/*
* Copyright (C) 2003-2010 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 <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>
#ifdef OLD_FFMPEG_INCLUDES
#include <avcodec.h>
#include <avformat.h>
#include <avio.h>
#else
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#endif
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "ffmpeg"
struct ffmpeg_context {
int audio_stream;
AVFormatContext *format_context;
AVCodecContext *codec_context;
struct decoder *decoder;
struct input_stream *input;
};
struct ffmpeg_stream {
/** hack - see url_to_struct() */
char url[64];
struct decoder *decoder;
struct input_stream *input;
};
/**
* Convert a faked mpd:// URL to a ffmpeg_stream structure. This is a
* hack because ffmpeg does not provide a nice API for passing a
* user-defined pointer to mpdurl_open().
*/
static struct ffmpeg_stream *url_to_struct(const char *url)
{
union {
const char *in;
struct ffmpeg_stream *out;
} u = { .in = url };
return u.out;
}
static int mpd_ffmpeg_open(URLContext *h, const char *filename,
G_GNUC_UNUSED int flags)
{
struct ffmpeg_stream *stream = url_to_struct(filename);
h->priv_data = stream;
h->is_streamed = stream->input->seekable ? 0 : 1;
return 0;
}
static int mpd_ffmpeg_read(URLContext *h, unsigned char *buf, int size)
{
struct ffmpeg_stream *stream = (struct ffmpeg_stream *) h->priv_data;
return decoder_read(stream->decoder, stream->input,
(void *)buf, size);
}
static int64_t mpd_ffmpeg_seek(URLContext *h, int64_t pos, int whence)
{
struct ffmpeg_stream *stream = (struct ffmpeg_stream *) h->priv_data;
bool ret;
if (whence == AVSEEK_SIZE)
return stream->input->size;
ret = input_stream_seek(stream->input, pos, whence, NULL);
if (!ret)
return -1;
return stream->input->offset;
}
static int mpd_ffmpeg_close(URLContext *h)
{
h->priv_data = NULL;
return 0;
}
static URLProtocol mpd_ffmpeg_fileops = {
.name = "mpd",
.url_open = mpd_ffmpeg_open,
.url_read = mpd_ffmpeg_read,
.url_seek = mpd_ffmpeg_seek,
.url_close = mpd_ffmpeg_close,
};
static bool
ffmpeg_init(G_GNUC_UNUSED const struct config_param *param)
{
av_register_all();
register_protocol(&mpd_ffmpeg_fileops);
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 ==
CODEC_TYPE_AUDIO)
return i;
return -1;
}
/**
* Append the suffix of the original URI to the virtual stream URI.
* Without this, libavformat cannot detect some of the codecs
* (e.g. "shorten").
*/
static void
append_uri_suffix(struct ffmpeg_stream *stream, const char *uri)
{
assert(stream != NULL);
assert(uri != NULL);
char *base = g_path_get_basename(uri);
const char *suffix = strrchr(base, '.');
if (suffix != NULL && suffix[1] != 0)
g_strlcat(stream->url, suffix, sizeof(stream->url));
g_free(base);
}
static bool
ffmpeg_helper(struct input_stream *input,
bool (*callback)(struct ffmpeg_context *ctx),
struct ffmpeg_context *ctx)
{
AVFormatContext *format_context;
AVCodecContext *codec_context;
AVCodec *codec;
int audio_stream;
struct ffmpeg_stream stream = {
.url = "mpd://X", /* only the mpd:// prefix matters */
};
bool ret;
if (input->uri != NULL)
append_uri_suffix(&stream, input->uri);
stream.input = input;
if (ctx && ctx->decoder) {
stream.decoder = ctx->decoder; //are we in decoding loop ?
} else {
stream.decoder = NULL;
}
//ffmpeg works with ours "fileops" helper
if (av_open_input_file(&format_context, stream.url, NULL, 0, NULL) != 0) {
g_warning("Open failed\n");
return false;
}
if (av_find_stream_info(format_context)<0) {
g_warning("Couldn't find stream info\n");
av_close_input_file(format_context);
return false;
}
audio_stream = ffmpeg_find_audio_stream(format_context);
if (audio_stream == -1) {
g_warning("No audio stream inside\n");
av_close_input_file(format_context);
return false;
}
codec_context = format_context->streams[audio_stream]->codec;
if (codec_context->codec_name[0] != 0)
g_debug("codec '%s'", codec_context->codec_name);
codec = avcodec_find_decoder(codec_context->codec_id);
if (!codec) {
g_warning("Unsupported audio codec\n");
av_close_input_file(format_context);
return false;
}
if (avcodec_open(codec_context, codec)<0) {
g_warning("Could not open codec\n");
av_close_input_file(format_context);
return false;
}
if (callback) {
ctx->audio_stream = audio_stream;
ctx->format_context = format_context;
ctx->codec_context = codec_context;
ret = callback(ctx);
} else
ret = true;
avcodec_close(codec_context);
av_close_input_file(format_context);
return ret;
}
/**
* 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;
}
static enum decoder_command
ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
const AVPacket *packet,
AVCodecContext *codec_context,
const AVRational *time_base)
{
enum decoder_command cmd = DECODE_COMMAND_NONE;
uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
int16_t *aligned_buffer;
size_t buffer_size;
int len, audio_size;
uint8_t *packet_data;
int packet_size;
packet_data = packet->data;
packet_size = packet->size;
buffer_size = sizeof(audio_buf);
aligned_buffer = align16(audio_buf, &buffer_size);
while ((packet_size > 0) && (cmd == DECODE_COMMAND_NONE)) {
