diff options
| author | Ronald S. Bultje <rsbultje@gmail.com> | 2010-04-21 17:57:48 +0000 |
|---|---|---|
| committer | Ronald S. Bultje <rsbultje@gmail.com> | 2010-04-21 17:57:48 +0000 |
| commit | b1078e9fe6b5d8f034d15a6ab91430fd41921fe2 (patch) | |
| tree | 5d16af4805e9567267f3f3daa96e015523f4216c /libavcodec/amrnbdec.c | |
| parent | 6858ce2ffc4c57a279b22ab6d8eb56ac3884796e (diff) | |
| download | ffmpeg-b1078e9fe6b5d8f034d15a6ab91430fd41921fe2.tar.gz | |
Move clipping of audio samples (for those codecs outputting float) from decoder
to the audio conversion routines.
Originally committed as revision 22937 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/amrnbdec.c')
| -rw-r--r-- | libavcodec/amrnbdec.c | 12 |
1 files changed, 3 insertions, 9 deletions
diff --git a/libavcodec/amrnbdec.c b/libavcodec/amrnbdec.c index 40cd91d110..cd2d95b11b 100644 --- a/libavcodec/amrnbdec.c +++ b/libavcodec/amrnbdec.c @@ -796,7 +796,7 @@ static int synthesis(AMRContext *p, float *lpc, float fixed_gain, const float *fixed_vector, float *samples, uint8_t overflow) { - int i, overflow_temp = 0; + int i; float excitation[AMR_SUBFRAME_SIZE]; // if an overflow has been detected, the pitch vector is scaled down by a @@ -831,12 +831,10 @@ static int synthesis(AMRContext *p, float *lpc, // detect overflow for (i = 0; i < AMR_SUBFRAME_SIZE; i++) if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) { - overflow_temp = 1; - samples[i] = av_clipf(samples[i], -AMR_SAMPLE_BOUND, - AMR_SAMPLE_BOUND); + return 1; } - return overflow_temp; + return 0; } /// @} @@ -1048,10 +1046,6 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, highpass_poles, highpass_gain, p->high_pass_mem, AMR_BLOCK_SIZE); - for (i = 0; i < AMR_BLOCK_SIZE; i++) - buf_out[i] = av_clipf(buf_out[i] * AMR_SAMPLE_SCALE, - -1.0, 32767.0 / 32768.0); - /* Update averaged lsf vector (used for fixed gain smoothing). * * Note that lsf_avg should not incorporate the current frame's LSFs |