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authorJames Almer <jamrial@gmail.com>2014-09-30 15:12:10 -0300
committerJames Almer <jamrial@gmail.com>2014-10-02 22:11:51 -0300
commitddb813b0ef8f135679020240fdca41c29976c23a (patch)
treeeb38ff7146c1949d6fc88272fb40c21c838a6dcc /libavcodec/mlpdec.c
parent5c378d6a6df8243f06c87962b873bd563e58cd39 (diff)
downloadffmpeg-ddb813b0ef8f135679020240fdca41c29976c23a.tar.gz
mlpdec: move rematrix_channels code to output_data()
Reviewed-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: James Almer <jamrial@gmail.com>
Diffstat (limited to 'libavcodec/mlpdec.c')
-rw-r--r--libavcodec/mlpdec.c45
1 files changed, 18 insertions, 27 deletions
diff --git a/libavcodec/mlpdec.c b/libavcodec/mlpdec.c
index e443f812bd..2c5426c242 100644
--- a/libavcodec/mlpdec.c
+++ b/libavcodec/mlpdec.c
@@ -1038,15 +1038,27 @@ static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
s->noisegen_seed = seed;
}
+/** Write the audio data into the output buffer. */
-/** Apply the channel matrices in turn to reconstruct the original audio
- * samples. */
-
-static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
+static int output_data(MLPDecodeContext *m, unsigned int substr,
+ AVFrame *frame, int *got_frame_ptr)
{
+ AVCodecContext *avctx = m->avctx;
SubStream *s = &m->substream[substr];
unsigned int mat;
unsigned int maxchan;
+ int ret;
+ int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
+
+ if (m->avctx->channels != s->max_matrix_channel + 1) {
+ av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (!s->blockpos) {
+ av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
+ return AVERROR_INVALIDDATA;
+ }
maxchan = s->max_matrix_channel;
if (!s->noise_type) {
@@ -1056,6 +1068,8 @@ static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
fill_noise_buffer(m, substr);
}
+ /* Apply the channel matrices in turn to reconstruct the original audio
+ * samples. */
for (mat = 0; mat < s->num_primitive_matrices; mat++) {
unsigned int dest_ch = s->matrix_out_ch[mat];
m->dsp.mlp_rematrix_channel(&m->sample_buffer[0][0],
@@ -1070,27 +1084,6 @@ static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
m->access_unit_size_pow2,
MSB_MASK(s->quant_step_size[dest_ch]));
}
-}
-
-/** Write the audio data into the output buffer. */
-
-static int output_data(MLPDecodeContext *m, unsigned int substr,
- AVFrame *frame, int *got_frame_ptr)
-{
- AVCodecContext *avctx = m->avctx;
- SubStream *s = &m->substream[substr];
- int ret;
- int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
-
- if (m->avctx->channels != s->max_matrix_channel + 1) {
- av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
- return AVERROR_INVALIDDATA;
- }
-
- if (!s->blockpos) {
- av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
- return AVERROR_INVALIDDATA;
- }
/* get output buffer */
frame->nb_samples = s->blockpos;
@@ -1298,8 +1291,6 @@ next_substr:
buf += substream_data_len[substr];
}
- rematrix_channels(m, m->max_decoded_substream);
-
if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
return ret;