Autres/audacity
1
0
Fork 0
mirror of https://github.com/cookiengineer/audacity synced 2025-04-30 07:39:42 +02:00
Code Issues Releases Wiki Activity
audacity/src/export/ExportFFmpeg.cpp

1252 lines
42 KiB
C++
Raw Blame History

/**********************************************************************
Audacity: A Digital Audio Editor
ExportFFmpeg.cpp
Audacity(R) is copyright (c) 1999-2009 Audacity Team.
License: GPL v2. See License.txt.
LRN
******************************************************************//**
\class ExportFFmpeg
\brief Controlling class for FFmpeg exporting. Creates the options
dialog of the appropriate type, adds tags and invokes the export
function.
*//*******************************************************************/
#include "../Audacity.h" // keep ffmpeg before wx because they interact // for USE_* macros
#include "../FFmpeg.h" // and Audacity.h before FFmpeg for config*.h
#include <wx/choice.h>
#include <wx/intl.h>
#include <wx/timer.h>
#include <wx/string.h>
#include <wx/textctrl.h>
#include <wx/listbox.h>
#include <wx/window.h>
#include <wx/spinctrl.h>
#include <wx/combobox.h>
#include "../FileFormats.h"
#include "../Mix.h"
#include "../ProjectSettings.h"
#include "../Tags.h"
#include "../Track.h"
#include "../widgets/AudacityMessageBox.h"
#include "../widgets/ErrorDialog.h"
#include "../widgets/ProgressDialog.h"
#include "../wxFileNameWrapper.h"
#include "Export.h"
#include "ExportFFmpegDialogs.h"
#if defined(WIN32) && _MSC_VER < 1900
#define snprintf _snprintf
#endif
#if defined(USE_FFMPEG)
// Define this to automatically resample audio to the nearest supported sample rate
#define FFMPEG_AUTO_RESAMPLE 1
static bool CheckFFmpegPresence(bool quiet = false)
{
bool result = true;
PickFFmpegLibs();
if (!FFmpegLibsInst()->ValidLibsLoaded())
{
if (!quiet)
{
AudacityMessageBox(XO(
"Properly configured FFmpeg is required to proceed.\nYou can configure it at Preferences > Libraries."));
}
result = false;
}
DropFFmpegLibs();
return result;
}
static int AdjustFormatIndex(int format)
{
int subFormat = -1;
for (int i = 0; i <= FMT_OTHER; i++)
{
if (ExportFFmpegOptions::fmts[i].compiledIn) subFormat++;
if (subFormat == format || i == FMT_OTHER)
{
subFormat = i;
break;
}
}
return subFormat;
}
//----------------------------------------------------------------------------
// ExportFFmpeg
//----------------------------------------------------------------------------
class ExportFFmpeg final : public ExportPlugin
{
public:
ExportFFmpeg();
~ExportFFmpeg() override;
/// Callback, called from GetFilename
bool CheckFileName(wxFileName &filename, int format = 0) override;
/// Format initialization
bool Init(const char *shortname, AudacityProject *project, const Tags *metadata, int subformat);
/// Codec initialization
bool InitCodecs(AudacityProject *project);
/// Writes metadata
bool AddTags(const Tags *metadata);
/// Sets individual metadata values
void SetMetadata(const Tags *tags, const char *name, const wxChar *tag);
/// Encodes audio
bool EncodeAudioFrame(int16_t *pFrame, size_t frameSize);
/// Flushes audio encoder
bool Finalize();
void FreeResources();
/// Creates options panel
///\param format - index of export type
void OptionsCreate(ShuttleGui &S, int format) override;
/// Check whether or not current project sample rate is compatible with the export codec
bool CheckSampleRate(int rate, int lowrate, int highrate, const int *sampRates);
/// Asks user to resample the project or cancel the export procedure
int AskResample(int bitrate, int rate, int lowrate, int highrate, const int *sampRates);
/// Exports audio
///\param project Audacity project
///\param fName output file name
///\param selectedOnly true if exporting only selected audio
///\param t0 audio start time
///\param t1 audio end time
///\param mixerSpec mixer
///\param metadata tags to write into file
///\param subformat index of export type
///\return true if export succeeded
ProgressResult Export(AudacityProject *project,
std::unique_ptr<ProgressDialog> &pDialog,
unsigned channels,
const wxFileNameWrapper &fName,
bool selectedOnly,
double t0,
double t1,
MixerSpec *mixerSpec = NULL,
const Tags *metadata = NULL,
int subformat = 0) override;
private:
AVOutputFormat * mEncFormatDesc{}; // describes our output file to libavformat
int default_frame_size{};
AVStream * mEncAudioStream{}; // the output audio stream (may remain NULL)
int mEncAudioFifoOutBufSiz{};
wxFileNameWrapper mName;
int mSubFormat{};
int mBitRate{};
int mSampleRate{};
unsigned mChannels{};
bool mSupportsUTF8{};
// Smart pointer fields, their order is the reverse in which they are reset in FreeResources():
AVFifoBufferHolder mEncAudioFifo; // FIFO to write incoming audio samples into
AVMallocHolder<int16_t> mEncAudioFifoOutBuf; // buffer to read _out_ of the FIFO into
AVFormatContextHolder mEncFormatCtx; // libavformat's context for our output file
UFileHolder mUfileCloser;
AVCodecContextHolder mEncAudioCodecCtx; // the encoder for the output audio stream
};
ExportFFmpeg::ExportFFmpeg()
: ExportPlugin()
{
mEncFormatDesc = NULL; // describes our output file to libavformat
mEncAudioStream = NULL; // the output audio stream (may remain NULL)
#define MAX_AUDIO_PACKET_SIZE (128 * 1024)
mEncAudioFifoOutBufSiz = 0;
mSampleRate = 0;
mSupportsUTF8 = true;
PickFFmpegLibs(); // DropFFmpegLibs() call is in ExportFFmpeg destructor
int avfver = FFmpegLibsInst()->ValidLibsLoaded() ? avformat_version() : 0;
int newfmt;
// Adds export types from the export type list
for (newfmt = 0; newfmt < FMT_LAST; newfmt++)
{
wxString shortname(ExportFFmpegOptions::fmts[newfmt].shortname);
//Don't hide export types when there's no av-libs, and don't hide FMT_OTHER
if (newfmt < FMT_OTHER && FFmpegLibsInst()->ValidLibsLoaded())
{
// Format/Codec support is compiled in?