audio_size = buffer_size;
len = avcodec_decode_audio2(codec_context,
aligned_buffer, &audio_size,
packet_data, packet_size);
if (len < 0) {
/* if error, we skip the frame */
g_message("decoding failed\n");
break;
}
packet_data += len;
packet_size -= len;
if (audio_size <= 0)
continue;
if (packet->pts != (int64_t)AV_NOPTS_VALUE)
decoder_timestamp(decoder,
av_rescale_q(packet->pts, *time_base,
(AVRational){1, 1}));
cmd = decoder_data(decoder, is,
aligned_buffer, audio_size,
codec_context->bit_rate / 1000);
}
return cmd;
}
static enum sample_format
ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context)
{
#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
int bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
/* XXX implement & test other sample formats */
switch (bits) {
case 16:
return SAMPLE_FORMAT_S16;
}
return SAMPLE_FORMAT_UNDEFINED;
#else
/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
return SAMPLE_FORMAT_S16;
#endif
}
static bool
ffmpeg_decode_internal(struct ffmpeg_context *ctx)
{
GError *error = NULL;
struct decoder *decoder = ctx->decoder;
AVCodecContext *codec_context = ctx->codec_context;
AVFormatContext *format_context = ctx->format_context;
AVPacket packet;
struct audio_format audio_format;
enum decoder_command cmd;
int total_time;
total_time = 0;
if (!audio_format_init_checked(&audio_format,
codec_context->sample_rate,
ffmpeg_sample_format(codec_context),
codec_context->channels, &error)) {
g_warning("%s", error->message);
g_error_free(error);
return false;
}
//there is some problem with this on some demux (mp3 at least)
if (format_context->duration != (int64_t)AV_NOPTS_VALUE) {
total_time = format_context->duration / AV_TIME_BASE;
}
decoder_initialized(decoder, &audio_format,
ctx->input->seekable, total_time);
do {
if (av_read_frame(format_context, &packet) < 0)
/* end of file */
break;
if (packet.stream_index == ctx->audio_stream)
cmd = ffmpeg_send_packet(decoder, ctx->input,
&packet, codec_context,
&format_context->streams[ctx->audio_stream]->time_base);
else
cmd = decoder_get_command(decoder);
av_free_packet(&packet);
if (cmd == DECODE_COMMAND_SEEK) {
int64_t where =
decoder_seek_where(decoder) * AV_TIME_BASE;
if (av_seek_frame(format_context, -1, where, 0) < 0)
decoder_seek_error(decoder);
else
decoder_command_finished(decoder);
}
} while (cmd != DECODE_COMMAND_STOP);
return true;
}
static void
ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
{
struct ffmpeg_context ctx;
ctx.input = input;
ctx.decoder = decoder;
ffmpeg_helper(input, ffmpeg_decode_internal, &ctx);
}
#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0)
static bool
ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m,
enum tag_type type, const char *name)
{
AVMetadataTag *mt = NULL;
while ((mt = av_metadata_get(m, name, mt, 0)) != NULL)
tag_add_item(tag, type, mt->value);
return mt != NULL;
}
#endif
//no tag reading in ffmpeg, check if playable
static struct tag *
ffmpeg_stream_tag(struct input_stream *is)
{
struct ffmpeg_stream stream = {
.url = "mpd://X", /* only the mpd:// prefix matters */
.decoder = NULL,
.input = is,
};
if (is->uri != NULL)
append_uri_suffix(&stream, is->uri);
AVFormatContext *f;
if (av_open_input_file(&f, stream.url, NULL, 0, NULL) != 0)
return NULL;
if (av_find_stream_info(f) < 0) {
av_close_input_file(f);
return NULL;
}
struct tag *tag = tag_new();
tag->time = f->duration != (int64_t)AV_NOPTS_VALUE
? f->duration / AV_TIME_BASE
: 0;
#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0)
av_metadata_conv(f, NULL, f->iformat->metadata_conv);
ffmpeg_copy_metadata(tag, f->metadata, TAG_TITLE, "title");
ffmpeg_copy_metadata(tag, f->metadata, TAG_ARTIST, "author");
ffmpeg_copy_metadata(tag, f->metadata, TAG_ALBUM, "album");
ffmpeg_copy_metadata(tag, f->metadata, TAG_COMMENT, "comment");
ffmpeg_copy_metadata(tag, f->metadata, TAG_GENRE, "genre");
ffmpeg_copy_metadata(tag, f->metadata, TAG_TRACK, "track");
ffmpeg_copy_metadata(tag, f->metadata, TAG_DATE, "year");
#else
if (f->author[0])
tag_add_item(tag, TAG_ARTIST, f->author);
if (f->title[0])
tag_add_item(tag, TAG_TITLE, f->title);
if (f->album[0])
tag_add_item(tag, TAG_ALBUM, f->album);
if (f->track > 0) {
char buffer[16];
snprintf(buffer, sizeof(buffer), "%d", f->track);
tag_add_item(tag, TAG_TRACK, buffer);
}
if (f->comment[0])
tag_add_item(tag, TAG_COMMENT, f->comment);
if (f->genre[0])
tag_add_item(tag, TAG_GENRE, f->genre);
if (f->year > 0) {
char buffer[16];
snprintf(buffer, sizeof(buffer), "%d", f->year);
tag_add_item(tag, TAG_DATE, buffer);
}
#endif
av_close_input_file(f);
return tag;
}
/**
* 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", "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", "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/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",
NULL
};
const struct decoder_plugin ffmpeg_decoder_plugin = {
.name = "ffmpeg",
.init = ffmpeg_init,
.stream_decode = ffmpeg_decode,
.stream_tag = ffmpeg_stream_tag,
.suffixes = ffmpeg_suffixes,
.mime_types = ffmpeg_mime_types
};