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authorJustin Ruggles <justin.ruggles@gmail.com>2012-03-23 17:42:17 -0400
committerJustin Ruggles <justin.ruggles@gmail.com>2012-04-24 21:28:27 -0400
commitc8af852b97447491823ff9b91413e32415e2babf (patch)
tree6c02f850cf954612c7077f266a75d663bb9cde57 /libavresample/audio_mix.c
parentc5671aeb77abb18a5a10ace314ab49e8fd3d0cb3 (diff)
downloadffmpeg-streaming-c8af852b97447491823ff9b91413e32415e2babf.zip
ffmpeg-streaming-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.c356
1 files changed, 356 insertions, 0 deletions
diff --git a/libavresample/audio_mix.c b/libavresample/audio_mix.c
new file mode 100644
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--- /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;
+}
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