AVOutputFormat *avoformat = av_guess_format(shortname.mb_str(), NULL, NULL);
AVCodec *avcodec = avcodec_find_encoder(ExportFFmpegOptions::fmts[newfmt].codecid);
if (avoformat == NULL || avcodec == NULL)
{
ExportFFmpegOptions::fmts[newfmt].compiledIn = false;
continue;
}
}
int fmtindex = AddFormat() - 1;
SetFormat(ExportFFmpegOptions::fmts[newfmt].name,fmtindex);
AddExtension(ExportFFmpegOptions::fmts[newfmt].extension,fmtindex);
// For some types add other extensions
switch(newfmt)
{
case FMT_M4A:
AddExtension(wxT("3gp"),fmtindex);
AddExtension(wxT("m4r"),fmtindex);
AddExtension(wxT("mp4"),fmtindex);
break;
case FMT_WMA2:
AddExtension(wxT("asf"),fmtindex);
AddExtension(wxT("wmv"),fmtindex);
break;
default:
break;
}
SetMaxChannels(ExportFFmpegOptions::fmts[newfmt].maxchannels,fmtindex);
SetDescription(ExportFFmpegOptions::fmts[newfmt].description, fmtindex);
int canmeta = ExportFFmpegOptions::fmts[newfmt].canmetadata;
if (canmeta && (canmeta == AV_CANMETA || canmeta <= avfver))
{
SetCanMetaData(true,fmtindex);
}
else
{
SetCanMetaData(false,fmtindex);
}
}
}
ExportFFmpeg::~ExportFFmpeg()
{
DropFFmpegLibs();
}
bool ExportFFmpeg::CheckFileName(wxFileName & WXUNUSED(filename), int WXUNUSED(format))
{
bool result = true;
// Show "Locate FFmpeg" dialog
if (!CheckFFmpegPresence(true))
{
FFmpegLibsInst()->FindLibs(NULL);
FFmpegLibsInst()->FreeLibs();
return LoadFFmpeg(true);
}
return result;
}
bool ExportFFmpeg::Init(const char *shortname, AudacityProject *project, const Tags *metadata, int subformat)
{
// This will undo the acquisition of resources along any early exit path:
auto deleter = [](ExportFFmpeg *This) {
if (This)
This->FreeResources();
};
std::unique_ptr<ExportFFmpeg, decltype(deleter)> cleanup{ this, deleter };
int err;
//FFmpegLibsInst()->LoadLibs(NULL,true); //Loaded at startup or from Prefs now
if (!FFmpegLibsInst()->ValidLibsLoaded())
return false;
av_log_set_callback(av_log_wx_callback);
// See if libavformat has modules that can write our output format. If so, mEncFormatDesc
// will describe the functions used to write the format (used internally by libavformat)
// and the default video/audio codecs that the format uses.
const auto path = mName.GetFullPath();
if ((mEncFormatDesc = av_guess_format(shortname, OSINPUT(path), NULL)) == NULL)
{
AudacityMessageBox(
XO(
"FFmpeg : ERROR - Can't determine format description for file \"%s\".")
.Format( path ),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION );
return false;
}
// mEncFormatCtx is used by libavformat to carry around context data re our output file.
mEncFormatCtx.reset(avformat_alloc_context());
if (!mEncFormatCtx)
{
AudacityMessageBox(
XO("FFmpeg : ERROR - Can't allocate output format context."),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
// Initialise the output format context.
mEncFormatCtx->oformat = mEncFormatDesc;
memcpy(mEncFormatCtx->filename, OSINPUT(path), strlen(OSINPUT(path))+1);
// At the moment Audacity can export only one audio stream
if ((mEncAudioStream = avformat_new_stream(mEncFormatCtx.get(), NULL)) == NULL)
{
AudacityMessageBox(
XO("FFmpeg : ERROR - Can't add audio stream to output file \"%s\".")
.Format( path ),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
// Documentation for avformat_new_stream says
// "User is required to call avcodec_close() and avformat_free_context() to clean
// up the allocation by avformat_new_stream()."
// We use smart pointers that ensure these cleanups either in their destructors or
// sooner if they are reset. These are std::unique_ptr with nondefault deleter
// template parameters.
// mEncFormatCtx takes care of avformat_free_context(), so
// mEncAudioStream can be a plain pointer.
// mEncAudioCodecCtx now becomes responsible for closing the codec:
mEncAudioCodecCtx.reset(mEncAudioStream->codec);
mEncAudioStream->id = 0;
// Open the output file.
if (!(mEncFormatDesc->flags & AVFMT_NOFILE))
{
if ((err = ufile_fopen(&mEncFormatCtx->pb, path, AVIO_FLAG_WRITE)) < 0)
{
AudacityMessageBox(
XO("FFmpeg : ERROR - Can't open output file \"%s\" to write. Error code is %d.")
.Format( path, err ),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
// Give mUfileCloser responsibility
mUfileCloser.reset(mEncFormatCtx->pb);
}
// Open the audio stream's codec and initialise any stream related data.
if (!InitCodecs(project))
return false;
if (metadata == NULL)
metadata = &Tags::Get( *project );
// Add metadata BEFORE writing the header.
