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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/audio_in.c
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/audio_in.c')
-rw-r--r--stream/audio_in.c198
1 files changed, 99 insertions, 99 deletions
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;
}
}