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-rw-r--r--src/inputPlugins/_flac_common.c17
-rw-r--r--src/inputPlugins/_flac_common.h6
-rw-r--r--src/inputPlugins/aac_plugin.c34
-rw-r--r--src/inputPlugins/audiofile_plugin.c35
-rw-r--r--src/inputPlugins/flac_plugin.c43
-rw-r--r--src/inputPlugins/mod_plugin.c28
-rw-r--r--src/inputPlugins/mp3_plugin.c74
-rw-r--r--src/inputPlugins/mp4_plugin.c41
-rw-r--r--src/inputPlugins/mpc_plugin.c53
-rw-r--r--src/inputPlugins/oggflac_plugin.c31
-rw-r--r--src/inputPlugins/oggvorbis_plugin.c47
-rw-r--r--src/inputPlugins/wavpack_plugin.c57
12 files changed, 222 insertions, 244 deletions
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index 3b351d3a7..a26163303 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -36,15 +36,13 @@
#include <FLAC/format.h>
#include <FLAC/metadata.h>
-void init_FlacData(FlacData * data, OutputBuffer * cb,
- DecoderControl * dc, InputStream * inStream)
+void init_FlacData(FlacData * data, OutputBuffer * cb, InputStream * inStream)
{
data->chunk_length = 0;
data->time = 0;
data->position = 0;
data->bitRate = 0;
data->cb = cb;
- data->dc = dc;
data->inStream = inStream;
data->replayGainInfo = NULL;
data->tag = NULL;
@@ -165,16 +163,15 @@ MpdTag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
FlacData * data)
{
- DecoderControl *dc = data->dc;
const FLAC__StreamMetadata_StreamInfo *si = &(block->data.stream_info);
switch (block->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
- dc->audioFormat.bits = (mpd_sint8)si->bits_per_sample;
- dc->audioFormat.sampleRate = si->sample_rate;
- dc->audioFormat.channels = (mpd_sint8)si->channels;
- dc->totalTime = ((float)si->total_samples) / (si->sample_rate);
- getOutputAudioFormat(&(dc->audioFormat),
+ dc.audioFormat.bits = (mpd_sint8)si->bits_per_sample;
+ dc.audioFormat.sampleRate = si->sample_rate;
+ dc.audioFormat.channels = (mpd_sint8)si->channels;
+ dc.totalTime = ((float)si->total_samples) / (si->sample_rate);
+ getOutputAudioFormat(&(dc.audioFormat),
&(data->cb->audioFormat));
break;
case FLAC__METADATA_TYPE_VORBIS_COMMENT:
@@ -188,7 +185,7 @@ void flac_error_common_cb(const char *plugin,
const FLAC__StreamDecoderErrorStatus status,
FlacData * data)
{
- if (data->dc->stop)
+ if (dc.stop)
return;
switch (status) {
diff --git a/src/inputPlugins/_flac_common.h b/src/inputPlugins/_flac_common.h
index faeccf1ca..37b5fdaae 100644
--- a/src/inputPlugins/_flac_common.h
+++ b/src/inputPlugins/_flac_common.h
@@ -149,15 +149,13 @@ typedef struct {
unsigned int bitRate;
FLAC__uint64 position;
OutputBuffer *cb;
- DecoderControl *dc;
InputStream *inStream;
ReplayGainInfo *replayGainInfo;
MpdTag *tag;
} FlacData;
/* initializes a given FlacData struct */
-void init_FlacData(FlacData * data, OutputBuffer * cb,
- DecoderControl * dc, InputStream * inStream);
+void init_FlacData(FlacData * data, OutputBuffer * cb, InputStream * inStream);
void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
FlacData * data);
void flac_error_common_cb(const char *plugin,
@@ -171,7 +169,7 @@ MpdTag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
static inline int flacSendChunk(FlacData * data)
{
if (sendDataToOutputBuffer(data->cb, data->inStream,
- data->dc, 1, data->chunk,
+ 1, data->chunk,
data->chunk_length, data->time,
data->bitRate,
data->replayGainInfo) ==
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index 2962b57c6..aeda10492 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -282,7 +282,7 @@ static int getAacTotalTime(char *file)
return file_time;
}
-static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
+static int aac_decode(OutputBuffer * cb, char *path)
{
float file_time;
float totalTime;
@@ -339,9 +339,9 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
return -1;
}
- dc->audioFormat.bits = 16;
+ dc.audioFormat.bits = 16;
- dc->totalTime = totalTime;
+ dc.totalTime = totalTime;
file_time = 0.0;
@@ -372,12 +372,12 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
sampleRate = frameInfo.samplerate;
#endif
- if (dc->state != DECODE_STATE_DECODE) {
- dc->audioFormat.channels = frameInfo.channels;
- dc->audioFormat.sampleRate = sampleRate;
- getOutputAudioFormat(&(dc->audioFormat),
+ if (dc.state != DECODE_STATE_DECODE) {
+ dc.audioFormat.channels = frameInfo.channels;
+ dc.audioFormat.sampleRate = sampleRate;
+ getOutputAudioFormat(&(dc.audioFormat),
&(cb->audioFormat));
- dc->state = DECODE_STATE_DECODE;
+ dc.state = DECODE_STATE_DECODE;
}
advanceAacBuffer(&b, frameInfo.bytesconsumed);
@@ -395,14 +395,14 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
sampleBufferLen = sampleCount * 2;
- sendDataToOutputBuffer(cb, NULL, dc, 0, sampleBuffer,
+ sendDataToOutputBuffer(cb, NULL, 0, sampleBuffer,
sampleBufferLen, file_time,
bitRate, NULL);
- if (dc->seek) {
- dc->seekError = 1;
- dc->seek = 0;
+ if (dc.seek) {
+ dc.seekError = 1;
+ dc.seek = 0;
decoder_wakeup_player();
- } else if (dc->stop) {
+ } else if (dc.stop) {
eof = 1;
break;
