diff options
author | wm4 <wm4@nowhere> | 2013-11-07 22:12:36 +0100 |
---|---|---|
committer | wm4 <wm4@nowhere> | 2013-11-07 22:12:36 +0100 |
commit | 91626b1c0606afb9bb582070e8a444a3ba8395ab (patch) | |
tree | ae7245c12aae01734bde1c68ab7f2cc26aeb5845 /audio/out | |
parent | aa48eeac9707f7e54468b55226af1188e7d72e30 (diff) | |
download | mpv-91626b1c0606afb9bb582070e8a444a3ba8395ab.tar.bz2 mpv-91626b1c0606afb9bb582070e8a444a3ba8395ab.tar.xz |
audio: replace af_fmt2str_short -> af_fmt_to_str
Also, remove all af_fmt2str usages.
Diffstat (limited to 'audio/out')
-rw-r--r-- | audio/out/ao_alsa.c | 2 | ||||
-rw-r--r-- | audio/out/ao_dsound.c | 4 | ||||
-rw-r--r-- | audio/out/ao_oss.c | 10 | ||||
-rw-r--r-- | audio/out/ao_pcm.c | 2 | ||||
-rw-r--r-- | audio/out/ao_wasapi.c | 8 |
5 files changed, 13 insertions, 13 deletions
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, |