From 4704ca52c0da802c89763391768eafa69f8a2f6e Mon Sep 17 00:00:00 2001
From: Chris Smowton <cs448@cam.ac.uk>
Date: Mon, 22 Sep 2014 00:11:16 +0100
Subject: [PATCH] Add M4A import support using MP4V2 and faad.
---
configure.ac | 23 ++++++++-
lib/rdaudioconvert.cpp | 74 ++++++++++++++++++++++++++++
lib/rdaudioconvert.h | 2 +
lib/rdwavefile.cpp | 107 +++++++++++++++++++++++++++++++++++++++++
lib/rdwavefile.h | 4 +-
5 files changed, 208 insertions(+), 2 deletions(-)
diff --git a/configure.ac b/configure.ac
index 04b39930..5c20fb9c 100644
--- a/configure.ac
+++ b/configure.ac
@@ -78,6 +78,8 @@ AC_ARG_ENABLE(lame,[ --disable-lame disable MPEG Layer 3 encode suppor
[LAME_DISABLED=yes],[])
AC_ARG_ENABLE(flac,[ --disable-flac disable FLAC encode/decode support],
[FLAC_DISABLED=yes],[])
+AC_ARG_ENABLE(mp4v2,[ --disable-mp4 disable M4A decode support],
+ [MP4V2_DISABLED=yes],[])
#
# Check for Qt
@@ -206,10 +208,24 @@ if test -z $FLAC_DISABLED ; then
AC_CHECK_LIB(FLAC,FLAC__metadata_get_tags,[FLAC_METADATA_FOUND=yes],[])
fi
+#
+# Check for MP4V2
+#
+if test -z $MP4V2_DISABLED ; then
+ AC_CHECK_HEADER(mp4v2/mp4v2.h,[MP4V2_HEADER_FOUND=yes],[])
+ if test $MP4V2_HEADER_FOUND ; then
+ AC_CHECK_LIB(mp4v2,MP4Read,[MP4V2_FOUND=yes],[])
+ fi
+ if test $MP4V2_FOUND ; then
+ MP4V2_LIBS="-lmp4v2"
+ AC_DEFINE(HAVE_MP4V2)
+ fi
+fi
+
#
# Set Hard Library Dependencies
#
-AC_SUBST(LIB_RDLIBS,"-lm -lpthread -lqui -lrd -lcurl -lid3 $FLAC_LIBS -lsndfile -lsamplerate -lcdda_interface -lcdda_paranoia -lcrypt -ldl -lpam -lSoundTouch")
+AC_SUBST(LIB_RDLIBS,"-lm -lpthread -lqui -lrd -lcurl -lid3 $FLAC_LIBS $MP4V2_LIBS -lsndfile -lsamplerate -lcdda_interface -lcdda_paranoia -lcrypt -ldl -lpam -lSoundTouch")
#
# Setup MPEG Dependencies
@@ -531,6 +547,11 @@ AC_MSG_NOTICE("| OggVorbis Encoding/Decoding Support ... No |")
else
AC_MSG_NOTICE("| OggVorbis Encoding/Decoding Support ... Yes |")
fi
+if test -z $MP4V2_FOUND ; then
+AC_MSG_NOTICE("| M4A Decoding Support ... No |")
+else
+AC_MSG_NOTICE("| M4A Decoding Support ... Yes |")
+fi
AC_MSG_NOTICE("| |")
AC_MSG_NOTICE("| Optional Components: |")
if test -z $USING_PAM ; then
diff --git a/lib/rdaudioconvert.cpp b/lib/rdaudioconvert.cpp
index a9767ca2..edd9bbbb 100644
--- a/lib/rdaudioconvert.cpp
+++ b/lib/rdaudioconvert.cpp
@@ -23,11 +23,13 @@
#include <stdlib.h>
#include <sys/types.h>
#include <sys/stat.h>
+#include <sys/wait.h>
#include <fcntl.h>
#include <unistd.h>
#include <math.h>
#include <dlfcn.h>
#include <errno.h>
+#include <unistd.h>
#include <sndfile.h>
#include <samplerate.h>
@@ -307,6 +309,11 @@ RDAudioConvert::ErrorCode RDAudioConvert::Stage1Convert(const QString &srcfile,
delete wave;
return err;
+ case RDWaveFile::M4A:
+ err=Stage1M4A(dstfile,wave);
+ delete wave;
+ return err;
+
case RDWaveFile::Aiff:
case RDWaveFile::Unknown:
break;
@@ -673,6 +680,73 @@ RDAudioConvert::ErrorCode RDAudioConvert::Stage1Mpeg(const QString &dstfile,
#endif // HAVE_MAD
}
+RDAudioConvert::ErrorCode RDAudioConvert::Stage1M4A(const QString &dstfile,
+ RDWaveFile *wave) {
+
+ const char* args[7];
+ int childstatus = 0;
+
+ pid_t child = fork();
+
+ QString tmpname = dstfile + ".m4a_temp.wav";
+
+ if(child == 0) {
+
+ freopen("/dev/null", "w", stdout);
+ freopen("/dev/null", "w", stderr);
+
+ args[0] = "faad";
+ args[1] = wave->getName();
+ args[2] = "-o";
+ args[3] = tmpname;
+ args[4] = "-b";
+ args[5] = "4"; // Emit a Float32 format wave file, like the other stage 1s.
