diff options
Diffstat (limited to 'libavcodec/resample.c')
| -rw-r--r-- | libavcodec/resample.c | 80 |
1 files changed, 68 insertions, 12 deletions
diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 20d7078113..dfaad66216 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -2,20 +2,20 @@ * samplerate conversion for both audio and video * Copyright (c) 2000 Fabrice Bellard * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * Libav is distributed in the hope that it will be useful, + * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -113,6 +113,39 @@ 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; + 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; @@ -152,6 +185,21 @@ static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) } } +#define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \ + ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0 + +static const uint8_t supported_resampling[MAX_CHANNELS] = { + // output ch: 1 2 3 4 5 6 7 8 + SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel + SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels + SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels + SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels + SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels + SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels + SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels + SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels +}; + ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, int output_rate, int input_rate, enum AVSampleFormat sample_fmt_out, @@ -167,12 +215,15 @@ ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, MAX_CHANNELS); return NULL; } - if (output_channels != input_channels && - (input_channels > 2 || - output_channels > 2 && - !(output_channels == 6 && input_channels == 2))) { - av_log(NULL, AV_LOG_ERROR, - "Resampling output channel count must be 1 or 2 for mono input; 1, 2 or 6 for stereo input; or N for N channel input.\n"); + if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) { + int i; + av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed " + "output channels for %d input channel%s", input_channels, + input_channels > 1 ? "s:" : ":"); + for (i = 0; i < MAX_CHANNELS; i++) + if (supported_resampling[input_channels-1] & (1<<i)) + av_log(NULL, AV_LOG_ERROR, " %d", i + 1); + av_log(NULL, AV_LOG_ERROR, "\n"); return NULL; } @@ -274,7 +325,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl input = s->buffer[0]; } - lenout = 4 * nb_samples * s->ratio + 16; + lenout= 2*s->output_channels*nb_samples * s->ratio + 16; if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) * @@ -308,6 +359,10 @@ 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]; @@ -337,7 +392,8 @@ 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); } |