diff options
Diffstat (limited to 'stream')
-rw-r--r-- | stream/ai_alsa1x.c | 104 | ||||
-rw-r--r-- | stream/ai_oss.c | 82 | ||||
-rw-r--r-- | stream/ai_sndio.c | 10 | ||||
-rw-r--r-- | stream/audio_in.c | 198 | ||||
-rw-r--r-- | stream/cookies.c | 62 | ||||
-rw-r--r-- | stream/dvb_tune.c | 368 | ||||
-rw-r--r-- | stream/dvbin.h | 76 | ||||
-rw-r--r-- | stream/frequencies.c | 1416 | ||||
-rw-r--r-- | stream/frequencies.h | 108 | ||||
-rw-r--r-- | stream/stream_dvb.c | 1200 | ||||
-rw-r--r-- | stream/stream_dvd.c | 8 | ||||
-rw-r--r-- | stream/stream_radio.c | 2 | ||||
-rw-r--r-- | stream/stream_smb.c | 2 | ||||
-rw-r--r-- | stream/stream_vcd.c | 2 | ||||
-rw-r--r-- | stream/tv.c | 610 | ||||
-rw-r--r-- | stream/tv.h | 136 | ||||
-rw-r--r-- | stream/tvi_def.h | 10 | ||||
-rw-r--r-- | stream/tvi_dummy.c | 70 | ||||
-rw-r--r-- | stream/tvi_v4l2.c | 10 | ||||
-rw-r--r-- | stream/vcd_read.h | 14 | ||||
-rw-r--r-- | stream/vcd_read_darwin.h | 216 | ||||
-rw-r--r-- | stream/vcd_read_fbsd.h | 12 | ||||
-rw-r--r-- | stream/vcd_read_win32.h | 48 |
23 files changed, 2382 insertions, 2382 deletions
diff --git a/stream/ai_alsa1x.c b/stream/ai_alsa1x.c index c1a7199c71..bf36443dfe 100644 --- a/stream/ai_alsa1x.c +++ b/stream/ai_alsa1x.c @@ -40,61 +40,61 @@ int ai_alsa_setup(audio_in_t *ai) err = snd_pcm_hw_params_any(ai->alsa.handle, params); if (err < 0) { - MP_ERR(ai, "Broken configuration for this PCM: no configurations available.\n"); - return -1; + MP_ERR(ai, "Broken configuration for this PCM: no configurations available.\n"); + return -1; } err = snd_pcm_hw_params_set_access(ai->alsa.handle, params, - SND_PCM_ACCESS_RW_INTERLEAVED); + SND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) { - MP_ERR(ai, "Access type not available.\n"); - return -1; + MP_ERR(ai, "Access type not available.\n"); + return -1; } err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE); if (err < 0) { - MP_ERR(ai, "Sample format not available.\n"); - return -1; + MP_ERR(ai, "Sample format not available.\n"); + return -1; } err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels); if (err < 0) { - snd_pcm_hw_params_get_channels(params, &ai->channels); - MP_ERR(ai, "Channel count not available - reverting to default: %d\n", - ai->channels); + snd_pcm_hw_params_get_channels(params, &ai->channels); + MP_ERR(ai, "Channel count not available - reverting to default: %d\n", + ai->channels); } else { - ai->channels = ai->req_channels; + ai->channels = ai->req_channels; } dir = 0; rate = ai->req_samplerate; err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, &rate, &dir); if (err < 0) { - MP_ERR(ai, "Cannot set samplerate.\n"); + MP_ERR(ai, "Cannot set samplerate.\n"); } ai->samplerate = rate; dir = 0; ai->alsa.buffer_time = 1000000; err = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params, - &ai->alsa.buffer_time, &dir); + &ai->alsa.buffer_time, &dir); if (err < 0) { - MP_ERR(ai, "Cannot set buffer time.\n"); + MP_ERR(ai, "Cannot set buffer time.