// At the moment that works with ffmpeg-git and ffmpeg-0.5 for MP4.
if (GetCanMetaData(subformat))
{
mSupportsUTF8 = ExportFFmpegOptions::fmts[mSubFormat].canutf8;
AddTags(metadata);
}
// Write headers to the output file.
if ((err = avformat_write_header(mEncFormatCtx.get(), NULL)) < 0)
{
AudacityMessageBox(
XO("FFmpeg : ERROR - Can't write headers to output file \"%s\". Error code is %d.")
.Format( path, err ),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
// Only now, we can keep all the resources until after Finalize().
// Cancel the local cleanup.
cleanup.release();
return true;
}
bool ExportFFmpeg::CheckSampleRate(int rate, int lowrate, int highrate, const int *sampRates)
{
if (lowrate && highrate)
{
if (rate < lowrate || rate > highrate)
{
return false;
}
}
if (sampRates)
{
for (int i = 0; sampRates[i] > 0; i++)
{
if (rate == sampRates[i])
{
return true;
}
}
}
return false;
}
static int set_dict_int(AVDictionary **dict, const char *key, int val)
{
char val_str[256];
snprintf(val_str, sizeof(val_str), "%d", val);
return av_dict_set(dict, key, val_str, 0);
}
bool ExportFFmpeg::InitCodecs(AudacityProject *project)
{
const auto &settings = ProjectSettings::Get( *project );
AVCodec *codec = NULL;
AVDictionary *options = NULL;
AVDictionaryCleanup cleanup{ &options };
// Get the sample rate from the passed settings if we haven't set it before.
// Doing this only when not set allows us to carry the sample rate from one
// iteration of ExportMultiple to the next. This prevents multiple resampling
// dialogs in the event the codec can't support the specified rate.
if (!mSampleRate)
{
mSampleRate = (int)settings.GetRate();
}
// Configure the audio stream's codec context.
mEncAudioCodecCtx->codec_id = ExportFFmpegOptions::fmts[mSubFormat].codecid;
mEncAudioCodecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
mEncAudioCodecCtx->codec_tag = av_codec_get_tag(mEncFormatCtx->oformat->codec_tag,mEncAudioCodecCtx->codec_id);
mEncAudioCodecCtx->global_quality = -99999; //quality mode is off by default;
// Each export type has its own settings
switch (mSubFormat)
{
case FMT_M4A:
{
int q = gPrefs->Read(wxT("/FileFormats/AACQuality"),-99999);
mEncAudioCodecCtx->global_quality = q;
q = wxClip( q, 98 * mChannels, 160 * mChannels);
// Set bit rate to between 98 kbps and 320 kbps (if two channels)
mEncAudioCodecCtx->bit_rate = q * 1000;
mEncAudioCodecCtx->profile = FF_PROFILE_AAC_LOW;
mEncAudioCodecCtx->cutoff = 0;
break;
}
case FMT_AC3:
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/AC3BitRate"), 192000);
if (!CheckSampleRate(mSampleRate,ExportFFmpegAC3Options::iAC3SampleRates[0], ExportFFmpegAC3Options::iAC3SampleRates[2], &ExportFFmpegAC3Options::iAC3SampleRates[0]))
mSampleRate = AskResample(mEncAudioCodecCtx->bit_rate,mSampleRate, ExportFFmpegAC3Options::iAC3SampleRates[0], ExportFFmpegAC3Options::iAC3SampleRates[2], &ExportFFmpegAC3Options::iAC3SampleRates[0]);
break;
case FMT_AMRNB:
mSampleRate = 8000;
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/AMRNBBitRate"), 12200);
break;
case FMT_OPUS:
av_dict_set(&options, "b", gPrefs->Read(wxT("/FileFormats/OPUSBitRate"), wxT("128000")).ToUTF8(), 0);
av_dict_set(&options, "vbr", gPrefs->Read(wxT("/FileFormats/OPUSVbrMode"), wxT("on")).ToUTF8(), 0);
av_dict_set(&options, "compression_level", gPrefs->Read(wxT("/FileFormats/OPUSCompression"), wxT("10")).ToUTF8(), 0);
av_dict_set(&options, "frame_duration", gPrefs->Read(wxT("/FileFormats/OPUSFrameDuration"), wxT("20")).ToUTF8(), 0);
av_dict_set(&options, "application", gPrefs->Read(wxT("/FileFormats/OPUSApplication"), wxT("audio")).ToUTF8(), 0);
av_dict_set(&options, "cutoff", gPrefs->Read(wxT("/FileFormats/OPUSCutoff"), wxT("0")).ToUTF8(), 0);
av_dict_set(&options, "mapping_family", mChannels <= 2 ? "0" : "255", 0);
break;
case FMT_WMA2:
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/WMABitRate"), 198000);
if (!CheckSampleRate(mSampleRate,ExportFFmpegWMAOptions::iWMASampleRates[0], ExportFFmpegWMAOptions::iWMASampleRates[4], &ExportFFmpegWMAOptions::iWMASampleRates[0]))
mSampleRate = AskResample(mEncAudioCodecCtx->bit_rate,mSampleRate, ExportFFmpegWMAOptions::iWMASampleRates[0], ExportFFmpegWMAOptions::iWMASampleRates[4], &ExportFFmpegWMAOptions::iWMASampleRates[0]);
break;
case FMT_OTHER:
av_dict_set(&mEncAudioStream->metadata, "language", gPrefs->Read(wxT("/FileFormats/FFmpegLanguage"),wxT("")).ToUTF8(), 0);
mEncAudioCodecCtx->sample_rate = gPrefs->Read(wxT("/FileFormats/FFmpegSampleRate"),(long)0);
if (mEncAudioCodecCtx->sample_rate != 0) mSampleRate = mEncAudioCodecCtx->sample_rate;
mEncAudioCodecCtx->bit_rate = gPrefs->Read(wxT("/FileFormats/FFmpegBitRate"), (long)0);
strncpy((char *)&mEncAudioCodecCtx->codec_tag,gPrefs->Read(wxT("/FileFormats/FFmpegTag"),wxT("")).mb_str(wxConvUTF8),4);
mEncAudioCodecCtx->global_quality = gPrefs->Read(wxT("/FileFormats/FFmpegQuality"),(long)-99999);
mEncAudioCodecCtx->cutoff = gPrefs->Read(wxT("/FileFormats/FFmpegCutOff"),(long)0);
mEncAudioCodecCtx->flags2 = 0;
if (gPrefs->Read(wxT("/FileFormats/FFmpegBitReservoir"),true))
av_dict_set(&options, "reservoir", "1", 0);
if (gPrefs->Read(wxT("/FileFormats/FFmpegVariableBlockLen"),true)) mEncAudioCodecCtx->flags2 |= 0x0004; //WMA only?
