[Scummvm-cvs-logs] SF.net SVN: scummvm:[46436] tools/branches/gsoc2009-gui/compress.cpp
sev at users.sourceforge.net
sev at users.sourceforge.net
Sun Dec 20 20:09:17 CET 2009
Revision: 46436
http://scummvm.svn.sourceforge.net/scummvm/?rev=46436&view=rev
Author: sev
Date: 2009-12-20 19:09:17 +0000 (Sun, 20 Dec 2009)
Log Message:
-----------
Whitespaces
Modified Paths:
--------------
tools/branches/gsoc2009-gui/compress.cpp
Modified: tools/branches/gsoc2009-gui/compress.cpp
===================================================================
--- tools/branches/gsoc2009-gui/compress.cpp 2009-12-20 16:35:37 UTC (rev 46435)
+++ tools/branches/gsoc2009-gui/compress.cpp 2009-12-20 19:09:17 UTC (rev 46436)
@@ -161,129 +161,129 @@
}
#ifdef DISABLE_BUILTIN_VORBIS
- if (compmode == AUDIO_VORBIS) {
- tmp += sprintf(tmp, "oggenc ");
- if (rawInput) {
- tmp += sprintf(tmp, "--raw ");
- tmp += sprintf(tmp, "--raw-chan=%d ", (rawAudioType.isStereo ? 2 : 1));
- tmp += sprintf(tmp, "--raw-bits=%d ", rawAudioType.bitsPerSample);
- tmp += sprintf(tmp, "--raw-rate=%d ", rawSamplerate);
- tmp += sprintf(tmp, "--raw-endianness=%d ", (rawAudioType.isLittleEndian ? 0 : 1));
- }
+ if (compmode == AUDIO_VORBIS) {
+ tmp += sprintf(tmp, "oggenc ");
+ if (rawInput) {
+ tmp += sprintf(tmp, "--raw ");
+ tmp += sprintf(tmp, "--raw-chan=%d ", (rawAudioType.isStereo ? 2 : 1));
+ tmp += sprintf(tmp, "--raw-bits=%d ", rawAudioType.bitsPerSample);
+ tmp += sprintf(tmp, "--raw-rate=%d ", rawSamplerate);
+ tmp += sprintf(tmp, "--raw-endianness=%d ", (rawAudioType.isLittleEndian ? 0 : 1));
+ }
- if (oggparms.nominalBitr != -1) {
- tmp += sprintf(tmp, "--bitrate=%d ", oggparms.nominalBitr);
- } else {
- tmp += sprintf(tmp, "--quality=%d ", oggparms.quality);
- }
+ if (oggparms.nominalBitr != -1) {
+ tmp += sprintf(tmp, "--bitrate=%d ", oggparms.nominalBitr);
+ } else {
+ tmp += sprintf(tmp, "--quality=%d ", oggparms.quality);
+ }
- if (oggparms.minBitr != -1) {
- tmp += sprintf(tmp, "--min-bitrate=%d ", oggparms.minBitr);
- }
+ if (oggparms.minBitr != -1) {
+ tmp += sprintf(tmp, "--min-bitrate=%d ", oggparms.minBitr);
+ }
- if (oggparms.maxBitr != -1) {
- tmp += sprintf(tmp, "--max-bitrate=%d ", oggparms.maxBitr);
- }
+ if (oggparms.maxBitr != -1) {
+ tmp += sprintf(tmp, "--max-bitrate=%d ", oggparms.maxBitr);
+ }
- if (oggparms.silent) {
- tmp += sprintf(tmp, "--quiet ");
- }
+ if (oggparms.silent) {
+ tmp += sprintf(tmp, "--quiet ");
+ }
- tmp += sprintf(tmp, "--output=\"%s\" ", outname);
- tmp += sprintf(tmp, "\"%s\" ", inname);
+ tmp += sprintf(tmp, "--output=\"%s\" ", outname);
+ tmp += sprintf(tmp, "\"%s\" ", inname);
- err = spawnSubprocess(fbuf) != 0;
+ err = spawnSubprocess(fbuf) != 0;
- if (err) {
- char buf[2048];
- sprintf(buf, "Error in Vorbis encoder. (check parameters)\nVorbis Encoder Commandline:%s\n", fbuf);
- throw ToolException(buf, err);
- } else {
- return;
- }
+ if (err) {
+ char buf[2048];
+ sprintf(buf, "Error in Vorbis encoder. (check parameters)\nVorbis Encoder Commandline:%s\n", fbuf);
+ throw ToolException(buf, err);
+ } else {
+ return;
}
+ }
#endif
#ifdef DISABLE_BUILTIN_FLAC
- if (compmode == AUDIO_FLAC) {
- /* --lax is needed to allow 11kHz, we dont need place for meta-tags, and no seektable */
- /* -f is reqired to force override of unremoved temp file. See bug #1294648 */
- tmp += sprintf(tmp, "flac -f --lax --no-padding --no-seektable --no-ogg ");
+ if (compmode == AUDIO_FLAC) {
+ /* --lax is needed to allow 11kHz, we dont need place for meta-tags, and no seektable */
+ /* -f is reqired to force override of unremoved temp file. See bug #1294648 */
+ tmp += sprintf(tmp, "flac -f --lax --no-padding --no-seektable --no-ogg ");
- if (rawInput) {
- tmp += sprintf(tmp, "--force-raw-format ");
- tmp += sprintf(tmp, "--sign=%s ", ((rawAudioType.bitsPerSample == 8) ? "unsigned" : "signed"));
- tmp += sprintf(tmp, "--channels=%d ", (rawAudioType.isStereo ? 2 : 1));
- tmp += sprintf(tmp, "--bps=%d ", rawAudioType.bitsPerSample);
- tmp += sprintf(tmp, "--sample-rate=%d ", rawSamplerate);
- tmp += sprintf(tmp, "--endian=%s ", (rawAudioType.isLittleEndian ? "little" : "big"));
- }
+ if (rawInput) {
