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mirror of https://github.com/cookiengineer/audacity synced 2025-09-19 17:40:51 +02:00
audacity/src/export/ExportFFmpeg.cpp
Paul Licameli b256a4fd1a Less code directly inside ShuttleGui code blocks...
... Trying to reduce that just to chained calls on S, or conditional and looping
logic for variations in layout.

Lift some declarations to higher scope; or use expressions that avoid local
variables; or even use lambdas for more complicated computation of arguments
for the member functions of S.
2019-11-29 15:16:51 -05:00

1102 lines
39 KiB
C++

/**********************************************************************
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/progdlg.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/ProgressDialog.h"
#include "Export.h"
#include "ExportFFmpegDialogs.h"
#if defined(WIN32) && _MSC_VER < 1900
#define snprintf _snprintf
#endif
#if defined(USE_FFMPEG)
extern FFmpegLibs *FFmpegLibsInst();
static bool CheckFFmpegPresence(bool quiet = false)
{
bool result = true;
PickFFmpegLibs();
if (!FFmpegLibsInst()->ValidLibsLoaded())
{
if (!quiet)
{
AudacityMessageBox(_("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 intialization
bool Init(const char *shortname, AudacityProject *project, const Tags *metadata, int subformat);
/// Codec intialization
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
wxWindow *OptionsCreate(wxWindow *parent, 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 succeded
ProgressResult Export(AudacityProject *project,
std::unique_ptr<ProgressDialog> &pDialog,
unsigned channels,
const wxString &fName,
bool selectedOnly,
double t0,
double t1,
MixerSpec *mixerSpec = NULL,
const Tags *metadata = NULL,
int subformat = 0) override;
virtual unsigned SequenceNumber() const override { return 70; }
private:
AVOutputFormat * mEncFormatDesc{}; // describes our output file to libavformat
int default_frame_size{};
AVStream * mEncAudioStream{}; // the output audio stream (may remain NULL)
int mEncAudioFifoOutBufSiz{};
wxString 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(wxString(wxT("3gp")),fmtindex);
AddExtension(wxString(wxT("m4r")),fmtindex);
AddExtension(wxString(wxT("mp4")),fmtindex);
break;
case FMT_WMA2:
AddExtension(wxString(wxT("asf")),fmtindex);
AddExtension(wxString(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.
if ((mEncFormatDesc = av_guess_format(shortname, OSINPUT(mName), NULL)) == NULL)
{
AudacityMessageBox(wxString::Format(_("FFmpeg : ERROR - Can't determine format description for file \"%s\"."), mName),
_("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(_("FFmpeg : ERROR - Can't allocate output format context."),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION);
return false;
}
// Initialise the output format context.
mEncFormatCtx->oformat = mEncFormatDesc;
memcpy(mEncFormatCtx->filename, OSINPUT(mName), strlen(OSINPUT(mName))+1);
// At the moment Audacity can export only one audio stream
if ((mEncAudioStream = avformat_new_stream(mEncFormatCtx.get(), NULL)) == NULL)
{
AudacityMessageBox(wxString::Format(_("FFmpeg : ERROR - Can't add audio stream to output file \"%s\"."), mName),
_("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, mName, AVIO_FLAG_WRITE)) < 0)
{
AudacityMessageBox(wxString::Format(_("FFmpeg : ERROR - Can't open output file \"%s\" to write. Error code is %d."), mName, err),
_("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(wxString::Format(_("FFmpeg : ERROR - Can't write headers to output file \"%s\". Error code is %d."), mName,err),
_("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 (rate < lowrate || rate > highrate) return false;
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 };
// 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);
mSampleRate = (int)settings.GetRate();
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;
if (!CheckSampleRate(mSampleRate,
ExportFFmpegOptions::iAACSampleRates[0],
ExportFFmpegOptions::iAACSampleRates[11],
&ExportFFmpegOptions::iAACSampleRates[0]))
{
mSampleRate = AskResample(mEncAudioCodecCtx->bit_rate,mSampleRate,
ExportFFmpegOptions::iAACSampleRates[0],
ExportFFmpegOptions::iAACSampleRates[11],
&ExportFFmpegOptions::iAACSampleRates[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_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 seleced 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(wxString::Format(_("FFmpeg cannot find audio codec 0x%x.\nSupport for this codec is probably not compiled in."), (unsigned int) mEncAudioCodecCtx->codec_id),
_("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 (mEncFormatCtx->oformat->flags & AVFMT_GLOBALHEADER)
{
mEncAudioCodecCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
mEncFormatCtx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
// Open the codec.
if (avcodec_open2(mEncAudioCodecCtx.get(), codec, &options) < 0)
{
AudacityMessageBox(wxString::Format(_("FFmpeg : ERROR - Can't open audio codec 0x%x."),mEncAudioCodecCtx->codec_id),
_("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(
_("FFmpeg : ERROR - Can't allocate buffer to read into from audio FIFO."),