mEncAudioCodecCtx->compression_level = gPrefs->Read(wxT("/FileFormats/FFmpegCompLevel"),-1);
mEncAudioCodecCtx->frame_size = gPrefs->Read(wxT("/FileFormats/FFmpegFrameSize"),(long)0);
//FIXME The list of supported options for the selected encoder should be extracted instead of a few hardcoded
set_dict_int(&options, "lpc_coeff_precision", gPrefs->Read(wxT("/FileFormats/FFmpegLPCCoefPrec"),(long)0));
set_dict_int(&options, "min_prediction_order", gPrefs->Read(wxT("/FileFormats/FFmpegMinPredOrder"),(long)-1));
set_dict_int(&options, "max_prediction_order", gPrefs->Read(wxT("/FileFormats/FFmpegMaxPredOrder"),(long)-1));
set_dict_int(&options, "min_partition_order", gPrefs->Read(wxT("/FileFormats/FFmpegMinPartOrder"),(long)-1));
set_dict_int(&options, "max_partition_order", gPrefs->Read(wxT("/FileFormats/FFmpegMaxPartOrder"),(long)-1));
set_dict_int(&options, "prediction_order_method", gPrefs->Read(wxT("/FileFormats/FFmpegPredOrderMethod"),(long)0));
set_dict_int(&options, "muxrate", gPrefs->Read(wxT("/FileFormats/FFmpegMuxRate"),(long)0));
mEncFormatCtx->packet_size = gPrefs->Read(wxT("/FileFormats/FFmpegPacketSize"),(long)0);
codec = avcodec_find_encoder_by_name(gPrefs->Read(wxT("/FileFormats/FFmpegCodec")).ToUTF8());
if (!codec)
mEncAudioCodecCtx->codec_id = mEncFormatDesc->audio_codec;
break;
default:
return false;
}
// This happens if user refused to resample the project
if (mSampleRate == 0) return false;
if (mEncAudioCodecCtx->global_quality >= 0)
{
mEncAudioCodecCtx->flags |= AV_CODEC_FLAG_QSCALE;
}
else mEncAudioCodecCtx->global_quality = 0;
mEncAudioCodecCtx->global_quality = mEncAudioCodecCtx->global_quality * FF_QP2LAMBDA;
mEncAudioCodecCtx->sample_rate = mSampleRate;
mEncAudioCodecCtx->channels = mChannels;
mEncAudioCodecCtx->channel_layout = av_get_default_channel_layout(mChannels);
mEncAudioCodecCtx->time_base.num = 1;
mEncAudioCodecCtx->time_base.den = mEncAudioCodecCtx->sample_rate;
mEncAudioCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
mEncAudioCodecCtx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
if (mEncAudioCodecCtx->codec_id == AV_CODEC_ID_AC3)
{
// As of Jan 4, 2011, the default AC3 encoder only accept SAMPLE_FMT_FLT samples.
// But, currently, Audacity only supports SAMPLE_FMT_S16. So, for now, look for the
// "older" AC3 codec. this is not a proper solution, but will suffice until other
// encoders no longer support SAMPLE_FMT_S16.
codec = avcodec_find_encoder_by_name("ac3_fixed");
}
if (!codec)
{
codec = avcodec_find_encoder(mEncAudioCodecCtx->codec_id);
}
// Is the required audio codec compiled into libavcodec?
if (codec == NULL)
{
AudacityMessageBox(
XO(
/* i18n-hint: "codec" is short for a "coder-decoder" algorithm */
"FFmpeg cannot find audio codec 0x%x.\nSupport for this codec is probably not compiled in.")
.Format( (unsigned int) mEncAudioCodecCtx->codec_id ),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
if (codec->sample_fmts) {
for (int i=0; codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; i++) {
enum AVSampleFormat fmt = codec->sample_fmts[i];
if ( fmt == AV_SAMPLE_FMT_U8
|| fmt == AV_SAMPLE_FMT_U8P
|| fmt == AV_SAMPLE_FMT_S16
|| fmt == AV_SAMPLE_FMT_S16P
|| fmt == AV_SAMPLE_FMT_S32
|| fmt == AV_SAMPLE_FMT_S32P
|| fmt == AV_SAMPLE_FMT_FLT
|| fmt == AV_SAMPLE_FMT_FLTP) {
mEncAudioCodecCtx->sample_fmt = fmt;
}
if ( fmt == AV_SAMPLE_FMT_S16
|| fmt == AV_SAMPLE_FMT_S16P)
break;
}
}
if (codec->supported_samplerates)
{
// Workaround for crash in bug #2378. Proper fix is to get a newer version of FFmpeg.