}
@@ -414,12 +414,12 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
if (b.buffer)
free(b.buffer);
- if (dc->state != DECODE_STATE_DECODE)
+ if (dc.state != DECODE_STATE_DECODE)
return -1;
- if (dc->seek) {
- dc->seekError = 1;
- dc->seek = 0;
+ if (dc.seek) {
+ dc.seekError = 1;
+ dc.seek = 0;
decoder_wakeup_player();
}
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 33ea54df9..4510ba46a 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -45,7 +45,7 @@ static int getAudiofileTotalTime(char *file)
return total_time;
}
-static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
+static int audiofile_decode(OutputBuffer * cb, char *path)
{
int fs, frame_count;
AFfilehandle af_fp;
@@ -67,41 +67,41 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, 16);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
- dc->audioFormat.bits = (mpd_uint8)bits;
- dc->audioFormat.sampleRate =
+ dc.audioFormat.bits = (mpd_uint8)bits;
+ dc.audioFormat.sampleRate =
(unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
- dc->audioFormat.channels =
+ dc.audioFormat.channels =
(mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
- getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+ getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
- dc->totalTime =
- ((float)frame_count / (float)dc->audioFormat.sampleRate);
+ dc.totalTime =
+ ((float)frame_count / (float)dc.audioFormat.sampleRate);
- bitRate = (mpd_uint16)(st.st_size * 8.0 / dc->totalTime / 1000.0 + 0.5);
+ bitRate = (mpd_uint16)(st.st_size * 8.0 / dc.totalTime / 1000.0 + 0.5);
- if (dc->audioFormat.bits != 8 && dc->audioFormat.bits != 16) {
+ if (dc.audioFormat.bits != 8 && dc.audioFormat.bits != 16) {
ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
- path, dc->audioFormat.bits);
+ path, dc.audioFormat.bits);
afCloseFile(af_fp);
return -1;
}
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
- dc->state = DECODE_STATE_DECODE;
+ dc.state = DECODE_STATE_DECODE;
{
int ret, eof = 0, current = 0;
char chunk[CHUNK_SIZE];
while (!eof) {
- if (dc->seek) {
+ if (dc.seek) {
clearOutputBuffer(cb);
- current = dc->seekWhere *
- dc->audioFormat.sampleRate;
+ current = dc.seekWhere *
+ dc.audioFormat.sampleRate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
- dc->seek = 0;
+ dc.seek = 0;
decoder_wakeup_player();
}
@@ -114,15 +114,14 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
current += ret;
sendDataToOutputBuffer(cb,
NULL,
- dc,
1,
chunk,
ret * fs,
(float)current /
- (float)dc->audioFormat.
+ (float)dc.audioFormat.
sampleRate, bitRate,
NULL);
- if (dc->stop)
+ if (dc.stop)
break;
}
}
diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c
index b5c18af75..f171ee457 100644
--- a/src/inputPlugins/flac_plugin.c
+++ b/src/inputPlugins/flac_plugin.c
@@ -41,15 +41,14 @@ static flac_read_status flacRead(const flac_decoder * flacDec,
while (1) {
r = readFromInputStream(data->inStream, (void *)buf, 1, *bytes);
- if (r == 0 && !inputStreamAtEOF(data->inStream) &&
- !data->dc->stop)
+ if (r == 0 && !inputStreamAtEOF(data->inStream) && !dc.stop)
my_usleep(10000);
else
break;
}
*bytes = r;
- if (r == 0 && !data->dc->stop) {
+ if (r == 0 && !dc.stop) {
if (inputStreamAtEOF(data->inStream))
return flac_read_status_eof;
else
@@ -248,7 +247,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
FLAC__uint32 samples = frame->header.blocksize;
unsigned int c_samp;
const unsigned int num_channels = frame->header.channels;
- const unsigned int bytes_per_sample = (data->dc->audioFormat.bits / 8);
+ const unsigned int bytes_per_sample = (dc.audioFormat.bits / 8);
const unsigned int bytes_per_channel =
bytes_per_sample * frame->header.channels;
const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel;
@@ -256,7 +255,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
float timeChange;
FLAC__uint64 newPosition = 0;
- assert(data->dc->audioFormat.bits > 0);
+ assert(dc.audioFormat.bits > 0);
timeChange = ((float)samples) / frame->header.sample_rate;
data->time += timeChange;
@@ -292,7 +291,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
data->chunk_length = 0;
- if (data->dc->seek) {
+ if (dc.seek) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
@@ -382,7 +381,7 @@ static MpdTag *flacTagDup(char *file)
return ret;
}
-static int flac_decode_internal(OutputBuffer * cb, DecoderControl * dc,
+static int flac_decode_internal(OutputBuffer * cb,
InputStream * inStream, int is_ogg)
{
flac_decoder *flacDec;
@@ -391,7 +390,7 @@ static int flac_decode_internal(OutputBuffer * cb, DecoderControl * dc,
if (!(flacDec = flac_new()))
return -1;
- init_FlacData(&data, cb, dc, inStream);
+ init_FlacData(&data, cb, inStream);
#if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
if(!FLAC__stream_decoder_set_metadata_respond(flacDec, FLAC__METADATA_TYPE_VORBIS_COMMENT))
@@ -421,33 +420,33 @@ static int flac_decode_internal(OutputBuffer * cb, DecoderControl * dc,
}
}
- dc->state = DECODE_STATE_DECODE;
