/*
* Copyright (C) 2003-2011 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "decoder_api.h"
#include "audio_check.h"
#include "ffmpeg_metadata.h"
#include "tag_handler.h"
#include <glib.h>
#include <assert.h>
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <string.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/avutil.h>
#include <libavutil/log.h>
#include <libavutil/mathematics.h>
#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,5,0)
#include <libavutil/dict.h>
#endif
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "ffmpeg"
static GLogLevelFlags
level_ffmpeg_to_glib(int level)
{
if (level <= AV_LOG_FATAL)
return G_LOG_LEVEL_CRITICAL;
if (level <= AV_LOG_ERROR)
return G_LOG_LEVEL_WARNING;
if (level <= AV_LOG_INFO)
return G_LOG_LEVEL_MESSAGE;
return G_LOG_LEVEL_DEBUG;
}
static void
mpd_ffmpeg_log_callback(G_GNUC_UNUSED void *ptr, int level,
const char *fmt, va_list vl)
{
const AVClass * cls = NULL;
if (ptr != NULL)
cls = *(const AVClass *const*)ptr;
if (cls != NULL) {
char *domain = g_strconcat(G_LOG_DOMAIN, "/", cls->item_name(ptr), NULL);
g_logv(domain, level_ffmpeg_to_glib(level), fmt, vl);
g_free(domain);
}
}
struct mpd_ffmpeg_stream {
struct decoder *decoder;
struct input_stream *input;
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0)
AVIOContext *io;
#else
ByteIOContext *io;
#endif
unsigned char buffer[8192];
};
static int
mpd_ffmpeg_stream_read(void *opaque, uint8_t *buf, int size)
{
struct mpd_ffmpeg_stream *stream = opaque;
return decoder_read(stream->decoder, stream->input,
(void *)buf, size);
}
static int64_t
mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence)
{
struct mpd_ffmpeg_stream *stream = opaque;
if (whence == AVSEEK_SIZE)
return stream->input->size;
if (!input_stream_lock_seek(stream->input, pos, whence, NULL))
return -1;
return stream->input->offset;
}
static struct mpd_ffmpeg_stream *
mpd_ffmpeg_stream_open(struct decoder *decoder, struct input_stream *input)
{
struct mpd_ffmpeg_stream *stream = g_new(struct mpd_ffmpeg_stream, 1);
stream->decoder = decoder;
stream->input = input;
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0)
stream->io = avio_alloc_context(stream->buffer, sizeof(stream->buffer),
false, stream,
mpd_ffmpeg_stream_read, NULL,
input->seekable
? mpd_ffmpeg_stream_seek : NULL);
#else
stream->io = av_alloc_put_byte(stream->buffer, sizeof(stream->buffer),
false, stream,
mpd_ffmpeg_stream_read, NULL,
input->seekable
? mpd_ffmpeg_stream_seek : NULL);
#endif
if (stream->io == NULL) {
g_free(stream);
return NULL;
}
return stream;
}
/**
* API compatibility wrapper for av_open_input_stream() and
* avformat_open_input().
*/
static int
mpd_ffmpeg_open_input(AVFormatContext **ic_ptr,
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0)
AVIOContext *pb,
#else
ByteIOContext *pb,
#endif
const char *filename,
AVInputFormat *fmt)
{
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,1,3)
AVFormatContext *context = avformat_alloc_context();
if (context == NULL)
return AVERROR(ENOMEM);
context->pb = pb;
*ic_ptr = context;
return avformat_open_input(ic_ptr, filename, fmt, NULL);
#else
return av_open_input_stream(ic_ptr, pb, filename, fmt, NULL);
#endif
}
static void
mpd_ffmpeg_stream_close(struct mpd_ffmpeg_stream *stream)
{
av_free(stream->io);
g_free(stream);
}
static bool
ffmpeg_init(G_GNUC_UNUSED const struct config_param *param)
{
av_log_set_callback(mpd_ffmpeg_log_callback);
av_register_all();
return true;
}
static int
ffmpeg_find_audio_stream(const AVFormatContext *format_context)
{
for (unsigned i = 0; i < format_context->nb_streams; ++i)
if (format_context->streams[i]->codec->codec_type ==
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 64, 0)
AVMEDIA_TYPE_AUDIO)
#else
CODEC_TYPE_AUDIO)
#endif
return i;
return -1;
}
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53,25,0)
/**
* On some platforms, libavcodec wants the output buffer aligned to 16
* bytes (because it uses SSE/Altivec internally). This function
* returns the aligned version of the specified buffer, and corrects
* the buffer size.
