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| author | Justin Ruggles <justin.ruggles@gmail.com> | 2011-10-09 13:36:01 -0400 |
|---|---|---|
| committer | Justin Ruggles <justin.ruggles@gmail.com> | 2011-10-26 11:50:17 -0400 |
| commit | 7080533cda2cccf980326b0942de8ff046619fd4 (patch) | |
| tree | 77d12cef269add9b7a5dc791a6dabfd00a1b0057 | |
| parent | cb50329fc5ea89240ea5e835a3ee72ce16cf052a (diff) | |
| download | ffmpeg-7080533cda2cccf980326b0942de8ff046619fd4.tar.gz | |
alacdec: remove the numchannels parameter from several functions.
They only operate on stereo content, so the extra param is not necessary and
also allows for simplifying the code.
| -rw-r--r-- | libavcodec/alac.c | 40 |
1 files changed, 13 insertions, 27 deletions
diff --git a/libavcodec/alac.c b/libavcodec/alac.c index 6563e52f30..3b79645ce1 100644 --- a/libavcodec/alac.c +++ b/libavcodec/alac.c @@ -299,8 +299,7 @@ static void predictor_decompress_fir_adapt(int32_t *error_buffer, } static void decorrelate_stereo(int32_t *buffer[MAX_CHANNELS], - int numchannels, int numsamples, - uint8_t interlacing_shift, + int numsamples, uint8_t interlacing_shift, uint8_t interlacing_leftweight) { int i; @@ -331,37 +330,25 @@ static void append_extra_bits(int32_t *buffer[MAX_CHANNELS], } static void interleave_stereo_16(int32_t *buffer[MAX_CHANNELS], - int16_t *buffer_out, int numchannels, - int numsamples) + int16_t *buffer_out, int numsamples) { int i; for (i = 0; i < numsamples; i++) { - int16_t left, right; - - left = buffer[0][i]; - right = buffer[1][i]; - - buffer_out[i*numchannels] = left; - buffer_out[i*numchannels + 1] = right; + *buffer_out++ = buffer[0][i]; + *buffer_out++ = buffer[1][i]; } } static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS], - int32_t *buffer_out, int numchannels, - int numsamples) + int32_t *buffer_out, int numsamples) { int i; - for (i = 0; i < numsamples; i++) { - int32_t left, right; - - left = buffer[0][i]; - right = buffer[1][i]; - - buffer_out[i * numchannels] = left << 8; - buffer_out[i * numchannels + 1] = right << 8; - } + for (i = 0; i < numsamples; i++) { + *buffer_out++ = buffer[0][i] << 8; + *buffer_out++ = buffer[1][i] << 8; + } } static int alac_decode_frame(AVCodecContext *avctx, @@ -521,9 +508,8 @@ static int alac_decode_frame(AVCodecContext *avctx, av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n"); if (channels == 2 && interlacing_leftweight) { - decorrelate_stereo(alac->outputsamples_buffer, alac->numchannels, - outputsamples, interlacing_shift, - interlacing_leftweight); + decorrelate_stereo(alac->outputsamples_buffer, outputsamples, + interlacing_shift, interlacing_leftweight); } if (alac->extra_bits) { @@ -535,7 +521,7 @@ static int alac_decode_frame(AVCodecContext *avctx, case 16: if (channels == 2) { interleave_stereo_16(alac->outputsamples_buffer, outbuffer, - alac->numchannels, outputsamples); + outputsamples); } else { int i; for (i = 0; i < outputsamples; i++) { @@ -546,7 +532,7 @@ static int alac_decode_frame(AVCodecContext *avctx, case 24: if (channels == 2) { interleave_stereo_24(alac->outputsamples_buffer, outbuffer, - alac->numchannels, outputsamples); + outputsamples); } else { int i; for (i = 0; i < outputsamples; i++) |