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authorwm4 <wm4@nowhere>2014-04-13 18:00:51 +0200
committerwm4 <wm4@nowhere>2014-04-13 18:03:01 +0200
commit78128bddda4bcea1f256fc13cc33fa2652ed277c (patch)
tree35bf6596cb8e2d7927618845833c3ee36534f890 /stream
parent44f382cf98564c0fe08bdc78579c284362cd6f3c (diff)
downloadmpv-78128bddda4bcea1f256fc13cc33fa2652ed277c.tar.bz2
mpv-78128bddda4bcea1f256fc13cc33fa2652ed277c.tar.xz
Kill all tabs
I hate tabs. This replaces all tabs in all source files with spaces. The only exception is old-makefile. The replacement was made by running the GNU coreutils "expand" command on every file. Since the replacement was automatic, it's possible that some formatting was destroyed (but perhaps only if it was assuming that the end of a tab does not correspond to aligning the end to multiples of 8 spaces).
Diffstat (limited to 'stream')
-rw-r--r--stream/ai_alsa1x.c104
-rw-r--r--stream/ai_oss.c82
-rw-r--r--stream/ai_sndio.c10
-rw-r--r--stream/audio_in.c198
-rw-r--r--stream/cookies.c62
-rw-r--r--stream/dvb_tune.c368
-rw-r--r--stream/dvbin.h76
-rw-r--r--stream/frequencies.c1416
-rw-r--r--stream/frequencies.h108
-rw-r--r--stream/stream_dvb.c1200
-rw-r--r--stream/stream_dvd.c8
-rw-r--r--stream/stream_radio.c2
-rw-r--r--stream/stream_smb.c2
-rw-r--r--stream/stream_vcd.c2
-rw-r--r--stream/tv.c610
-rw-r--r--stream/tv.h136
-rw-r--r--stream/tvi_def.h10
-rw-r--r--stream/tvi_dummy.c70
-rw-r--r--stream/tvi_v4l2.c10
-rw-r--r--stream/vcd_read.h14
-rw-r--r--stream/vcd_read_darwin.h216
-rw-r--r--stream/vcd_read_fbsd.h12
-rw-r--r--stream/vcd_read_win32.h48
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)
{
- i