diff options
Diffstat (limited to 'libao2/ao_win32.c')
-rw-r--r-- | libao2/ao_win32.c | 11 |
1 files changed, 5 insertions, 6 deletions
diff --git a/libao2/ao_win32.c b/libao2/ao_win32.c index ff07a9e73c..49159b4324 100644 --- a/libao2/ao_win32.c +++ b/libao2/ao_win32.c @@ -147,7 +147,6 @@ static int init(int rate,int channels,int format,int flags) MMRESULT result; unsigned char* buffer; int i; - char buf[128]; switch(format){ case AF_FORMAT_AC3: @@ -156,7 +155,7 @@ static int init(int rate,int channels,int format,int flags) case AF_FORMAT_S8: break; default: - mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str(format, &buf, 128)); + mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format)); format=AF_FORMAT_S16_LE; } //fill global ao_data @@ -168,11 +167,11 @@ static int init(int rate,int channels,int format,int flags) ao_data.bps*=2; if(ao_data.buffersize==-1) { - ao_data.buffersize=audio_out_format_bits(format)/8; + ao_data.buffersize=af_fmt2bits(format)/8; ao_data.buffersize*= channels; ao_data.buffersize*= SAMPLESIZE; } - mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, audio_out_format_name(format)); + mp_msg(MSGT_AO, MSGL_V,"ao_win32: Samplerate:%iHz Channels:%i Format:%s\n",rate, channels, af_fmt2str_short(format)); mp_msg(MSGT_AO, MSGL_V,"ao_win32: Buffersize:%d\n",ao_data.buffersize); //fill waveformatex @@ -189,14 +188,14 @@ static int init(int rate,int channels,int format,int flags) else { wformat.Format.wFormatTag = (channels>2)?WAVE_FORMAT_EXTENSIBLE:WAVE_FORMAT_PCM; - wformat.Format.wBitsPerSample = audio_out_format_bits(format); + wformat.Format.wBitsPerSample = af_fmt2bits(format); wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3); } if(channels>2) { wformat.dwChannelMask = channel_mask[channels-3]; wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; - wformat.Samples.wValidBitsPerSample=audio_out_format_bits(format); + wformat.Samples.wValidBitsPerSample=af_fmt2bits(format); } wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign; |