*/
static void *
align16(void *p, size_t *length_p)
{
unsigned add = 16 - (size_t)p % 16;
*length_p -= add;
return (char *)p + add;
}
#endif
G_GNUC_CONST
static double
time_from_ffmpeg(int64_t t, const AVRational time_base)
{
assert(t != (int64_t)AV_NOPTS_VALUE);
return (double)av_rescale_q(t, time_base, (AVRational){1, 1024})
/ (double)1024;
}
G_GNUC_CONST
static int64_t
time_to_ffmpeg(double t, const AVRational time_base)
{
return av_rescale_q((int64_t)(t * 1024), (AVRational){1, 1024},
time_base);
}
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
static void
copy_interleave_frame2(uint8_t *dest, uint8_t **src,
unsigned nframes, unsigned nchannels,
unsigned sample_size)
{
for (unsigned frame = 0; frame < nframes; ++frame) {
for (unsigned channel = 0; channel < nchannels; ++channel) {
memcpy(dest, src[channel] + frame * sample_size,
sample_size);
dest += sample_size;
}
}
}
/**
* Copy PCM data from a AVFrame to an interleaved buffer.
*/
static int
copy_interleave_frame(const AVCodecContext *codec_context,
const AVFrame *frame,
uint8_t *buffer, size_t buffer_size)
{
int plane_size;
const int data_size =
av_samples_get_buffer_size(&plane_size,
codec_context->channels,
frame->nb_samples,
codec_context->sample_fmt, 1);
if (buffer_size < (size_t)data_size)
/* buffer is too small - shouldn't happen */
return AVERROR(EINVAL);
if (av_sample_fmt_is_planar(codec_context->sample_fmt) &&
codec_context->channels > 1) {
copy_interleave_frame2(buffer, frame->extended_data,
frame->nb_samples,
codec_context->channels,
av_get_bytes_per_sample(codec_context->sample_fmt));
} else {
memcpy(buffer, frame->extended_data[0], data_size);
}
return data_size;
}
#endif
static enum decoder_command
ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
const AVPacket *packet,
AVCodecContext *codec_context,
const AVRational *time_base)
{
if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE)
decoder_timestamp(decoder,
time_from_ffmpeg(packet->pts, *time_base));
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
AVPacket packet2 = *packet;
#else
const uint8_t *packet_data = packet->data;
int packet_size = packet->size;
#endif
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
uint8_t aligned_buffer[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
const size_t buffer_size = sizeof(aligned_buffer);
#else
/* libavcodec < 0.8 needs an aligned buffer */
uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
size_t buffer_size = sizeof(audio_buf);
int16_t *aligned_buffer = align16(audio_buf, &buffer_size);
#endif
enum decoder_command cmd = DECODE_COMMAND_NONE;
while (
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
packet2.size > 0 &&
#else
packet_size > 0 &&
#endif
cmd == DECODE_COMMAND_NONE) {
int audio_size = buffer_size;
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
AVFrame *frame = avcodec_alloc_frame();
if (frame == NULL) {
g_warning("Could not allocate frame");
break;
}
int got_frame = 0;
int len = avcodec_decode_audio4(codec_context,
frame, &got_frame,
&packet2);
if (len >= 0 && got_frame) {
audio_size = copy_interleave_frame(codec_context,
frame,
aligned_buffer,
buffer_size);
if (audio_size < 0)
len = audio_size;
} else if (len >= 0)
len = -1;
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0)
avcodec_free_frame(&frame);
#else
av_freep(&frame);
#endif
#elif LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
int len = avcodec_decode_audio3(codec_context,
aligned_buffer, &audio_size,
&packet2);
#else
int len = avcodec_decode_audio2(codec_context,
aligned_buffer, &audio_size,
packet_data, packet_size);
#endif
if (len < 0) {
/* if error, we skip the frame */
g_message("decoding failed, frame skipped\n");
break;
}
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
packet2.data += len;
packet2.size -= len;
#else
packet_data += len;
packet_size -= len;
#endif
if (audio_size <= 0)
continue;
cmd = decoder_data(decoder, is,