+ dc.state = DECODE_STATE_DECODE;
while (1) {
if (!flac_process_single(flacDec))
break;
if (flac_get_state(flacDec) == flac_decoder_eof)
break;
- if (dc->seek) {
- FLAC__uint64 sampleToSeek = dc->seekWhere *
- dc->audioFormat.sampleRate + 0.5;
+ if (dc.seek) {
+ FLAC__uint64 sampleToSeek = dc.seekWhere *
+ dc.audioFormat.sampleRate + 0.5;
if (flac_seek_absolute(flacDec, sampleToSeek)) {
clearOutputBuffer(cb);
data.time = ((float)sampleToSeek) /
- dc->audioFormat.sampleRate;
+ dc.audioFormat.sampleRate;
data.position = 0;
} else
- dc->seekError = 1;
- dc->seek = 0;
+ dc.seekError = 1;
+ dc.seek = 0;
decoder_wakeup_player();
}
}
- if (!dc->stop) {
+ if (!dc.stop) {
flacPrintErroredState(flac_get_state(flacDec));
flac_finish(flacDec);
}
/* send last little bit */
- if (data.chunk_length > 0 && !dc->stop) {
+ if (data.chunk_length > 0 && !dc.stop) {
flacSendChunk(&data);
flushOutputBuffer(data.cb);
}
@@ -466,10 +465,9 @@ fail:
return 0;
}
-static int flac_decode(OutputBuffer * cb, DecoderControl * dc,
- InputStream * inStream)
+static int flac_decode(OutputBuffer * cb, InputStream * inStream)
{
- return flac_decode_internal(cb, dc, inStream, 0);
+ return flac_decode_internal(cb, inStream, 0);
}
#if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
@@ -508,10 +506,9 @@ out:
return ret;
}
-static int oggflac_decode(OutputBuffer * cb, DecoderControl * dc,
- InputStream * inStream)
+static int oggflac_decode(OutputBuffer * cb, InputStream * inStream)
{
- return flac_decode_internal(cb, dc, inStream, 1);
+ return flac_decode_internal(cb, inStream, 1);
}
static unsigned int oggflac_try_decode(InputStream * inStream)
diff --git a/src/inputPlugins/mod_plugin.c b/src/inputPlugins/mod_plugin.c
index 25cedf04b..728f42d6f 100644
--- a/src/inputPlugins/mod_plugin.c
+++ b/src/inputPlugins/mod_plugin.c
@@ -163,7 +163,7 @@ static void mod_close(mod_Data * data)
free(data);
}
-static int mod_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
+static int mod_decode(OutputBuffer * cb, char *path)
{
mod_Data *data;
float total_time = 0.0;
@@ -179,25 +179,25 @@ static int mod_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
return -1;
}
- dc->totalTime = 0;
- dc->audioFormat.bits = 16;
- dc->audioFormat.sampleRate = 44100;
- dc->audioFormat.channels = 2;
- getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+ dc.totalTime = 0;
+ dc.audioFormat.bits = 16;
+ dc.audioFormat.sampleRate = 44100;
+ dc.audioFormat.channels = 2;
+ getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
secPerByte =
- 1.0 / ((dc->audioFormat.bits * dc->audioFormat.channels / 8.0) *
- (float)dc->audioFormat.sampleRate);
+ 1.0 / ((dc.audioFormat.bits * dc.audioFormat.channels / 8.0) *
+ (float)dc.audioFormat.sampleRate);
- dc->state = DECODE_STATE_DECODE;
+ dc.state = DECODE_STATE_DECODE;
while (1) {
- if (dc->seek) {
- dc->seekError = 1;
- dc->seek = 0;
+ if (dc.seek) {
+ dc.seekError = 1;
+ dc.seek = 0;
decoder_wakeup_player();
}
- if (dc->stop)
+ if (dc.stop)
break;
if (!Player_Active())
@@ -205,7 +205,7 @@ static int mod_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
ret = VC_WriteBytes(data->audio_buffer, MIKMOD_FRAME_SIZE);
total_time += ret * secPerByte;
- sendDataToOutputBuffer(cb, NULL, dc, 0,
+ sendDataToOutputBuffer(cb, NULL, 0,
(char *)data->audio_buffer, ret,
total_time, 0, NULL);
}
diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
index 875ec1482..ea33ad5ad 100644
--- a/src/inputPlugins/mp3_plugin.c
+++ b/src/inputPlugins/mp3_plugin.c
@@ -674,7 +674,7 @@ static int parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen)
return 1;
}
-static int decodeFirstFrame(mp3DecodeData * data, DecoderControl * dc,
+static int decodeFirstFrame(mp3DecodeData * data,
MpdTag ** tag, ReplayGainInfo ** replayGainInfo)
{
struct xing xing;
@@ -689,13 +689,13 @@ static int decodeFirstFrame(mp3DecodeData * data, DecoderControl * dc,
while (1) {
while ((ret = decodeNextFrameHeader(data, tag, replayGainInfo)) == DECODE_CONT &&
- (!dc || !dc->stop));
- if (ret == DECODE_BREAK || (dc && dc->stop)) return -1;
+ !dc.stop);
+ if (ret == DECODE_BREAK || dc.stop) return -1;
if (ret == DECODE_SKIP) continue;
while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
- (!dc || !dc->stop));
- if (ret == DECODE_BREAK || (dc && dc->stop)) return -1;
+ !dc.stop);
+ if (ret == DECODE_BREAK || dc.stop) return -1;
if (ret == DECODE_OK) break;
}
@@ -787,7 +787,7 @@ static int getMp3TotalTime(char *file)
if (openInputStream(&inStream, file) < 0)
return -1;
initMp3DecodeData(&data, &inStream);
- if (decodeFirstFrame(&data, NULL, NULL, NULL) < 0)
+ if (decodeFirstFrame(&data, NULL, NULL) < 0)
ret = -1;
else
ret = data.totalTime + 0.5;
@@ -798,12 +798,12 @@ static int getMp3TotalTime(char *file)
}
static int openMp3FromInputStream(InputStream * inStream, mp3DecodeData * data,
- DecoderControl * dc, MpdTag ** tag,
+ MpdTag ** tag,
ReplayGainInfo ** replayGainInfo)
{
initMp3DecodeData(data, inStream);
*tag = NULL;
