obs-studio/libobs/media-io/audio-resampler-ffmpeg.c

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/******************************************************************************
Copyright (C) 2013 by Hugh Bailey <obs.jim@gmail.com>
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 2 of the License, or
(at your option) any later version.
This program 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 General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
******************************************************************************/
#include "../util/bmem.h"
#include "audio-resampler.h"
#include "audio-io.h"
#include <libavutil/avutil.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
struct audio_resampler {
struct SwrContext *context;
bool opened;
uint32_t input_freq;
uint64_t input_layout;
enum AVSampleFormat input_format;
uint8_t *output_buffer[MAX_AV_PLANES];
uint64_t output_layout;
enum AVSampleFormat output_format;
int output_size;
uint32_t output_ch;
uint32_t output_freq;
uint32_t output_planes;
};
static inline enum AVSampleFormat convert_audio_format(enum audio_format format)
{
switch (format) {
case AUDIO_FORMAT_UNKNOWN: return AV_SAMPLE_FMT_S16;
case AUDIO_FORMAT_U8BIT: return AV_SAMPLE_FMT_U8;
case AUDIO_FORMAT_16BIT: return AV_SAMPLE_FMT_S16;
case AUDIO_FORMAT_32BIT: return AV_SAMPLE_FMT_S32;
case AUDIO_FORMAT_FLOAT: return AV_SAMPLE_FMT_FLT;
case AUDIO_FORMAT_U8BIT_PLANAR: return AV_SAMPLE_FMT_U8P;
case AUDIO_FORMAT_16BIT_PLANAR: return AV_SAMPLE_FMT_S16P;
case AUDIO_FORMAT_32BIT_PLANAR: return AV_SAMPLE_FMT_S32P;
case AUDIO_FORMAT_FLOAT_PLANAR: return AV_SAMPLE_FMT_FLTP;
}
/* shouldn't get here */
return AV_SAMPLE_FMT_S16;
}
static inline uint64_t convert_speaker_layout(enum speaker_layout layout)
{
switch (layout) {
case SPEAKERS_UNKNOWN: return 0;
case SPEAKERS_MONO: return AV_CH_LAYOUT_MONO;
case SPEAKERS_STEREO: return AV_CH_LAYOUT_STEREO;
case SPEAKERS_2POINT1: return AV_CH_LAYOUT_2_1;
case SPEAKERS_QUAD: return AV_CH_LAYOUT_QUAD;
case SPEAKERS_4POINT1: return AV_CH_LAYOUT_4POINT1;
case SPEAKERS_5POINT1: return AV_CH_LAYOUT_5POINT1;
case SPEAKERS_5POINT1_SURROUND: return AV_CH_LAYOUT_5POINT1_BACK;
case SPEAKERS_7POINT1: return AV_CH_LAYOUT_7POINT1;
case SPEAKERS_7POINT1_SURROUND: return AV_CH_LAYOUT_7POINT1_WIDE_BACK;
case SPEAKERS_SURROUND: return AV_CH_LAYOUT_SURROUND;
}
/* shouldn't get here */
return 0;
}
audio_resampler_t *audio_resampler_create(const struct resample_info *dst,
const struct resample_info *src)
{
struct audio_resampler *rs = bzalloc(sizeof(struct audio_resampler));
int errcode;
rs->opened = false;
rs->input_freq = src->samples_per_sec;
rs->input_layout = convert_speaker_layout(src->speakers);
rs->input_format = convert_audio_format(src->format);
rs->output_size = 0;
rs->output_ch = get_audio_channels(dst->speakers);
rs->output_freq = dst->samples_per_sec;
rs->output_layout = convert_speaker_layout(dst->speakers);
rs->output_format = convert_audio_format(dst->format);
rs->output_planes = is_audio_planar(dst->format) ? rs->output_ch : 1;
rs->context = swr_alloc_set_opts(NULL,
rs->output_layout, rs->output_format, dst->samples_per_sec,
rs->input_layout, rs->input_format, src->samples_per_sec,
0, NULL);
if (!rs->context) {
blog(LOG_ERROR, "swr_alloc_set_opts failed");
audio_resampler_destroy(rs);
return NULL;
}
errcode = swr_init(rs->context);
if (errcode != 0) {
blog(LOG_ERROR, "avresample_open failed: error code %d",
errcode);
audio_resampler_destroy(rs);
return NULL;
}
return rs;
}
void audio_resampler_destroy(audio_resampler_t *rs)
{
if (rs) {
if (rs->context)
swr_free(&rs->context);
if (rs->output_buffer)
av_freep(&rs->output_buffer[0]);
bfree(rs);
}
}
bool audio_resampler_resample(audio_resampler_t *rs,
uint8_t *output[], uint32_t *out_frames, uint64_t *ts_offset,
const uint8_t *const input[], uint32_t in_frames)
{
2014-02-23 21:39:33 -08:00
if (!rs) return false;
struct SwrContext *context = rs->context;
int ret;
int64_t delay = swr_get_delay(context, rs->input_freq);
int estimated = (int)av_rescale_rnd(
delay + (int64_t)in_frames,
(int64_t)rs->output_freq, (int64_t)rs->input_freq,
AV_ROUND_UP);
*ts_offset = (uint64_t)swr_get_delay(context, 1000000000);
/* resize the buffer if bigger */
if (estimated > rs->output_size) {
if (rs->output_buffer[0])
av_freep(&rs->output_buffer[0]);
av_samples_alloc(rs->output_buffer, NULL, rs->output_ch,
estimated, rs->output_format, 0);
rs->output_size = estimated;
}
ret = swr_convert(context,
rs->output_buffer, rs->output_size,
(const uint8_t**)input, in_frames);
if (ret < 0) {
blog(LOG_ERROR, "swr_convert failed: %d", ret);
return false;
}
for (uint32_t i = 0; i < rs->output_planes; i++)
output[i] = rs->output_buffer[i];
*out_frames = (uint32_t)ret;
return true;
}