aligned_buffer, audio_size,
codec_context->bit_rate / 1000);
}
return cmd;
}
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(52, 94, 1)
#define AVSampleFormat SampleFormat
#endif
G_GNUC_CONST
static enum sample_format
ffmpeg_sample_format(enum AVSampleFormat sample_fmt)
{
switch (sample_fmt) {
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1)
case AV_SAMPLE_FMT_S16:
#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0)
case AV_SAMPLE_FMT_S16P:
#endif
#else
case SAMPLE_FMT_S16:
#endif
return SAMPLE_FORMAT_S16;
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1)
case AV_SAMPLE_FMT_S32:
#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0)
case AV_SAMPLE_FMT_S32P:
#endif
#else
case SAMPLE_FMT_S32:
#endif
return SAMPLE_FORMAT_S32;
#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0)
case AV_SAMPLE_FMT_FLTP:
return SAMPLE_FORMAT_FLOAT;
#endif
default:
break;
}
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1)
char buffer[64];
const char *name = av_get_sample_fmt_string(buffer, sizeof(buffer),
sample_fmt);
if (name != NULL)
g_warning("Unsupported libavcodec SampleFormat value: %s (%d)",
name, sample_fmt);
else
#endif
g_warning("Unsupported libavcodec SampleFormat value: %d",
sample_fmt);
return SAMPLE_FORMAT_UNDEFINED;
}
static AVInputFormat *
ffmpeg_probe(struct decoder *decoder, struct input_stream *is)
{
enum {
BUFFER_SIZE = 16384,
PADDING = 16,
};
unsigned char *buffer = g_malloc(BUFFER_SIZE);
size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE);
if (nbytes <= PADDING ||
!input_stream_lock_seek(is, 0, SEEK_SET, NULL)) {
g_free(buffer);
return NULL;
}
/* some ffmpeg parsers (e.g. ac3_parser.c) read a few bytes
beyond the declared buffer limit, which makes valgrind
angry; this workaround removes some padding from the buffer
size */
nbytes -= PADDING;
AVProbeData avpd = {
.buf = buffer,
.buf_size = nbytes,
.filename = is->uri,
};
AVInputFormat *format = av_probe_input_format(&avpd, true);
g_free(buffer);
return format;
}
static void
ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
{
AVInputFormat *input_format = ffmpeg_probe(decoder, input);
if (input_format == NULL)
return;
g_debug("detected input format '%s' (%s)",
input_format->name, input_format->long_name);
struct mpd_ffmpeg_stream *stream =
mpd_ffmpeg_stream_open(decoder, input);
if (stream == NULL) {
g_warning("Failed to open stream");
return;
}
//ffmpeg works with ours "fileops" helper
AVFormatContext *format_context = NULL;
if (mpd_ffmpeg_open_input(&format_context, stream->io, input->uri,
input_format) != 0) {
g_warning("Open failed\n");
mpd_ffmpeg_stream_close(stream);
return;
}
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,6,0)
const int find_result =
avformat_find_stream_info(format_context, NULL);
#else
const int find_result = av_find_stream_info(format_context);
#endif
if (find_result < 0) {
g_warning("Couldn't find stream info\n");
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
avformat_close_input(&format_context);
#else
av_close_input_stream(format_context);
#endif
mpd_ffmpeg_stream_close(stream);
return;
}
int audio_stream = ffmpeg_find_audio_stream(format_context);
if (audio_stream == -1) {
g_warning("No audio stream inside\n");
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
avformat_close_input(&format_context);
#else
av_close_input_stream(format_context);
#endif
mpd_ffmpeg_stream_close(stream);
return;
}
AVStream *av_stream = format_context->streams[audio_stream];
AVCodecContext *codec_context = av_stream->codec;
if (codec_context->codec_name[0] != 0)
g_debug("codec '%s'", codec_context->codec_name);
AVCodec *codec = avcodec_find_decoder(codec_context->codec_id);
if (!codec) {
g_warning("Unsupported audio codec\n");
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
avformat_close_input(&format_context);
#else
av_close_input_stream(format_context);
#endif
mpd_ffmpeg_stream_close(stream);
return;
}
const enum sample_format sample_format =