+ tmp += sprintf(tmp, "--force-raw-format ");
+ tmp += sprintf(tmp, "--sign=%s ", ((rawAudioType.bitsPerSample == 8) ? "unsigned" : "signed"));
+ tmp += sprintf(tmp, "--channels=%d ", (rawAudioType.isStereo ? 2 : 1));
+ tmp += sprintf(tmp, "--bps=%d ", rawAudioType.bitsPerSample);
+ tmp += sprintf(tmp, "--sample-rate=%d ", rawSamplerate);
+ tmp += sprintf(tmp, "--endian=%s ", (rawAudioType.isLittleEndian ? "little" : "big"));
+ }
- if (flacparms.silent) {
- tmp += sprintf(tmp, "--silent ");
- }
+ if (flacparms.silent) {
+ tmp += sprintf(tmp, "--silent ");
+ }
- if (flacparms.verify) {
- tmp += sprintf(tmp, "--verify ");
- }
+ if (flacparms.verify) {
+ tmp += sprintf(tmp, "--verify ");
+ }
- tmp += sprintf(tmp, "--compression-level-%d ", flacparms.compressionLevel);
- tmp += sprintf(tmp, "-b %d ", flacparms.blocksize);
- tmp += sprintf(tmp, "-o \"%s\" ", outname);
- tmp += sprintf(tmp, "\"%s\" ", inname);
+ tmp += sprintf(tmp, "--compression-level-%d ", flacparms.compressionLevel);
+ tmp += sprintf(tmp, "-b %d ", flacparms.blocksize);
+ tmp += sprintf(tmp, "-o \"%s\" ", outname);
+ tmp += sprintf(tmp, "\"%s\" ", inname);
- err = spawnSubprocess(fbuf) != 0;
+ err = spawnSubprocess(fbuf) != 0;
- if (err) {
- char buf[2048];
- sprintf(buf, "Error in FLAC encoder. (check parameters)\nFLAC Encoder Commandline:%s\n", fbuf);
- throw ToolException(buf, err);
- } else {
- return;
- }
+ if (err) {
+ char buf[2048];
+ sprintf(buf, "Error in FLAC encoder. (check parameters)\nFLAC Encoder Commandline:%s\n", fbuf);
+ throw ToolException(buf, err);
+ } else {
+ return;
}
+ }
#endif
- if (rawInput) {
- long length;
- char *rawData;
+ if (rawInput) {
+ long length;
+ char *rawData;
- File inputRaw(inname, "rb");
- length = inputRaw.size();
- rawData = (char *)malloc(length);
- inputRaw.read_throwsOnError(rawData, length);
+ File inputRaw(inname, "rb");
+ length = inputRaw.size();
+ rawData = (char *)malloc(length);
+ inputRaw.read_throwsOnError(rawData, length);
- encodeRaw(rawData, length, rawSamplerate, outname, compmode);
+ encodeRaw(rawData, length, rawSamplerate, outname, compmode);
- free(rawData);
- } else {
- int fmtHeaderSize, length, numChannels, sampleRate, bitsPerSample;
- char *wavData;
+ free(rawData);
+ } else {
+ int fmtHeaderSize, length, numChannels, sampleRate, bitsPerSample;
+ char *wavData;
- File inputWav(inname, "rb");
+ File inputWav(inname, "rb");
- /* Standard PCM fmt header is 16 bits, but at least Simon 1 and 2 use 18 bits */
- inputWav.seek(16, SEEK_SET);
- fmtHeaderSize = inputWav.readUint32LE();
+ /* Standard PCM fmt header is 16 bits, but at least Simon 1 and 2 use 18 bits */
+ inputWav.seek(16, SEEK_SET);
+ fmtHeaderSize = inputWav.readUint32LE();
- inputWav.seek(22, SEEK_SET);
- numChannels = inputWav.readUint16LE();
- sampleRate = inputWav.readUint32LE();
+ inputWav.seek(22, SEEK_SET);
+ numChannels = inputWav.readUint16LE();
+ sampleRate = inputWav.readUint32LE();
- inputWav.seek(34, SEEK_SET);
- bitsPerSample = inputWav.readUint16LE();
+ inputWav.seek(34, SEEK_SET);
+ bitsPerSample = inputWav.readUint16LE();
- /* The size of the raw audio is after the RIFF chunk (12 bytes), fmt chunk (8 + fmtHeaderSize bytes), and data chunk id (4 bytes) */
- inputWav.seek(24 + fmtHeaderSize, SEEK_SET);
- length = inputWav.readUint32LE();
+ /* The size of the raw audio is after the RIFF chunk (12 bytes), fmt chunk (8 + fmtHeaderSize bytes), and data chunk id (4 bytes) */
+ inputWav.seek(24 + fmtHeaderSize, SEEK_SET);
+ length = inputWav.readUint32LE();
- wavData = (char *)malloc(length);
- inputWav.read_throwsOnError(wavData, length);
+ wavData = (char *)malloc(length);
+ inputWav.read_throwsOnError(wavData, length);
- setRawAudioType(true, numChannels == 2, (uint8)bitsPerSample);
- encodeRaw(wavData, length, sampleRate, outname, compmode);
+ setRawAudioType(true, numChannels == 2, (uint8)bitsPerSample);
+ encodeRaw(wavData, length, sampleRate, outname, compmode);
- free(wavData);
- }
+ free(wavData);
+ }
}
void CompressionTool::encodeRaw(char *rawData, int length, int samplerate, const char *outname, AudioFormat compmode) {
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