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Diffstat (limited to 'libavcodec/resample.c')
| -rw-r--r-- | libavcodec/resample.c | 443 |
1 files changed, 443 insertions, 0 deletions
diff --git a/libavcodec/resample.c b/libavcodec/resample.c new file mode 100644 index 0000000000..c45aa16cd1 --- /dev/null +++ b/libavcodec/resample.c @@ -0,0 +1,443 @@ +/* + * 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 <string.h> + +#include "avcodec.h" +#include "audioconvert.h" +#include "libavutil/opt.h" +#include "libavutil/mem.h" +#include "libavutil/samplefmt.h" + +#if FF_API_AVCODEC_RESAMPLE + +#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) { + int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) * + s->output_channels; + output_bak = output; + + if (!s->buffer_size[1] || s->buffer_size[1] < out_size) { + av_free(s->buffer[1]); + s->buffer_size[1] = out_size; + 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_array((nb_samples + s->temp_len), sizeof(short)); + bufout[i] = av_malloc_array(lenout, sizeof(short)); + + if (!bufin[i] || !bufout[i]) { + av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); + nb_samples1 = 0; + goto fail; + } + + memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); + buftmp2[i] = bufin[i] + s->temp_len; + } + + 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_array(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 conversion failed\n"); + return 0; + } + } + +fail: + 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); +} + +#endif |