ffmpeg_sample_format(codec_context->sample_fmt);
if (sample_format == SAMPLE_FORMAT_UNDEFINED)
return;
GError *error = NULL;
struct audio_format audio_format;
if (!audio_format_init_checked(&audio_format,
codec_context->sample_rate,
sample_format,
codec_context->channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
avformat_close_input(&format_context);
#else
av_close_input_stream(format_context);
#endif
mpd_ffmpeg_stream_close(stream);
return;
}
/* the audio format must be read from AVCodecContext by now,
because avcodec_open() has been demonstrated to fill bogus
values into AVCodecContext.channels - a change that will be
reverted later by avcodec_decode_audio3() */
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,6,0)
const int open_result = avcodec_open2(codec_context, codec, NULL);
#else
const int open_result = avcodec_open(codec_context, codec);
#endif
if (open_result < 0) {
g_warning("Could not open codec\n");
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
avformat_close_input(&format_context);
#else
av_close_input_stream(format_context);
#endif
mpd_ffmpeg_stream_close(stream);
return;
}
int total_time = format_context->duration != (int64_t)AV_NOPTS_VALUE
? format_context->duration / AV_TIME_BASE
: 0;
decoder_initialized(decoder, &audio_format,
input->seekable, total_time);
enum decoder_command cmd;
do {
AVPacket packet;
if (av_read_frame(format_context, &packet) < 0)
/* end of file */
break;
if (packet.stream_index == audio_stream)
cmd = ffmpeg_send_packet(decoder, input,
&packet, codec_context,
&av_stream->time_base);
else
cmd = decoder_get_command(decoder);
av_free_packet(&packet);
if (cmd == DECODE_COMMAND_SEEK) {
int64_t where =
time_to_ffmpeg(decoder_seek_where(decoder),
av_stream->time_base);
if (av_seek_frame(format_context, audio_stream, where,
AV_TIME_BASE) < 0)
decoder_seek_error(decoder);
else {
avcodec_flush_buffers(codec_context);
decoder_command_finished(decoder);
}
}
} while (cmd != DECODE_COMMAND_STOP);
avcodec_close(codec_context);
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
avformat_close_input(&format_context);
#else
av_close_input_stream(format_context);
#endif
mpd_ffmpeg_stream_close(stream);
}
//no tag reading in ffmpeg, check if playable
static bool
ffmpeg_scan_stream(struct input_stream *is,
const struct tag_handler *handler, void *handler_ctx)
{
AVInputFormat *input_format = ffmpeg_probe(NULL, is);
if (input_format == NULL)
return false;
struct mpd_ffmpeg_stream *stream = mpd_ffmpeg_stream_open(NULL, is);
if (stream == NULL)
return false;
AVFormatContext *f = NULL;
if (mpd_ffmpeg_open_input(&f, stream->io, is->uri,
input_format) != 0) {
mpd_ffmpeg_stream_close(stream);
return false;
}
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,6,0)
const int find_result =
avformat_find_stream_info(f, NULL);
#else
const int find_result = av_find_stream_info(f);
#endif
if (find_result < 0) {
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
avformat_close_input(&f);
#else
av_close_input_stream(f);
#endif
mpd_ffmpeg_stream_close(stream);
return false;
}
if (f->duration != (int64_t)AV_NOPTS_VALUE)
tag_handler_invoke_duration(handler, handler_ctx,
f->duration / AV_TIME_BASE);
#if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(52,101,0)
av_metadata_conv(f, NULL, f->iformat->metadata_conv);
#endif
ffmpeg_scan_dictionary(f->metadata, handler, handler_ctx);
int idx = ffmpeg_find_audio_stream(f);
if (idx >= 0)
ffmpeg_scan_dictionary(f->streams[idx]->metadata,
handler, handler_ctx);
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,17,0)
avformat_close_input(&f);
#else
av_close_input_stream(f);
#endif
mpd_ffmpeg_stream_close(stream);
return true;
}
/**
* A list of extensions found for the formats supported by ffmpeg.
* This list is current as of 02-23-09; To find out if there are more
* supported formats, check the ffmpeg changelog since this date for
* more formats.
