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https://github.com/asterisk/asterisk.git
synced 2025-10-24 05:38:11 +00:00
removed GAIN preprocessor definition, removed needsgain from struct wav_desc, removed unnecessary gain code from wav_read() and wav_write()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -53,7 +53,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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struct wav_desc { /* format-specific parameters */
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struct wav_desc { /* format-specific parameters */
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int bytes;
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int bytes;
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int needsgain;
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int lasttimeout;
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int lasttimeout;
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int maxlen;
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int maxlen;
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struct timeval last;
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struct timeval last;
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@@ -61,9 +60,6 @@ struct wav_desc { /* format-specific parameters */
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#define BLOCKSIZE 160
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#define BLOCKSIZE 160
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#define GAIN 0 /* 2^GAIN is the multiple to increase the volume by. The original value of GAIN was 2, or 4x (12 dB),
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* but there were many reports of the clipping of loud signal peaks (issue 5823 for example). */
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#if __BYTE_ORDER == __LITTLE_ENDIAN
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#if __BYTE_ORDER == __LITTLE_ENDIAN
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#define htoll(b) (b)
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#define htoll(b) (b)
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#define htols(b) (b)
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#define htols(b) (b)
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@@ -387,21 +383,6 @@ static struct ast_frame *wav_read(struct ast_filestream *s, int *whennext)
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tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8);
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tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8);
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#endif
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#endif
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if (fs->needsgain) {
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for (x=0; x < samples; x++) {
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if (tmp[x] & ((1 << GAIN) - 1)) {
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/* If it has data down low, then it's not something we've artificially increased gain
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on, so we don't need to gain adjust it */
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fs->needsgain = 0;
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break;
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}
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}
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if (fs->needsgain) {
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for (x=0; x < samples; x++)
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tmp[x] = tmp[x] >> GAIN;
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}
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}
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*whennext = samples;
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*whennext = samples;
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return &s->fr;
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return &s->fr;
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}
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}
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@@ -409,8 +390,9 @@ static struct ast_frame *wav_read(struct ast_filestream *s, int *whennext)
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static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
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static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
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{
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{
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int x;
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int x;
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#if __BYTE_ORDER == __BIG_ENDIAN
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short tmp[8000], *tmpi;
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short tmp[8000], *tmpi;
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float tmpf;
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#endif
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struct wav_desc *s = (struct wav_desc *)fs->private;
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struct wav_desc *s = (struct wav_desc *)fs->private;
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int res;
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int res;
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@@ -422,34 +404,24 @@ static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
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ast_log(LOG_WARNING, "Asked to write non-SLINEAR frame (%d)!\n", f->subclass);
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ast_log(LOG_WARNING, "Asked to write non-SLINEAR frame (%d)!\n", f->subclass);
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return -1;
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return -1;
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}
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}
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if (!f->datalen)
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return -1;
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#if __BYTE_ORDER == __BIG_ENDIAN
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/* swap and write */
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if (f->datalen > sizeof(tmp)) {
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if (f->datalen > sizeof(tmp)) {
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ast_log(LOG_WARNING, "Data length is too long\n");
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ast_log(LOG_WARNING, "Data length is too long\n");
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return -1;
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return -1;
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}
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}
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if (!f->datalen)
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return -1;
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#if 0
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printf("Data Length: %d\n", f->datalen);
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#endif
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tmpi = f->data;
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tmpi = f->data;
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/* Volume adjust here to accomodate */
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for (x=0; x < f->datalen/2; x++)
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for (x=0;x<f->datalen/2;x++) {
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tmp[x] = (tmpi[x] << 8) | ((tmpi[x] & 0xff00) >> 8);
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tmpf = ((float)tmpi[x]) * ((float)(1 << GAIN));
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if (tmpf > 32767.0)
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tmpf = 32767.0;
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if (tmpf < -32768.0)
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tmpf = -32768.0;
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tmp[x] = tmpf;
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tmp[x] &= ~((1 << GAIN) - 1);
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#if __BYTE_ORDER == __BIG_ENDIAN
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tmp[x] = (tmp[x] << 8) | ((tmp[x] & 0xff00) >> 8);
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#endif
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}
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if ((res = fwrite(tmp, 1, f->datalen, fs->f)) != f->datalen ) {
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if ((res = fwrite(tmp, 1, f->datalen, fs->f)) != f->datalen ) {
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#else
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/* just write */
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if ((res = fwrite(f->data, 1, f->datalen, fs->f)) != f->datalen ) {
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#endif
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ast_log(LOG_WARNING, "Bad write (%d): %s\n", res, strerror(errno));
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ast_log(LOG_WARNING, "Bad write (%d): %s\n", res, strerror(errno));
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return -1;
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return -1;
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}
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}
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