- if (decodeFirstFrame(data, dc, tag, replayGainInfo) < 0) {
+ if (decodeFirstFrame(data, tag, replayGainInfo) < 0) {
mp3DecodeDataFinalize(data);
if (tag && *tag)
freeMpdTag(*tag);
@@ -813,7 +813,7 @@ static int openMp3FromInputStream(InputStream * inStream, mp3DecodeData * data,
return 0;
}
-static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
+static int mp3Read(mp3DecodeData * data, OutputBuffer * cb,
ReplayGainInfo ** replayGainInfo)
{
int samplesPerFrame;
@@ -852,11 +852,11 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
data->muteFrame = 0;
break;
case MUTEFRAME_SEEK:
- if (dc->seekWhere <= data->elapsedTime) {
+ if (dc.seekWhere <= data->elapsedTime) {
data->outputPtr = data->outputBuffer;
clearOutputBuffer(cb);
data->muteFrame = 0;
- dc->seek = 0;
+ dc.seek = 0;
decoder_wakeup_player();
}
break;
@@ -931,7 +931,6 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
if (data->outputPtr >= data->outputBufferEnd) {
ret = sendDataToOutputBuffer(cb,
data->inStream,
- dc,
data->inStream->seekable,
data->outputBuffer,
data->outputPtr - data->outputBuffer,
@@ -952,10 +951,10 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
data->decodedFirstFrame = 1;
- if (dc->seek && data->inStream->seekable) {
+ if (dc.seek && data->inStream->seekable) {
long j = 0;
data->muteFrame = MUTEFRAME_SEEK;
- while (j < data->highestFrame && dc->seekWhere >
+ while (j < data->highestFrame && dc.seekWhere >
((float)mad_timer_count(data->times[j],
MAD_UNITS_MILLISECONDS))
/ 1000) {
@@ -969,14 +968,14 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
clearOutputBuffer(cb);
data->currentFrame = j;
} else
- dc->seekError = 1;
+ dc.seekError = 1;
data->muteFrame = 0;
- dc->seek = 0;
+ dc.seek = 0;
decoder_wakeup_player();
}
- } else if (dc->seek && !data->inStream->seekable) {
- dc->seek = 0;
- dc->seekError = 1;
+ } else if (dc.seek && !data->inStream->seekable) {
+ dc.seek = 0;
+ dc.seekError = 1;
decoder_wakeup_player();
}
}
@@ -986,22 +985,22 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
while ((ret =
decodeNextFrameHeader(data, NULL,
replayGainInfo)) == DECODE_CONT
- && !dc->stop) ;
- if (ret == DECODE_BREAK || dc->stop || dc->seek)
+ && !dc.stop) ;
+ if (ret == DECODE_BREAK || dc.stop || dc.seek)
break;
else if (ret == DECODE_SKIP)
skip = 1;
if (!data->muteFrame) {
while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
- !dc->stop && !dc->seek) ;
- if (ret == DECODE_BREAK || dc->stop || dc->seek)
+ !dc.stop && !dc.seek) ;
+ if (ret == DECODE_BREAK || dc.stop || dc.seek)
break;
}
if (!skip && ret == DECODE_OK)
break;
}
- if (dc->stop)
+ if (dc.stop)
return DECODE_BREAK;
return ret;
@@ -1015,16 +1014,15 @@ static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data,
af->channels = MAD_NCHANNELS(&(data->frame).header);
}
-static int mp3_decode(OutputBuffer * cb, DecoderControl * dc,
- InputStream * inStream)
+static int mp3_decode(OutputBuffer * cb, InputStream * inStream)
{
mp3DecodeData data;
MpdTag *tag = NULL;
ReplayGainInfo *replayGainInfo = NULL;
- if (openMp3FromInputStream(inStream, &data, dc, &tag, &replayGainInfo) <
+ if (openMp3FromInputStream(inStream, &data, &tag, &replayGainInfo) <
0) {
- if (!dc->stop) {
+ if (!dc.stop) {
ERROR
("Input does not appear to be a mp3 bit stream.\n");
return -1;
@@ -1032,10 +1030,10 @@ static int mp3_decode(OutputBuffer * cb, DecoderControl * dc,
return 0;
}
- initAudioFormatFromMp3DecodeData(&data, &(dc->audioFormat));
- getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+ initAudioFormatFromMp3DecodeData(&data, &(dc.audioFormat));
+ getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
- dc->totalTime = data.totalTime;
+ dc.totalTime = data.totalTime;
if (inStream->metaTitle) {
if (tag)
@@ -1062,12 +1060,12 @@ static int mp3_decode(OutputBuffer * cb, DecoderControl * dc,
freeMpdTag(tag);
}
- dc->state = DECODE_STATE_DECODE;
+ dc.state = DECODE_STATE_DECODE;
- while (mp3Read(&data, cb, dc, &replayGainInfo) != DECODE_BREAK) ;
- /* send last little bit if not dc->stop */
- if (!dc->stop && data.outputPtr != data.outputBuffer && data.flush) {
- sendDataToOutputBuffer(cb, NULL, dc,
+ while (mp3Read(&data, cb, &replayGainInfo) != DECODE_BREAK) ;
+ /* send last little bit if not dc.stop */
+ if (!dc.stop && data.outputPtr != data.outputBuffer && data.flush) {
+ sendDataToOutputBuffer(cb, NULL,
data.inStream->seekable,
data.outputBuffer,
data.outputPtr - data.outputBuffer,
@@ -1078,9 +1076,9 @@ static int mp3_decode(OutputBuffer * cb, DecoderControl * dc,
if (replayGainInfo)
freeReplayGainInfo(replayGainInfo);
- if (dc->seek && data.muteFrame == MUTEFRAME_SEEK) {
+ if (dc.seek && data.muteFrame == MUTEFRAME_SEEK) {
clearOutputBuffer(cb);
- dc->seek = 0;
+ dc.seek = 0;
decoder_wakeup_player();
}
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index 1e755b95c..fb8c71020 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -84,8 +84,7 @@ static uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position)
