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authorMichael Niedermayer <michaelni@gmx.at>2011-10-22 01:03:27 +0200
committerMichael Niedermayer <michaelni@gmx.at>2011-10-22 01:16:41 +0200
commitaedc908601de7396751a9a4504e064782d9f6a0b (patch)
tree8f04b899142439893bac426ac83d05c4068b099c /libavcodec/libspeexdec.c
parent1a7090bfafe986d4470ba8059c815939171ddb74 (diff)
parentf4b51d061f0f34e36be876b562b8abe47f4b9c1c (diff)
downloadffmpeg-aedc908601de7396751a9a4504e064782d9f6a0b.tar.gz
Merge remote-tracking branch 'qatar/master'
* qatar/master: (35 commits) flvdec: Do not call parse_keyframes_index with a NULL stream libspeexdec: include system headers before local headers libspeexdec: return meaningful error codes libspeexdec: cosmetics: reindent libspeexdec: decode one frame at a time. swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables() Move timefilter code from lavf to lavd. mov: add support for hdvd and pgapmetadata atoms mov: rename function _stik, some indentation cosmetics mov: rename function _int8 to remove ambiguity, some indentation cosmetics mov: parse the gnre atom mp3on4: check for allocation failures in decode_init_mp3on4() mp3on4: create a separate flush function for MP3onMP4. mp3on4: ensure that the frame channel count does not exceed the codec channel count. mp3on4: set channel layout mp3on4: fix the output channel order mp3on4: allocate temp buffer with av_malloc() instead of on the stack. mp3on4: copy MPADSPContext from first context to all contexts. fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm ... Conflicts: libavcodec/arm/h264dsp_init_arm.c libavcodec/h264.c libavcodec/h264.h libavcodec/h264_cabac.c libavcodec/h264_cavlc.c libavcodec/h264_ps.c libavcodec/h264dsp_template.c libavcodec/h264idct_template.c libavcodec/h264pred.c libavcodec/h264pred_template.c libavcodec/x86/h264dsp_mmx.c libavdevice/Makefile libavdevice/jack_audio.c libavformat/Makefile libavformat/flvdec.c libavformat/flvenc.c libavutil/pixfmt.h libswscale/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/libspeexdec.c')
-rw-r--r--libavcodec/libspeexdec.c68
1 files changed, 43 insertions, 25 deletions
diff --git a/libavcodec/libspeexdec.c b/libavcodec/libspeexdec.c
index 7ee53b04e5..91f190525d 100644
--- a/libavcodec/libspeexdec.c
+++ b/libavcodec/libspeexdec.c
@@ -18,11 +18,11 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#include "avcodec.h"
#include <speex/speex.h>
#include <speex/speex_header.h>
#include <speex/speex_stereo.h>
#include <speex/speex_callbacks.h>
+#include "avcodec.h"
typedef struct {
SpeexBits bits;
@@ -60,14 +60,14 @@ static av_cold int libspeex_decode_init(AVCodecContext *avctx)
mode = speex_lib_get_mode(s->header->mode);
if (!mode) {
av_log(avctx, AV_LOG_ERROR, "Unknown Speex mode %d", s->header->mode);
- return -1;
+ return AVERROR_INVALIDDATA;
}
} else
av_log(avctx, AV_LOG_INFO, "Missing Speex header, assuming defaults.\n");
if (avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Only stereo and mono are supported.\n");
- return -1;
+ return AVERROR(EINVAL);
}
speex_bits_init(&s->bits);
@@ -99,32 +99,42 @@ static int libspeex_decode_frame(AVCodecContext *avctx,
uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
LibSpeexContext *s = avctx->priv_data;
- int16_t *output = data, *end;
- int i, num_samples;
-
- num_samples = s->frame_size * avctx->channels;
- end = output + *data_size / sizeof(*output);
-
- speex_bits_read_from(&s->bits, buf, buf_size);
-
- for (i = 0; speex_bits_remaining(&s->bits) && output + num_samples < end; i++) {
- int ret = speex_decode_int(s->dec_state, &s->bits, output);
- if (ret <= -2) {
- av_log(avctx, AV_LOG_ERROR, "Error decoding Speex frame.\n");
- return -1;
- } else if (ret == -1)
- // end of stream
- break;
+ int16_t *output = data;
+ int out_size, ret, consumed = 0;
+
+ /* check output buffer size */
+ out_size = s->frame_size * avctx->channels *
+ av_get_bytes_per_sample(avctx->sample_fmt);
+ if (*data_size < out_size) {
+ av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+ return AVERROR(EINVAL);
+ }
- if (avctx->channels == 2)
- speex_decode_stereo_int(output, s->frame_size, &s->stereo);
+ /* if there is not enough data left for the smallest possible frame,
+ reset the libspeex buffer using the current packet, otherwise ignore
+ the current packet and keep decoding frames from the libspeex buffer. */
+ if (speex_bits_remaining(&s->bits) < 43) {
+ /* check for flush packet */
+ if (!buf || !buf_size) {
+ *data_size = 0;
+ return buf_size;
+ }
+ /* set new buffer */
+ speex_bits_read_from(&s->bits, buf, buf_size);
+ consumed = buf_size;
+ }
- output += num_samples;
+ /* decode a single frame */
+ ret = speex_decode_int(s->dec_state, &s->bits, output);
+ if (ret <= -2) {
+ av_log(avctx, AV_LOG_ERROR, "Error decoding Speex frame.\n");
+ return AVERROR_INVALIDDATA;
}
+ if (avctx->channels == 2)
+ speex_decode_stereo_int(output, s->frame_size, &s->stereo);
- avctx->frame_size = s->frame_size * i;
- *data_size = avctx->channels * avctx->frame_size * sizeof(*output);
- return buf_size;
+ *data_size = out_size;
+ return consumed;
}
static av_cold int libspeex_decode_close(AVCodecContext *avctx)
@@ -138,6 +148,12 @@ static av_cold int libspeex_decode_close(AVCodecContext *avctx)
return 0;
}
+static av_cold void libspeex_decode_flush(AVCodecContext *avctx)
+{
+ LibSpeexContext *s = avctx->priv_data;
+ speex_bits_reset(&s->bits);
+}
+
AVCodec ff_libspeex_decoder = {
.name = "libspeex",
.type = AVMEDIA_TYPE_AUDIO,
@@ -146,5 +162,7 @@ AVCodec ff_libspeex_decoder = {
.init = libspeex_decode_init,
.close = libspeex_decode_close,
.decode = libspeex_decode_frame,
+ .flush = libspeex_decode_flush,
+ .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY,
.long_name = NULL_IF_CONFIG_SMALL("libspeex Speex"),
};