\n"); } dir = 0; ai->alsa.period_time = ai->alsa.buffer_time / 4; err = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params, - &ai->alsa.period_time, &dir); + &ai->alsa.period_time, &dir); if (err < 0) { - MP_ERR(ai, "Cannot set period time.\n"); + MP_ERR(ai, "Cannot set period time.\n"); } err = snd_pcm_hw_params(ai->alsa.handle, params); if (err < 0) { - MP_ERR(ai, "Unable to install hardware parameters: %s", snd_strerror(err)); - snd_pcm_hw_params_dump(params, ai->alsa.log); - return -1; + MP_ERR(ai, "Unable to install hardware parameters: %s", snd_strerror(err)); + snd_pcm_hw_params_dump(params, ai->alsa.log); + return -1; } dir = -1; @@ -102,8 +102,8 @@ int ai_alsa_setup(audio_in_t *ai) snd_pcm_hw_params_get_buffer_size(params, &buffer_size); ai->alsa.chunk_size = period_size; if (period_size == buffer_size) { - MP_ERR(ai, "Can't use period equal to buffer size (%u == %lu)\n", ai->alsa.chunk_size, (long)buffer_size); - return -1; + MP_ERR(ai, "Can't use period equal to buffer size (%u == %lu)\n", ai->alsa.chunk_size, (long)buffer_size); + return -1; } snd_pcm_sw_params_current(ai->alsa.handle, swparams); @@ -113,13 +113,13 @@ int ai_alsa_setup(audio_in_t *ai) err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size); if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) { - MP_ERR(ai, "Unable to install software parameters:\n"); - snd_pcm_sw_params_dump(swparams, ai->alsa.log); - return -1; + MP_ERR(ai, "Unable to install software parameters:\n"); + snd_pcm_sw_params_dump(swparams, ai->alsa.log); + return -1; } if (mp_msg_test(ai->log, MSGL_V)) { - snd_pcm_dump(ai->alsa.handle, ai->alsa.log); + snd_pcm_dump(ai->alsa.handle, ai->alsa.log); } ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE); @@ -137,14 +137,14 @@ int ai_alsa_init(audio_in_t *ai) err = snd_pcm_open(&ai->alsa.handle, ai->alsa.device, SND_PCM_STREAM_CAPTURE, 0); if (err < 0) { - MP_ERR(ai, "Error opening audio: %s\n", snd_strerror(err)); - return -1; + MP_ERR(ai, "Error opening audio: %s\n", snd_strerror(err)); + return -1; } err = snd_output_stdio_attach(&ai->alsa.log, stderr, 0); if (err < 0) { - return -1; + return -1; } err = ai_alsa_setup(ai); @@ -153,14 +153,14 @@ int ai_alsa_init(audio_in_t *ai) } #ifndef timersub -#define timersub(a, b, result) \ +#define timersub(a, b, result) \ do { \ - (result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \ - (result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \ - if ((result)->tv_usec < 0) { \ - --(result)->tv_sec; \ - (result)->tv_usec += 1000000; \ - } \ + (result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \ + (result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \ + if ((result)->tv_usec < 0) { \ + --(result)->tv_sec; \ + (result)->tv_usec += 1000000; \ + } \ } while (0) #endif @@ -171,25 +171,25 @@ int ai_alsa_xrun(audio_in_t *ai) snd_pcm_status_alloca(&status); if ((res = snd_pcm_status(ai->alsa.handle, status))<0) { - MP_ERR(ai, "ALSA status error: %s", snd_strerror(res)); - return -1; + MP_ERR(ai, "ALSA status error: %s", snd_strerror(res)); + return -1; } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) { - struct timeval now, diff, tstamp; - gettimeofday(&now, 0); - snd_pcm_status_get_trigger_tstamp(status, &tstamp); - timersub(&now, &tstamp, &diff); - MP_ERR(ai, "ALSA xrun!!! (at least %.3f ms long)\n", - diff.tv_sec * 1000 + diff.tv_usec / 1000.0); - if (mp_msg_test(ai->log, MSGL_V)) { - MP_ERR(ai, "ALSA Status:\n"); - snd_pcm_status_dump(status, ai->alsa.log); - } - if ((res = snd_pcm_prepare(ai->alsa.handle))<0) { - MP_ERR(ai, "ALSA xrun: prepare error: %s", snd_strerror(res)); - return -1; - } - return 0; /* ok, data should be accepted again */ + struct timeval now, diff, tstamp; + gettimeofday(&now, 0); + snd_pcm_status_get_trigger_tstamp(status, &tstamp); + timersub(&now, &tstamp, &diff); + MP_ERR(ai, "ALSA xrun!!! (at least %.3f ms long)\n", + diff.tv_sec * 1000 + diff.tv_usec / 1000.0); + if (mp_msg_test(ai->log, MSGL_V)) { + MP_ERR(ai, "ALSA Status:\n"); + snd_pcm_status_dump(status, ai->alsa.log); + } + if ((res = snd_pcm_prepare(ai->alsa.handle))<0) { + MP_ERR(ai, "ALSA xrun: prepare error: %s", snd_strerror(res)); + return -1; + } + return 0; /* ok, data should be accepted again */ } MP_ERR(ai, "ALSA read/write error"); return -1; diff --git a/stream/ai_oss.c b/stream/ai_oss.c index 8672d13fc0..b7a7988bde 100644 --- a/stream/ai_oss.c +++ b/stream/ai_oss.c @@ -56,28 +56,28 @@ int ai_oss_set_channels(audio_in_t *ai) if (ai->req_channels > 2) { - ioctl_param = ai->req_channels; - MP_VERBOSE(ai, "ioctl dsp channels: %d\n", - err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_CHANNELS, &ioctl_param)); - if (err < 0) { - MP_ERR(ai, "Unable to set channel count: %d\n", - ai->req_channels); - return -1; - } - ai->channels = ioctl_param; + ioctl_param = ai->req_channels; + MP_VERBOSE(ai, "ioctl dsp channels: %d\n", + err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_CHANNELS, &ioctl_param)); + if (err < 0) { + MP_ERR(ai, "Unable to set channel count: %d\n", + ai->req_channels); + return -1; + } + ai->channels = ioctl_param; } else { - ioctl_param = (ai->req_channels == 2); - MP_VERBOSE(ai, "ioctl dsp stereo: %d (req: %d)\n", - err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_STEREO, &ioctl_param), - ioctl_param); - if (err < 0) { - MP_ERR(ai, "Unable to set stereo: %d\n", - ai->req_channels == 2); - return -1; - } - ai->channels = ioctl_param ? 2 : 1; + ioctl_param = (ai->req_channels == 2); + MP_VERBOSE(ai, "ioctl dsp stereo: %d (req: %d)\n", + err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_STEREO, &ioctl_param), + ioctl_param); + if (err < 0) { + MP_ERR(ai, "Unable to set stereo: %d\n", + ai->req_channels == 2); + return -1; + } + ai->channels = ioctl_param ? 2 : 1; } return 0; } @@ -90,65 +90,65 @@ int ai_oss_init(audio_in_t *ai) ai->oss.audio_fd = open(ai->oss.device, O_RDONLY | O_CLOEXEC); if (ai->oss.audio_fd < 0) { - MP_ERR(ai, "Unable to open '%s': %s\n", - ai->oss.device, strerror(errno)); - return -1; + MP_ERR(ai, "Unable to open '%s': %s\n", + ai->oss.device, strerror(errno)); + return -1; } ioctl_param = 0 ; MP_VERBOSE(ai, "ioctl dsp getfmt: %d\n", - ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETFMTS, &ioctl_param)); + ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETFMTS, &ioctl_param)); MP_VERBOSE(ai, "Supported formats: %x\n", ioctl_param); if (!