return seekInputStream((InputStream *) inStream, position, SEEK_SET);
}
-static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
- InputStream * inStream)
+static int mp4_decode(OutputBuffer * cb, InputStream * inStream)
{
mp4ff_t *mp4fh;
mp4ff_callback_t *mp4cb;
@@ -146,7 +145,7 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
#endif
faacDecSetConfiguration(decoder, config);
- dc->audioFormat.bits = 16;
+ dc.audioFormat.bits = 16;
mp4Buffer = NULL;
mp4BufferSize = 0;
@@ -161,8 +160,8 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
return -1;
}
- dc->audioFormat.sampleRate = sampleRate;
- dc->audioFormat.channels = channels;
+ dc.audioFormat.sampleRate = sampleRate;
+ dc.audioFormat.channels = channels;
file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
scale = mp4ff_time_scale(mp4fh, track);
@@ -176,7 +175,7 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
free(mp4cb);
return -1;
}
- dc->totalTime = ((float)file_time) / scale;
+ dc.totalTime = ((float)file_time) / scale;
numSamples = mp4ff_num_samples(mp4fh, track);
@@ -185,13 +184,13 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
seekTable = xmalloc(sizeof(float) * numSamples);
for (sampleId = 0; sampleId < numSamples && !eof; sampleId++) {
- if (dc->seek)
+ if (dc.seek)
seeking = 1;
if (seeking && seekTableEnd > 1 &&
- seekTable[seekTableEnd] >= dc->seekWhere) {
+ seekTable[seekTableEnd] >= dc.seekWhere) {
int i = 2;
- while (seekTable[i] < dc->seekWhere)
+ while (seekTable[i] < dc.seekWhere)
i++;
sampleId = i - 1;
file_time = seekTable[sampleId];
@@ -213,14 +212,14 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
dur -= offset;
file_time += ((float)dur) / scale;
- if (seeking && file_time > dc->seekWhere)
+ if (seeking && file_time > dc.seekWhere)
seekPositionFound = 1;
if (seeking && seekPositionFound) {
seekPositionFound = 0;
clearOutputBuffer(cb);
seeking = 0;
- dc->seek = 0;
+ dc.seek = 0;
decoder_wakeup_player();
}
@@ -248,16 +247,16 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
break;
}
- if (dc->state != DECODE_STATE_DECODE) {
+ if (dc.state != DECODE_STATE_DECODE) {
channels = frameInfo.channels;
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
scale = frameInfo.samplerate;
#endif
- dc->audioFormat.sampleRate = scale;
- dc->audioFormat.channels = frameInfo.channels;
- getOutputAudioFormat(&(dc->audioFormat),
+ dc.audioFormat.sampleRate = scale;
+ dc.audioFormat.channels = frameInfo.channels;
+ getOutputAudioFormat(&(dc.audioFormat),
&(cb->audioFormat));
- dc->state = DECODE_STATE_DECODE;
+ dc.state = DECODE_STATE_DECODE;
}
if (channels * (unsigned long)(dur + offset) > frameInfo.samples) {
@@ -278,10 +277,10 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
sampleBuffer += offset * channels * 2;
- sendDataToOutputBuffer(cb, inStream, dc, 1, sampleBuffer,
+ sendDataToOutputBuffer(cb, inStream, 1, sampleBuffer,
sampleBufferLen, file_time,
bitRate, NULL);
- if (dc->stop) {
+ if (dc.stop) {
eof = 1;
break;
}
@@ -292,12 +291,12 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
mp4ff_close(mp4fh);
free(mp4cb);
- if (dc->state != DECODE_STATE_DECODE)
+ if (dc.state != DECODE_STATE_DECODE)
return -1;
- if (dc->seek && seeking) {
+ if (dc.seek && seeking) {
clearOutputBuffer(cb);
- dc->seek = 0;
+ dc.seek = 0;
decoder_wakeup_player();
}
flushOutputBuffer(cb);
diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
index 9b6e862ff..867965688 100644
--- a/src/inputPlugins/mpc_plugin.c
+++ b/src/inputPlugins/mpc_plugin.c
@@ -33,7 +33,6 @@
typedef struct _MpcCallbackData {
InputStream *inStream;
- DecoderControl *dc;
} MpcCallbackData;
static mpc_int32_t mpc_read_cb(void *vdata, void *ptr, mpc_int32_t size)
@@ -43,10 +42,9 @@ static mpc_int32_t mpc_read_cb(void *vdata, void *ptr, mpc_int32_t size)
while (1) {
ret = readFromInputStream(data->inStream, ptr, 1, size);
- if (ret == 0 && !inputStreamAtEOF(data->inStream) &&
- (data->dc && !data->dc->stop)) {
+ if (ret == 0 && !inputStreamAtEOF(data->inStream) && !dc.stop)
my_usleep(10000);
- } else
+ else
break;
}
@@ -113,8 +111,7 @@ static inline mpd_sint16 convertSample(MPC_SAMPLE_FORMAT sample)
return val;
}
-static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
- InputStream * inStream)
+static int mpc_decode(OutputBuffer * cb, InputStream * inStream)
{
mpc_decoder decoder;
mpc_reader reader;
@@ -139,7 +136,6 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
ReplayGainInfo *replayGainInfo = NULL;
data.inStream = inStream;
- data.dc = dc;
reader.read = mpc_read_cb;
reader.seek = mpc_seek_cb;
@@ -151,7 +147,7 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
mpc_streaminfo_init(&info);
if ((ret = mpc_streaminfo_read(&info, &reader)) != ERROR_CODE_OK) {
- if (!dc->stop) {
+ if (!dc.stop) {
ERROR("Not a valid musepack stream\n");
return -1;
}
@@ -161,20 +157,20 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
mpc_decoder_setup(&decoder, &reader);
if (!mpc_decoder_initialize(&decoder, &info)) {
