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author | Michael Niedermayer <michaelni@gmx.at> | 2013-02-13 12:58:24 +0100 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2013-02-13 12:58:29 +0100 |
commit | 845fa2f5c9643a2f80b29614351609f1f1d3994d (patch) | |
tree | 7a3b1136b62cfaeb8889e1709a225c52ab207edb /libavresample | |
parent | 91043de8256fa87a0a20cc3a139f668110257dbe (diff) | |
parent | 157542ebc15dd175e5b4d14ffa92afd74ab4a991 (diff) | |
download | ffmpeg-845fa2f5c9643a2f80b29614351609f1f1d3994d.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
lavr: fix mixing matrix reduction when normalization is disabled
lavr: fix matrix reduction for upmixing in certain cases
lavr: cosmetics: reindent
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavresample')
-rw-r--r-- | libavresample/audio_mix.c | 49 |
1 files changed, 35 insertions, 14 deletions
diff --git a/libavresample/audio_mix.c b/libavresample/audio_mix.c index fc37eac6a3..b69bfbcf3e 100644 --- a/libavresample/audio_mix.c +++ b/libavresample/audio_mix.c @@ -572,11 +572,22 @@ static void reduce_matrix(AudioMix *am, const double *matrix, int stride) int skip = 1; for (o = 0; o < am->out_channels; o++) { + int i0; if ((o != i && matrix[o * stride + i] != 0.0) || (o == i && matrix[o * stride + i] != 1.0)) { skip = 0; break; } + /* if the input contributes fully to the output, also check that no + other inputs contribute to this output */ + if (o == i) { + for (i0 = 0; i0 < am->in_channels; i0++) { + if (i0 != i && matrix[o * stride + i0] != 0.0) { + skip = 0; + break; + } + } + } } if (skip) { am->input_skip[i] = 1; @@ -607,6 +618,7 @@ static void reduce_matrix(AudioMix *am, const double *matrix, int stride) corresponding input channel */ for (o = 0; o < FFMIN(am->in_channels, am->out_channels); o++) { int skip = 1; + int o0; for (i = 0; i < am->in_channels; i++) { if ((o != i && matrix[o * stride + i] != 0.0) || @@ -615,6 +627,15 @@ static void reduce_matrix(AudioMix *am, const double *matrix, int stride) break; } } + /* check if the corresponding input channel makes a contribution to + any other output channel */ + i = o; + for (o0 = 0; o0 < am->out_channels; o0++) { + if (o0 != i && matrix[o0 * stride + i] != 0.0) { + skip = 0; + break; + } + } if (skip) { am->output_skip[o] = 1; am->out_matrix_channels--; @@ -673,20 +694,20 @@ int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride) am->matrix = (void **)am->matrix_## type; if (am->in_matrix_channels && am->out_matrix_channels) { - switch (am->coeff_type) { - case AV_MIX_COEFF_TYPE_Q8: - CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v))) - break; - case AV_MIX_COEFF_TYPE_Q15: - CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v))) - break; - case AV_MIX_COEFF_TYPE_FLT: - CONVERT_MATRIX(flt, v) - break; - default: - av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n"); - return AVERROR(EINVAL); - } + switch (am->coeff_type) { + case AV_MIX_COEFF_TYPE_Q8: + CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v))) + break; + case AV_MIX_COEFF_TYPE_Q15: + CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v))) + break; + case AV_MIX_COEFF_TYPE_FLT: + CONVERT_MATRIX(flt, v) + break; + default: + av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n"); + return AVERROR(EINVAL); + } } ret = mix_function_init(am); |