if (codec->id == AV_CODEC_ID_AAC)
{
std::vector<int> rates;
int i = 0;
while (codec->supported_samplerates[i] && codec->supported_samplerates[i] != 7350)
{
rates.push_back(codec->supported_samplerates[i++]);
}
rates.push_back(0);
if (!CheckSampleRate(mSampleRate, 0, 0, rates.data()))
{
mEncAudioCodecCtx->sample_rate = mSampleRate = AskResample(0, mSampleRate, 0, 0, rates.data());
}
}
else
{
if (!CheckSampleRate(mSampleRate, 0, 0, codec->supported_samplerates))
{
mEncAudioCodecCtx->sample_rate = mSampleRate = AskResample(0, mSampleRate, 0, 0, codec->supported_samplerates);
}
}
// This happens if user refused to resample the project
if (mSampleRate == 0)
{
return false;
}
}
if (mEncFormatCtx->oformat->flags & AVFMT_GLOBALHEADER)
{
mEncAudioCodecCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
mEncFormatCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
// Open the codec.
int rc = avcodec_open2(mEncAudioCodecCtx.get(), codec, &options);
if (rc < 0)
{
TranslatableString errmsg;
switch (rc)
{
case -EPERM:
errmsg = XO("The codec reported a generic error (EPERM)");
break;
case -EINVAL:
errmsg = XO("The codec reported an invalid parameter (EINVAL)");
break;
default:
char buf[AV_ERROR_MAX_STRING_SIZE];
av_strerror(rc, buf, sizeof(buf));
errmsg = Verbatim(buf);
}
AudacityMessageBox(
/* i18n-hint: "codec" is short for a "coder-decoder" algorithm */
XO("Can't open audio codec \"%s\" (0x%x)\n\n%s")
.Format(codec->name, mEncAudioCodecCtx->codec_id, errmsg),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
default_frame_size = mEncAudioCodecCtx->frame_size;
if (default_frame_size == 0)
default_frame_size = 1024; // arbitrary non zero value;
wxLogDebug(wxT("FFmpeg : Audio Output Codec Frame Size: %d samples."), mEncAudioCodecCtx->frame_size);
// The encoder may require a minimum number of raw audio samples for each encoding but we can't
// guarantee we'll get this minimum each time an audio frame is decoded from the input file so
// we use a FIFO to store up incoming raw samples until we have enough for one call to the codec.
mEncAudioFifo.reset(av_fifo_alloc(1024));
mEncAudioFifoOutBufSiz = 2*MAX_AUDIO_PACKET_SIZE;
// Allocate a buffer to read OUT of the FIFO into. The FIFO maintains its own buffer internally.
mEncAudioFifoOutBuf.reset(static_cast<int16_t*>(av_malloc(mEncAudioFifoOutBufSiz)));
if (!mEncAudioFifoOutBuf)
{
AudacityMessageBox(
XO("FFmpeg : ERROR - Can't allocate buffer to read into from audio FIFO."),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION
);
return false;
}
return true;
}
// Returns 0 if no more output, 1 if more output, negative if error
static int encode_audio(AVCodecContext *avctx, AVPacket *pkt, int16_t *audio_samples, int nb_samples)
{
// Assume *pkt is already initialized.
int i, ch, buffer_size, ret, got_output = 0;
AVMallocHolder<uint8_t> samples;
AVFrameHolder frame;
if (audio_samples) {
frame.reset(av_frame_alloc());
if (!frame)
return AVERROR(ENOMEM);
frame->nb_samples = nb_samples;
frame->format = avctx->sample_fmt;
#if !defined(DISABLE_DYNAMIC_LOADING_FFMPEG) || (LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 13, 0))
frame->channel_layout = avctx->channel_layout;
#endif
buffer_size = av_samples_get_buffer_size(NULL, avctx->channels, frame->nb_samples,
avctx->sample_fmt, 0);
if (buffer_size < 0) {
AudacityMessageBox(
XO("FFmpeg : ERROR - Could not get sample buffer size"),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION
);
return buffer_size;
}
samples.reset(static_cast<uint8_t*>(av_malloc(buffer_size)));
if (!samples) {
AudacityMessageBox(
XO("FFmpeg : ERROR - Could not allocate bytes for samples buffer"),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION
);
return AVERROR(ENOMEM);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame.get(), avctx->channels, avctx->sample_fmt,
samples.get(), buffer_size, 0);
if (ret < 0) {
AudacityMessageBox(
XO("FFmpeg : ERROR - Could not setup audio frame"),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION
);
return ret;
}
for (ch = 0; ch < avctx->channels; ch++) {
for (i = 0; i < frame->nb_samples; i++) {
switch(avctx->sample_fmt) {
case AV_SAMPLE_FMT_U8:
((uint8_t*)(frame->data[0]))[ch + i*avctx->channels] = audio_samples[ch + i*avctx->channels]/258 + 128;
break;
case AV_SAMPLE_FMT_U8P:
((uint8_t*)(frame->data[ch]))[i] = audio_samples[ch + i*avctx->channels]/258 + 128;
break;
case AV_SAMPLE_FMT_S16:
((int16_t*)(frame->data[0]))[ch + i*avctx->channels] = audio_samples[ch + i*avctx->channels];
break;
case AV_SAMPLE_FMT_S16P:
((int16_t*)(frame->data[ch]))[i] = audio_samples[ch + i*avctx->channels];
break;
case AV_SAMPLE_FMT_S32:
((int32_t*)(frame->data[0]))[ch + i*avctx->channels] = audio_samples[ch + i*avctx->channels]<<16;
break;
case AV_SAMPLE_FMT_S32P:
((int32_t*)(frame->data[ch]))[i] = audio_samples[ch + i*avctx->channels]<<16;
break;
case AV_SAMPLE_FMT_FLT:
((float*)(frame->data[0]))[ch + i*avctx->channels] = audio_samples[ch + i*avctx->channels] / 32767.0;
break;
case AV_SAMPLE_FMT_FLTP:
((float*)(frame->data[ch]))[i] = audio_samples[ch + i*avctx->channels] / 32767.;
break;
case AV_SAMPLE_FMT_NONE:
case AV_SAMPLE_FMT_DBL:
case AV_SAMPLE_FMT_DBLP:
case AV_SAMPLE_FMT_NB:
wxASSERT(false);
break;
}
}
}
}
pkt->data = NULL; // packet data will be allocated by the encoder
pkt->size = 0;
ret = avcodec_encode_audio2(avctx, pkt, frame.get(), &got_output);
if (ret < 0) {
AudacityMessageBox(
XO("FFmpeg : ERROR - encoding frame failed"),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION
);
return ret;
}
pkt->dts = pkt->pts = AV_NOPTS_VALUE; // we dont set frame.pts thus dont trust the AVPacket ts
return got_output;
}
bool ExportFFmpeg::Finalize()
{
// Flush the audio FIFO and encoder.
