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MP4 support v2 (undebugged). This replaces callouts to faad with use of libfaad (aka neaacdec) and loads all components dynamically, much like existing MP1/2/3 support.

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This commit is contained in:
Chris Smowton
2014-12-21 20:13:58 +00:00
parent aae6763f4f
commit f63b14cca9
8 changed files with 388 additions and 104 deletions
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206
lib/rdaudioconvert.cpp
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@@ -42,6 +42,10 @@
#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>
@@ -680,72 +684,160 @@ 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) {
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);
RDWaveFile *wave)
{
#ifdef HAVE_MP4_LIBS
SNDFILE *sf_dst=NULL;
SF_INFO sf_dst_info;
int ret = RDAudioConvert::ErrorOk;
if(!LoadMP4Libs()) {
return RDAudioConvert::ErrorFormatNotSupported;
}
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;
//
// Open source
//
MP4FileHandle f = dlmp4.MP4Read(getName());
if(f == MP4_INVALID_FILE_HANDLE)
return RDAudioConvert::ErrorNoSource;
MP4TrackId audioTrack = dlmp4.getMP4AACTrack(f);
MP4SampleId firstSample = 0;
MP4SampleId lastSample = dlmp4.MP4GetTrackNumberOfSamples(f, audioTrack) - 1;
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;
}
}
uint32_t aacBufSize = dlmp4.MP4GetTrackMaxSampleSize(f, audioTrack);
uint8_t* aacBuf = malloc(aacBufSize);
if(!aacBufSize || !aacBuf) {
// Probably the source's fault for specifying a massive buffer.
ret = RDAudioConvert::ErrorInvalidSource;
goto out_mp4;
}
uint8_t* aacConfigBuffer;
uint32_t* aacConfigSize;
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
//
NeAACDecHandle hDecoder = dlmp4.NeAACDecOpen();
NeAACDecConfigurationPtr 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;
}
unsigned long foundSampleRate;
unsigned char foundChannels;
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: %lu/%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(sample_buffer, frameInfo.samples);
if(sf_write_float(dst_sf, sample_buffer, frameInfo.samples) != 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,