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|
/*
* 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>
#include <libavutil/log.h>
#endif
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "ffmpeg"
#ifndef OLD_FFMPEG_INCLUDES
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);
}
}
#endif /* !OLD_FFMPEG_INCLUDES */
#ifndef AV_VERSION_INT
#define AV_VERSION_INT(a, b, c) (a<<16 | b<<8 | c)
#endif
struct mpd_ffmpeg_stream {
struct decoder *decoder;
struct input_stream *input;
ByteIOContext *io;
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;
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 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;
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);
if (stream->io == NULL) {
g_free(stream);
return NULL;
}
return stream;
}
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)
{
#ifndef OLD_FFMPEG_INCLUDES
av_log_set_callback(mpd_ffmpeg_log_callback);
#endif
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 ==
CODEC_TYPE_AUDIO)
return i;
return -1;
}
/**
* 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;
if (packet->pts != (int64_t)AV_NOPTS_VALUE)
decoder_timestamp(decoder,
av_rescale_q(packet->pts, *time_base,
(AVRational){1, 1}));
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;
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 >= AV_VERSION_INT(51, 41, 0)
switch (codec_context->sample_fmt) {
case SAMPLE_FMT_S16:
return SAMPLE_FORMAT_S16;
case SAMPLE_FMT_S32:
return SAMPLE_FORMAT_S32;
default:
g_warning("Unsupported libavcodec SampleFormat value: %d",
codec_context->sample_fmt);
return SAMPLE_FORMAT_UNDEFINED;
}
#else
/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
return SAMPLE_FORMAT_S16;
#endif
}
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_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;
}
AVFormatContext *format_context;
AVCodecContext *codec_context;
AVCodec *codec;
int audio_stream;
//ffmpeg works with ours "fileops" helper
if (av_open_input_stream(&format_context, stream->io, input->uri,
input_format, NULL) != 0) {
g_warning("Open failed\n");
mpd_ffmpeg_stream_close(stream);
return;
}
if (av_find_stream_info(format_context)<0) {
g_warning("Couldn't find stream info\n");
av_close_input_stream(format_context);
mpd_ffmpeg_stream_close(stream);
return;
}
audio_stream = ffmpeg_find_audio_stream(format_context);
if (audio_stream == -1) {
g_warning("No audio stream inside\n");
av_close_input_stream(format_context);
mpd_ffmpeg_stream_close(stream);
return;
}
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_stream(format_context);
mpd_ffmpeg_stream_close(stream);
return;
}
if (avcodec_open(codec_context, codec)<0) {
g_warning("Could not open codec\n");
av_close_input_stream(format_context);
mpd_ffmpeg_stream_close(stream);
return;
}
GError *error = NULL;
struct audio_format audio_format;
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);
avcodec_close(codec_context);
av_close_input_stream(format_context);
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,
&format_context->streams[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);
avcodec_close(codec_context);
av_close_input_stream(format_context);
mpd_ffmpeg_stream_close(stream);
}
#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(31<<8)+0)
typedef struct ffmpeg_tag_map {
enum tag_type type;
const char *name;
} ffmpeg_tag_map;
static const ffmpeg_tag_map ffmpeg_tag_maps[] = {
{ TAG_TITLE, "title" },
#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(50<<8))
{ TAG_ARTIST, "artist" },
{ TAG_DATE, "date" },
#else
{ TAG_ARTIST, "author" },
{ TAG_DATE, "year" },
#endif
{ TAG_ALBUM, "album" },
{ TAG_COMMENT, "comment" },
{ TAG_GENRE, "genre" },
{ TAG_TRACK, "track" },
{ TAG_ARTIST_SORT, "author-sort" },
{ TAG_ALBUM_ARTIST, "album_artist" },
{ TAG_ALBUM_ARTIST_SORT, "album_artist-sort" },
{ TAG_COMPOSER, "composer" },
{ TAG_PERFORMER, "performer" },
{ TAG_DISC, "disc" },
};
static bool
ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m,
const ffmpeg_tag_map tag_map)
{
AVMetadataTag *mt = NULL;
while ((mt = av_metadata_get(m, tag_map.name, mt, 0)) != NULL)
tag_add_item(tag, tag_map.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)
{
AVInputFormat *input_format = ffmpeg_probe(NULL, is);
if (input_format == NULL)
return NULL;
struct mpd_ffmpeg_stream *stream = mpd_ffmpeg_stream_open(NULL, is);
if (stream == NULL)
return NULL;
AVFormatContext *f;
if (av_open_input_stream(&f, stream->io, is->uri,
input_format, NULL) != 0) {
mpd_ffmpeg_stream_close(stream);
return NULL;
}
if (av_find_stream_info(f) < 0) {
av_close_input_stream(f);
mpd_ffmpeg_stream_close(stream);
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);
for (unsigned i = 0; i < sizeof(ffmpeg_tag_maps)/sizeof(ffmpeg_tag_map); i++) {
int idx = ffmpeg_find_audio_stream(f);
ffmpeg_copy_metadata(tag, f->metadata, ffmpeg_tag_maps[i]);
if (idx >= 0)
ffmpeg_copy_metadata(tag, f->streams[idx]->metadata, ffmpeg_tag_maps[i]);
}
#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_stream(f);
mpd_ffmpeg_stream_close(stream);
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", "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", "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",
/* 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,
.stream_tag = ffmpeg_stream_tag,
.suffixes = ffmpeg_suffixes,
.mime_types = ffmpeg_mime_types
};
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