diff options
Diffstat (limited to 'media/ffvpx/libavcodec/resample.c')
| -rw-r--r-- | media/ffvpx/libavcodec/resample.c | 439 |
1 files changed, 0 insertions, 439 deletions
diff --git a/media/ffvpx/libavcodec/resample.c b/media/ffvpx/libavcodec/resample.c deleted file mode 100644 index 4c5eb9f10e..0000000000 --- a/media/ffvpx/libavcodec/resample.c +++ /dev/null @@ -1,439 +0,0 @@ -/* - * 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 -FF_DISABLE_DEPRECATION_WARNINGS - -#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->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); -} - -FF_ENABLE_DEPRECATION_WARNINGS -#endif |