(ioctl_param & AFMT_S16_LE)) - MP_ERR(ai, "unsupported format\n"); + MP_ERR(ai, "unsupported format\n"); ioctl_param = AFMT_S16_LE; MP_VERBOSE(ai, "ioctl dsp setfmt: %d\n", - err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETFMT, &ioctl_param)); + err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETFMT, &ioctl_param)); if (err < 0) { - MP_ERR(ai, "Unable to set audio format."); - return -1; + MP_ERR(ai, "Unable to set audio format."); + return -1; } if (ai_oss_set_channels(ai) < 0) return -1; ioctl_param = ai->req_samplerate; MP_VERBOSE(ai, "ioctl dsp speed: %d\n", - err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SPEED, &ioctl_param)); + err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SPEED, &ioctl_param)); if (err < 0) { - MP_ERR(ai, "Unable to set samplerate: %d\n", - ai->req_samplerate); - return -1; + MP_ERR(ai, "Unable to set samplerate: %d\n", + ai->req_samplerate); + return -1; } ai->samplerate = ioctl_param; MP_VERBOSE(ai, "ioctl dsp trigger: %d\n", - ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETTRIGGER, &ioctl_param)); + ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETTRIGGER, &ioctl_param)); MP_VERBOSE(ai, "trigger: %x\n", ioctl_param); ioctl_param = PCM_ENABLE_INPUT; MP_VERBOSE(ai, "ioctl dsp trigger: %d\n", - err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETTRIGGER, &ioctl_param)); + err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETTRIGGER, &ioctl_param)); if (err < 0) { - MP_ERR(ai, "Unable to set trigger: %d\n", - PCM_ENABLE_INPUT); + MP_ERR(ai, "Unable to set trigger: %d\n", + PCM_ENABLE_INPUT); } ai->blocksize = 0; MP_VERBOSE(ai, "ioctl dsp getblocksize: %d\n", - err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETBLKSIZE, &ai->blocksize)); + err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETBLKSIZE, &ai->blocksize)); if (err < 0) { - MP_ERR(ai, "Unable to get block size!\n"); + MP_ERR(ai, "Unable to get block size!\n"); } MP_VERBOSE(ai, "blocksize: %d\n", ai->blocksize); // correct the blocksize to a reasonable value if (ai->blocksize <= 0) { - ai->blocksize = 4096*ai->channels*2; - MP_ERR(ai, "Audio block size is zero, setting to %d!\n", ai->blocksize); + ai->blocksize = 4096*ai->channels*2; + MP_ERR(ai, "Audio block size is zero, setting to %d!\n", ai->blocksize); } else if (ai->blocksize < 4096*ai->channels*2) { - ai->blocksize *= 4096*ai->channels*2/ai->blocksize; - MP_ERR(ai, "Audio block size too low, setting to %d!\n", ai->blocksize); + ai->blocksize *= 4096*ai->channels*2/ai->blocksize; + MP_ERR(ai, "Audio block size too low, setting to %d!