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Merged 04bd883b65 from master

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This commit is contained in:
Fred Gleason
2015-09-15 08:06:20 -04:00
parent 6936746e34 04bd883b65
commit c0446ea5f7
103 changed files with 1645 additions and 205 deletions
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170
lib/rdaudioconvert.cpp
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@@ -22,11 +22,13 @@
#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>
@@ -39,6 +41,10 @@
#include <FLAC++/encoder.h>
#include <rdflacdecode.h>
#endif // HAVE_FLAC
#ifdef HAVE_MP4_LIBS
#include <mp4v2/mp4v2.h>
#include <neaacdec.h>
#endif // HAVE_MP4_LIBS
#include <id3/tag.h>
#include <qfile.h>
@@ -306,6 +312,11 @@ 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;
@@ -672,6 +683,165 @@ RDAudioConvert::ErrorCode RDAudioConvert::Stage1Mpeg(const QString &dstfile,
#endif // HAVE_MAD
}
// Based on libfaad's frontend/main.c, but using libmp4v2 for MP4 access.
RDAudioConvert::ErrorCode RDAudioConvert::Stage1M4A(const QString &dstfile,
RDWaveFile *wave)
{
#ifdef HAVE_MP4_LIBS
SNDFILE *sf_dst=NULL;
SF_INFO sf_dst_info;
MP4FileHandle f;
MP4TrackId audioTrack;
MP4SampleId firstSample, lastSample;
uint32_t aacBufSize, aacConfigSize;
uint8_t *aacBuf, *aacConfigBuffer;
NeAACDecHandle hDecoder;
NeAACDecConfigurationPtr config;
unsigned long foundSampleRate;
unsigned char foundChannels;
RDAudioConvert::ErrorCode ret = RDAudioConvert::ErrorOk;
if(!dlmp4.load()) {
return RDAudioConvert::ErrorFormatNotSupported;
}
//
// Open source
//
f = dlmp4.MP4Read(wave->getName());
if(f == MP4_INVALID_FILE_HANDLE)
return RDAudioConvert::ErrorNoSource;
audioTrack = dlmp4.getMP4AACTrack(f);
firstSample = 1;
lastSample = dlmp4.MP4GetTrackNumberOfSamples(f, audioTrack);
if(conv_start_point > 0) {
double startsecs = ((double)conv_start_point) / 1000;
MP4Timestamp startts = (MP4Timestamp)(startsecs * wave->getSamplesPerSec());
firstSample = dlmp4.MP4GetSampleIdFromTime(f, audioTrack, startts, /*need_sync=*/false);
if(firstSample == MP4_INVALID_SAMPLE_ID) {
ret = RDAudioConvert::ErrorInvalidSource;
goto out_mp4;
}
}
if(conv_end_point > 0) {
double stopsecs = ((double)conv_end_point) / 1000;
MP4Timestamp stopts = (MP4Timestamp)(stopsecs * wave->getSamplesPerSec());
lastSample = dlmp4.MP4GetSampleIdFromTime(f, audioTrack, stopts, /*need_sync=*/false);
if(lastSample == MP4_INVALID_SAMPLE_ID) {
ret = RDAudioConvert::ErrorInvalidSource;
goto out_mp4;
}
}
aacBufSize = dlmp4.MP4GetTrackMaxSampleSize(f, audioTrack);
aacBuf = (uint8_t*)malloc(aacBufSize);
if(!aacBufSize || !aacBuf) {
// Probably the source's fault for specifying a massive buffer.
ret = RDAudioConvert::ErrorInvalidSource;
goto out_mp4;
}
dlmp4.MP4GetTrackESConfiguration(f, audioTrack, &aacConfigBuffer, &aacConfigSize);
if(!aacConfigBuffer) {
ret = RDAudioConvert::ErrorInvalidSource;
goto out_mp4_buf;
}
//
// Open Destination
//
memset(&sf_dst_info,0,sizeof(sf_dst_info));
sf_dst_info.format=SF_FORMAT_WAV|SF_FORMAT_FLOAT;
sf_dst_info.channels=wave->getChannels();
sf_dst_info.samplerate=wave->getSamplesPerSec();
if((sf_dst=sf_open(dstfile,SFM_WRITE,&sf_dst_info))==NULL) {
ret = RDAudioConvert::ErrorNoDestination;
goto out_mp4_configbuf;
}
sf_command(sf_dst,SFC_SET_NORM_DOUBLE,NULL,SF_FALSE);
//
// Initialize Decoder
//
hDecoder = dlmp4.NeAACDecOpen();
config = dlmp4.NeAACDecGetCurrentConfiguration(hDecoder);
config->outputFormat = FAAD_FMT_FLOAT;
config->downMatrix = 1; // Downmix >2 channels to stereo.
if(!dlmp4.NeAACDecSetConfiguration(hDecoder, config)) {
ret = RDAudioConvert::ErrorInvalidSource;
goto out_decoder;
}
if(dlmp4.NeAACDecInit2(hDecoder, aacConfigBuffer, aacConfigSize, &foundSampleRate, &foundChannels) < 0) {
ret = RDAudioConvert::ErrorInvalidSource;
goto out_decoder;
}
if(foundSampleRate != wave->getSamplesPerSec() || foundChannels != wave->getChannels()) {
fprintf(stderr, "M4A header information inconsistent with actual file? Header: %u/%u; file: %lu/%u\n",
wave->getSamplesPerSec(), (unsigned)wave->getChannels(), foundSampleRate, (unsigned)foundChannels);
ret = RDAudioConvert::ErrorInvalidSource;
goto out_decoder;
}
//
// Decode
//
for(MP4SampleId i = firstSample; i <= lastSample; ++i) {
uint32_t aacBytes = aacBufSize;
if(!dlmp4.MP4ReadSample(f, audioTrack, i, &aacBuf, &aacBytes, 0, 0, 0, 0)) {
ret = RDAudioConvert::ErrorInvalidSource;
break;
}
NeAACDecFrameInfo frameInfo;
// The library docs are not clear about the lifetime or cleanup of sample_buffer.
// I hope it lives until the next NeAACDecDecode call, and is cleaned up by NeAACDecClose
void* sample_buffer = dlmp4.NeAACDecDecode(hDecoder, &frameInfo, aacBuf, aacBytes);
if(!sample_buffer) {
ret = RDAudioConvert::ErrorInvalidSource;
break;
}
UpdatePeak((const float*)sample_buffer, frameInfo.samples);
if(sf_write_float(sf_dst, (const float*)sample_buffer, frameInfo.samples) != (sf_count_t)frameInfo.samples) {
ret = RDAudioConvert::ErrorInternal;
break;
}
}
//
// Cleanup
//
out_decoder:
dlmp4.NeAACDecClose(hDecoder);
// out_sf:
sf_close(sf_dst);
out_mp4_configbuf:
free(aacConfigBuffer);
out_mp4_buf:
free(aacBuf);
out_mp4:
dlmp4.MP4Close(f, 0);
return ret;
#else
return RDAudioConvert::ErrorFormatNotSupported;
#endif
}
RDAudioConvert::ErrorCode RDAudioConvert::Stage1SndFile(const QString &dstfile,
SNDFILE *sf_src,