/*
 * Copyright (C) 2003-2014 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.
 */

/* necessary because libavutil/common.h uses UINT64_C */
#define __STDC_CONSTANT_MACROS

#include "config.h"
#include "FfmpegDecoderPlugin.hxx"
#include "lib/ffmpeg/Domain.hxx"
#include "../DecoderAPI.hxx"
#include "FfmpegMetaData.hxx"
#include "tag/TagHandler.hxx"
#include "input/InputStream.hxx"
#include "CheckAudioFormat.hxx"
#include "util/Error.hxx"
#include "util/Domain.hxx"
#include "LogV.hxx"

extern "C" {
#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_MAJOR >= 53
#include <libavutil/frame.h>
#endif
}

#include <assert.h>
#include <string.h>

/* suppress the ffmpeg compatibility macro */
#ifdef SampleFormat
#undef SampleFormat
#endif

static LogLevel
import_ffmpeg_level(int level)
{
	if (level <= AV_LOG_FATAL)
		return LogLevel::ERROR;

	if (level <= AV_LOG_WARNING)
		return LogLevel::WARNING;

	if (level <= AV_LOG_INFO)
		return LogLevel::INFO;

	return LogLevel::DEBUG;
}

static void
mpd_ffmpeg_log_callback(gcc_unused void *ptr, int level,
			const char *fmt, va_list vl)
{
	const AVClass * cls = nullptr;

	if (ptr != nullptr)
		cls = *(const AVClass *const*)ptr;

	if (cls != nullptr) {
		char domain[64];
		snprintf(domain, sizeof(domain), "%s/%s",
			 ffmpeg_domain.GetName(), cls->item_name(ptr));
		const Domain d(domain);
		LogFormatV(d, import_ffmpeg_level(level), fmt, vl);
	}
}

struct AvioStream {
	Decoder *const decoder;
	InputStream &input;

	AVIOContext *io;

	AvioStream(Decoder *_decoder, InputStream &_input)
		:decoder(_decoder), input(_input), io(nullptr) {}

	~AvioStream() {
		if (io != nullptr) {
			av_free(io->buffer);
			av_free(io);
		}
	}

	bool Open();
};

static int
mpd_ffmpeg_stream_read(void *opaque, uint8_t *buf, int size)
{
	AvioStream *stream = (AvioStream *)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)
{
	AvioStream *stream = (AvioStream *)opaque;

	switch (whence) {
	case SEEK_SET:
		break;

	case SEEK_CUR:
		pos += stream->input.GetOffset();
		break;

	case SEEK_END:
		if (!stream->input.KnownSize())
			return -1;

		pos += stream->input.GetSize();
		break;

	case AVSEEK_SIZE:
		if (!stream->input.KnownSize())
			return -1;

		return stream->input.GetSize();

	default:
		return -1;
	}

	if (!stream->input.LockSeek(pos, IgnoreError()))
		return -1;

	return stream->input.GetOffset();
}

bool
AvioStream::Open()
{
	constexpr size_t BUFFER_SIZE = 8192;
	auto buffer = (unsigned char *)av_malloc(BUFFER_SIZE);
	if (buffer == nullptr)
		return false;

	io = avio_alloc_context(buffer, BUFFER_SIZE,
				false, this,
				mpd_ffmpeg_stream_read, nullptr,
				input.IsSeekable()
				? mpd_ffmpeg_stream_seek : nullptr);
	/* If avio_alloc_context() fails, who frees the buffer?  The
	   libavformat API documentation does not specify this, it
	   only says that AVIOContext.buffer must be freed in the end,
	   however no AVIOContext exists in that failure code path. */
	return io != nullptr;
}

/**
 * API compatibility wrapper for av_open_input_stream() and
 * avformat_open_input().
 */
static int
mpd_ffmpeg_open_input(AVFormatContext **ic_ptr,
		      AVIOContext *pb,
		      const char *filename,
		      AVInputFormat *fmt)
{
	AVFormatContext *context = avformat_alloc_context();
	if (context == nullptr)
		return AVERROR(ENOMEM);

	context->pb = pb;
	*ic_ptr = context;
	return avformat_open_input(ic_ptr, filename, fmt, nullptr);
}

static bool
ffmpeg_init(gcc_unused const 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 ==
		    AVMEDIA_TYPE_AUDIO)
			return i;

	return -1;
}

gcc_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;
}

template<typename Ratio>
static constexpr AVRational
RatioToAVRational()
{
	return { Ratio::num, Ratio::den };
}

gcc_const
static int64_t
time_to_ffmpeg(SongTime t, const AVRational time_base)
{
	return av_rescale_q(t.count(),
			    RatioToAVRational<SongTime::period>(),
			    time_base);
}

