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author | jan gerber <j@v2v.cc> | 2011-05-23 17:22:02 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2011-05-23 17:22:02 +0200 |
commit | 02260ccc3b7f8b88c11af4739252f5c5f97fa6e7 (patch) | |
tree | 03986041d02fff905c1008679b36b88e4b6def7b /libavcodec/resample.c | |
parent | 01a73d6cef70dfa90fdf79f7afd55b817b4c9342 (diff) | |
download | ffmpeg-02260ccc3b7f8b88c11af4739252f5c5f97fa6e7.tar.gz |
add 5.1 to stereo downmix to resample.c
this is based on previous 6to2channel-resample.patch from ffmpeg2theora
but updated to work with trunk and using av_clip_int16.
Diffstat (limited to 'libavcodec/resample.c')
-rw-r--r-- | libavcodec/resample.c | 40 |
1 files changed, 39 insertions, 1 deletions
diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 9e6defefdf..8c4eebe3c4 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -108,6 +108,39 @@ static void mono_to_stereo(short *output, short *input, int n1) } } +/* +5.1 to stereo input: [fl, fr, c, lfe, rl, rr] +- Left = front_left + rear_gain * rear_left + center_gain * center +- Right = front_right + rear_gain * rear_right + center_gain * center +Where rear_gain is usually around 0.5-1.0 and + center_gain is almost always 0.7 (-3 dB) +*/ +static void surround_to_stereo(short **output, short *input, int channels, int samples) +{ + int i; + short l, r; + + for (i = 0; i < samples; i++) { + int fl,fr,c,rl,rr,lfe; + fl = input[0]; + fr = input[1]; + c = input[2]; + lfe = input[3]; + rl = input[4]; + rr = input[5]; + + l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c)); + r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c)); + + /* output l & r. */ + *output[0]++ = l; + *output[1]++ = r; + + /* increment input. */ + input += channels; + } +} + static void deinterleave(short **output, short *input, int channels, int samples) { int i, j; @@ -301,6 +334,10 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl } else if (s->output_channels >= 2 && s->input_channels == 1) { buftmp3[0] = bufout[0]; memcpy(buftmp2[0], input, nb_samples * sizeof(short)); + } else if (s->input_channels == 6 && s->output_channels ==2) { + buftmp3[0] = bufout[0]; + buftmp3[1] = bufout[1]; + surround_to_stereo(buftmp2, input, s->input_channels, nb_samples); } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) { for (i = 0; i < s->input_channels; i++) { buftmp3[i] = bufout[i]; @@ -330,7 +367,8 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl mono_to_stereo(output, buftmp3[0], nb_samples1); } else if (s->output_channels == 6 && s->input_channels == 2) { ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); - } else if (s->output_channels == s->input_channels && s->input_channels >= 2) { + } else if ((s->output_channels == s->input_channels && s->input_channels >= 2) || + (s->output_channels == 2 && s->input_channels == 6)) { interleave(output, buftmp3, s->output_channels, nb_samples1); } |