obs-studio/plugins/aja/audio-repack.c
Paul Hindt 003241511c aja: Add audio channel selection to capture
Adapted audio repacker for 32-bit samples,

with an ifdef to use SIMD or C version.
2023-02-15 10:38:39 -05:00

269 lines
7.1 KiB
C

#include "audio-repack.h"
#define USE_SIMD
#ifdef USE_SIMD
#include <util/sse-intrin.h>
#endif
#define NUM_CHANNELS 8 /* max until OBS supports higher channel counts */
int check_buffer(struct audio_repack *repack, uint32_t frame_count)
{
const uint32_t new_size =
frame_count * repack->base_dst_size + repack->pad_dst_size;
if (repack->packet_size < new_size) {
repack->packet_buffer =
brealloc(repack->packet_buffer, new_size);
if (!repack->packet_buffer)
return -1;
repack->packet_size = new_size;
}
return 0;
}
#ifdef USE_SIMD
/*
* Squash array of 8ch to new channel count
* 16-bit PCM, SIMD version
* For instance:
* 2.1:
*
* | FL | FR | LFE | emp | emp | emp |emp |emp |
* | | |
* | FL | FR | LFE |
*/
int repack_squash16(struct audio_repack *repack, const uint8_t *bsrc,
uint32_t frame_count)
{
if (check_buffer(repack, frame_count) < 0)
return -1;
int squash = repack->squash_count;
const __m128i *src = (__m128i *)bsrc;
const __m128i *end = src + frame_count;
uint16_t *dst = (uint16_t *)repack->packet_buffer;
/* Audio needs squashing in order to avoid resampling issues.
* The condition checks for 7.1 audio for which no squash is needed.
*/
if (squash > 0) {
while (src != end) {
__m128i target = _mm_load_si128(src++);
_mm_storeu_si128((__m128i *)dst, target);
dst += NUM_CHANNELS - squash;
}
}
return 0;
}
/*
* Squash array of 8ch and swap Front Center channel with LFE
* 16-bit PCM, SIMD version
* For instance:
* 2.1:
*
* | FL | FR | FC | LFE | RL | RR | LC | RC |
* | | |
* | FL | FR | LFE | FC | RL | RR | LC | RC |
*/
int repack_squash_swap16(struct audio_repack *repack, const uint8_t *bsrc,
uint32_t frame_count)
{
if (check_buffer(repack, frame_count) < 0)
return -1;
int squash = repack->squash_count;
const __m128i *src = (__m128i *)bsrc;
const __m128i *end = src + frame_count;
uint16_t *dst = (uint16_t *)repack->packet_buffer;
while (src != end) {
__m128i target = _mm_load_si128(src++);
__m128i buf =
_mm_shufflelo_epi16(target, _MM_SHUFFLE(2, 3, 1, 0));
_mm_storeu_si128((__m128i *)dst, buf);
dst += NUM_CHANNELS - squash;
}
return 0;
}
/*
* Squash array of 8ch to new channel count
* 32-bit PCM, SIMD version
*/
int repack_squash32(struct audio_repack *repack, const uint8_t *bsrc,
uint32_t frame_count)
{
if (check_buffer(repack, frame_count) < 0)
return -1;
int squash = repack->squash_count;
const __m128i *src = (__m128i *)bsrc;
const __m128i *end =
(__m128i *)(bsrc + (frame_count * repack->base_src_size));
uint32_t *dst = (uint32_t *)repack->packet_buffer;
if (squash > 0) {
while (src != end) {
__m128i tgt_lo = _mm_loadu_si128(src++);
__m128i tgt_hi = _mm_loadu_si128(src++);
_mm_storeu_si128((__m128i *)dst, tgt_lo);
dst += 4;
_mm_storeu_si128((__m128i *)dst, tgt_hi);
dst += 4;
dst -= squash;
}
}
return 0;
}
/*
* Squash array of 8ch to new channel count and swap FC with LFE
* 32-bit PCM, SIMD version
*/
int repack_squash_swap32(struct audio_repack *repack, const uint8_t *bsrc,
uint32_t frame_count)
{
if (check_buffer(repack, frame_count) < 0)
return -1;
int squash = repack->squash_count;
const __m128i *src = (__m128i *)bsrc;
const __m128i *end =
(__m128i *)(bsrc + (frame_count * repack->base_src_size));
uint32_t *dst = (uint32_t *)repack->packet_buffer;
while (src != end) {
__m128i tgt_lo = _mm_shuffle_epi32(_mm_loadu_si128(src++),
_MM_SHUFFLE(2, 3, 1, 0));