for (;;)
{
AVPacketEx pkt;
const int nFifoBytes = av_fifo_size(mEncAudioFifo.get()); // any bytes left in audio FIFO?
int encodeResult = 0;
// Flush the audio FIFO first if necessary. It won't contain a _full_ audio frame because
// if it did we'd have pulled it from the FIFO during the last encodeAudioFrame() call
if (nFifoBytes > 0)
{
const int nAudioFrameSizeOut = default_frame_size * mEncAudioCodecCtx->channels * sizeof(int16_t);
if (nAudioFrameSizeOut > mEncAudioFifoOutBufSiz || nFifoBytes > mEncAudioFifoOutBufSiz) {
AudacityMessageBox(
XO("FFmpeg : ERROR - Too much remaining data."),
XO("FFmpeg Error"),
wxOK | wxCENTER | wxICON_EXCLAMATION
);
return false;
}
// We have an incomplete buffer of samples left, encode it.
// If codec supports CODEC_CAP_SMALL_LAST_FRAME, we can feed it with smaller frame
// Or if frame_size is 1, then it's some kind of PCM codec, they don't have frames and will be fine with the samples
// Otherwise we'll send a full frame of audio + silence padding to ensure all audio is encoded
int frame_size = default_frame_size;
if (mEncAudioCodecCtx->codec->capabilities & AV_CODEC_CAP_SMALL_LAST_FRAME ||
frame_size == 1)
frame_size = nFifoBytes / (mEncAudioCodecCtx->channels * sizeof(int16_t));
wxLogDebug(wxT("FFmpeg : Audio FIFO still contains %d bytes, writing %d sample frame ..."),
nFifoBytes, frame_size);
// Fill audio buffer with zeroes. If codec tries to read the whole buffer,
// it will just read silence. If not - who cares?
memset(mEncAudioFifoOutBuf.get(), 0, mEncAudioFifoOutBufSiz);
//const AVCodec *codec = mEncAudioCodecCtx->codec;
// Pull the bytes out from the FIFO and feed them to the encoder.
if (av_fifo_generic_read(mEncAudioFifo.get(), mEncAudioFifoOutBuf.get(), nFifoBytes, NULL) == 0)
{
encodeResult = encode_audio(mEncAudioCodecCtx.get(), &pkt, mEncAudioFifoOutBuf.get(), frame_size);
}
else
{
wxLogDebug(wxT("FFmpeg : Reading from Audio FIFO failed, aborting"));
// TODO: more precise message
ShowExportErrorDialog("FFmpeg:825");
return false;
}
}
else
{
// Fifo is empty, flush encoder. May be called multiple times.
encodeResult = encode_audio(mEncAudioCodecCtx.get(), &pkt, NULL, 0);
}
if (encodeResult < 0) {
// TODO: more precise message
ShowExportErrorDialog("FFmpeg:837");
return false;
}
else if (encodeResult == 0)
break;
// We have a packet, send to the muxer
pkt.stream_index = mEncAudioStream->index;
// Set presentation time of frame (currently in the codec's timebase) in the stream timebase.
if (pkt.pts != int64_t(AV_NOPTS_VALUE))
pkt.pts = av_rescale_q(pkt.pts, mEncAudioCodecCtx->time_base, mEncAudioStream->time_base);
if (pkt.dts != int64_t(AV_NOPTS_VALUE))
pkt.dts = av_rescale_q(pkt.dts, mEncAudioCodecCtx->time_base, mEncAudioStream->time_base);
if (pkt.duration)
pkt.duration = av_rescale_q(pkt.duration, mEncAudioCodecCtx->time_base, mEncAudioStream->time_base);
if (av_interleaved_write_frame(mEncFormatCtx.get(), &pkt) != 0)
{
AudacityMessageBox(
XO("FFmpeg : ERROR - Couldn't write last audio frame to output file."),
XO("FFmpeg Error"),
wxOK | wxCENTER | wxICON_EXCLAMATION
);
return false;
}
}
// Write any file trailers.
if (av_write_trailer(mEncFormatCtx.get()) != 0) {
// TODO: more precise message
ShowExportErrorDialog("FFmpeg:868");
return false;
}
return true;
}
void ExportFFmpeg::FreeResources()
{
// Close the codecs.
mEncAudioCodecCtx.reset();
// Close the output file if we created it.
mUfileCloser.reset();
// Free any buffers or structures we allocated.
