X-Git-Url: http://git.tuebingen.mpg.de/?a=blobdiff_plain;f=osx_writer.c;h=54a1acd1e98e3c3ce597982cf7f05d56842e698b;hb=b5ff683a297e5223357c4edd968cc8e4ec17ed4f;hp=4b505f0ee5de74be4cf8ec3902e74bb774820945;hpb=96bdf2d7b6206bb042031f2b13075a56bb8921a3;p=paraslash.git diff --git a/osx_writer.c b/osx_writer.c index 4b505f0e..54a1acd1 100644 --- a/osx_writer.c +++ b/osx_writer.c @@ -50,6 +50,8 @@ struct private_osx_writer_data { sem_t *semaphore; osx_buffer *from; /* Current buffers */ osx_buffer *to; + unsigned samplerate; + unsigned channels; }; @@ -181,6 +183,8 @@ static int osx_writer_open(struct writer_node *wn) AudioStreamBasicDescription format; char s[10]; int m, ret; + struct writer_node_group *wng = wn->wng; + struct osx_write_args_info *conf = wn->conf; wn->private_data = powd; /* where did that default audio output go? */ @@ -209,19 +213,25 @@ static int osx_writer_open(struct writer_node *wn) * AND you want the DefaultOutputUnit to do any format conversions * necessary from your format to the device's format. */ - format.mSampleRate = 44100.0; /* The sample rate of the audio stream */ + if (!conf->samplerate_given && wng->samplerate) + powd->samplerate = *wng->samplerate; + else + powd->samplerate = conf->samplerate_arg; + format.mSampleRate = powd->samplerate; /* The specific encoding type of audio stream*/ format.mFormatID = kAudioFormatLinearPCM; /* flags specific to each format */ format.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian; - /* - * We produce 2-channel audio. Now if we have a mega-super-hyper card for our - * audio, it is its problem to convert it to 8-, 16-, 32- or 1024-channel data. - */ - format.mBytesPerFrame = (format.mFramesPerPacket = 1) - * (format.mBytesPerPacket = (format.mChannelsPerFrame = 2) * sizeof(float)); + if (!conf->channels_given && wng->channels) + powd->channels = *wng->channels; + else + powd->channels = conf->channels_arg; + format.mChannelsPerFrame = powd->channels; + format.mFramesPerPacket = 1; + format.mBytesPerPacket = format.mChannelsPerFrame * sizeof(float); + format.mBytesPerFrame = format.mFramesPerPacket * format.mBytesPerPacket; /* one of the most constant constants of the whole computer history */ format.mBitsPerChannel = sizeof(float) * 8; ret = -E_STREAM_FORMAT; @@ -284,20 +294,30 @@ static void osx_writer_close(struct writer_node *wn) free(powd); } +static int need_new_buffer(struct writer_node *wn) +{ + struct writer_node_group *wng = wn->wng; + struct private_osx_writer_data *powd = wn->private_data; + + if (*wng->loaded < sizeof(short)) + return 0; + if (powd->to->remaining) /* Non empty buffer, must still be playing */ + return 0; + return 1; +} + static int osx_write_post_select(__a_unused struct sched *s, struct writer_node *wn) { struct private_osx_writer_data *powd = wn->private_data; struct writer_node_group *wng = wn->wng; - short *data = (short*)wng->buf + wn->written; + short *data = (short*)wng->buf; - if (*wng->loaded <= wn->written) - return 1; - if (powd->to->remaining) /* Non empty buffer, must still be playing */ + if (!need_new_buffer(wn)) return 1; - fill_buffer(powd->to, data, (*wng->loaded - wn->written) / sizeof(short)); + fill_buffer(powd->to, data, *wng->loaded / sizeof(short)); powd->to = powd->to->next; - wn->written += (*wng->loaded - wn->written); + wn->written = *wng->loaded; if (!powd->play) { if (AudioOutputUnitStart(powd->output)) return -1;