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Fix a few places where frame data was used directly.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jason Parker
2008-05-22 17:10:53 +00:00
parent 5ceec8b052
commit f7eb823a7a
4 changed files with 11 additions and 11 deletions
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8
apps/app_jack.c
View File

@@ -484,7 +484,7 @@ static int queue_voice_frame(struct jack_data *jack_data, struct ast_frame *f)
float f_buf[f->samples * 8];
size_t f_buf_used = 0;
int i;
int16_t *s_buf = f->data;
int16_t *s_buf = f->data.ptr;
size_t res;
memset(f_buf, 0, sizeof(f_buf));
@@ -572,7 +572,7 @@ static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_d
.frametype = AST_FRAME_VOICE,
.subclass = AST_FORMAT_SLINEAR,
.src = "JACK",
.data = buf,
.data.ptr = buf,
.datalen = sizeof(buf),
.samples = ARRAY_LEN(buf),
};
@@ -582,7 +582,7 @@ static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_d
char *read_buf;
read_len = out_frame ? out_frame->datalen : sizeof(buf);
read_buf = out_frame ? out_frame->data : buf;
read_buf = out_frame ? out_frame->data.ptr : buf;
res = jack_ringbuffer_read_space(jack_data->input_rb);
@@ -590,7 +590,7 @@ static void handle_jack_audio(struct ast_channel *chan, struct jack_data *jack_d
/* Not enough data ready for another frame, move on ... */
if (out_frame) {
ast_debug(1, "Sending an empty frame for the JACK_HOOK\n");
memset(out_frame->data, 0, out_frame->datalen);
memset(out_frame->data.ptr, 0, out_frame->datalen);
}
break;
}

8
codecs/codec_speex.c
View File

@@ -165,7 +165,7 @@ static struct ast_frame *lintospeex_sample(void)
f.mallocd = 0;
f.offset = 0;
f.src = __PRETTY_FUNCTION__;
f.data = slin_speex_ex;
f.data.ptr = slin_speex_ex;
return &f;
}
@@ -180,7 +180,7 @@ static struct ast_frame *speextolin_sample(void)
f.mallocd = 0;
f.offset = 0;
f.src = __PRETTY_FUNCTION__;
f.data = speex_slin_ex;
f.data.ptr = speex_slin_ex;
return &f;
}
@@ -220,7 +220,7 @@ static int speextolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
}
/* Read in bits */
speex_bits_read_from(&tmp->bits, f->data, f->datalen);
speex_bits_read_from(&tmp->bits, f->data.ptr, f->datalen);
for (;;) {
#ifdef _SPEEX_TYPES_H
res = speex_decode_int(tmp->speex, &tmp->bits, fout);
@@ -249,7 +249,7 @@ static int lintospeex_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
/* XXX We should look at how old the rest of our stream is, and if it
is too old, then we should overwrite it entirely, otherwise we can
get artifacts of earlier talk that do not belong */
memcpy(tmp->buf + pvt->samples, f->data, f->datalen);
memcpy(tmp->buf + pvt->samples, f->data.ptr, f->datalen);
pvt->samples += f->samples;
return 0;
}

4
formats/format_ogg_vorbis.c
View File

@@ -291,7 +291,7 @@ static int ogg_vorbis_write(struct ast_filestream *fs, struct ast_frame *f)
if (!f->datalen)
return -1;
data = (short *) f->data;
data = (short *) f->data.ptr;
buffer = vorbis_analysis_buffer(&s->vd, f->samples);
@@ -433,7 +433,7 @@ static struct ast_frame *ogg_vorbis_read(struct ast_filestream *fs,
fs->fr.subclass = AST_FORMAT_SLINEAR;
fs->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
buf = (short *)(fs->fr.data); /* SLIN data buffer */
buf = (short *)(fs->fr.data.ptr); /* SLIN data buffer */
while (samples_out != SAMPLES_MAX) {
float **pcm;

2
funcs/func_speex.c
View File

@@ -135,7 +135,7 @@ static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *c
speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_DENOISE, &sdi->denoise);
}
speex_preprocess(sdi->state, frame->data, NULL);
speex_preprocess(sdi->state, frame->data.ptr, NULL);
return 0;
}