__m128i tgt_hi = _mm_loadu_si128(src++);
_mm_storeu_si128((__m128i *)dst, tgt_lo);
dst += 4;
_mm_storeu_si128((__m128i *)dst, tgt_hi);
dst += 4;
dst -= squash;
}
return 0;
}
#else
/*
* Squash array of 8ch to new channel count
* 16-bit or 32-bit PCM, C version
*/
int repack_squash(struct audio_repack *repack, const uint8_t *bsrc,
uint32_t frame_count)
{
if (check_buffer(repack, frame_count) < 0)
return -1;
int squash = repack->squash_count;
const uint8_t *src = bsrc;
const uint8_t *end = src + frame_count * repack->base_src_size;
uint8_t *dst = repack->packet_buffer;
uint32_t new_channel_count = NUM_CHANNELS - squash;
if (squash > 0) {
while (src != end) {
memcpy(dst, src,
repack->bytes_per_sample * new_channel_count);
dst += (new_channel_count * repack->bytes_per_sample);
src += NUM_CHANNELS * repack->bytes_per_sample;
}
}
return 0;
}
void shuffle_8ch(uint8_t *dst, const uint8_t *src, size_t szb, int ch1, int ch2,
int ch3, int ch4, int ch5, int ch6, int ch7, int ch8)
{
/* shuffle 8 channels of audio */
for (size_t i = 0; i < szb; i++) {
dst[0 * szb + i] = src[ch1 * szb + i];
dst[1 * szb + i] = src[ch2 * szb + i];
dst[2 * szb + i] = src[ch3 * szb + i];
dst[3 * szb + i] = src[ch4 * szb + i];
dst[4 * szb + i] = src[ch5 * szb + i];
dst[5 * szb + i] = src[ch6 * szb + i];
dst[6 * szb + i] = src[ch7 * szb + i];
dst[7 * szb + i] = src[ch8 * szb + i];
}
}
/*
* Squash array of 8ch to new channel count and swap FC with LFE
* 16-bit or 32-bit PCM, C version
*/
int repack_squash_swap(struct audio_repack *repack, const uint8_t *bsrc,
uint32_t frame_count)
{
if (check_buffer(repack, frame_count) < 0)
return -1;
int squash = repack->squash_count;
const uint8_t *src = bsrc;
const uint8_t *end = src + (frame_count * repack->base_src_size);
uint8_t *dst = repack->packet_buffer;
uint32_t new_channel_count = NUM_CHANNELS - squash;
if (squash > 0) {
while (src != end) {
shuffle_8ch(dst, src, 4, 0, 1, 3, 2, 4, 5, 6, 7);
dst += (new_channel_count * repack->bytes_per_sample);
src += (NUM_CHANNELS * repack->bytes_per_sample);
}
}
return 0;
}
#endif
int audio_repack_init(struct audio_repack *repack,
audio_repack_mode_t repack_mode, uint8_t bits_per_sample)
{
memset(repack, 0, sizeof(*repack));
if (bits_per_sample != 16 && bits_per_sample != 32)
return -1;
int _audio_repack_ch[9] = {2, 3, 4, 5, 6, 5, 6, 8, 8};
int bytes_per_sample = (bits_per_sample / 8);
repack->bytes_per_sample = bytes_per_sample;
repack->base_src_size = NUM_CHANNELS * bytes_per_sample;
repack->base_dst_size =
_audio_repack_ch[repack_mode] * bytes_per_sample;
uint32_t squash_count = NUM_CHANNELS - _audio_repack_ch[repack_mode];
repack->pad_dst_size = squash_count * bytes_per_sample;
repack->squash_count = squash_count;
#ifdef USE_SIMD
if (bits_per_sample == 16) {
repack->repack_func = &repack_squash16;
if (repack_mode == repack_mode_8to5ch_swap ||
repack_mode == repack_mode_8to6ch_swap ||
repack_mode == repack_mode_8ch_swap)
repack->repack_func = &repack_squash_swap16;
} else if (bits_per_sample == 32) {
repack->repack_func = &repack_squash32;
if (repack_mode == repack_mode_8to5ch_swap ||
repack_mode == repack_mode_8to6ch_swap ||
repack_mode == repack_mode_8ch_swap)
repack->repack_func = &repack_squash_swap32;
}
#else
repack->repack_func = &repack_squash;
if (repack_mode == repack_mode_8to5ch_swap ||
repack_mode == repack_mode_8to6ch_swap ||
repack_mode == repack_mode_8ch_swap)
repack->repack_func = &repack_squash_swap;
#endif
return 0;
}
void audio_repack_free(struct audio_repack *repack)
{
if (repack->packet_buffer)
bfree(repack->packet_buffer);
memset(repack, 0, sizeof(*repack));
}