/* * samplerate conversion for both audio and video * Copyright (c) 2000 Fabrice Bellard * * This file is part of FFmpeg. * * 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. * * 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * samplerate conversion for both audio and video */ #include "avcodec.h" #include "audioconvert.h" #include "libavutil/opt.h" #include "libavutil/samplefmt.h" #define MAX_CHANNELS 8 struct AVResampleContext; static const char *context_to_name(void *ptr) { return "audioresample"; } static const AVOption options[] = {{NULL}}; static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT }; struct ReSampleContext { struct AVResampleContext *resample_context; short *temp[MAX_CHANNELS]; int temp_len; float ratio; /* channel convert */ int input_channels, output_channels, filter_channels; AVAudioConvert *convert_ctx[2]; enum AVSampleFormat sample_fmt[2]; ///< input and output sample format unsigned sample_size[2]; ///< size of one sample in sample_fmt short *buffer[2]; ///< buffers used for conversion to S16 unsigned buffer_size[2]; ///< sizes of allocated buffers }; /* n1: number of samples */ static void stereo_to_mono(short *output, short *input, int n1) { short *p, *q; int n = n1; p = input; q = output; while (n >= 4) { q[0] = (p[0] + p[1]) >> 1; q[1] = (p[2] + p[3]) >> 1; q[2] = (p[4] + p[5]) >> 1; q[3] = (p[6] + p[7]) >> 1; q += 4; p += 8; n -= 4; } while (n > 0) { q[0] = (p[0] + p[1]) >> 1; q++; p += 2; n--; } } /* n1: number of samples */ static void mono_to_stereo(short *output, short *input, int n1) { short *p, *q; int n = n1; int v; p = input; q = output; while (n >= 4) { v = p[0]; q[0] = v; q[1] = v; v = p[1]; q[2] = v; q[3] = v; v = p[2]; q[4] = v; q[5] = v; v = p[3]; q[6] = v; q[7] = v; q += 8; p += 4; n -= 4; } while (n > 0) { v = p[0]; q[0] = v; q[1] = v; q += 2; p += 1; n--; } } /* 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; for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *output[j]++ = *input++; } } } static void interleave(short *output, short **input, int channels, int samples) { int i, j; for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *output++ = *input[j]++; } } } static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) { int i; short l, r; for (i = 0; i < n; i++) { l = *input1++; r = *input2++; *output++ = l; /* left */ *output++ = (l / 2) + (r / 2); /* center */ *output++ = r; /* right */ *output++ = 0; /* left surround */ *output++ = 0; /* right surroud */ *output++ = 0; /* low freq */ } } #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, enum AVSampleFormat sample_fmt_in, int filter_length, int log2_phase_count, int linear, double cutoff) { ReSampleContext *s; if (input_channels > MAX_CHANNELS) { av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than %d is unsupported.\n", MAX_CHANNELS); return NULL; } 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; } s = av_mallocz(sizeof(ReSampleContext)); if (!s) { av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n"); return NULL; } s->ratio = (float)output_rate / (float)input_rate; s->input_channels = input_channels; s->output_channels = output_channels; s->filter_channels = s->input_channels; if (s->output_channels < s->filter_channels) s->filter_channels = s->output_channels; s->sample_fmt[0] = sample_fmt_in; s->sample_fmt[1] = sample_fmt_out; s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]); s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]); if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) { if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1, s->sample_fmt[0], 1, NULL, 0))) { av_log(s, AV_LOG_ERROR, "Cannot convert %s sample format to s16 sample format\n", av_get_sample_fmt_name(s->sample_fmt[0])); av_free(s); return NULL; } } if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1, AV_SAMPLE_FMT_S16, 1, NULL, 0))) { av_log(s, AV_LOG_ERROR, "Cannot convert s16 sample format to %s sample format\n", av_get_sample_fmt_name(s->sample_fmt[1])); av_audio_convert_free(s->convert_ctx[0]); av_free(s); return NULL; } } s->resample_context = av_resample_init(output_rate, input_rate, filter_length, log2_phase_count, linear, cutoff); *(const AVClass**)s->resample_context = &audioresample_context_class; return s; } /* resample audio. 'nb_samples' is the number of input samples */ /* XXX: optimize it ! */ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) { int i, nb_samples1; short *bufin[MAX_CHANNELS]; short *bufout[MAX_CHANNELS]; short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS]; short *output_bak = NULL; int lenout; if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) { /* nothing to do */ memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); return nb_samples; } if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) { int istride[1] = { s->sample_size[0] }; int ostride[1] = { 2 }; const void *ibuf[1] = { input }; void *obuf[1]; unsigned input_size = nb_samples * s->input_channels * 2; if (!s->buffer_size[0] || s->buffer_size[0] < input_size) { av_free(s->buffer[0]); s->buffer_size[0] = input_size; s->buffer[0] = av_malloc(s->buffer_size[0]); if (!s->buffer[0]) { av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); return 0; } } obuf[0] = s->buffer[0]; if (av_audio_convert(s->convert_ctx[0], obuf, ostride, ibuf, istride, nb_samples * s->input_channels) < 0) { av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n"); return 0; } input = s->buffer[0]; } lenout= 2*s->output_channels*nb_samples * s->ratio + 16; if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { output_bak = output; if (!s->buffer_size[1] || s->buffer_size[1] < 2*lenout) { av_free(s->buffer[1]); s->buffer_size[1] = 2*lenout; s->buffer[1] = av_malloc(s->buffer_size[1]); if (!s->buffer[1]) { av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); return 0; } } output = s->buffer[1]; } /* XXX: move those malloc to resample init code */ for (i = 0; i < s->filter_channels; i++) { bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short)); memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); buftmp2[i] = bufin[i] + s->temp_len; bufout[i] = av_malloc(lenout * sizeof(short)); } if (s->input_channels == 2 && s->output_channels == 1) { buftmp3[0] = output; stereo_to_mono(buftmp2[0], input, nb_samples); } 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]; } deinterleave(buftmp2, input, s->input_channels, nb_samples); } else { buftmp3[0] = output; memcpy(buftmp2[0], input, nb_samples * sizeof(short)); } nb_samples += s->temp_len; /* resample each channel */ nb_samples1 = 0; /* avoid warning */ for (i = 0; i < s->filter_channels; i++) { int consumed; int is_last = i + 1 == s->filter_channels; nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last); s->temp_len = nb_samples - consumed; s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short)); memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short)); } if (s->output_channels == 2 && s->input_channels == 1) { 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) || (s->output_channels == 2 && s->input_channels == 6)) { interleave(output, buftmp3, s->output_channels, nb_samples1); } if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { int istride[1] = { 2 }; int ostride[1] = { s->sample_size[1] }; const void *ibuf[1] = { output }; void *obuf[1] = { output_bak }; if (av_audio_convert(s->convert_ctx[1], obuf, ostride, ibuf, istride, nb_samples1 * s->output_channels) < 0) { av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n"); return 0; } } for (i = 0; i < s->filter_channels; i++) { av_free(bufin[i]); av_free(bufout[i]); } return nb_samples1; } void audio_resample_close(ReSampleContext *s) { int i; av_resample_close(s->resample_context); for (i = 0; i < s->filter_channels; i++) av_freep(&s->temp[i]); av_freep(&s->buffer[0]); av_freep(&s->buffer[1]); av_audio_convert_free(s->convert_ctx[0]); av_audio_convert_free(s->convert_ctx[1]); av_free(s); }