\n", ai->blocksize); } ai->samplesize = 16; diff --git a/stream/ai_sndio.c b/stream/ai_sndio.c index 3cd68e5ee1..dc3c66279d 100644 --- a/stream/ai_sndio.c +++ b/stream/ai_sndio.c @@ -18,11 +18,11 @@ int ai_sndio_setup(audio_in_t *ai) par.le = 1; par.rchan = ai->req_channels; par.rate = ai->req_samplerate; - par.appbufsz = ai->req_samplerate; /* 1 sec */ + par.appbufsz = ai->req_samplerate; /* 1 sec */ if (!sio_setpar(ai->sndio.hdl, &par) || !sio_getpar(ai->sndio.hdl, &par)) { - MP_ERR(ai, "could not configure sndio audio"); - return -1; + MP_ERR(ai, "could not configure sndio audio"); + return -1; } ai->channels = par.rchan; @@ -39,8 +39,8 @@ int ai_sndio_init(audio_in_t *ai) int err; if ((ai->sndio.hdl = sio_open(ai->sndio.device, SIO_REC, 0)) == NULL) { - MP_ERR(ai, "could not open sndio audio"); - return -1; + MP_ERR(ai, "could not open sndio audio"); + return -1; } err = ai_sndio_setup(ai); diff --git a/stream/audio_in.c b/stream/audio_in.c index 6592735aa9..8e956630b7 100644 --- a/stream/audio_in.c +++ b/stream/audio_in.c @@ -43,25 +43,25 @@ int audio_in_init(audio_in_t *ai, struct mp_log *log, int type) switch (ai->type) { #if HAVE_ALSA case AUDIO_IN_ALSA: - ai->alsa.handle = NULL; - ai->alsa.log = NULL; - ai->alsa.device = strdup("default"); - return 0; + ai->alsa.handle = NULL; + ai->alsa.log = NULL; + ai->alsa.device = strdup("default"); + return 0; #endif #if HAVE_OSS_AUDIO case AUDIO_IN_OSS: - ai->oss.audio_fd = -1; - ai->oss.device = strdup("/dev/dsp"); - return 0; + ai->oss.audio_fd = -1; + ai->oss.device = strdup("/dev/dsp"); + return 0; #endif #if HAVE_SNDIO case AUDIO_IN_SNDIO: - ai->sndio.hdl = NULL; - ai->sndio.device = strdup("default"); - return 0; + ai->sndio.hdl = NULL; + ai->sndio.device = strdup("default"); + return 0; #endif default: - return -1; + return -1; } } @@ -71,24 +71,24 @@ int audio_in_setup(audio_in_t *ai) switch (ai->type) { #if HAVE_ALSA case AUDIO_IN_ALSA: - if (ai_alsa_init(ai) < 0) return -1; - ai->setup = 1; - return 0; + if (ai_alsa_init(ai) < 0) return -1; + ai->setup = 1; + return 0; #endif #if HAVE_OSS_AUDIO case AUDIO_IN_OSS: - if (ai_oss_init(ai) < 0) return -1; - ai->setup = 1; - return 0; + if (ai_oss_init(ai) < 0) return -1; + ai->setup = 1; + return 0; #endif #if HAVE_SNDIO case AUDIO_IN_SNDIO: - if (ai_sndio_init(ai) < 0) return -1; - ai->setup = 1; - return 0; + if (ai_sndio_init(ai) < 0) return -1; + ai->setup = 1; + return 0; #endif default: - return -1; + return -1; } } @@ -97,27 +97,27 @@ int audio_in_set_samplerate(audio_in_t *ai, int rate) switch (ai->type) { #if HAVE_ALSA case AUDIO_IN_ALSA: - ai->req_samplerate = rate; - if (!ai->setup) return 0; - if (ai_alsa_setup(ai) < 0) return -1; - return ai->samplerate; + ai->req_samplerate = rate; + if (!ai->setup) return 0; + if (ai_alsa_setup(ai) < 0) return -1; + return ai->samplerate; #endif #if HAVE_OSS_AUDIO case AUDIO_IN_OSS: - ai->req_samplerate = rate; - if (!ai->setup) return 0; - if (ai_oss_set_samplerate(ai) < 0) return -1; - return ai->samplerate; + ai->req_samplerate = rate; + if (!ai->setup) return 0; + if (ai_oss_set_samplerate(ai) < 0) return -1; + return ai->samplerate; #endif #if HAVE_SNDIO case AUDIO_IN_SNDIO: - ai->req_samplerate = rate; - if (!ai->setup) return 0; - if (ai_sndio_setup(ai) < 0) return -1; - return ai->samplerate; + ai->req_samplerate = rate; + if (!ai->setup) return 0; + if (ai_sndio_setup(ai) < 0) return -1; + return ai->samplerate; #endif default: - return -1; + return -1; } } @@ -126,17 +126,17 @@ int audio_in_set_channels(audio_in_t *ai, int channels) switch (ai->type) { #if HAVE_ALSA case AUDIO_IN_ALSA: - ai->req_channels = channels; - if (!ai->setup) return 0; - if (ai_alsa_setup(ai) < 0) return -1; - return ai->channels; + ai->req_channels = channels; + if (!ai->setup) return 0; + if (ai_alsa_setup(ai) < 0) return -1; + return ai->channels; #endif #if HAVE_OSS_AUDIO case AUDIO_IN_OSS: - ai->req_channels = channels; - if (!ai->setup) return 0; - if (ai_oss_set_channels(ai) < 0) return -1; - return ai->channels; + ai->req_channels = channels; + if (!ai->setup) return 0; + if (ai_oss_set_channels(ai) < 0) return -1; + return ai->channels; #endif #if HAVE_SNDIO case AUDIO_IN_SNDIO: @@ -146,7 +146,7 @@ int audio_in_set_channels(audio_in_t *ai, int channels) return ai->channels; #endif default: - return -1; + return -1; } } @@ -159,19 +159,19 @@ int audio_in_set_device(audio_in_t *ai, char *device) switch (ai->type) { #if HAVE_ALSA case AUDIO_IN_ALSA: - free(ai->alsa.device); - ai->alsa.device = strdup(device); - /* mplayer cannot handle colons in arguments */ - for (i = 0; i < (int)strlen(ai->alsa.device); i++) { - if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; - } - return 0; + free(ai->alsa.device); + ai->alsa.device = strdup(device); + /* mplayer cannot handle colons in arguments */ + for (i = 0; i < (int)strlen(ai->alsa.device); i++) { + if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; + } + return 0; #endif #if HAVE_OSS_AUDIO case AUDIO_IN_OSS: - free(ai->oss.device); - ai->oss.device = strdup(device); - return 0; + free(ai->oss.device); + ai->oss.device = strdup(device); + return 0; #endif #if HAVE_SNDIO case AUDIO_IN_SNDIO: @@ -180,29 +180,29 @@ int audio_in_set_device(audio_in_t *ai, char *device) return 0; #endif default: - return -1; + return -1; } } int audio_in_uninit(audio_in_t *ai) { if (ai->setup) { - switch (ai->type) { + switch (ai->type) { #if HAVE_ALSA - case AUDIO_IN_ALSA: - if (ai->alsa.log) - snd_output_close(ai->alsa.log); - if (ai->alsa.handle) { - snd_pcm_close(ai->alsa.handle); - } - ai->setup = 0; - return 0; + case AUDIO_IN_ALSA: + if (ai->alsa.log) + snd_output_close(ai->alsa.log); + if (ai->alsa.handle) { + snd_pcm_close(ai->alsa.handle); + } + ai->setup = 0; + return 0; #endif #if HAVE_OSS_AUDIO - case AUDIO_IN_OSS: - close(ai->oss.audio_fd); - ai->setup = 0; - return 0; + case AUDIO_IN_OSS: + close(ai->oss.audio_fd); + ai->setup = 0; + return 0; #endif #if HAVE_SNDIO case AUDIO_IN_SNDIO: @@ -211,7 +211,7 @@ int audio_in_uninit(audio_in_t *ai) ai->setup = 0; return 0; #endif - } + } } return -1; } @@ -221,11 +221,11 @@ int audio_in_start_capture(audio_in_t *ai) switch (ai->type) { #if HAVE_ALSA case AUDIO_IN_ALSA: - return snd_pcm_start(ai->alsa.handle); + return