diff options
Diffstat (limited to 'src/inputPlugins')
-rw-r--r-- | src/inputPlugins/_flac_common.c | 6 | ||||
-rw-r--r-- | src/inputPlugins/_flac_common.h | 5 | ||||
-rw-r--r-- | src/inputPlugins/aac_plugin.c | 8 | ||||
-rw-r--r-- | src/inputPlugins/audiofile_plugin.c | 11 | ||||
-rw-r--r-- | src/inputPlugins/flac_plugin.c | 17 | ||||
-rw-r--r-- | src/inputPlugins/mod_plugin.c | 8 | ||||
-rw-r--r-- | src/inputPlugins/mp3_plugin.c | 22 | ||||
-rw-r--r-- | src/inputPlugins/mp4_plugin.c | 31 | ||||
-rw-r--r-- | src/inputPlugins/mpc_plugin.c | 12 | ||||
-rw-r--r-- | src/inputPlugins/oggflac_plugin.c | 8 | ||||
-rw-r--r-- | src/inputPlugins/oggvorbis_plugin.c | 16 | ||||
-rw-r--r-- | src/inputPlugins/wavpack_plugin.c | 19 |
12 files changed, 77 insertions, 86 deletions
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c index a26163303..80b1210d1 100644 --- a/src/inputPlugins/_flac_common.c +++ b/src/inputPlugins/_flac_common.c @@ -36,13 +36,12 @@ #include <FLAC/format.h> #include <FLAC/metadata.h> -void init_FlacData(FlacData * data, OutputBuffer * cb, InputStream * inStream) +void init_FlacData(FlacData * data, InputStream * inStream) { data->chunk_length = 0; data->time = 0; data->position = 0; data->bitRate = 0; - data->cb = cb; data->inStream = inStream; data->replayGainInfo = NULL; data->tag = NULL; @@ -171,8 +170,7 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block, 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)); + getOutputAudioFormat(&(dc.audioFormat), &(cb.audioFormat)); break; case FLAC__METADATA_TYPE_VORBIS_COMMENT: flacParseReplayGain(block, data); diff --git a/src/inputPlugins/_flac_common.h b/src/inputPlugins/_flac_common.h index 37b5fdaae..18e51d587 100644 --- a/src/inputPlugins/_flac_common.h +++ b/src/inputPlugins/_flac_common.h @@ -148,14 +148,13 @@ typedef struct { float time; unsigned int bitRate; FLAC__uint64 position; - OutputBuffer *cb; InputStream *inStream; ReplayGainInfo *replayGainInfo; MpdTag *tag; } FlacData; /* initializes a given FlacData struct */ -void init_FlacData(FlacData * data, OutputBuffer * cb, InputStream * inStream); +void init_FlacData(FlacData * data, InputStream * inStream); void flac_metadata_common_cb(const FLAC__StreamMetadata * block, FlacData * data); void flac_error_common_cb(const char *plugin, @@ -168,7 +167,7 @@ MpdTag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block, /* keep this inlined, this is just macro but prettier :) */ static inline int flacSendChunk(FlacData * data) { - if (sendDataToOutputBuffer(data->cb, data->inStream, + if (sendDataToOutputBuffer(data->inStream, 1, data->chunk, data->chunk_length, data->time, data->bitRate, diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c index aeda10492..ebf402be1 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, char *path) +static int aac_decode(char *path) { float file_time; float totalTime; @@ -376,7 +376,7 @@ static int aac_decode(OutputBuffer * cb, char *path) dc.audioFormat.channels = frameInfo.channels; dc.audioFormat.sampleRate = sampleRate; getOutputAudioFormat(&(dc.audioFormat), - &(cb->audioFormat)); + &(cb.audioFormat)); dc.state = DECODE_STATE_DECODE; } @@ -395,7 +395,7 @@ static int aac_decode(OutputBuffer * cb, char *path) sampleBufferLen = sampleCount * 2; - sendDataToOutputBuffer(cb, NULL, 0, sampleBuffer, + sendDataToOutputBuffer(NULL, 