/**
 * Replace #AV_NOPTS_VALUE with the given fallback.
 */
static constexpr int64_t
timestamp_fallback(int64_t t, int64_t fallback)
{
	return gcc_likely(t != int64_t(AV_NOPTS_VALUE))
		? t
		: fallback;
}

/**
 * Accessor for AVStream::start_time that replaces AV_NOPTS_VALUE with
 * zero.  We can't use AV_NOPTS_VALUE in calculations, and we simply
 * assume that the stream's start time is zero, which appears to be
 * the best way out of that situation.
 */
static int64_t
start_time_fallback(const AVStream &stream)
{
	return timestamp_fallback(stream.start_time, 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 **output_buffer,
		      uint8_t **global_buffer, int *global_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 (data_size <= 0)
		return data_size;

	if (av_sample_fmt_is_planar(codec_context->sample_fmt) &&
	    codec_context->channels > 1) {
		if(*global_buffer_size < data_size) {
			av_freep(global_buffer);

			*global_buffer = (uint8_t*)av_malloc(data_size);

			if (!*global_buffer)
				/* Not enough memory - shouldn't happen */
				return AVERROR(ENOMEM);
			*global_buffer_size = data_size;
		}
		*output_buffer = *global_buffer;
		copy_interleave_frame2(*output_buffer, frame->extended_data,
				       frame->nb_samples,
				       codec_context->channels,
				       av_get_bytes_per_sample(codec_context->sample_fmt));
	} else {
		*output_buffer = frame->extended_data[0];
	}

	return data_size;
}

/**
 * Convert AVPacket::pts to a stream-relative time stamp (still in
 * AVStream::time_base units).  Returns a negative value on error.
 */
gcc_pure
static int64_t
StreamRelativePts(const AVPacket &packet, const AVStream &stream)
{
	auto pts = packet.pts;
	if (pts < 0 || pts == int64_t(AV_NOPTS_VALUE))
		return -1;

	auto start = start_time_fallback(stream);
	return pts - start;
}

/**
 * Convert a non-negative stream-relative time stamp in
 * AVStream::time_base units to a PCM frame number.
 */
gcc_pure
static uint64_t
PtsToPcmFrame(uint64_t pts, const AVStream &stream,
	      const AVCodecContext &codec_context)
{
	return av_rescale_q(pts, stream.time_base, codec_context.time_base);
}

/**
 * @param min_frame skip all data before this PCM frame number; this
 * is used after seeking to skip data in an AVPacket until the exact
 * desired time stamp has been reached
 */
static DecoderCommand
ffmpeg_send_packet(Decoder &decoder, InputStream &is,
		   const AVPacket *packet,
		   AVCodecContext *codec_context,
		   const AVStream *stream,
		   AVFrame *frame,
		   uint64_t min_frame, size_t pcm_frame_size,
		   uint8_t **buffer, int *buffer_size)
{
	size_t skip_bytes = 0;

	const auto pts = StreamRelativePts(*packet, *stream);
	if (pts >= 0) {
		if (min_frame > 0) {
			auto cur_frame = PtsToPcmFrame(pts, *stream,
						       *codec_context);
			if (cur_frame < min_frame)
				skip_bytes = pcm_frame_size * (min_frame - cur_frame);
		} else
			decoder_timestamp(decoder,
					  time_from_ffmpeg(pts, stream->time_base));
	}

	AVPacket packet2 = *packet;

	uint8_t *output_buffer;

	DecoderCommand cmd = DecoderCommand::NONE;
	while (packet2.size > 0 && cmd == DecoderCommand::NONE) {
		int audio_size = 0;
		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,
							   &output_buffer,
							   buffer, buffer_size);
			if (audio_size < 0)
				len = audio_size;
		}

		if (len < 0) {
			/* if error, we skip the frame */
			LogDefault(ffmpeg_domain,
				   "decoding failed, frame skipped");
			break;
		}

		packet2.data += len;
		packet2.size -= len;

		if (audio_size <= 0)
			continue;

		const uint8_t *data = output_buffer;
		if (skip_bytes > 0) {
			if (skip_bytes >= size_t(audio_size)) {
				skip_bytes -= audio_size;
				continue;
			}

			data += skip_bytes;
			audio_size -= skip_bytes;
			skip_bytes = 0;
		}

		cmd = decoder_data(decoder, is,
				   data, audio_size,
				   codec_context->bit_rate / 1000);
	}
	return cmd;
}