snd_pcm_start(ai->alsa.handle); #endif #if HAVE_OSS_AUDIO case AUDIO_IN_OSS: - return 0; + return 0; #endif #if HAVE_SNDIO case AUDIO_IN_SNDIO: @@ -234,7 +234,7 @@ int audio_in_start_capture(audio_in_t *ai) return 0; #endif default: - return -1; + return -1; } } @@ -245,27 +245,27 @@ int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) switch (ai->type) { #if HAVE_ALSA case AUDIO_IN_ALSA: - ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); - if (ret != ai->alsa.chunk_size) { - if (ret < 0) { - MP_ERR(ai, "\nError reading audio: %s\n", snd_strerror(ret)); - if (ret == -EPIPE) { - if (ai_alsa_xrun(ai) == 0) { - MP_ERR(ai, "Recovered from cross-run, some frames may be left out!\n"); - } else { - MP_ERR(ai, "Fatal error, cannot recover!\n"); - } - } - } else { - MP_ERR(ai, "\nNot enough audio samples!\n"); - } - return -1; - } - return ret; + ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); + if (ret != ai->alsa.chunk_size) { + if (ret < 0) { + MP_ERR(ai, "\nError reading audio: %s\n", snd_strerror(ret)); + if (ret == -EPIPE) { + if (ai_alsa_xrun(ai) == 0) { + MP_ERR(ai, "Recovered from cross-run, some frames may be left out!\n"); + } else { + MP_ERR(ai, "Fatal error, cannot recover!\n"); + } + } + } else { + MP_ERR(ai, "\nNot enough audio samples!\n"); + } + return -1; + } + return ret; #endif #if HAVE_OSS_AUDIO case AUDIO_IN_OSS: - ret = read(ai->oss.audio_fd, buffer, ai->blocksize); + ret = read(ai->oss.audio_fd, buffer, ai->blocksize); if (ret != ai->blocksize) { if (ret < 0) { MP_ERR(ai, "\nError reading audio: %s\n", strerror(errno)); @@ -280,17 +280,17 @@ int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) #if HAVE_SNDIO case AUDIO_IN_SNDIO: ret = sio_read(ai->sndio.hdl, buffer, ai->blocksize); - if (ret != ai->blocksize) { - if (ret < 0) { - MP_ERR(ai, "\nError reading audio: %s\n", strerror(errno)); - } else { - MP_ERR(ai, "\nNot enough audio samples!\n"); - } - return -1; - } - return ret; + if (ret != ai->blocksize) { + if (ret < 0) { + MP_ERR(ai, "\nError reading audio: %s\n", strerror(errno)); + } else { + MP_ERR(ai, "\nNot enough audio samples!\n"); + } + return -1; + } + return ret; #endif default: - return -1; + return -1; } } diff --git a/stream/cookies.c b/stream/cookies.c index a12122f0ac..f8bc852259 100644 --- a/stream/cookies.c +++ b/stream/cookies.c @@ -55,7 +55,7 @@ static char *col_dup(void *talloc_ctx, const char *src) { int length = 0; while (src[length] > 31) - length++; + length++; return talloc_strndup(talloc_ctx, src, length); } @@ -67,13 +67,13 @@ static int parse_line(char **ptr, char *cols[7]) cols[0] = *ptr; for (col = 1; col < 7; col++) { - for (; (**ptr) > 31; (*ptr)++); - if (**ptr == 0) - return 0; - (*ptr)++; - if ((*ptr)[-1] != 9) - return 0; - cols[col] = (*ptr); + for (; (**ptr) > 31; (*ptr)++); + if (**ptr == 0) + return 0; + (*ptr)++; + if ((*ptr)[-1] != 9) + return 0; + cols[col] = (*ptr); } return 1; @@ -89,32 +89,32 @@ static char *load_file(struct mp_log *log, const char *filename, int64_t * lengt fd = open(filename, O_RDONLY | O_CLOEXEC); if (fd < 0) { - mp_verbose(log, "Could not open"); - goto err_out; + mp_verbose(log, "Could not open"); + goto err_out; } *length = lseek(fd, 0, SEEK_END); if (*length < 0) { - mp_verbose(log, "Could not find EOF"); - goto err_out; + mp_verbose(log, "Could not find EOF"); + goto err_out; } if (*length > SIZE_MAX - 1) { - mp_verbose(log, "File too big, could not malloc."); - goto err_out; + mp_verbose(log, "File too big, could not malloc."); + goto err_out; } lseek(fd, 0, SEEK_SET); if (!(buffer = malloc(*length + 1))) { - mp_verbose(log, "Could not malloc."); - goto err_out; + mp_verbose(log, "Could not malloc."); + goto err_out; } if (read(fd, buffer, *length) != *length) { - mp_verbose(log, "Read is behaving funny."); - goto err_out; + mp_verbose(log, "Read is behaving funny."); + goto err_out; } close(fd); buffer[*length] = 0; @@ -137,22 +137,22 @@ static struct cookie_list_type *load_cookies_from(void *ctx, ptr = file = load_file(log, filename, &length); if (!ptr) - return NULL; + return NULL; struct cookie_list_type *list = NULL; while (*ptr) { - char *cols[7]; - if (parse_line(&ptr, cols)) { - struct cookie_list_type *new; - new = talloc_zero(ctx, cookie_list_t); - new->name = col_dup(new, cols[5]); - new->value = col_dup(new, cols[6]); - new->path = col_dup(new, cols[2]); - new->domain = col_dup(new, cols[0]); - new->secure = (*(cols[3]) == 't') || (*(cols[3]) == 'T'); - new->next = list; - list = new; - } + char *cols[7]; + if (parse_line(&ptr, cols)) { + struct cookie_list_type *new; + new = talloc_zero(ctx, cookie_list_t); + new->name = col_dup(new, cols[5]); + new->value = col_dup(new, cols[6]); + new->path = col_dup(new, cols[2]); + new->domain = col_dup(new, cols[0]); + new->secure = (*(cols[3]) == 't') || (*(cols[3]) == 'T'); + new->next = list; + list = new; + } } free(file); return list; diff --git a/stream/dvb_tune.c b/stream/dvb_tune.c index 0cf19a8fba..7065a77aa3 100644 --- a/stream/dvb_tune.c +++ b/stream/dvb_tune.c @@ -52,225 +52,225 @@ int dvb_get_tuner_type(int fe_fd, struct mp_log *log) res = ioctl(fe_fd, FE_GET_INFO, &fe_info); if(res < 0) { - mp_err(log, "FE_GET_INFO error: %d, FD: %d\n\n", errno, fe_fd); - return 0; + mp_err(log, "FE_GET_INFO error: %d, FD: %d\n\n", errno, fe_fd); + return 0; } switch(fe_info.type) { - case FE_OFDM: + case FE_OFDM: mp_verbose(log, "TUNER TYPE SEEMS TO BE DVB-T\n"); - return TUNER_TER; + return TUNER_TER; - case FE_QPSK: + case FE_QPSK: mp_verbose(log, "TUNER TYPE SEEMS TO BE DVB-S\n"); - return TUNER_SAT; + return TUNER_SAT; - case FE_QAM: + case FE_QAM: mp_verbose(log, "TUNER TYPE SEEMS TO BE DVB-C\n"); - return TUNER_CBL; + return TUNER_CBL; #ifdef DVB_ATSC - case FE_ATSC: + case FE_ATSC: mp_verbose(log, "TUNER TYPE SEEMS TO BE DVB-ATSC\n"); - return TUNER_ATSC; + return TUNER_ATSC; #endif - default: - mp_err(log, "UNKNOWN TUNER TYPE\n"); - return 0; + default: + mp_err(log, "UNKNOWN TUNER TYPE\n"); + return 0; } } int dvb_open_devices(dvb_priv_t *priv, int n, int demux_cnt) { - int i; - char fronte |