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author | Michael Niedermayer <michaelni@gmx.at> | 2011-10-22 01:03:27 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2011-10-22 01:16:41 +0200 |
commit | aedc908601de7396751a9a4504e064782d9f6a0b (patch) | |
tree | 8f04b899142439893bac426ac83d05c4068b099c /libavcodec/libspeexdec.c | |
parent | 1a7090bfafe986d4470ba8059c815939171ddb74 (diff) | |
parent | f4b51d061f0f34e36be876b562b8abe47f4b9c1c (diff) | |
download | ffmpeg-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.c | 68 |
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"), }; |