removed int16 decoding (was a SoundPlay workaround)
git-svn-id: file:///srv/svn/repos/haiku/trunk/current@6803 a95241bf-73f2-0310-859d-f6bbb57e9c96
This commit is contained in:
parent
34cee003fb
commit
fb3053bc25
@ -4,7 +4,6 @@
|
||||
#include <Locker.h>
|
||||
#include <MediaFormats.h>
|
||||
#include <MediaRoster.h>
|
||||
#include <Roster.h>
|
||||
#include <vector>
|
||||
#include "vorbisCodecPlugin.h"
|
||||
#include "OggVorbisFormats.h"
|
||||
@ -49,9 +48,6 @@ VorbisDecoder::VorbisDecoder()
|
||||
fStartTime = 0;
|
||||
fFrameSize = 0;
|
||||
fOutputBufferSize = 0;
|
||||
app_info info;
|
||||
fSoundplayLossage = (be_roster->GetActiveAppInfo(&info) == B_OK) &&
|
||||
(strcmp(info.signature, "application/x-vnd.marcone-soundplay") == 0);
|
||||
}
|
||||
|
||||
|
||||
@ -132,9 +128,6 @@ VorbisDecoder::NegotiateOutputFormat(media_format *ioDecodedFormat)
|
||||
// Be R5 behavior seems to be that we can never fail. If we
|
||||
// don't support the requested format, just return one we do.
|
||||
media_format format = vorbis_decoded_media_format();
|
||||
if (fSoundplayLossage) {
|
||||
format.u.raw_audio.format = media_raw_audio_format::B_AUDIO_SHORT;
|
||||
}
|
||||
format.u.raw_audio.frame_rate = (float)fInfo.rate;
|
||||
format.u.raw_audio.channel_count = fInfo.channels;
|
||||
format.u.raw_audio.channel_mask = B_CHANNEL_LEFT | (fInfo.channels != 1 ? B_CHANNEL_RIGHT : 0);
|
||||
@ -212,25 +205,10 @@ VorbisDecoder::Decode(void *buffer, int64 *frameCount,
|
||||
// reduce samples to the amount of samples we will actually consume
|
||||
samples = min_c(samples,out_bytes_needed/fFrameSize);
|
||||
total_samples += samples;
|
||||
if (fSoundplayLossage) {
|
||||
for (int sample = 0; sample < samples ; sample++) {
|
||||
for (int channel = 0; channel < fInfo.channels; channel++) {
|
||||
int32 thesample = (int32)(pcm[channel][sample] * 32767.0f);
|
||||
if (thesample > 32767)
|
||||
*(int16*)out_buffer = 32767;
|
||||
else if (thesample < -32767)
|
||||
*(int16*)out_buffer = -32767;
|
||||
else
|
||||
*(int16*)out_buffer = thesample;
|
||||
out_buffer += 2;
|
||||
}
|
||||
}
|
||||
} else {
|
||||
for (int sample = 0; sample < samples ; sample++) {
|
||||
for (int channel = 0; channel < fInfo.channels; channel++) {
|
||||
*((float*)out_buffer) = pcm[channel][sample];
|
||||
out_buffer += sizeof(float);
|
||||
}
|
||||
for (int sample = 0; sample < samples ; sample++) {
|
||||
for (int channel = 0; channel < fInfo.channels; channel++) {
|
||||
*((float*)out_buffer) = pcm[channel][sample];
|
||||
out_buffer += sizeof(float);
|
||||
}
|
||||
}
|
||||
out_bytes_needed -= samples * fFrameSize;
|
||||
|
@ -33,7 +33,6 @@ private:
|
||||
bigtime_t fStartTime;
|
||||
int fFrameSize;
|
||||
int fOutputBufferSize;
|
||||
bool fSoundplayLossage;
|
||||
};
|
||||
|
||||
|
||||
|
Loading…
Reference in New Issue
Block a user