0, sampleBuffer, sampleBufferLen, file_time, bitRate, NULL); if (dc.seek) { @@ -408,7 +408,7 @@ static int aac_decode(OutputBuffer * cb, char *path) } } - flushOutputBuffer(cb); + flushOutputBuffer(); faacDecClose(decoder); if (b.buffer) diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c index 4510ba46a..d661278b1 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, char *path) +static int audiofile_decode(char *path) { int fs, frame_count; AFfilehandle af_fp; @@ -72,7 +72,7 @@ static int audiofile_decode(OutputBuffer * cb, char *path) (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK); 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); @@ -97,7 +97,7 @@ static int audiofile_decode(OutputBuffer * cb, char *path) while (!eof) { if (dc.seek) { - clearOutputBuffer(cb); + clearOutputBuffer(); current = dc.seekWhere * dc.audioFormat.sampleRate; afSeekFrame(af_fp, AF_DEFAULT_TRACK, current); @@ -112,8 +112,7 @@ static int audiofile_decode(OutputBuffer * cb, char *path) eof = 1; else { current += ret; - sendDataToOutputBuffer(cb, - NULL, + sendDataToOutputBuffer(NULL, 1, chunk, ret * fs, @@ -126,7 +125,7 @@ static int audiofile_decode(OutputBuffer * cb, char *path) } } - flushOutputBuffer(cb); + flushOutputBuffer(); } afCloseFile(af_fp); diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c index f171ee457..70b5c7a80 100644 --- a/src/inputPlugins/flac_plugin.c +++ b/src/inputPlugins/flac_plugin.c @@ -381,8 +381,7 @@ static MpdTag *flacTagDup(char *file) return ret; } -static int flac_decode_internal(OutputBuffer * cb, - InputStream * inStream, int is_ogg) +static int flac_decode_internal(InputStream * inStream, int is_ogg) { flac_decoder *flacDec; FlacData data; @@ -390,7 +389,7 @@ static int flac_decode_internal(OutputBuffer * cb, if (!(flacDec = flac_new())) return -1; - init_FlacData(&data, cb, inStream); + init_FlacData(&data, 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)) @@ -431,7 +430,7 @@ static int flac_decode_internal(OutputBuffer * cb, FLAC__uint64 sampleToSeek = dc.seekWhere * dc.audioFormat.sampleRate + 0.5; if (flac_seek_absolute(flacDec, sampleToSeek)) { - clearOutputBuffer(cb); + clearOutputBuffer(); data.time = ((float)sampleToSeek) / dc.audioFormat.sampleRate; data.position = 0; @@ -448,7 +447,7 @@ static int flac_decode_internal(OutputBuffer * cb, /* send last little bit */ if (data.chunk_length > 0 && !dc.stop) { flacSendChunk(&data); - flushOutputBuffer(data.cb); + flushOutputBuffer(); } fail: @@ -465,9 +464,9 @@ fail: return 0; } -static int flac_decode(OutputBuffer * cb, InputStream * inStream) +static int flac_decode(InputStream * inStream) { - return flac_decode_internal(cb, inStream, 0); + return flac_decode_internal(inStream, 0); } #if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7 @@ -506,9 +505,9 @@ out: return ret; } -static int oggflac_decode(OutputBuffer * cb, InputStream * inStream) +static int oggflac_decode(InputStream * inStream) { - return flac_decode_internal(cb, inStream, 1); + return flac_decode_internal(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 728f42d6f..31ffa9a3d 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, char *path) +static int mod_decode(char *path) { mod_Data *data; float total_time = 0.0; @@ -183,7 +183,7 @@ static int mod_decode(OutputBuffer * cb, char *path) dc.audioFormat.bits = 16; dc.audioFormat.sampleRate = 44100; dc.audioFormat.channels = 2; - getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat)); + getOutputAudioFormat(&(dc.audioFormat), &(cb.audioFormat)); secPerByte = 1.0 / ((dc.audioFormat.bits * dc.audioFormat.channels / 8.0) * @@ -205,12 +205,12 @@ static int mod_decode(OutputBuffer * cb, char *path) ret = VC_WriteBytes(data->audio_buffer, MIKMOD_FRAME_SIZE); total_time += ret * secPerByte; - sendDataToOutputBuffer(cb, NULL, 0, + sendDataToOutputBuffer(NULL, 0, (char *)data->audio_buffer, ret, total_time, 0, NULL); } - flushOutputBuffer(cb); + flushOutputBuffer(); mod_close(data); diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c index ea33ad5ad..ee26385d9 100644 --- a/src/inputPlugins/mp3_plugin.c +++ b/src/inputPlugins/mp3_plugin.c @@ -813,8 +813,7 @@ static int openMp3FromInputStream(InputStream * inStream, mp3DecodeData * data, return 0; } -static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, - ReplayGainInfo ** replayGainInfo) +static int mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo) { int samplesPerFrame; int samplesLeft; @@ -854,7 +853,7 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, case MUTEFRAME_SEEK: if (dc.seekWhere <= data->elapsedTime) { data->outputPtr = data->outputBuffer; - clearOutputBuffer(cb); + clearOutputBuffer(); data->muteFrame = 0; dc.seek = 0; decoder_wakeup_player(); @@ -929,8 +928,7 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, } if (data->outputPtr >= data->outputBufferEnd) { - ret = sendDataToOutputBuffer(cb, - data->inStream, + ret = sendDataToOutputBuffer(data->inStream, data->inStream->seekable, data->outputBuffer, data->outputPtr - data->outputBuffer, @@ -965,7 +963,7 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, data->frameOffset[j]) == 0) { data->outputPtr = data->outputBuffer; - clearOutputBuffer(cb); + clearOutputBuffer(); data->currentFrame = j; } else dc.seekError = 1; @@ -1014,7 +1012,7 @@ static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data, af->channels = MAD_NCHANNELS(&(data->frame).header); } -static int mp3_decode(OutputBuffer * cb, InputStream * inStream) +static int mp3_decode(InputStream * inStream) { mp3DecodeData data; MpdTag *tag = NULL; @@ -1031,7 +1029,7 @@ static int mp3_decode(OutputBuffer * cb, InputStream * inStream) } initAudioFormatFromMp3DecodeData(&data, &(dc.audioFormat)); - getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat)); + getOutputAudioFormat(&(dc.audioFormat), &(cb.audioFormat)); dc.totalTime = data.totalTime; @@ -1062,10 +1060,10 @@ static int mp3_decode(OutputBuffer * cb, InputStream * inStream) dc.state = DECODE_STATE_DECODE; - while (mp3Read(&data, cb, &replayGainInfo) != DECODE_BREAK) ; + while (mp3Read(&data, &replayGainInfo) != DECODE_BREAK) ; /* send last little bit if not dc.stop */ if (!dc.stop && data.outputPtr != data.outputBuffer && data.flush) { - sendDataToOutputBuffer(cb, NULL, + sendDataToOutputBuffer(NULL, data.inStream->seekable, data.outputBuffer, data.outputPtr - data.outputBuffer, @@ -1077,12 +1075,12 @@ static int mp3_decode(OutputBuffer * cb, InputStream * inStream) freeReplayGainInfo(replayGainInfo); if (dc.seek && data.muteFrame == MUTEFRAME_SEEK) { - clearOutputBuffer(cb); + clearOutputBuffer(); dc.seek = 0; decoder_wakeup_player(); } - flushOutputBuffer(cb); + flushOutputBuffer(); mp3DecodeDataFinalize(&data); return 0; diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c index fb8c71020..1dd418b2d 100644 --- a/src/inputPlugins/mp4_plugin.c +++ b/src/inputPlugins/mp4_plugin.c @@ -84,7 +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, InputStream * inStream) +static int mp4_decode(InputStream * inStream) { mp4ff_t *mp4fh; mp4ff_callback_t *mp4cb; @@ -217,7 +217,7 @@ static int mp4_decode(OutputBuffer * cb, InputStream * inStream) if (seeking && seekPositionFound) { seekPositionFound = 0; - clearOutputBuffer(cb); + clearOutputBuffer(); seeking = 0; dc.seek = 0; decoder_wakeup_player(); @@ -255,7 +255,7 @@ static int mp4_decode(OutputBuffer * cb, InputStream * inStream) dc.audioFormat.sampleRate = scale; dc.audioFormat.channels = frameInfo.channels; getOutputAudioFormat(&(dc.audioFormat), - &(cb->audioFormat)); + &(cb.audioFormat)); dc.state = DECODE_STATE_DECODE; } @@ -277,7 +277,7 @@ static int mp4_decode(OutputBuffer * cb, InputStream * inStream) sampleBuffer += offset * channels * 2; - sendDataToOutputBuffer(cb, inStream, 1, sampleBuffer, + sendDataToOutputBuffer(inStream, 1, sampleBuffer, sampleBufferLen, file_time, bitRate, NULL); if (dc.stop) { @@ -295,11 +295,11 @@ static int mp4_decode(OutputBuffer * cb, InputStream * inStream) return -1; if (dc.seek && seeking) { - clearOutputBuffer(cb); + clearOutputBuffer(); dc.seek = 0; decoder_wakeup_player(); } - flushOutputBuffer(cb); + flushOutputBuffer(); return 0; } @@ -309,7 +309,7 @@ static MpdTag *mp4DataDup(char *file, int *mp4MetadataFound) MpdTag *ret = NULL; InputStream inStream; mp4ff_t *mp4fh; - mp4ff_callback_t *cb; + mp4ff_callback_t *callback; int32_t track; int32_t file_time; int32_t scale; @@ -322,14 +322,14 @@ static MpdTag *mp4DataDup(char *file, int *mp4MetadataFound) return NULL; } - cb = xmalloc(sizeof(mp4ff_callback_t)); - cb->read = mp4_inputStreamReadCallback; - cb->seek = mp4_inputStreamSeekCallback; - cb->user_data = &inStream; + callback = xmalloc(sizeof(mp4ff_callback_t)); + callback->read = mp4_inputStreamReadCallback; + callback->seek = mp4_inputStreamSeekCallback; + callback->user_data = &inStream; - mp4fh = mp4ff_open_read(cb); + mp4fh = mp4ff_open_read(callback); if (!mp4fh) { - free(cb); + free(callback); closeInputStream(&inStream); return NULL; } @@ -338,7 +338,7 @@ static MpdTag *mp4DataDup(char *file, int *mp4MetadataFound) if (track < 0) { mp4ff_close(mp4fh); closeInputStream(&inStream); - free(cb); + free(callback); return NULL; } @@ -348,7 +348,7 @@ static MpdTag *mp4DataDup(char *file, int *mp4MetadataFound) if (scale < 0) { mp4ff_close(mp4fh); closeInputStream(&inStream); - free(cb); + free(callback); freeMpdTag(ret); return NULL; } @@ -389,7 +389,6 @@ static MpdTag *mp4DataDup(char *file, int *mp4MetadataFound) mp4ff_close(mp4fh); closeInputStream(&inStream); - free(cb); return ret; } diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c index 867965688..77ca07b30 100644 --- a/src/inputPlugins/mpc_plugin.c +++ b/src/inputPlugins/mpc_plugin.c @@ -111,7 +111,7 @@ static inline mpd_sint16 convertSample(MPC_SAMPLE_FORMAT sample) return val; } -static int mpc_decode(OutputBuffer * cb, InputStream * inStream) +static int mpc_decode(InputStream * inStream) { mpc_decoder decoder; mpc_reader reader; @@ -170,7 +170,7 @@ static int mpc_decode(OutputBuffer * cb, InputStream * inStream) 