*/
static const char *const ffmpeg_suffixes[] = {
"16sv", "3g2", "3gp", "4xm", "8svx", "aa3", "aac", "ac3", "afc", "aif",
"aifc", "aiff", "al", "alaw", "amr", "anim", "apc", "ape", "asf",
"atrac", "au", "aud", "avi", "avm2", "avs", "bap", "bfi", "c93", "cak",
"cin", "cmv", "cpk", "daud", "dct", "divx", "dts", "dv", "dvd", "dxa",
"eac3", "film", "flac", "flc", "fli", "fll", "flx", "flv", "g726",
"gsm", "gxf", "iss", "m1v", "m2v", "m2t", "m2ts",
"m4a", "m4b", "m4v",
"mad",
"mj2", "mjpeg", "mjpg", "mka", "mkv", "mlp", "mm", "mmf", "mov", "mp+",
"mp1", "mp2", "mp3", "mp4", "mpc", "mpeg", "mpg", "mpga", "mpp", "mpu",
"mve", "mvi", "mxf", "nc", "nsv", "nut", "nuv", "oga", "ogm", "ogv",
"ogx", "oma", "ogg", "omg", "psp", "pva", "qcp", "qt", "r3d", "ra",
"ram", "rl2", "rm", "rmvb", "roq", "rpl", "rvc", "shn", "smk", "snd",
"sol", "son", "spx", "str", "swf", "tgi", "tgq", "tgv", "thp", "ts",
"tsp", "tta", "xa", "xvid", "uv", "uv2", "vb", "vid", "vob", "voc",
"vp6", "vmd", "wav", "webm", "wma", "wmv", "wsaud", "wsvga", "wv",
"wve",
NULL
};
static const char *const ffmpeg_mime_types[] = {
"application/m4a",
"application/mp4",
"application/octet-stream",
"application/ogg",
"application/x-ms-wmz",
"application/x-ms-wmd",
"application/x-ogg",
"application/x-shockwave-flash",
"application/x-shorten",
"audio/8svx",
"audio/16sv",
"audio/aac",
"audio/ac3",
"audio/aiff"
"audio/amr",
"audio/basic",
"audio/flac",
"audio/m4a",
"audio/mp4",
"audio/mpeg",
"audio/musepack",
"audio/ogg",
"audio/qcelp",
"audio/vorbis",
"audio/vorbis+ogg",
"audio/x-8svx",
"audio/x-16sv",
"audio/x-aac",
"audio/x-ac3",
"audio/x-aiff"
"audio/x-alaw",
"audio/x-au",
"audio/x-dca",
"audio/x-eac3",
"audio/x-flac",
"audio/x-gsm",
"audio/x-mace",
"audio/x-matroska",
"audio/x-monkeys-audio",
"audio/x-mpeg",
"audio/x-ms-wma",
"audio/x-ms-wax",
"audio/x-musepack",
"audio/x-ogg",
"audio/x-vorbis",
"audio/x-vorbis+ogg",
"audio/x-pn-realaudio",
"audio/x-pn-multirate-realaudio",
"audio/x-speex",
"audio/x-tta"
"audio/x-voc",
"audio/x-wav",
"audio/x-wma",
"audio/x-wv",
"video/anim",
"video/quicktime",
"video/msvideo",
"video/ogg",
"video/theora",
"video/webm",
"video/x-dv",
"video/x-flv",
"video/x-matroska",
"video/x-mjpeg",
"video/x-mpeg",
"video/x-ms-asf",
"video/x-msvideo",
"video/x-ms-wmv",
"video/x-ms-wvx",
"video/x-ms-wm",
"video/x-ms-wmx",
"video/x-nut",
"video/x-pva",
"video/x-theora",
"video/x-vid",
"video/x-wmv",
"video/x-xvid",
/* special value for the "ffmpeg" input plugin: all streams by
the "ffmpeg" input plugin shall be decoded by this
plugin */
"audio/x-mpd-ffmpeg",
NULL
};
const struct decoder_plugin ffmpeg_decoder_plugin = {
.name = "ffmpeg",
.init = ffmpeg_init,
.stream_decode = ffmpeg_decode,
.scan_stream = ffmpeg_scan_stream,
.suffixes = ffmpeg_suffixes,
.mime_types = ffmpeg_mime_types
};