mEncFormatCtx.reset();
mEncAudioFifoOutBuf.reset();
mEncAudioFifoOutBufSiz = 0;
mEncAudioFifo.reset();
av_log_set_callback(av_log_default_callback);
}
// All paths in this that fail must report their error to the user.
bool ExportFFmpeg::EncodeAudioFrame(int16_t *pFrame, size_t frameSize)
{
int nBytesToWrite = 0;
uint8_t *pRawSamples = NULL;
int nAudioFrameSizeOut = default_frame_size * mEncAudioCodecCtx->channels * sizeof(int16_t);
int ret;
nBytesToWrite = frameSize;
pRawSamples = (uint8_t*)pFrame;
if (av_fifo_realloc2(mEncAudioFifo.get(), av_fifo_size(mEncAudioFifo.get()) + frameSize) < 0) {
ShowExportErrorDialog("FFmpeg:905");
return false;
}
// Put the raw audio samples into the FIFO.
ret = av_fifo_generic_write(mEncAudioFifo.get(), pRawSamples, nBytesToWrite,NULL);
if (ret != nBytesToWrite) {
ShowExportErrorDialog("FFmpeg:913");
return false;
}
if (nAudioFrameSizeOut > mEncAudioFifoOutBufSiz) {
AudacityMessageBox(
XO("FFmpeg : ERROR - nAudioFrameSizeOut too large."),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION
);
return false;
}
// Read raw audio samples out of the FIFO in nAudioFrameSizeOut byte-sized groups to encode.
while ( av_fifo_size(mEncAudioFifo.get()) >= nAudioFrameSizeOut)
{
ret = av_fifo_generic_read(mEncAudioFifo.get(), mEncAudioFifoOutBuf.get(), nAudioFrameSizeOut, NULL);
AVPacketEx pkt;
ret= encode_audio(mEncAudioCodecCtx.get(),
&pkt, // out
mEncAudioFifoOutBuf.get(), // in
default_frame_size);
if (ret < 0)
{
AudacityMessageBox(
XO("FFmpeg : ERROR - Can't encode audio frame."),
XO("FFmpeg Error"),
wxOK|wxCENTER|wxICON_EXCLAMATION
);
return false;
}
if (ret == 0)
continue;
// Rescale from the codec time_base to the AVStream time_base.
if (pkt.pts != int64_t(AV_NOPTS_VALUE))
pkt.pts = av_rescale_q(pkt.pts, mEncAudioCodecCtx->time_base, mEncAudioStream->time_base);
if (pkt.dts != int64_t(AV_NOPTS_VALUE))
pkt.dts = av_rescale_q(pkt.dts, mEncAudioCodecCtx->time_base, mEncAudioStream->time_base);
//wxLogDebug(wxT("FFmpeg : (%d) Writing audio frame with PTS: %lld."), mEncAudioCodecCtx->frame_number, (long long) pkt.pts);
pkt.stream_index = mEncAudioStream->index;
// Write the encoded audio frame to the output file.
if ((ret = av_interleaved_write_frame(mEncFormatCtx.get(), &pkt)) < 0)
{
ShowDiskFullExportErrorDialog();
return false;
}
}
return true;
}
ProgressResult ExportFFmpeg::Export(AudacityProject *project,
std::unique_ptr<ProgressDialog> &pDialog,
unsigned channels, const wxFileNameWrapper &fName,
bool selectionOnly, double t0, double t1,
MixerSpec *mixerSpec, const Tags *metadata, int subformat)
{
if (!CheckFFmpegPresence())
return ProgressResult::Cancelled;
mChannels = channels;
// subformat index may not correspond directly to fmts[] index, convert it
mSubFormat = AdjustFormatIndex(subformat);
if (channels > ExportFFmpegOptions::fmts[mSubFormat].maxchannels)
{
AudacityMessageBox(
XO(
"Attempted to export %d channels, but maximum number of channels for selected output format is %d")
.Format(
channels,
ExportFFmpegOptions::fmts[mSubFormat].maxchannels ),
XO("Error"));
return ProgressResult::Cancelled;
}
mName = fName;
const auto &tracks = TrackList::Get( *project );
bool ret = true;
if (mSubFormat >= FMT_LAST) {
// TODO: more precise message
ShowExportErrorDialog("FFmpeg:996");
return ProgressResult::Cancelled;
}
wxString shortname(ExportFFmpegOptions::fmts[mSubFormat].shortname);
if (mSubFormat == FMT_OTHER)
shortname = gPrefs->Read(wxT("/FileFormats/FFmpegFormat"),wxT("matroska"));
ret = Init(shortname.mb_str(),project, metadata, subformat);
auto cleanup = finally ( [&] { FreeResources(); } );
if (!ret) {
// TODO: more precise message
ShowExportErrorDialog("FFmpeg:1008");
return ProgressResult::Cancelled;
}
size_t pcmBufferSize = 1024;
auto mixer = CreateMixer(tracks, selectionOnly,
t0, t1,
channels, pcmBufferSize, true,
mSampleRate, int16Sample, true, mixerSpec);
auto updateResult = ProgressResult::Success;
{
InitProgress( pDialog, fName,
selectionOnly
? XO("Exporting selected audio as %s")
.Format( ExportFFmpegOptions::fmts[mSubFormat].description )
: XO("Exporting the audio as %s")
.Format( ExportFFmpegOptions::fmts[mSubFormat].description ) );
auto &progress = *pDialog;
while (updateResult == ProgressResult::Success) {
auto pcmNumSamples = mixer->Process(pcmBufferSize);
if (pcmNumSamples == 0)
break;
short *pcmBuffer = (short *)mixer->GetBuffer();
if (!EncodeAudioFrame(
pcmBuffer, (pcmNumSamples)*sizeof(int16_t)*mChannels)) {
// All errors should already have been reported.