gcc_const
static SampleFormat
ffmpeg_sample_format(enum AVSampleFormat sample_fmt)
{
	switch (sample_fmt) {
	case AV_SAMPLE_FMT_S16:
	case AV_SAMPLE_FMT_S16P:
		return SampleFormat::S16;

	case AV_SAMPLE_FMT_S32:
	case AV_SAMPLE_FMT_S32P:
		return SampleFormat::S32;

	case AV_SAMPLE_FMT_FLT:
	case AV_SAMPLE_FMT_FLTP:
		return SampleFormat::FLOAT;

	default:
		break;
	}

	char buffer[64];
	const char *name = av_get_sample_fmt_string(buffer, sizeof(buffer),
						    sample_fmt);
	if (name != nullptr)
		FormatError(ffmpeg_domain,
			    "Unsupported libavcodec SampleFormat value: %s (%d)",
			    name, sample_fmt);
	else
		FormatError(ffmpeg_domain,
			    "Unsupported libavcodec SampleFormat value: %d",
			    sample_fmt);
	return SampleFormat::UNDEFINED;
}

static AVInputFormat *
ffmpeg_probe(Decoder *decoder, InputStream &is)
{
	enum {
		BUFFER_SIZE = 16384,
		PADDING = 16,
	};

	unsigned char buffer[BUFFER_SIZE];
	size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE);
	if (nbytes <= PADDING || !is.LockRewind(IgnoreError()))
		return nullptr;

	/* 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;

	/* new versions of ffmpeg may add new attributes, and leaving
	   them uninitialized may crash; hopefully, zero-initializing
	   everything we don't know is ok */
	memset(&avpd, 0, sizeof(avpd));

	avpd.buf = buffer;
	avpd.buf_size = nbytes;
	avpd.filename = is.GetURI();

#ifdef AVPROBE_SCORE_MIME
#if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(56, 5, 1)
	/* this attribute was added in libav/ffmpeg version 11, but
	   unfortunately it's "uint8_t" instead of "char", and it's
	   not "const" - wtf? */
	avpd.mime_type = (uint8_t *)const_cast<char *>(is.GetMimeType());
#else
	/* API problem fixed in FFmpeg 2.5 */
	avpd.mime_type = is.GetMimeType();
#endif
#endif

	return av_probe_input_format(&avpd, true);
}

static void
ffmpeg_decode(Decoder &decoder, InputStream &input)
{
	AVInputFormat *input_format = ffmpeg_probe(&decoder, input);
	if (input_format == nullptr)
		return;

	FormatDebug(ffmpeg_domain, "detected input format '%s' (%s)",
		    input_format->name, input_format->long_name);

	AvioStream stream(&decoder, input);
	if (!stream.Open()) {
		LogError(ffmpeg_domain, "Failed to open stream");
		return;
	}

	//ffmpeg works with ours "fileops" helper
	AVFormatContext *format_context = nullptr;
	if (mpd_ffmpeg_open_input(&format_context, stream.io,
				  input.GetURI(),
				  input_format) != 0) {
		LogError(ffmpeg_domain, "Open failed");
		return;
	}

	const int find_result =
		avformat_find_stream_info(format_context, nullptr);
	if (find_result < 0) {
		LogError(ffmpeg_domain, "Couldn't find stream info");
		avformat_close_input(&format_context);
		return;
	}

	int audio_stream = ffmpeg_find_audio_stream(format_context);
	if (audio_stream == -1) {
		LogError(ffmpeg_domain, "No audio stream inside");
		avformat_close_input(&format_context);
		return;
	}

	AVStream *av_stream = format_context->streams[audio_stream];

	AVCodecContext *codec_context = av_stream->codec;

#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 25, 0)
	const AVCodecDescriptor *codec_descriptor =
		avcodec_descriptor_get(codec_context->codec_id);
	if (codec_descriptor != nullptr)
		FormatDebug(ffmpeg_domain, "codec '%s'",
			    codec_descriptor->name);
#else
	if (codec_context->codec_name[0] != 0)
		FormatDebug(ffmpeg_domain, "codec '%s'",
			    codec_context->codec_name);
#endif

	AVCodec *codec = avcodec_find_decoder(codec_context->codec_id);

	if (!codec) {
		LogError(ffmpeg_domain, "Unsupported audio codec");
		avformat_close_input(&format_context);
		return;
	}

	const SampleFormat sample_format =
		ffmpeg_sample_format(codec_context->sample_fmt);
	if (sample_format == SampleFormat::UNDEFINED) {
		// (error message already done by ffmpeg_sample_format())
		avformat_close_input(&format_context);
		return;
	}