_("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(
_("FFmpeg : ERROR - Could not get sample buffer size"),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION
);
return buffer_size;
}
samples.reset(static_cast<uint8_t*>(av_malloc(buffer_size)));
if (!samples) {
AudacityMessageBox(
_("FFmpeg : ERROR - Could not allocate bytes for samples buffer"),
_("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(
_("FFmpeg : ERROR - Could not setup audio frame"),
_("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(
_("FFmpeg : ERROR - encoding frame failed"),
_("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(
_("FFmpeg : ERROR - Too much remaining data."),
_("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
AudacityMessageBox(_("Unable to export"));
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
AudacityMessageBox(_("Unable to export"));
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(
_("FFmpeg : ERROR - Couldn't write last audio frame to output file."),
_("FFmpeg Error"), wxOK | wxCENTER | wxICON_EXCLAMATION
);
return false;
}
}
// Write any file trailers.
if (av_write_trailer(mEncFormatCtx.get()) != 0) {
// TODO: more precise message
AudacityMessageBox(_("Unable to export"));
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);
}
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)
return false;
// Put the raw audio samples into the FIFO.
ret = av_fifo_generic_write(mEncAudioFifo.get(), pRawSamples, nBytesToWrite,NULL);
if(ret != nBytesToWrite)
return false;
if (nAudioFrameSizeOut > mEncAudioFifoOutBufSiz) {
AudacityMessageBox(
_("FFmpeg : ERROR - nAudioFrameSizeOut too large."),
_("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(
_("FFmpeg : ERROR - Can't encode audio frame."),
_("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)
{
AudacityMessageBox(
_("FFmpeg : ERROR - Failed to write audio frame to file."),
_("FFmpeg Error"), wxOK|wxCENTER|wxICON_EXCLAMATION
);
return false;
}
}
return true;
}
ProgressResult ExportFFmpeg::Export(AudacityProject *project,
std::unique_ptr<ProgressDialog> &pDialog,
unsigned channels, const wxString &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(
wxString::Format(
_("Attempted to export %d channels, but maximum number of channels for selected output format is %d"),
channels,
ExportFFmpegOptions::fmts[mSubFormat].maxchannels),
_("Error"));
return ProgressResult::Cancelled;
}
mName = fName;
const auto &tracks = TrackList::Get( *project );
bool ret = true;
if (mSubFormat >= FMT_LAST) {
// TODO: more precise message
AudacityMessageBox(_("Unable to export"));
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
AudacityMessageBox(_("Unable to export"));
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, wxFileName(fName).GetName(),
selectionOnly
? wxString::Format(_("Exporting selected audio as %s"),
ExportFFmpegOptions::fmts[mSubFormat].Description())
: wxString::Format(_("Exporting the audio as %s"),
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)) {
// TODO: more precise message, and fix redundancy with messages
// already given on some of the failure paths of the above call
AudacityMessageBox(_("Unable to export"));
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
AudacityMessageBox(_("Unable to export"));
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)
{
wxDialogWrapper d(nullptr, wxID_ANY, wxString(_("Invalid sample rate")));
d.SetName(d.GetTitle());
wxChoice *choice;
ShuttleGui S(&d, eIsCreating);
int selected = -1;
S.StartVerticalLay();
{
S.SetBorder(10);
S.StartStatic(_("Resample"));
{
S.StartHorizontalLay(wxALIGN_CENTER, false);
{
S.AddTitle(
(bitrate == 0
? wxString::Format(
_("The project sample rate (%d) is not supported by the current output\nfile format. "),
rate)
: wxString::Format(
_("The project sample rate (%d) and bit rate (%d kbps) combination is not\nsupported by the current output file format. "),
rate, bitrate/1024))
+ _("You may resample to one of the rates below.")
);
}
S.EndHorizontalLay();
S.StartHorizontalLay(wxALIGN_CENTER, false);
{
choice = S.AddChoice(_("Sample Rates"),
[&]{
wxArrayStringEx choices;
for (int i = 0; sampRates[i] > 0; i++)
{
int label = sampRates[i];
if (label >= lowrate && label <= highrate)
{
wxString name = wxString::Format(wxT("%d"),label);
choices.push_back(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());
}
wxWindow *ExportFFmpeg::OptionsCreate(wxWindow *parent, int format)
{
wxASSERT(parent); // to justify safenew
// subformat index may not correspond directly to fmts[] index, convert it
mSubFormat = AdjustFormatIndex(format);
if (mSubFormat == FMT_M4A)
{
return safenew ExportFFmpegAACOptions(parent, format);
}
else if (mSubFormat == FMT_AC3)
{
return safenew ExportFFmpegAC3Options(parent, format);
}
else if (mSubFormat == FMT_AMRNB)
{
return safenew ExportFFmpegAMRNBOptions(parent, format);
}
else if (mSubFormat == FMT_WMA2)
{
return safenew ExportFFmpegWMAOptions(parent, format);
}
else if (mSubFormat == FMT_OTHER)
{
return safenew ExportFFmpegCustomOptions(parent, format);
}
return ExportPlugin::OptionsCreate(parent, format);
}
static Exporter::RegisteredExportPlugin
sRegisteredPlugin{ []{ return std::make_unique< ExportFFmpeg >(); } };
#endif