* 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
*/
}
}
+/*
+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;
}
}
+#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] = {
+ //ouput channels: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,
MAX_CHANNELS);
return NULL;
}
- if (output_channels > 2 &&
- !(output_channels == 6 && input_channels == 2) &&
- output_channels != input_channels) {
- 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;
}
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) {
output_bak = output;
} 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];
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);
}
OSDN Git Service