+ args[6] = 0;
+ execvp("faad", (char* const*)args);
+ exit(109);
+
+ }
+ else {
+
+ waitpid(child, &childstatus, 0);
+
+ // Killed by a signal (e.g. OOM?)
+ if(!WIFEXITED(childstatus)) {
+ unlink(tmpname);
+ return RDAudioConvert::ErrorInternal;
+ }
+
+ // Returned 109, probably because we could't find faad?
+ if(WEXITSTATUS(childstatus) == 109)
+ return RDAudioConvert::ErrorFormatNotSupported;
+
+ // Two elements are missing:
+ // 1. need to measure the peak amplitude;
+ // 2. need to trim if a subrange was requested.
+ // For now just use the sndfile import path to accomplish both tasks.
+
+ {
+
+ SF_INFO sf_src_info;
+ SNDFILE* sf_src;
+
+ memset(&sf_src_info, 0, sizeof(sf_src_info));
+ // If this fails it is likely because faad could not decode.
+ if((sf_src = sf_open(tmpname, SFM_READ, &sf_src_info)) == NULL)
+ return RDAudioConvert::ErrorFormatError;
+
+ RDAudioConvert::ErrorCode err = Stage1SndFile(dstfile, sf_src, &sf_src_info);
+
+ sf_close(sf_src);
+ unlink(tmpname);
+
+ return err;
+
+ }
+
+ }
+
+}
RDAudioConvert::ErrorCode RDAudioConvert::Stage1SndFile(const QString &dstfile,
SNDFILE *sf_src,
diff --git a/lib/rdaudioconvert.h b/lib/rdaudioconvert.h
index 3b833c9a..14eeb976 100644
--- a/lib/rdaudioconvert.h
+++ b/lib/rdaudioconvert.h
@@ -70,6 +70,8 @@ class RDAudioConvert : public QObject
RDWaveFile *wave);
RDAudioConvert::ErrorCode Stage1Mpeg(const QString &dstfile,
RDWaveFile *wave);
+ RDAudioConvert::ErrorCode Stage1M4A(const QString &dstfile,
+ RDWaveFile *wave);
RDAudioConvert::ErrorCode Stage1SndFile(const QString &dstfile,
SNDFILE *sf_src,
SF_INFO *sf_src_info);
diff --git a/lib/rdwavefile.cpp b/lib/rdwavefile.cpp
index c09f9985..5c2408de 100644
--- a/lib/rdwavefile.cpp
+++ b/lib/rdwavefile.cpp
@@ -48,6 +48,10 @@
#include <rdwavefile.h>
#include <rdconf.h>
+#ifdef HAVE_MP4V2
+#include <mp4v2/mp4v2.h>
+#endif
+
RDWaveFile::RDWaveFile(QString file_name)
{
//
@@ -322,6 +326,94 @@ bool RDWaveFile::openWave(RDWaveData *data)
ReadId3Metadata();
break;
+ case RDWaveFile::M4A:
+ {
+#ifdef HAVE_MP4V2
+
+ format_tag=WAVE_FORMAT_M4A;
+
+ MP4FileHandle f = MP4Read(getName());
+ if(f == MP4_INVALID_FILE_HANDLE)
+ return false;
+
+ // Find an audio track, and populate sample rate, bits/sample etc.