- if (!dc->stop) {
+ if (!dc.stop) {
ERROR("Not a valid musepack stream\n");
return -1;
}
return 0;
}
- dc->totalTime = mpc_streaminfo_get_length(&info);
+ dc.totalTime = mpc_streaminfo_get_length(&info);
- dc->audioFormat.bits = 16;
- dc->audioFormat.channels = info.channels;
- dc->audioFormat.sampleRate = info.sample_freq;
+ dc.audioFormat.bits = 16;
+ dc.audioFormat.channels = info.channels;
+ dc.audioFormat.sampleRate = info.sample_freq;
- getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+ getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
replayGainInfo = newReplayGainInfo();
replayGainInfo->albumGain = info.gain_album * 0.01;
@@ -182,18 +178,18 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
replayGainInfo->trackGain = info.gain_title * 0.01;
replayGainInfo->trackPeak = info.peak_title / 32767.0;
- dc->state = DECODE_STATE_DECODE;
+ dc.state = DECODE_STATE_DECODE;
while (!eof) {
- if (dc->seek) {
- samplePos = dc->seekWhere * dc->audioFormat.sampleRate;
+ if (dc.seek) {
+ samplePos = dc.seekWhere * dc.audioFormat.sampleRate;
if (mpc_decoder_seek_sample(&decoder, samplePos)) {
clearOutputBuffer(cb);
s16 = (mpd_sint16 *) chunk;
chunkpos = 0;
} else
- dc->seekError = 1;
- dc->seek = 0;
+ dc.seekError = 1;
+ dc.seek = 0;
decoder_wakeup_player();
}
@@ -202,7 +198,7 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
ret = mpc_decoder_decode(&decoder, sample_buffer,
&vbrUpdateAcc, &vbrUpdateBits);
- if (ret <= 0 || dc->stop) {
+ if (ret <= 0 || dc.stop) {
eof = 1;
break;
}
@@ -220,12 +216,12 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
if (chunkpos >= MPC_CHUNK_SIZE) {
total_time = ((float)samplePos) /
- dc->audioFormat.sampleRate;
+ dc.audioFormat.sampleRate;
bitRate = vbrUpdateBits *
- dc->audioFormat.sampleRate / 1152 / 1000;
+ dc.audioFormat.sampleRate / 1152 / 1000;
- sendDataToOutputBuffer(cb, inStream, dc,
+ sendDataToOutputBuffer(cb, inStream,
inStream->seekable,
chunk, chunkpos,
total_time,
@@ -233,7 +229,7 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
chunkpos = 0;
s16 = (mpd_sint16 *) chunk;
- if (dc->stop) {
+ if (dc.stop) {
eof = 1;
break;
}
@@ -241,13 +237,13 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
}
}
- if (!dc->stop && chunkpos > 0) {
- total_time = ((float)samplePos) / dc->audioFormat.sampleRate;
+ if (!dc.stop && chunkpos > 0) {
+ total_time = ((float)samplePos) / dc.audioFormat.sampleRate;
bitRate =
- vbrUpdateBits * dc->audioFormat.sampleRate / 1152 / 1000;
+ vbrUpdateBits * dc.audioFormat.sampleRate / 1152 / 1000;
- sendDataToOutputBuffer(cb, NULL, dc, inStream->seekable,
+ sendDataToOutputBuffer(cb, NULL, inStream->seekable,
chunk, chunkpos, total_time, bitRate,
replayGainInfo);
}
@@ -269,7 +265,6 @@ static float mpcGetTime(char *file)
MpcCallbackData data;
data.inStream = &inStream;
- data.dc = NULL;
reader.read = mpc_read_cb;
reader.seek = mpc_seek_cb;
diff --git a/src/inputPlugins/oggflac_plugin.c b/src/inputPlugins/oggflac_plugin.c
index 6638362b6..070404e26 100644
--- a/src/inputPlugins/oggflac_plugin.c
+++ b/src/inputPlugins/oggflac_plugin.c
@@ -56,14 +56,14 @@ static OggFLAC__SeekableStreamDecoderReadStatus of_read_cb(const
while (1) {
r = readFromInputStream(data->inStream, (void *)buf, 1, *bytes);
if (r == 0 && !inputStreamAtEOF(data->inStream) &&
- !data->dc->stop)
+ !dc.stop)
my_usleep(10000);
else
break;
}
*bytes = r;
- if (r == 0 && !inputStreamAtEOF(data->inStream) && !data->dc->stop)
+ if (r == 0 && !inputStreamAtEOF(data->inStream) && !dc.stop)
return OggFLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR;
return OggFLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK;
@@ -193,14 +193,14 @@ static FLAC__StreamDecoderWriteStatus oggflacWrite(const
c_chan++) {
u16 = buf[c_chan][c_samp];
uc = (unsigned char *)&u16;
- for (i = 0; i < (data->dc->audioFormat.bits / 8); i++) {
+ for (i = 0; i < (dc.audioFormat.bits / 8); i++) {
if (data->chunk_length >= FLAC_CHUNK_SIZE) {
if (flacSendChunk(data) < 0) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
data->chunk_length = 0;
- if (data->dc->seek) {
+ if (dc.seek) {
return
FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
@@ -336,21 +336,20 @@ static unsigned int oggflac_try_decode(InputStream * inStream)
return (ogg_stream_type_detect(inStream) == FLAC) ? 1 : 0;
}
-static int oggflac_decode(OutputBuffer * cb, DecoderControl * dc,
- InputStream * inStream)
+static int oggflac_decode(OutputBuffer * cb, InputStream * inStream)
{
OggFLAC__SeekableStreamDecoder *decoder = NULL;
FlacData data;
int ret = 0;
- init_FlacData(&data, cb, dc, inStream);
+ init_FlacData(&data, cb, inStream);
if (!(decoder = full_decoder_init_and_read_metadata(&data, 0))) {
ret = -1;
goto fail;
}
- dc->state = DECODE_STATE_DECODE;
+ dc.state = DECODE_STATE_DECODE;
while (1) {
OggFLAC__seekable_stream_decoder_process_single(decoder);
@@ -358,29 +357,29 @@ static int oggflac_decode(OutputBuffer * cb, DecoderControl * dc,
OggFLAC__SEEKABLE_STREAM_DECODER_OK) {
