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/**********************************************************************
SPDX-License-Identifier: GPL-2.0-or-later
Audacity: A Digital Audio Editor
ExportOpus.cpp
Dmitry Vedenko
**********************************************************************/
#include "Export.h"
#include <random>
#include <string_view>
#include <ogg/ogg.h>
#include <opus/opus.h>
#include <opus/opus_multistream.h>
#include "wxFileNameWrapper.h"
#include "Mix.h"
#include "MemoryX.h"
#include "Track.h"
#include "Tags.h"
#include "ExportPluginHelpers.h"
#include "ExportOptionsEditor.h"
#include "ExportPluginRegistry.h"
#include "PlainExportOptionsEditor.h"
#include "CodeConversions.h"
namespace
{
TranslatableString GetOpusEncErrorString(int error)
{
switch (error)
{
case OPUS_OK:
return XO("no error");
case OPUS_BAD_ARG:
return XO("invalid argument");
case OPUS_BUFFER_TOO_SMALL:
return XO("buffer too small");
case OPUS_INTERNAL_ERROR:
return XO("internal error");
case OPUS_INVALID_PACKET:
return XO("invalid packet");
case OPUS_UNIMPLEMENTED:
return XO("not implemented");
case OPUS_INVALID_STATE:
return XO("invalid state");
case OPUS_ALLOC_FAIL:
return XO("memory allocation has failed");
default:
return XO("Unknown error");
}
}
[[noreturn]] void FailExport(const TranslatableString& title, int errorCode = 0)
{
if (errorCode != 0)
{
throw ExportException(Verbatim("%s: %s")
.Format(title, GetOpusEncErrorString(errorCode))
.Translation());
}
throw ExportException(title.Translation());
}
/* i18n-hint: kbps abbreviates "thousands of bits per second" */
TranslatableString n_kbps(int n)
{
return XO("%d kbps").Format(n);
}
enum : int
{
OPUSOptionIDBitRate = 0,
OPUSOptionIDQuality,
OPUSOptionIDFrameDuration,
OPUSOptionIDVBRMode,
OPUSOptionIDApplication,
OPUSOptionIDCutoff
};
namespace VBRMode
{
enum : int
{
CBR,
VBR,
CVBR
};
}
const std::initializer_list<PlainExportOptionsEditor::OptionDesc> OPUSOptions {
{
{
OPUSOptionIDBitRate, XO("Bit Rate"),
OPUS_AUTO,
ExportOption::TypeEnum,
{
6000,
8000,
16000,
24000,
32000,
40000,
48000,
64000,
80000,
96000,
128000,
160000,
192000,
256000,
OPUS_AUTO,
OPUS_BITRATE_MAX,
},
{
n_kbps( 6 ),
n_kbps( 8 ),
n_kbps( 16 ),
n_kbps( 24 ),
n_kbps( 32 ),
n_kbps( 40 ),
n_kbps( 48 ),
n_kbps( 64 ),
n_kbps( 80 ),
n_kbps( 96 ),
n_kbps( 128 ),
n_kbps( 160 ),
n_kbps( 192 ),
n_kbps( 256 ),
XO("Auto"),
XO("Maximum")
}
}, wxT("/FileFormats/OPUS/Bitrate")
},
{
{
OPUSOptionIDQuality, XO("Quality"),
10,
ExportOption::TypeRange,
{ 0, 10 }
}, wxT("/FileFormats/OPUS/Quality")
},
{
{
OPUSOptionIDFrameDuration, XO("Frame Duration"),
200,
ExportOption::TypeEnum,
{
25,
50,
100,
200,
400,
600,
},
{
XO("2.5 ms"),
XO("5 ms"),
XO("10 ms"),
XO("20 ms"),
XO("40 ms"),
XO("60 ms"),
}
}, wxT("/FileFormats/OPUS/FrameDuration")
},
{
{
OPUSOptionIDVBRMode, XO("VBR Mode"),
VBRMode::VBR,
ExportOption::TypeEnum,
{ VBRMode::CBR, VBRMode::VBR, VBRMode::CVBR },
{ XO("Off"), XO("On"), XO("Constrained") }
}, wxT("/FileFormats/OPUS/VbrMode")
},
{
{
OPUSOptionIDApplication, XO("Optimize for"),
OPUS_APPLICATION_AUDIO,
