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author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-03-23 17:42:17 -0400 |
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committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-04-24 21:28:27 -0400 |
commit | c8af852b97447491823ff9b91413e32415e2babf (patch) | |
tree | 6c02f850cf954612c7077f266a75d663bb9cde57 /libavresample/audio_mix.c | |
parent | c5671aeb77abb18a5a10ace314ab49e8fd3d0cb3 (diff) | |
download | ffmpeg-c8af852b97447491823ff9b91413e32415e2babf.tar.gz |
Add libavresample
This is a new library for audio sample format, channel layout, and sample rate
conversion.
Diffstat (limited to 'libavresample/audio_mix.c')
-rw-r--r-- | libavresample/audio_mix.c | 356 |
1 files changed, 356 insertions, 0 deletions
diff --git a/libavresample/audio_mix.c b/libavresample/audio_mix.c new file mode 100644 index 0000000000..34252bf68d --- /dev/null +++ b/libavresample/audio_mix.c @@ -0,0 +1,356 @@ +/* + * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> + * + * This file is part of Libav. + * + * Libav 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, + * 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 + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <stdint.h> + +#include "libavutil/libm.h" +#include "libavutil/samplefmt.h" +#include "avresample.h" +#include "internal.h" +#include "audio_data.h" +#include "audio_mix.h" + +static const char *coeff_type_names[] = { "q6", "q15", "flt" }; + +void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt, + enum AVMixCoeffType coeff_type, int in_channels, + int out_channels, int ptr_align, int samples_align, + const char *descr, void *mix_func) +{ + if (fmt == am->fmt && coeff_type == am->coeff_type && + ( in_channels == am->in_channels || in_channels == 0) && + (out_channels == am->out_channels || out_channels == 0)) { + char chan_str[16]; + am->mix = mix_func; + am->func_descr = descr; + am->ptr_align = ptr_align; + am->samples_align = samples_align; + if (ptr_align == 1 && samples_align == 1) { + am->mix_generic = mix_func; + am->func_descr_generic = descr; + } else { + am->has_optimized_func = 1; + } + if (in_channels) { + if (out_channels) + snprintf(chan_str, sizeof(chan_str), "[%d to %d] ", + in_channels, out_channels); + else + snprintf(chan_str, sizeof(chan_str), "[%d to any] ", + in_channels); + } else if (out_channels) { + snprintf(chan_str, sizeof(chan_str), "[any to %d] ", + out_channels); + } + av_log(am->avr, AV_LOG_DEBUG, "audio_mix: found function: [fmt=%s] " + "[c=%s] %s(%s)\n", av_get_sample_fmt_name(fmt), + coeff_type_names[coeff_type], + (in_channels || out_channels) ? chan_str : "", descr); + } +} + +#define MIX_FUNC_NAME(fmt, cfmt) mix_any_ ## fmt ##_## cfmt ##_c + +#define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr) \ +static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \ + int len, int out_ch, int in_ch) \ +{ \ + int i, in, out; \ + stype temp[AVRESAMPLE_MAX_CHANNELS]; \ + for (i = 0; i < len; i++) { \ + for (out = 0; out < out_ch; out++) { \ + sumtype sum = 0; \ + for (in = 0; in < in_ch; in++) \ + sum += samples[in][i] * matrix[out][in]; \ + temp[out] = expr; \ + } \ + for (out = 0; out < out_ch; out++) \ + samples[out][i] = temp[out]; \ + } \ +} + +MIX_FUNC_GENERIC(FLTP, FLT, float, float, float, sum) +MIX_FUNC_GENERIC(S16P, FLT, int16_t, float, float, av_clip_int16(lrintf(sum))) +MIX_FUNC_GENERIC(S16P, Q15, int16_t, int32_t, int64_t, av_clip_int16(sum >> 15)) +MIX_FUNC_GENERIC(S16P, Q6, int16_t, int16_t, int32_t, av_clip_int16(sum >> 6)) + +/* TODO: templatize the channel-specific C functions */ + +static void mix_2_to_1_fltp_flt_c(float **samples, float **matrix, int len, + int out_ch, int in_ch) +{ + float *src0 = samples[0]; + float *src1 = samples[1]; + float *dst = src0; + float m0 = matrix[0][0]; + float m1 = matrix[0][1]; + + while (len > 4) { + *dst++ = *src0++ * m0 + *src1++ * m1; + *dst++ = *src0++ * m0 + *src1++ * m1; + *dst++ = *src0++ * m0 + *src1++ * m1; + *dst++ = *src0++ * m0 + *src1++ * m1; + len -= 4; + } + while (len > 0) { + *dst++ = *src0++ * m0 + *src1++ * m1; + len--; + } +} + +static void mix_1_to_2_fltp_flt_c(float **samples, float **matrix, int len, + int out_ch, int in_ch) +{ + float v; + float *dst0 = samples[0]; + float *dst1 = samples[1]; + float *src = dst0; + float m0 = matrix[0][0]; + float m1 = matrix[1][0]; + + while (len > 4) { + v = *src++; + *dst0++ = v * m1; + *dst1++ = v * m0; + v = *src++; + *dst0++ = v * m1; + *dst1++ = v * m0; + v = *src++; + *dst0++ = v * m1; + *dst1++ = v * m0; + v = *src++; + *dst0++ = v * m1; + *dst1++ = v * m0; + len -= 4; + } + while (len > 0) { + v = *src++; + *dst0++ = v * m1; + *dst1++ = v * m0; + len--; + } +} + +static void mix_6_to_2_fltp_flt_c(float **samples, float **matrix, int len, + int out_ch, int in_ch) +{ + float v0, v1; + float *src0 = samples[0]; + float *src1 = samples[1]; + float *src2 = samples[2]; + float *src3 = samples[3]; + float *src4 = samples[4]; + float *src5 = samples[5]; + float *dst0 = src0; + float *dst1 = src1; + float *m0 = matrix[0]; + float *m1 = matrix[1]; + + while (len > 0) { + v0 = *src0++; + v1 = *src1++; + *dst0++ = v0 * m0[0] + + v1 * m0[1] + + *src2 * m0[2] + + *src3 * m0[3] + + *src4 * m0[4] + + *src5 * m0[5]; + *dst1++ = v0 * m1[0] + + v1 * m1[1] + + *src2++ * m1[2] + + *src3++ * m1[3] + + *src4++ * m1[4] + + *src5++ * m1[5]; + len--; + } +} + +static void mix_2_to_6_fltp_flt_c(float **samples, float **matrix, int len, + int out_ch, int in_ch) +{ + float v0, v1; + float *dst0 = samples[0]; + float *dst1 = samples[1]; + float *dst2 = samples[2]; + float *dst3 = samples[3]; + float *dst4 = samples[4]; + float *dst5 = samples[5]; + float *src0 = dst0; + float *src1 = dst1; + + while (len > 0) { + v0 = *src0++; + v1 = *src1++; + *dst0++ = v0 * matrix[0][0] + v1 * matrix[0][1]; + *dst1++ = v0 * matrix[1][0] + v1 * matrix[1][1]; + *dst2++ = v0 * matrix[2][0] + v1 * matrix[2][1]; + *dst3++ = v0 * matrix[3][0] + v1 * matrix[3][1]; + *dst4++ = v0 * matrix[4][0] + v1 * matrix[4][1]; + *dst5++ = v0 * matrix[5][0] + v1 * matrix[5][1]; + len--; + } +} + +static int mix_function_init(AudioMix *am) +{ + /* any-to-any C versions */ + + ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, + 0, 0, 1, 1, "C", MIX_FUNC_NAME(FLTP, FLT)); + + ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT, + 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, FLT)); + + ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q15, + 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q15)); + + ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q6, + 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q6)); + + /* channel-specific C versions */ + + ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, + 2, 1, 1, 1, "C", mix_2_to_1_fltp_flt_c); + + ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, + 