @@ 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; } @@ -847,15 +867,15 @@ 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; } @@ -875,14 +895,14 @@ 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 feb2919f1d..b2dc1efd3a 100644 --- a/libao2/ao_alsa5.c +++ b/libao2/ao_alsa5.c @@ -65,7 +65,7 @@ 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 +75,7 @@ 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; } @@ -125,7 +125,7 @@ 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; @@ -177,17 +177,17 @@ 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) @@ -196,7 +196,7 @@ 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; } @@ -220,7 +220,7 @@ static int init(int rate_hz, int channels, int format, int flags) if ((err = snd_pcm_channel_params(alsa_handler, ¶ms)) < 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; } @@ -233,13 +233,13 @@ 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; } @@ -256,19 +256,19 @@ 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; } } @@ -280,19 +280,19 @@ 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; } } @@ -304,13 +304,13 @@ 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; } } @@ -321,7 +321,7 @@ 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; } } @@ -341,21 +341,21 @@ 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 df44db4f02..5e55d6f6b7 100644 --- a/libao2/ao_arts.c +++ b/libao2/ao_arts.c @@ -59,10 +59,10 @@ 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 +93,7 @@ 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 +104,11 @@ 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; @@ -147,4 +147,3 @@ static float get_delay(void) return ((float) (ao_data.buffersize - arts_stream_get(stream, ARTS_P_BUFFER_SPACE))) / ((float) ao_data.bps); } - diff --git a/libao2/ao_dxr2.c b/libao2/ao_dxr2.c index 136efd1a9e..e8aa65aab5 100644 --- a/libao2/ao_dxr2.c +++ b/libao2/ao_dxr2.c @@ -73,7 +73,7 @@ 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 +135,7 @@ 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; } @@ -220,4 +220,3 @@ static float get_delay(void){ return 0.0; } - diff --git a/libao2/ao_esd.c b/libao2/ao_esd.c index 593d5a934a..28a0009b80 100644 --- a/libao2/ao_esd.c +++ b/libao2/ao_esd.c @@ -168,7 +168,7 @@ 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 +245,14 @@ 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..6b6bb8816c 100644 --- a/libao2/ao_ivtv.c +++ b/libao2/ao_ivtv.c @@ -86,7 +86,7 @@ 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 +147,7 @@ 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 e92786c28d..14bb1c90dd 100644 --- a/libao2/ao_jack.c +++ b/libao2/ao_jack.c @@ -360,4 +360,3 @@ static float get_delay(void) { } return (float)buffered / (float)ao_data.bps + in_jack; } - diff --git a/libao2/ao_mpegpes.c b/libao2/ao_mpegpes.c index 64cb05fa03..096f7bade5 100644 --- a/libao2/ao_mpegpes.c +++ b/libao2/ao_mpegpes.c @@ -97,7 +97,7 @@ 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; } @@ -208,7 +208,7 @@ static int preinit(const char *arg) return vo_mpegpes_fd2; } -static int my_ao_write(unsigned char* data,int len){ +static int my_ao_write(const unsigned char* data,int len){ int orig_len = len; #ifdef CONFIG_DVB #define NFD 1 @@ -263,7 +263,7 @@ 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 n |