	Error error;
	AudioFormat audio_format;
	if (!audio_format_init_checked(audio_format,
				       codec_context->sample_rate,
				       sample_format,
				       codec_context->channels, error)) {
		LogError(error);
		avformat_close_input(&format_context);
		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() */

	const int open_result = avcodec_open2(codec_context, codec, nullptr);
	if (open_result < 0) {
		LogError(ffmpeg_domain, "Could not open codec");
		avformat_close_input(&format_context);
		return;
	}

	const SignedSongTime total_time =
		format_context->duration != (int64_t)AV_NOPTS_VALUE
		? SignedSongTime::FromScale<uint64_t>(format_context->duration,
						      AV_TIME_BASE)
		: SignedSongTime::Negative();

	decoder_initialized(decoder, audio_format,
			    input.IsSeekable(), total_time);

#if LIBAVUTIL_VERSION_MAJOR >= 53
	AVFrame *frame = av_frame_alloc();
#else
	AVFrame *frame = avcodec_alloc_frame();
#endif
	if (!frame) {
		LogError(ffmpeg_domain, "Could not allocate frame");
		avformat_close_input(&format_context);
		return;
	}

	uint8_t *interleaved_buffer = nullptr;
	int interleaved_buffer_size = 0;

	uint64_t min_frame = 0;

	DecoderCommand 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,
						 frame,
						 min_frame, audio_format.GetFrameSize(),
						 &interleaved_buffer, &interleaved_buffer_size);
			min_frame = 0;
		} else
			cmd = decoder_get_command(decoder);

		av_free_packet(&packet);

		if (cmd == DecoderCommand::SEEK) {
			int64_t where =
				time_to_ffmpeg(decoder_seek_time(decoder),
					       av_stream->time_base) +
				start_time_fallback(*av_stream);

			/* AVSEEK_FLAG_BACKWARD asks FFmpeg to seek to
			   the packet boundary before the seek time
			   stamp, not after */

			if (av_seek_frame(format_context, audio_stream, where,
					  AVSEEK_FLAG_ANY|AVSEEK_FLAG_BACKWARD) < 0)
				decoder_seek_error(decoder);
			else {
				avcodec_flush_buffers(codec_context);
				min_frame = decoder_seek_where_frame(decoder);
				decoder_command_finished(decoder);
			}
		}
	} while (cmd != DecoderCommand::STOP);

#if LIBAVUTIL_VERSION_MAJOR >= 53
	av_frame_free(&frame);
#elif LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0)
	avcodec_free_frame(&frame);
#else
	av_freep(&frame);
#endif
	av_freep(&interleaved_buffer);

	avcodec_close(codec_context);
	avformat_close_input(&format_context);
}

//no tag reading in ffmpeg, check if playable
static bool
ffmpeg_scan_stream(InputStream &is,
		   const struct tag_handler *handler, void *handler_ctx)
{
	AVInputFormat *input_format = ffmpeg_probe(nullptr, is);
	if (input_format == nullptr)
		return false;

	AvioStream stream(nullptr, is);
	if (!stream.Open())
		return false;

	AVFormatContext *f = nullptr;
	if (mpd_ffmpeg_open_input(&f, stream.io, is.GetURI(),
				  input_format) != 0)
		return false;

	const int find_result =
		avformat_find_stream_info(f, nullptr);
	if (find_result < 0) {
		avformat_close_input(&f);
		return false;
	}

	if (f->duration != (int64_t)AV_NOPTS_VALUE) {
		const auto duration =
			SongTime::FromScale<uint64_t>(f->duration,
						      AV_TIME_BASE);
		tag_handler_invoke_duration(handler, handler_ctx, duration);
	}

	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);

	avformat_close_input(&f);
	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", "opus", "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",
	nullptr
};

static const char *const ffmpeg_mime_types[] = {
	"application/flv",
	"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/aacp",
	"audio/ac3",
	"audio/aiff"
	"audio/amr",
	"audio/basic",
	"audio/flac",
	"audio/m4a",
	"audio/mp4",
	"audio/mpeg",
	"audio/musepack",
	"audio/ogg",
	"audio/opus",
	"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",

	nullptr
};

const struct DecoderPlugin ffmpeg_decoder_plugin = {
	"ffmpeg",
	ffmpeg_init,
	nullptr,
	ffmpeg_decode,
	nullptr,
	nullptr,
	ffmpeg_scan_stream,
	nullptr,
	ffmpeg_suffixes,
	ffmpeg_mime_types
};