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:
beveloper 2004-02-29 17:58:32 +00:00
parent 34cee003fb
commit fb3053bc25
2 changed files with 4 additions and 27 deletions

View File

@ -4,7 +4,6 @@
#include <Locker.h> #include <Locker.h>
#include <MediaFormats.h> #include <MediaFormats.h>
#include <MediaRoster.h> #include <MediaRoster.h>
#include <Roster.h>
#include <vector> #include <vector>
#include "vorbisCodecPlugin.h" #include "vorbisCodecPlugin.h"
#include "OggVorbisFormats.h" #include "OggVorbisFormats.h"
@ -49,9 +48,6 @@ VorbisDecoder::VorbisDecoder()
fStartTime = 0; fStartTime = 0;
fFrameSize = 0; fFrameSize = 0;
fOutputBufferSize = 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 // Be R5 behavior seems to be that we can never fail. If we
// don't support the requested format, just return one we do. // don't support the requested format, just return one we do.
media_format format = vorbis_decoded_media_format(); 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.frame_rate = (float)fInfo.rate;
format.u.raw_audio.channel_count = fInfo.channels; format.u.raw_audio.channel_count = fInfo.channels;
format.u.raw_audio.channel_mask = B_CHANNEL_LEFT | (fInfo.channels != 1 ? B_CHANNEL_RIGHT : 0); format.u.raw_audio.channel_mask = B_CHANNEL_LEFT | (fInfo.channels != 1 ? B_CHANNEL_RIGHT : 0);
@ -212,27 +205,12 @@ VorbisDecoder::Decode(void *buffer, int64 *frameCount,
// reduce samples to the amount of samples we will actually consume // reduce samples to the amount of samples we will actually consume
samples = min_c(samples,out_bytes_needed/fFrameSize); samples = min_c(samples,out_bytes_needed/fFrameSize);
total_samples += samples; 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 sample = 0; sample < samples ; sample++) {
for (int channel = 0; channel < fInfo.channels; channel++) { for (int channel = 0; channel < fInfo.channels; channel++) {
*((float*)out_buffer) = pcm[channel][sample]; *((float*)out_buffer) = pcm[channel][sample];
out_buffer += sizeof(float); out_buffer += sizeof(float);
} }
} }
}
out_bytes_needed -= samples * fFrameSize; out_bytes_needed -= samples * fFrameSize;
// report back how many samples we consumed // report back how many samples we consumed
vorbis_synthesis_read(&fDspState,samples); vorbis_synthesis_read(&fDspState,samples);

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@ -33,7 +33,6 @@ private:
bigtime_t fStartTime; bigtime_t fStartTime;
int fFrameSize; int fFrameSize;
int fOutputBufferSize; int fOutputBufferSize;
bool fSoundplayLossage;
}; };