ExportOption::TypeEnum,
{ OPUS_APPLICATION_VOIP, OPUS_APPLICATION_AUDIO, OPUS_APPLICATION_RESTRICTED_LOWDELAY },
{ XO("Speech"), XO("Audio"), XO("Low Delay") }
}, wxT("/FileFormats/OPUS/Application")
},
{
{
OPUSOptionIDCutoff, XO("Cutoff"),
OPUS_AUTO,
ExportOption::TypeEnum,
{
OPUS_AUTO,
OPUS_BANDWIDTH_NARROWBAND,
OPUS_BANDWIDTH_MEDIUMBAND,
OPUS_BANDWIDTH_WIDEBAND,
OPUS_BANDWIDTH_SUPERWIDEBAND,
OPUS_BANDWIDTH_FULLBAND,
},
{
XO("Auto"),
XO("Narrowband"),
XO("Mediumband"),
XO("Wideband"),
XO("Super Wideband"),
XO("Fullband")
}
}, wxT("/FileFormats/OPUS/Cutoff")
},
};
constexpr int supportedSampleRates[] = { 8000, 12000, 16000, 24000, 48000 };
bool IsValidSampleRate(int sampleRate) noexcept
{
for (auto sr : supportedSampleRates)
if (sr == sampleRate)
return true;
return false;
}
}
class OpusExportProcessor final : public ExportProcessor
{
struct OggPacket final
{
using byte_type = std::remove_pointer_t<decltype(ogg_packet::packet)>;
static_assert(sizeof(byte_type) == 1);
OggPacket(int64_t packetNo, bool resizable)
: resizable { resizable }
{
packet.packetno = packetNo;
}
explicit OggPacket(int64_t packetNo)
: OggPacket { packetNo, false }
{
}
OggPacket(int64_t packetNo, long size, bool resizable)
: OggPacket { packetNo, resizable }
{
Resize(size);
}
void Resize(long size)
{
buffer.resize(size);
packet.packet = buffer.data();
}
void Reset() noexcept
{
packet.bytes = 0;
}
void MarkBOS() noexcept
{
packet.b_o_s = 1;
}
void MarkEOS() noexcept
{
packet.e_o_s = 1;
}
void Write(const void* data, const long length)
{
const auto nextPos = packet.bytes + length;
if (nextPos > buffer.size())
{
if (resizable)
Resize(std::max<size_t>(1024, buffer.size() * 2));
else
FailExport(
XO("Buffer overflow in OGG packet"), OPUS_BUFFER_TOO_SMALL);
}
std::copy(
reinterpret_cast<const byte_type*>(data),
reinterpret_cast<const byte_type*>(data) + length,
buffer.data() + packet.bytes);
packet.bytes = nextPos;
}
template<typename IntType>
void Write(IntType value)
{
static_assert(std::is_integral_v<IntType>);
if constexpr (sizeof (IntType) == 1)
{
Write(&value, 1);
}
else
{
if (IsLittleEndian ())
{
Write(&value, sizeof (IntType));
}
else
{
IntType swapped = SwapIntBytes(value);
Write(&swapped, sizeof (IntType));
}
}
}
byte_type* GetBuffer()
{
return buffer.data();
}
size_t GetBufferSize() const
{
return buffer.size();
}
ogg_packet packet {};
private:
std::vector<byte_type> buffer;
bool resizable { false };
};
struct
{
TranslatableString status;
int32_t sampleRate {};
double t0 {};
double t1 {};
unsigned numChannels {};
wxFileNameWrapper fName;
wxFile outFile;
std::unique_ptr<Mixer> mixer;
std::unique_ptr<Tags> metadata;
// Encoder properties
struct OpusState final
{
~OpusState()
{
if (encoder != nullptr)
opus_multistream_encoder_destroy(encoder);
}
OpusMSEncoder* encoder {};
int32_t frameSize {};
int32_t sampleRateFactor {};
uint16_t preskip {};
uint8_t channelMapping {};
uint8_t nbStreams {};
uint8_t nbCoupled {};
uint8_t streamMap[255] {};
} opus;
// Bitstream properties
struct OggState final
{
OggState()
// Audio always starts in the packet #3.
