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add audio resampling, add ffmpeg support

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jp9000
2013-10-30 17:07:01 -07:00
parent 3adfaf7f9f
commit 60e6ddbf71
9 changed files with 556 additions and 37 deletions
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35
libobs/media-io/audio-io.h
View File

@@ -34,16 +34,15 @@ struct audio_line;
typedef struct audio_output *audio_t;
typedef struct audio_line *audio_line_t;
enum audio_type {
enum audio_format {
AUDIO_FORMAT_UNKNOWN,
AUDIO_FORMAT_8BIT,
AUDIO_FORMAT_U8BIT,
AUDIO_FORMAT_16BIT,
AUDIO_FORMAT_24BIT,
AUDIO_FORMAT_32BIT,
AUDIO_FORMAT_FLOAT,
};
enum speaker_setup {
enum speaker_layout {
SPEAKERS_UNKNOWN,
SPEAKERS_MONO,
SPEAKERS_STEREO,
@@ -58,22 +57,21 @@ enum speaker_setup {
};
struct audio_data {
const void *data;
uint32_t frames;
uint64_t timestamp;
const void *data;
uint32_t frames;
uint64_t timestamp;
};
struct audio_info {
const char *name;
const char *format;
const char *name;
const char *format;
uint32_t samples_per_sec;
enum audio_type type;
enum speaker_setup speakers;
uint32_t samples_per_sec;
enum audio_format type;
enum speaker_layout speakers;
};
static inline uint32_t get_audio_channels(enum speaker_setup speakers)
static inline uint32_t get_audio_channels(enum speaker_layout speakers)
{
switch (speakers) {
case SPEAKERS_MONO: return 1;
@@ -92,12 +90,11 @@ static inline uint32_t get_audio_channels(enum speaker_setup speakers)
return 0;
}
static inline size_t get_audio_bytes_per_channel(enum audio_type type)
static inline size_t get_audio_bytes_per_channel(enum audio_format type)
{
switch (type) {
case AUDIO_FORMAT_8BIT: return 1;
case AUDIO_FORMAT_U8BIT: return 1;
case AUDIO_FORMAT_16BIT: return 2;
case AUDIO_FORMAT_24BIT: return 3;
case AUDIO_FORMAT_FLOAT:
case AUDIO_FORMAT_32BIT: return 4;
case AUDIO_FORMAT_UNKNOWN: return 0;
@@ -106,8 +103,8 @@ static inline size_t get_audio_bytes_per_channel(enum audio_type type)
return 0;
}
static inline size_t get_audio_size(enum audio_type type,
enum speaker_setup speakers, uint32_t frames)
static inline size_t get_audio_size(enum audio_format type,
enum speaker_layout speakers, uint32_t frames)
{
return get_audio_channels(speakers) *
get_audio_bytes_per_channel(type) *

170
libobs/media-io/audio-resampler-ffmpeg.c Normal file
View File

@@ -0,0 +1,170 @@
/******************************************************************************
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 3 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 <libswresample/swresample.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.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;
uint64_t output_layout;
enum AVSampleFormat output_format;
int output_size;
uint32_t output_ch;
uint32_t output_freq;
};
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;
}
/* 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(struct resample_info *dst,
struct resample_info *src)
{
struct audio_resampler *rs = bmalloc(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_buffer = NULL;
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->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;
}
av_opt_set_channel_layout(rs->context, "in_channel_layout",
rs->input_layout, 0);
av_opt_set_channel_layout(rs->context, "out_channel_layout",
rs->output_layout, 0);
av_opt_set_int(rs->context, "in_sample_rate", src->samples_per_sec, 0);
av_opt_set_int(rs->context, "out_sample_rate", dst->samples_per_sec, 0);
av_opt_set_sample_fmt(rs->context, "in_sample_fmt",
rs->input_format, 0);
av_opt_set_sample_fmt(rs->context, "out_sample_fmt",
rs->output_format, 0);
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);
bfree(rs);
}
}
bool audio_resampler_resample(audio_resampler_t rs,
void **output, uint32_t *out_frames,
void *input, uint32_t in_frames)
{
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);
/* resize the buffer if bigger */
if (estimated > rs->output_size) {
if (rs->output_buffer)
av_freep(&rs->output_buffer);
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,
input, in_frames);
if (ret < 0) {
blog(LOG_ERROR, "swr_convert failed: %d", ret);
return false;
}
*output = rs->output_buffer;
*out_frames = (uint32_t)ret;
return true;
}

38
libobs/media-io/audio-resampler.h Normal file
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@@ -0,0 +1,38 @@
/******************************************************************************
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 3 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/>.
******************************************************************************/
#pragma once
#include "../util/c99defs.h"
#include "audio-io.h"
struct audio_resampler;
typedef struct audio_resampler *audio_resampler_t;
struct resample_info {
uint32_t samples_per_sec;
enum audio_format format;
enum speaker_layout speakers;
};
EXPORT audio_resampler_t audio_resampler_create(struct resample_info *dst,
struct resample_info *src);
EXPORT void audio_resampler_destroy(audio_resampler_t resampler);
EXPORT bool audio_resampler_resample(audio_resampler_t resampler,
void **output, uint32_t *out_frames,
void *input, uint32_t in_frames);