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authorGravatar wm4 <wm4@nowhere>2013-11-07 22:12:36 +0100
committerGravatar wm4 <wm4@nowhere>2013-11-07 22:12:36 +0100
commit91626b1c0606afb9bb582070e8a444a3ba8395ab (patch)
treeae7245c12aae01734bde1c68ab7f2cc26aeb5845 /audio
parentaa48eeac9707f7e54468b55226af1188e7d72e30 (diff)
audio: replace af_fmt2str_short -> af_fmt_to_str
Also, remove all af_fmt2str usages.
Diffstat (limited to 'audio')
-rw-r--r--audio/audio.c2
-rw-r--r--audio/decode/dec_audio.c2
-rw-r--r--audio/filter/af_bs2b.c3
-rw-r--r--audio/filter/af_dummy.c2
-rw-r--r--audio/filter/af_lavcac3enc.c3
-rw-r--r--audio/fmt-conversion.c2
-rw-r--r--audio/format.c11
-rw-r--r--audio/format.h5
-rw-r--r--audio/out/ao_alsa.c2
-rw-r--r--audio/out/ao_dsound.c4
-rw-r--r--audio/out/ao_oss.c10
-rw-r--r--audio/out/ao_pcm.c2
-rw-r--r--audio/out/ao_wasapi.c8
13 files changed, 22 insertions, 34 deletions
diff --git a/audio/audio.c b/audio/audio.c
index e1d3a76978..9d41928436 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -65,7 +65,7 @@ char *mp_audio_fmt_to_str(int srate, const struct mp_chmap *chmap, int format)
{
char *chstr = mp_chmap_to_str(chmap);
char *res = talloc_asprintf(NULL, "%dHz %s %dch %s", srate, chstr,
- chmap->num, af_fmt2str_short(format));
+ chmap->num, af_fmt_to_str(format));
talloc_free(chstr);
return res;
}
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index b9ca71692f..127139ff60 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -160,7 +160,7 @@ int init_best_audio_codec(sh_audio_t *sh_audio, char *audio_decoders)
mp_msg(MSGT_DECAUDIO, MSGL_V,
"AUDIO: %d Hz, %d ch, %s\n",
sh_audio->samplerate, sh_audio->channels.num,
- af_fmt2str_short(sh_audio->sample_format));
+ af_fmt_to_str(sh_audio->sample_format));
mp_msg(MSGT_IDENTIFY, MSGL_INFO,
"ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels.num);
diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c
index aa92b8be23..0e77b3e4eb 100644
--- a/audio/filter/af_bs2b.c
+++ b/audio/filter/af_bs2b.c
@@ -74,7 +74,6 @@ static int control(struct af_instance *af, int cmd, void *arg)
switch (cmd) {
case AF_CONTROL_REINIT: {
int format;
- char buf[256];
// Sanity check
if (!arg) return AF_ERROR;
@@ -150,7 +149,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
}
bs2b_set_srate(s->filter, (long)af->data->rate);
mp_msg(MSGT_AFILTER, MSGL_V, "[bs2b] using format %s\n",
- af_fmt2str(af->data->format,buf,256));
+ af_fmt_to_str(af->data->format));
return af_test_output(af,(struct mp_audio*)arg);
}
diff --git a/audio/filter/af_dummy.c b/audio/filter/af_dummy.c
index 03c9aa9e99..ab601ba9bb 100644
--- a/audio/filter/af_dummy.c
+++ b/audio/filter/af_dummy.c
@@ -34,7 +34,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
case AF_CONTROL_REINIT: ;
*af->data = *(struct mp_audio*)arg;
mp_msg(MSGT_AFILTER, MSGL_V, "[dummy] Was reinitialized: %iHz/%ich/%s\n",
- af->data->rate,af->data->nch,af_fmt2str_short(af->data->format));
+ af->data->rate,af->data->nch,af_fmt_to_str(af->data->format));
return AF_OK;
}
return AF_UNKNOWN;
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index cd3548ea8f..9ca193017a 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -362,9 +362,8 @@ static int af_open(struct af_instance* af){
break;
}
}
- char buf[100];
mp_msg(MSGT_AFILTER, MSGL_V, "[af_lavcac3enc]: in sample format: %s\n",
- af_fmt2str(s->in_sampleformat, buf, 100));
+ af_fmt_to_str(s->in_sampleformat));
s->pending_data_size = AF_NCH * AC3_FRAME_SIZE *