//ShowDiskFullExportErrorDialog();
updateResult = ProgressResult::Cancelled;
break;
}
updateResult = progress.Update(mixer->MixGetCurrentTime() - t0, t1 - t0);
}
}
if ( updateResult != ProgressResult::Cancelled )
if ( !Finalize() ) // Finalize makes its own messages
return ProgressResult::Cancelled;
if ( mUfileCloser.close() != 0 ) {
// TODO: more precise message
ShowExportErrorDialog("FFmpeg:1056");
return ProgressResult::Cancelled;
}
return updateResult;
}
void AddStringTagUTF8(char field[], int size, wxString value)
{
memset(field,0,size);
memcpy(field,value.ToUTF8(),(int)strlen(value.ToUTF8()) > size -1 ? size -1 : strlen(value.ToUTF8()));
}
void AddStringTagANSI(char field[], int size, wxString value)
{
memset(field,0,size);
memcpy(field,value.mb_str(),(int)strlen(value.mb_str()) > size -1 ? size -1 : strlen(value.mb_str()));
}
bool ExportFFmpeg::AddTags(const Tags *tags)
{
if (tags == NULL)
{
return false;
}
SetMetadata(tags, "author", TAG_ARTIST);
SetMetadata(tags, "album", TAG_ALBUM);
SetMetadata(tags, "comment", TAG_COMMENTS);
SetMetadata(tags, "genre", TAG_GENRE);
SetMetadata(tags, "title", TAG_TITLE);
SetMetadata(tags, "year", TAG_YEAR);
SetMetadata(tags, "track", TAG_TRACK);
return true;
}
void ExportFFmpeg::SetMetadata(const Tags *tags, const char *name, const wxChar *tag)
{
if (tags->HasTag(tag))
{
wxString value = tags->GetTag(tag);
av_dict_set(&mEncFormatCtx->metadata, name, mSupportsUTF8 ? value.ToUTF8() : value.mb_str(), 0);
}
}
//----------------------------------------------------------------------------
// AskResample dialog
//----------------------------------------------------------------------------
int ExportFFmpeg::AskResample(int bitrate, int rate, int lowrate, int highrate, const int *sampRates)
{
#if defined(FFMPEG_AUTO_RESAMPLE)
std::vector<int> rates;
for (int i = 0; sampRates[i]; ++i)
{
rates.push_back(sampRates[i]);
}
std::sort(rates.begin(), rates.end());
int bestRate = 0;
for (auto i : rates)
{
bestRate = i;
if (i > rate)
{
break;
}
}
return bestRate;
#else
wxDialogWrapper d(nullptr, wxID_ANY, XO("Invalid sample rate"));
d.SetName();
wxChoice *choice;
ShuttleGui S(&d, eIsCreating);
int selected = -1;
S.StartVerticalLay();
{
S.SetBorder(10);
S.StartStatic(XO("Resample"));
{
S.StartHorizontalLay(wxALIGN_CENTER, false);
{
S.AddTitle(
(bitrate == 0
? XO(
"The project sample rate (%d) is not supported by the current output\nfile format. ")
.Format( rate )
: XO(
"The project sample rate (%d) and bit rate (%d kbps) combination is not\nsupported by the current output file format. ")
.Format( rate, bitrate/1000))
+ XO("You may resample to one of the rates below.")
);
}
S.EndHorizontalLay();
S.StartHorizontalLay(wxALIGN_CENTER, false);
{
choice = S.AddChoice(XO("Sample Rates"),
[&]{
TranslatableStrings choices;
for (int i = 0; sampRates[i] > 0; i++)
{
int label = sampRates[i];
if ((!lowrate || label >= lowrate) && (!highrate || label <= highrate))
{
wxString name = wxString::Format(wxT("%d"),label);
choices.push_back( Verbatim( name ) );
if (label <= rate)
selected = i;
}
}
return choices;
}(),
std::max( 0, selected )
);
}
S.EndHorizontalLay();
}
S.EndStatic();
S.AddStandardButtons();
}
S.EndVerticalLay();
d.Layout();
d.Fit();
d.SetMinSize(d.GetSize());
d.Center();
if (d.ShowModal() == wxID_CANCEL) {
return 0;
}
return wxAtoi(choice->GetStringSelection());
#endif
}
void ExportFFmpeg::OptionsCreate(ShuttleGui &S, int format)
{
// subformat index may not correspond directly to fmts[] index, convert it
mSubFormat = AdjustFormatIndex(format);
if (mSubFormat == FMT_M4A)
{
S.AddWindow(
safenew ExportFFmpegAACOptions{ S.GetParent(), format } );
return;
}
else if (mSubFormat == FMT_AC3)
{
S.AddWindow(
safenew ExportFFmpegAC3Options{ S.GetParent(), format } );
return;
}
else if (mSubFormat == FMT_AMRNB)
{
S.AddWindow(
safenew ExportFFmpegAMRNBOptions{ S.GetParent(), format } );
return;
}
else if (mSubFormat == FMT_OPUS)
{
S.AddWindow(
safenew ExportFFmpegOPUSOptions{ S.GetParent(), format });
return;
}
else if (mSubFormat == FMT_WMA2)
{
S.AddWindow(
safenew ExportFFmpegWMAOptions{ S.GetParent(), format } );
return;
}
else if (mSubFormat == FMT_OTHER)
{
S.AddWindow(
safenew ExportFFmpegCustomOptions{ S.GetParent(), format } );
return;
}
ExportPlugin::OptionsCreate(S, format);
}
static Exporter::RegisteredExportPlugin sRegisteredPlugin{ "FFmpeg",
[]{ return std::make_unique< ExportFFmpeg >(); }
};
#endif
Reference in New Issue View Git Blame Copy Permalink