break;
}
- if (dc->seek) {
- FLAC__uint64 sampleToSeek = dc->seekWhere *
- dc->audioFormat.sampleRate + 0.5;
+ if (dc.seek) {
+ FLAC__uint64 sampleToSeek = dc.seekWhere *
+ dc.audioFormat.sampleRate + 0.5;
if (OggFLAC__seekable_stream_decoder_seek_absolute
(decoder, sampleToSeek)) {
clearOutputBuffer(cb);
data.time = ((float)sampleToSeek) /
- dc->audioFormat.sampleRate;
+ dc.audioFormat.sampleRate;
data.position = 0;
} else
- dc->seekError = 1;
- dc->seek = 0;
+ dc.seekError = 1;
+ dc.seek = 0;
decoder_wakeup_player();
}
}
- if (!dc->stop) {
+ if (!dc.stop) {
oggflacPrintErroredState
(OggFLAC__seekable_stream_decoder_get_state(decoder));
OggFLAC__seekable_stream_decoder_finish(decoder);
}
/* send last little bit */
- if (data.chunk_length > 0 && !dc->stop) {
+ if (data.chunk_length > 0 && !dc.stop) {
flacSendChunk(&data);
flushOutputBuffer(data.cb);
}
diff --git a/src/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c
index c79e7893c..afcef3e08 100644
--- a/src/inputPlugins/oggvorbis_plugin.c
+++ b/src/inputPlugins/oggvorbis_plugin.c
@@ -56,7 +56,6 @@
typedef struct _OggCallbackData {
InputStream *inStream;
- DecoderControl *dc;
} OggCallbackData;
static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata)
@@ -67,7 +66,7 @@ static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata)
while (1) {
ret = readFromInputStream(data->inStream, ptr, size, nmemb);
if (ret == 0 && !inputStreamAtEOF(data->inStream) &&
- !data->dc->stop) {
+ !dc.stop) {
my_usleep(10000);
} else
break;
@@ -81,7 +80,7 @@ static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata)
static int ogg_seek_cb(void *vdata, ogg_int64_t offset, int whence)
{
const OggCallbackData *data = (const OggCallbackData *) vdata;
- if(data->dc->stop)
+ if (dc.stop)
return -1;
return seekInputStream(data->inStream, offset, whence);
}
@@ -217,8 +216,7 @@ static void putOggCommentsIntoOutputBuffer(OutputBuffer * cb, char *streamName,
}
/* public */
-static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
- InputStream * inStream)
+static int oggvorbis_decode(OutputBuffer * cb, InputStream * inStream)
{
OggVorbis_File vf;
ov_callbacks callbacks;
@@ -236,14 +234,13 @@ static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
const char *errorStr;
data.inStream = inStream;
- data.dc = dc;
callbacks.read_func = ogg_read_cb;
callbacks.seek_func = ogg_seek_cb;
callbacks.close_func = ogg_close_cb;
callbacks.tell_func = ogg_tell_cb;
if ((ret = ov_open_callbacks(&data, &vf, NULL, 0, callbacks)) < 0) {
- if (!dc->stop) {
+ if (!dc.stop) {
switch (ret) {
case OV_EREAD:
errorStr = "read error";
@@ -270,19 +267,19 @@ static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
}
return 0;
}
- dc->totalTime = ov_time_total(&vf, -1);
- if (dc->totalTime < 0)
- dc->totalTime = 0;
- dc->audioFormat.bits = 16;
+ dc.totalTime = ov_time_total(&vf, -1);
+ if (dc.totalTime < 0)
+ dc.totalTime = 0;
+ dc.audioFormat.bits = 16;
while (1) {
- if (dc->seek) {
- if (0 == ov_time_seek_page(&vf, dc->seekWhere)) {
+ if (dc.seek) {
+ if (0 == ov_time_seek_page(&vf, dc.seekWhere)) {
clearOutputBuffer(cb);
chunkpos = 0;
} else
- dc->seekError = 1;
- dc->seek = 0;
+ dc.seekError = 1;
+ dc.seek = 0;
decoder_wakeup_player();
}
ret = ov_read(&vf, chunk + chunkpos,
@@ -291,12 +288,12 @@ static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
if (current_section != prev_section) {
/*printf("new song!\n"); */
vorbis_info *vi = ov_info(&vf, -1);
- dc->audioFormat.channels = vi->channels;
- dc->audioFormat.sampleRate = vi->rate;
- if (dc->state == DECODE_STATE_START) {
- getOutputAudioFormat(&(dc->audioFormat),
+ dc.audioFormat.channels = vi->channels;
+ dc.audioFormat.sampleRate = vi->rate;
+ if (dc.state == DECODE_STATE_START) {
+ getOutputAudioFormat(&(dc.audioFormat),
&(cb->audioFormat));
- dc->state = DECODE_STATE_DECODE;
+ dc.state = DECODE_STATE_DECODE;
}
comments = ov_comment(&vf, -1)->user_comments;
putOggCommentsIntoOutputBuffer(cb, inStream->metaName,
@@ -319,20 +316,20 @@ static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
if ((test = ov_bitrate_instant(&vf)) > 0) {
bitRate = test / 1000;
}
- sendDataToOutputBuffer(cb, inStream, dc,
+ sendDataToOutputBuffer(cb, inStream,
inStream->seekable,
chunk, chunkpos,
ov_pcm_tell(&vf) /
- dc->audioFormat.sampleRate,
+ dc.audioFormat.sampleRate,
bitRate, replayGainInfo);
chunkpos = 0;
- if (dc->stop)
+ if (dc.stop)
break;
}
}
- if (!dc->stop && chunkpos > 0) {
- sendDataToOutputBuffer(cb, NULL, dc, inStream->seekable,
+ if (!dc.stop && chunkpos > 0) {
+ sendDataToOutputBuffer(cb, NULL, inStream->seekable,
chunk, chunkpos,
ov_time_tell(&vf), bitRate,
replayGainInfo);
diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
index c46cdeb75..bae5f6acb 100644
--- a/src/inputPlugins/wavpack_plugin.c
+++ b/src/inputPlugins/wavpack_plugin.c
@@ -128,7 +128,7 @@ static void format_samples_float(int Bps, void *buffer, uint32_t samcnt)
* This does the main decoding thing.