af_fmt2bits(s->in_sampleformat) / 8;
s->pending_data = malloc(s->pending_data_size);
diff --git a/audio/fmt-conversion.c b/audio/fmt-conversion.c
index ce2a9054bf..bb32361ce5 100644
--- a/audio/fmt-conversion.c
+++ b/audio/fmt-conversion.c
@@ -45,7 +45,7 @@ enum AVSampleFormat af_to_avformat(int fmt)
sample_fmt = audio_conversion_map[i].sample_fmt;
if (sample_fmt == AF_FORMAT_UNKNOWN)
mp_msg(MSGT_GLOBAL, MSGL_V, "Unsupported sample format: %s\n",
- af_fmt2str_short(fmt));
+ af_fmt_to_str(fmt));
return sample_fmt;
}
diff --git a/audio/format.c b/audio/format.c
index 9b0967ae53..01c9a431a1 100644
--- a/audio/format.c
+++ b/audio/format.c
@@ -62,15 +62,6 @@ int af_fmt_change_bits(int format, int bits)
return af_fmt_is_valid(format) ? format : 0;
}
-/* Convert format to str input str is a buffer for the
- converted string, size is the size of the buffer */
-char *af_fmt2str(int format, char* str, int size)
-{
- const char *name = af_fmt2str_short(format);
- snprintf(str, size, "%s", name);
- return str;
-}
-
const struct af_fmt_entry af_fmtstr_table[] = {
{ "mpeg2", AF_FORMAT_MPEG2 },
{ "ac3le", AF_FORMAT_AC3_LE },
@@ -119,7 +110,7 @@ bool af_fmt_is_valid(int format)
return false;
}
-const char *af_fmt2str_short(int format)
+const char *af_fmt_to_str(int format)
{
for (int i = 0; af_fmtstr_table[i].name; i++) {
if (af_fmtstr_table[i].format == format)
diff --git a/audio/format.h b/audio/format.h
index 9a898fe55c..6ec435b6b7 100644
--- a/audio/format.h
+++ b/audio/format.h
@@ -130,15 +130,14 @@ struct af_fmt_entry {
extern const struct af_fmt_entry af_fmtstr_table[];
int af_str2fmt_short(bstr str);
+const char *af_fmt_to_str(int format);
+
int af_fmt2bits(int format);
int af_fmt_change_bits(int format, int bits);
// Amount of bytes that contain audio of the given duration, aligned to frames.
int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate);
-char *af_fmt2str(int format, char* str, int size);
-const char *af_fmt2str_short(int format);
-
bool af_fmt_is_valid(int format);
#endif /* MPLAYER_AF_FORMAT_H */
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index 1ad7598075..1f7dc71d3d 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -442,7 +442,7 @@ static int init(struct ao *ao)
err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
if (err < 0) {
MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n",
- af_fmt2str_short(ao->format));
+ af_fmt_to_str(ao->format));
p->alsa_fmt = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao->format))
ao->format = AF_FORMAT_AC3_LE;
diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index 034f5d8446..464947c0dc 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -407,7 +407,7 @@ static int init(struct ao *ao)
break;
default:
MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
- af_fmt2str_short(format));
+ af_fmt_to_str(format));
format = AF_FORMAT_S16_LE;
}
//set our audio parameters
@@ -416,7 +416,7 @@ static int init(struct ao *ao)
ao->bps = ao->channels.num * rate * (af_fmt2bits(format) >> 3);
int buffersize = ao->bps; // space for 1 sec
MP_VERBOSE(ao, "Samplerate:%iHz Channels:%i Format:%s\n", rate,
- ao->channels.num, af_fmt2str_short(format));
+ ao->channels.num, af_fmt_to_str(format));
MP_VERBOSE(ao, "Buffersize:%d bytes (%d msec)\n",
buffersize, buffersize / ao->bps * 1000);
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index 283b960255..1a28267613 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -205,7 +205,7 @@ static int init(struct ao *ao)