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; @@ -184,7 +184,7 @@ static int mpc_decode(OutputBuffer * cb, InputStream * inStream) if (dc.seek) { samplePos = dc.seekWhere * dc.audioFormat.sampleRate; if (mpc_decoder_seek_sample(&decoder, samplePos)) { - clearOutputBuffer(cb); + clearOutputBuffer(); s16 = (mpd_sint16 *) chunk; chunkpos = 0; } else @@ -221,7 +221,7 @@ static int mpc_decode(OutputBuffer * cb, InputStream * inStream) bitRate = vbrUpdateBits * dc.audioFormat.sampleRate / 1152 / 1000; - sendDataToOutputBuffer(cb, inStream, + sendDataToOutputBuffer(inStream, inStream->seekable, chunk, chunkpos, total_time, @@ -243,12 +243,12 @@ static int mpc_decode(OutputBuffer * cb, InputStream * inStream) bitRate = vbrUpdateBits * dc.audioFormat.sampleRate / 1152 / 1000; - sendDataToOutputBuffer(cb, NULL, inStream->seekable, + sendDataToOutputBuffer(NULL, inStream->seekable, chunk, chunkpos, total_time, bitRate, replayGainInfo); } - flushOutputBuffer(cb); + flushOutputBuffer(); freeReplayGainInfo(replayGainInfo); diff --git a/src/inputPlugins/oggflac_plugin.c b/src/inputPlugins/oggflac_plugin.c index 070404e26..003b057d9 100644 --- a/src/inputPlugins/oggflac_plugin.c +++ b/src/inputPlugins/oggflac_plugin.c @@ -336,13 +336,13 @@ static unsigned int oggflac_try_decode(InputStream * inStream) return (ogg_stream_type_detect(inStream) == FLAC) ? 1 : 0; } -static int oggflac_decode(OutputBuffer * cb, InputStream * inStream) +static int oggflac_decode(InputStream * inStream) { OggFLAC__SeekableStreamDecoder *decoder = NULL; FlacData data; int ret = 0; - init_FlacData(&data, cb, inStream); + init_FlacData(&data, inStream); if (!(decoder = full_decoder_init_and_read_metadata(&data, 0))) { ret = -1; @@ -362,7 +362,7 @@ static int oggflac_decode(OutputBuffer * cb, InputStream * inStream) dc.audioFormat.sampleRate + 0.5; if (OggFLAC__seekable_stream_decoder_seek_absolute (decoder, sampleToSeek)) { - clearOutputBuffer(cb); + clearOutputBuffer(); data.time = ((float)sampleToSeek) / dc.audioFormat.sampleRate; data.position = 0; @@ -381,7 +381,7 @@ static int oggflac_decode(OutputBuffer * cb, InputStream * inStream) /* send last little bit */ if (data.chunk_length > 0 && !dc.stop) { flacSendChunk(&data); - flushOutputBuffer(data.cb); + flushOutputBuffer(); } fail: diff --git a/src/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c index afcef3e08..eb44b5c6e 100644 --- a/src/inputPlugins/oggvorbis_plugin.c +++ b/src/inputPlugins/oggvorbis_plugin.c @@ -195,7 +195,7 @@ static MpdTag *oggCommentsParse(char **comments) return tag; } -static void putOggCommentsIntoOutputBuffer(OutputBuffer * cb, char *streamName, +static void putOggCommentsIntoOutputBuffer(char *streamName, char **comments) { MpdTag *tag; @@ -216,7 +216,7 @@ static void putOggCommentsIntoOutputBuffer(OutputBuffer * cb, char *streamName, } /* public */ -static int oggvorbis_decode(OutputBuffer * cb, InputStream * inStream) +static int oggvorbis_decode(InputStream * inStream) { OggVorbis_File vf; ov_callbacks callbacks; @@ -275,7 +275,7 @@ static int oggvorbis_decode(OutputBuffer * cb, InputStream * inStream) while (1) { if (dc.seek) { if (0 == ov_time_seek_page(&vf, dc.seekWhere)) { - clearOutputBuffer(cb); + clearOutputBuffer(); chunkpos = 0; } else dc.seekError = 1; @@ -292,11 +292,11 @@ static int oggvorbis_decode(OutputBuffer * cb, InputStream * inStream) dc.audioFormat.sampleRate = vi->rate; if (dc.state == DECODE_STATE_START) { getOutputAudioFormat(&(dc.audioFormat), - &(cb->audioFormat)); + &(cb.audioFormat)); dc.state = DECODE_STATE_DECODE; } comments = ov_comment(&vf, -1)->user_comments; - putOggCommentsIntoOutputBuffer(cb, inStream->metaName, + putOggCommentsIntoOutputBuffer(inStream->metaName, comments); ogg_getReplayGainInfo(comments, &replayGainInfo); } @@ -316,7 +316,7 @@ static int oggvorbis_decode(OutputBuffer * cb, InputStream * inStream) if ((test = ov_bitrate_instant(&vf)) > 0) { bitRate = test / 1000; } - sendDataToOutputBuffer(cb, inStream, + sendDataToOutputBuffer(inStream, inStream->seekable, chunk, chunkpos, ov_pcm_tell(&vf) / @@ -329,7 +329,7 @@ static int oggvorbis_decode(OutputBuffer * cb, InputStream * inStream) } if (!dc.stop && chunkpos > 0) { - sendDataToOutputBuffer(cb, NULL, inStream->seekable, + sendDataToOutputBuffer(NULL, inStream->seekable, chunk, chunkpos, ov_time_tell(&vf), bitRate, replayGainInfo); @@ -340,7 +340,7 @@ static int oggvorbis_decode(OutputBuffer * cb, InputStream * inStream) ov_clear(&vf); - flushOutputBuffer(cb); + flushOutputBuffer(); return 0; } diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c index bae5f6acb..13f10a1e9 100644 --- a/src/inputPlugins/wavpack_plugin.c +++ b/src/inputPlugins/wavpack_plugin.c @@ -128,8 +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, - WavpackContext *wpc, int canseek, +static void wavpack_decode(WavpackContext *wpc, int canseek, ReplayGainInfo *replayGainInfo) { void (*format_samples)(int Bps, void *buffer, uint32_t samcnt); @@ -167,7 +166,7 @@ static void wavpack_decode(OutputBuffer *cb, 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; @@ -180,7 +179,7 @@ static void wavpack_decode(OutputBuffer *cb, if (canseek) { int where; - clearOutputBuffer(cb); + clearOutputBuffer(); where = dc.seekWhere * dc.audioFormat.sampleRate; @@ -211,14 +210,14 @@ static void wavpack_decode(OutputBuffer *cb, format_samples(Bps, chunk, samplesgot * dc.audioFormat.channels); - sendDataToOutputBuffer(cb, NULL, 0, chunk, + sendDataToOutputBuffer(NULL, 0, chunk, samplesgot * outsamplesize, file_time, bitrate, replayGainInfo); } } while (samplesgot == samplesreq); - flushOutputBuffer(cb); + flushOutputBuffer(); } static char *wavpack_tag(WavpackContext *wpc, char *key) @@ -442,7 +441,7 @@ static unsigned int wavpack_trydecode(InputStream *is) /* * Decodes a stream. */ -static int wavpack_streamdecode(OutputBuffer *cb, InputStream *is) +static int wavpack_streamdecode(InputStream *is) { char error[ERRORLEN]; WavpackContext *wpc; @@ -541,7 +540,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, InputStream *is) return -1; } - wavpack_decode(cb, wpc, canseek, NULL); + wavpack_decode(wpc, canseek, NULL); WavpackCloseFile(wpc); if (wvc_url != NULL) { @@ -556,7 +555,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, InputStream *is) /* * Decodes a file. */ -static int wavpack_filedecode(OutputBuffer *cb, char *fname) +static int wavpack_filedecode(char *fname) { char error[ERRORLEN]; WavpackContext *wpc; @@ -572,7 +571,7 @@ static int wavpack_filedecode(OutputBuffer *cb, char *fname) replayGainInfo = wavpack_replaygain(wpc); - wavpack_decode(cb, wpc, 1, replayGainInfo); + wavpack_decode(wpc, 1, replayGainInfo); if (replayGainInfo) freeReplayGainInfo(replayGainInfo); |