+ uint32_t nTracks = MP4GetNumberOfTracks(f);
+
+ MP4TrackId audioTrack = MP4_INVALID_TRACK_ID;
+ for(uint32_t trackIndex = 0; trackIndex < nTracks && audioTrack == MP4_INVALID_TRACK_ID; ++trackIndex) {
+
+ MP4TrackId thisTrack = MP4FindTrackId(f, trackIndex);
+ const char* trackType = MP4GetTrackType(f, thisTrack);
+ if(trackType && !strcmp(trackType, MP4_AUDIO_TRACK_TYPE)) {
+
+ const char* dataName = MP4GetTrackMediaDataName(f, thisTrack);
+ // The M4A format is only currently useful for AAC in an M4A container:
+ if(dataName &&
+ (!strcasecmp(dataName, "mp4a")) &&
+ MP4GetTrackEsdsObjectTypeId(f, thisTrack) == MP4_MPEG4_AUDIO_TYPE) {
+
+ audioTrack = thisTrack;
+
+ }
+
+ }
+
+ }
+
+ if(audioTrack == MP4_INVALID_TRACK_ID) {
+ MP4Close(f);
+ return false;
+ }
+
+ // Found audio track. Get audio data:
+ avg_bytes_per_sec = MP4GetTrackBitRate(f, audioTrack);
+ channels = MP4GetTrackAudioChannels(f, audioTrack);
+
+ MP4Duration trackDuration = MP4GetTrackDuration(f, audioTrack);
+ ext_time_length = (unsigned)MP4ConvertFromTrackDuration(f, audioTrack, trackDuration,
+ MP4_MSECS_TIME_SCALE);
+ time_length = ext_time_length / 1000;
+ samples_per_sec = MP4GetTrackTimeScale(f, audioTrack);
+ bits_per_sample = 16;
+ data_start = 0;
+ sample_length = MP4GetTrackNumberOfSamples(f, audioTrack);
+ data_length = sample_length * 2 * channels;
+ data_chunk = true;
+ format_chunk = true;
+ wave_type = RDWaveFile::M4A;
+
+ // Now extract metadata (title, artist, etc)
+
+ if(wave_data) {
+
+ const MP4Tags* tags = MP4TagsAlloc();
+ MP4TagsFetch(tags, f);
+
+ wave_data->setMetadataFound(true);
+
+ if(tags->name)
+ wave_data->setTitle(tags->name);
+ if(tags->artist)
+ wave_data->setArtist(tags->artist);
+ if(tags->composer)
+ wave_data->setComposer(tags->composer);
+ if(tags->album)
+ wave_data->setAlbum(tags->album);
+
+ MP4TagsFree(tags);
+
+ }
+
+ MP4Close(f);
+
+ return true;
+
+#else
+ return false;
+#endif
+ break;
+ }
+
case RDWaveFile::Ogg:
#ifdef HAVE_VORBIS
format_tag=WAVE_FORMAT_VORBIS;
@@ -2020,6 +2112,9 @@ RDWaveFile::Type RDWaveFile::GetType(int fd)
if(IsOgg(fd)) {
return RDWaveFile::Ogg;
}
+ if(IsM4A(fd)) {
+ return RDWaveFile::M4A;
+ }
if(IsMpeg(fd)) {
return RDWaveFile::Mpeg;
}
@@ -2211,6 +2306,18 @@ bool RDWaveFile::IsAiff(int fd)
return true;
}
+bool RDWaveFile::IsM4A(int fd)
+{
+#ifdef HAVE_MP4V2
+ MP4FileHandle f = MP4Read(getName());
+ bool ret = f != MP4_INVALID_FILE_HANDLE;
+ if(ret)
+ MP4Close(f);
+ return ret;
+#else
+ return false;
+#endif
+}
off_t RDWaveFile::FindChunk(int fd,const char *chunk_name,unsigned *chunk_size,
bool big_end)
diff --git a/lib/rdwavefile.h b/lib/rdwavefile.h
index fbe4cfc1..c4a632fc 100644
--- a/lib/rdwavefile.h
+++ b/lib/rdwavefile.h
@@ -116,7 +116,7 @@ class RDWaveFile
enum Format {Pcm8=0,Pcm16=1,Float32=2,MpegL1=3,MpegL2=4,MpegL3=5,
DolbyAc2=6,DolbyAc3=7,Vorbis=8};
enum Type {Unknown=0,Wave=1,Mpeg=2,Ogg=3,Atx=4,Tmc=5,Flac=6,Ambos=7,
- Aiff=8};
+ Aiff=8,M4A=9};
enum MpegID {NonMpeg=0,Mpeg1=1,Mpeg2=2};
/**
@@ -1046,6 +1046,7 @@ class RDWaveFile
bool IsTmc(int fd);
bool IsFlac(int fd);
bool IsAiff(int fd);
+ bool IsM4A(int fd);
off_t FindChunk(int fd,const char *chunk_name,unsigned *chunk_size,
bool big_end=false);
bool GetChunk(int fd,const char *chunk_name,unsigned *chunk_size,
@@ -1378,6 +1379,7 @@ class RDWaveFile
*/
#define WAVE_FORMAT_VORBIS 0xFFFF
#define WAVE_FORMAT_FLAC 0xFFFE
+#define WAVE_FORMAT_M4A 0xFFFD
/*
* Proprietary Format Categories