mchan = p->cfg_oss_mixer_channel;
MP_VERBOSE(ao, "%d Hz %d chans %s\n", ao->samplerate,
- ao->channels.num, af_fmt2str_short(ao->format));
+ ao->channels.num, af_fmt_to_str(ao->format));
if (mchan) {
int fd, devs, i;
@@ -274,7 +274,7 @@ ac3_retry:
oss_format = format2oss(ao->format);
if (oss_format == -1) {
MP_VERBOSE(ao, "Unknown/not supported internal format: %s\n",
- af_fmt2str_short(ao->format));
+ af_fmt_to_str(ao->format));
#if BYTE_ORDER == BIG_ENDIAN
oss_format = AFMT_S16_BE;
#else
@@ -286,8 +286,8 @@ ac3_retry:
oss_format != format2oss(ao->format))
{
MP_WARN(ao, "Can't set audio device %s to %s output, trying %s...\n",
- p->dsp, af_fmt2str_short(ao->format),
- af_fmt2str_short(AF_FORMAT_S16_NE));
+ p->dsp, af_fmt_to_str(ao->format),
+ af_fmt_to_str(AF_FORMAT_S16_NE));
ao->format = AF_FORMAT_S16_NE;
goto ac3_retry;
}
@@ -298,7 +298,7 @@ ac3_retry:
return -1;
}
- MP_VERBOSE(ao, "sample format: %s\n", af_fmt2str_short(ao->format));
+ MP_VERBOSE(ao, "sample format: %s\n", af_fmt_to_str(ao->format));
if (!AF_FORMAT_IS_AC3(ao->format)) {
struct mp_chmap_sel sel = {0};
diff --git a/audio/out/ao_pcm.c b/audio/out/ao_pcm.c
index d2f5f791ec..f7d793700d 100644
--- a/audio/out/ao_pcm.c
+++ b/audio/out/ao_pcm.c
@@ -145,7 +145,7 @@ static int init(struct ao *ao)
MP_INFO(ao, "File: %s (%s)\nPCM: Samplerate: %d Hz Channels: %d Format: %s\n",
priv->outputfilename,
priv->waveheader ? "WAVE" : "RAW PCM", ao->samplerate,
- ao->channels.num, af_fmt2str_short(ao->format));
+ ao->channels.num, af_fmt_to_str(ao->format));
MP_INFO(ao, "Info: Faster dumping is achieved with -no-video\n");
MP_INFO(ao, "Info: To write WAVE files use -ao pcm:waveheader (default).\n");
diff --git a/audio/out/ao_wasapi.c b/audio/out/ao_wasapi.c
index 47c6fcfdb7..c605e1cd5d 100644
--- a/audio/out/ao_wasapi.c
+++ b/audio/out/ao_wasapi.c
@@ -325,7 +325,7 @@ static int try_format(struct wasapi_state *state,
EnterCriticalSection(&state->print_lock);
mp_msg(MSGT_AO, MSGL_V, "ao-wasapi: trying %dch %s @ %dhz\n",
- channels.num, af_fmt2str_short(af_format), samplerate);
+ channels.num, af_fmt_to_str(af_format), samplerate);
LeaveCriticalSection(&state->print_lock);
union WAVEFMT u;
@@ -351,7 +351,7 @@ static int try_format(struct wasapi_state *state,
if (set_ao_format(state, ao, wformat)) {
EnterCriticalSection(&state->print_lock);
mp_msg(MSGT_AO, MSGL_V, "ao-wasapi: accepted as %dch %s @ %dhz\n",
- ao->channels.num, af_fmt2str_short(ao->format), ao->samplerate);
+ ao->channels.num, af_fmt_to_str(ao->format), ao->samplerate);
LeaveCriticalSection(&state->print_lock);
return 1;
@@ -361,7 +361,7 @@ static int try_format(struct wasapi_state *state,
if (set_ao_format(state, ao, wformat)) {
EnterCriticalSection(&state->print_lock);
mp_msg(MSGT_AO, MSGL_V, "ao-wasapi: %dch %s @ %dhz accepted\n",
- ao->channels.num, af_fmt2str_short(af_format), samplerate);
+ ao->channels.num, af_fmt_to_str(af_format), samplerate);
LeaveCriticalSection(&state->print_lock);
return 1;
}
@@ -418,7 +418,7 @@ static int try_passthrough(struct wasapi_state *state,
EnterCriticalSection(&state->print_lock);
mp_msg(MSGT_AO, MSGL_V, "ao-wasapi: trying passthrough for %s...\n",
- af_fmt2str_short((ao->format&~AF_FORMAT_END_MASK) | AF_FORMAT_LE));
+ af_fmt_to_str((ao->format&~AF_FORMAT_END_MASK) | AF_FORMAT_LE));
LeaveCriticalSection(&state->print_lock);
HRESULT hr = IAudioClient_IsFormatSupported(state->pAudioClient,