/*
* 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;
unsigned char buffer[8192];
AvioStream(Decoder *_decoder, InputStream &_input)
:decoder(_decoder), input(_input), io(nullptr) {}
~AvioStream() {
if (io != nullptr)
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()
{
io = avio_alloc_context(buffer, sizeof(buffer),
false, this,
mpd_ffmpeg_stream_read, nullptr,
input.IsSeekable()
? mpd_ffmpeg_stream_seek : nullptr);
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 ¶m)
{
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
};