1, 2, 1, 1, "C", mix_1_to_2_fltp_flt_c); + + ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, + 6, 2, 1, 1, "C", mix_6_to_2_fltp_flt_c); + + ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, + 2, 6, 1, 1, "C", mix_2_to_6_fltp_flt_c); + + if (ARCH_X86) + ff_audio_mix_init_x86(am); + + if (!am->mix) { + av_log(am->avr, AV_LOG_ERROR, "audio_mix: NO FUNCTION FOUND: [fmt=%s] " + "[c=%s] [%d to %d]\n", av_get_sample_fmt_name(am->fmt), + coeff_type_names[am->coeff_type], am->in_channels, + am->out_channels); + return AVERROR_PATCHWELCOME; + } + return 0; +} + +int ff_audio_mix_init(AVAudioResampleContext *avr) +{ + int ret; + + /* build matrix if the user did not already set one */ + if (!avr->am->matrix) { + int i, j; + char in_layout_name[128]; + char out_layout_name[128]; + double *matrix_dbl = av_mallocz(avr->out_channels * avr->in_channels * + sizeof(*matrix_dbl)); + if (!matrix_dbl) + return AVERROR(ENOMEM); + + ret = avresample_build_matrix(avr->in_channel_layout, + avr->out_channel_layout, + avr->center_mix_level, + avr->surround_mix_level, + avr->lfe_mix_level, 1, matrix_dbl, + avr->in_channels); + if (ret < 0) { + av_free(matrix_dbl); + return ret; + } + + av_get_channel_layout_string(in_layout_name, sizeof(in_layout_name), + avr->in_channels, avr->in_channel_layout); + av_get_channel_layout_string(out_layout_name, sizeof(out_layout_name), + avr->out_channels, avr->out_channel_layout); + av_log(avr, AV_LOG_DEBUG, "audio_mix: %s to %s\n", + in_layout_name, out_layout_name); + for (i = 0; i < avr->out_channels; i++) { + for (j = 0; j < avr->in_channels; j++) { + av_log(avr, AV_LOG_DEBUG, " %0.3f ", + matrix_dbl[i * avr->in_channels + j]); + } + av_log(avr, AV_LOG_DEBUG, "\n"); + } + + ret = avresample_set_matrix(avr, matrix_dbl, avr->in_channels); + if (ret < 0) { + av_free(matrix_dbl); + return ret; + } + av_free(matrix_dbl); + } + + avr->am->fmt = avr->internal_sample_fmt; + avr->am->coeff_type = avr->mix_coeff_type; + avr->am->in_layout = avr->in_channel_layout; + avr->am->out_layout = avr->out_channel_layout; + avr->am->in_channels = avr->in_channels; + avr->am->out_channels = avr->out_channels; + + ret = mix_function_init(avr->am); + if (ret < 0) + return ret; + + return 0; +} + +void ff_audio_mix_close(AudioMix *am) +{ + if (!am) + return; + if (am->matrix) { + av_free(am->matrix[0]); + am->matrix = NULL; + } + memset(am->matrix_q6, 0, sizeof(am->matrix_q6 )); + memset(am->matrix_q15, 0, sizeof(am->matrix_q15)); + memset(am->matrix_flt, 0, sizeof(am->matrix_flt)); +} + +int ff_audio_mix(AudioMix *am, AudioData *src) +{ + int use_generic = 1; + int len = src->nb_samples; + + /* determine whether to use the optimized function based on pointer and + samples alignment in both the input and output */ + if (am->has_optimized_func) { + int aligned_len = FFALIGN(len, am->samples_align); + if (!(src->ptr_align % am->ptr_align) && + src->samples_align >= aligned_len) { + len = aligned_len; + use_generic = 0; + } + } + av_dlog(am->avr, "audio_mix: %d samples - %d to %d channels (%s)\n", + src->nb_samples, am->in_channels, am->out_channels, + use_generic ? am->func_descr_generic : am->func_descr); + + if (use_generic) + am->mix_generic(src->data, am->matrix, len, am->out_channels, + am->in_channels); + else + am->mix(src->data, am->matrix, len, am->out_channels, am->in_channels); + + ff_audio_data_set_channels(src, am->out_channels); + + return 0; +} |