* Requires an already opened WavpackContext.
*/
-static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
+static void wavpack_decode(OutputBuffer *cb,
WavpackContext *wpc, int canseek,
ReplayGainInfo *replayGainInfo)
{
@@ -140,12 +140,12 @@ static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
int position, outsamplesize;
int Bps;
- dc->audioFormat.sampleRate = WavpackGetSampleRate(wpc);
- dc->audioFormat.channels = WavpackGetReducedChannels(wpc);
- dc->audioFormat.bits = WavpackGetBitsPerSample(wpc);
+ dc.audioFormat.sampleRate = WavpackGetSampleRate(wpc);
+ dc.audioFormat.channels = WavpackGetReducedChannels(wpc);
+ dc.audioFormat.bits = WavpackGetBitsPerSample(wpc);
- if (dc->audioFormat.bits > 16)
- dc->audioFormat.bits = 16;
+ if (dc.audioFormat.bits > 16)
+ dc.audioFormat.bits = 16;
if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
format_samples = format_samples_float;
@@ -163,40 +163,40 @@ static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
outsamplesize = Bps;
if (outsamplesize > 2)
outsamplesize = 2;
- outsamplesize *= dc->audioFormat.channels;
+ outsamplesize *= dc.audioFormat.channels;
- samplesreq = sizeof(chunk) / (4 * dc->audioFormat.channels);
+ samplesreq = sizeof(chunk) / (4 * dc.audioFormat.channels);
- getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+ getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
- dc->totalTime = (float)allsamples / dc->audioFormat.sampleRate;
- dc->state = DECODE_STATE_DECODE;
- dc->seekable = canseek;
+ dc.totalTime = (float)allsamples / dc.audioFormat.sampleRate;
+ dc.state = DECODE_STATE_DECODE;
+ dc.seekable = canseek;
position = 0;
do {
- if (dc->seek) {
+ if (dc.seek) {
if (canseek) {
int where;
clearOutputBuffer(cb);
- where = dc->seekWhere *
- dc->audioFormat.sampleRate;
+ where = dc.seekWhere *
+ dc.audioFormat.sampleRate;
if (WavpackSeekSample(wpc, where))
position = where;
else
- dc->seekError = 1;
+ dc.seekError = 1;
} else {
- dc->seekError = 1;
+ dc.seekError = 1;
}
- dc->seek = 0;
+ dc.seek = 0;
decoder_wakeup_player();
}
- if (dc->stop)
+ if (dc.stop)
break;
samplesgot = WavpackUnpackSamples(wpc,
@@ -206,12 +206,12 @@ static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
1000 + 0.5);
position += samplesgot;
file_time = (float)position /
- dc->audioFormat.sampleRate;
+ dc.audioFormat.sampleRate;
format_samples(Bps, chunk,
- samplesgot * dc->audioFormat.channels);
+ samplesgot * dc.audioFormat.channels);
- sendDataToOutputBuffer(cb, NULL, dc, 0, chunk,
+ sendDataToOutputBuffer(cb, NULL, 0, chunk,
samplesgot * outsamplesize,
file_time, bitrate,
replayGainInfo);
@@ -442,8 +442,7 @@ static unsigned int wavpack_trydecode(InputStream *is)
/*
* Decodes a stream.
*/
-static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
- InputStream *is)
+static int wavpack_streamdecode(OutputBuffer *cb, InputStream *is)
{
char error[ERRORLEN];
WavpackContext *wpc;
@@ -462,7 +461,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
err = 1;
/*
- * As we use dc->utf8url, this function will be bad for
+ * As we use dc.utf8url, this function will be bad for
* single files. utf8url is not absolute file path :/
*/
utf8url = get_song_url(tmp, pc.current_song);
@@ -507,7 +506,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
break;
}
- if (dc->stop) {
+ if (dc.stop) {
break;
}
@@ -542,7 +541,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
return -1;
}
- wavpack_decode(cb, dc, wpc, canseek, NULL);
+ wavpack_decode(cb, wpc, canseek, NULL);
WavpackCloseFile(wpc);
if (wvc_url != NULL) {
@@ -557,7 +556,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
/*
* Decodes a file.
*/
-static int wavpack_filedecode(OutputBuffer *cb, DecoderControl *dc, char *fname)
+static int wavpack_filedecode(OutputBuffer *cb, char *fname)
{
char error[ERRORLEN];
WavpackContext *wpc;
@@ -573,7 +572,7 @@ static int wavpack_filedecode(OutputBuffer *cb, DecoderControl *dc, char *fname)
replayGainInfo = wavpack_replaygain(wpc);
- wavpack_decode(cb, dc, wpc, 1, replayGainInfo);
+ wavpack_decode(cb, wpc, 1, replayGainInfo);
if (replayGainInfo)
freeReplayGainInfo(replayGainInfo);
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-08 13:06:16 +0000