// The first one is for opus header, the second one is for tags,
// both are mandatory
: audioStreamPacket(2)
{
// As per OGG docs - stream serialno should be a random
// number. It is used to stitch the streams, this doesn't
// matter much for Audacity
std::mt19937 gen(std::time(nullptr));
ogg_stream_init(&stream, gen());
}
void PacketIn(const OggPacket& packet)
{
ogg_stream_packetin(&stream,
// C libraries are not always const-correct
const_cast<ogg_packet*>(&packet.packet));
}
void WriteOut(wxFile& outputStream)
{
ogg_page page {};
while (ogg_stream_pageout(&stream, &page))
WritePage(outputStream, page);
}
void Flush(wxFile& outputStream)
{
ogg_page page {};
while (ogg_stream_flush(&stream, &page))
WritePage(outputStream, page);
}
ogg_stream_state stream;
OggPacket audioStreamPacket;
private:
void WritePage(wxFile& outputStream, const ogg_page& page)
{
if (
outputStream.Write(page.header, page.header_len) !=
page.header_len)
FailExport(XO("Unable to write OGG page header"));
if (outputStream.Write(page.body, page.body_len) != page.body_len)
FailExport(XO("Unable to write OGG page"));
}
} ogg;
std::vector<float> encodeBuffer;
} context;
void WriteOpusHeader();
void WriteTags();
int32_t GetBestFrameSize(int32_t samplesCount) const noexcept
{
static const int32_t multipliers[] = {
25, 50, 100, 200, 400, 600,
};
const auto sampleRate = context.sampleRate;
for (auto multiplier : multipliers)
{
const auto frameSize = multiplier * sampleRate / 10000;
if (samplesCount <= frameSize)
return frameSize;
}
return 60 * sampleRate / 1000;
}
public:
~OpusExportProcessor();
bool Initialize(AudacityProject& project,
const Parameters& parameters,
const wxFileNameWrapper& filename,
double t0, double t1, bool selectedOnly,
double sampleRate, unsigned channels,
MixerOptions::Downmix* mixerSpec,
const Tags* tags) override;
ExportResult Process(ExportProcessorDelegate& delegate) override;
};
class ExportOpus final : public ExportPlugin
{
public:
ExportOpus();
int GetFormatCount() const override;
FormatInfo GetFormatInfo(int) const override;
std::vector<std::string> GetMimeTypes(int) const override;
std::unique_ptr<ExportOptionsEditor>
CreateOptionsEditor(int, ExportOptionsEditor::Listener*) const override;
std::unique_ptr<ExportProcessor> CreateProcessor(int format) const override;
};
ExportOpus::ExportOpus() = default;
int ExportOpus::GetFormatCount() const
{
return 1;
}
FormatInfo ExportOpus::GetFormatInfo(int) const
{
return {
wxT("Opus"), XO("Opus Files"), { wxT("opus") }, 255, true
};
}
std::vector<std::string> ExportOpus::GetMimeTypes(int) const
{
return { "audio/opus" };
}
std::unique_ptr<ExportOptionsEditor>
ExportOpus::CreateOptionsEditor(int, ExportOptionsEditor::Listener* listener) const
{
return std::make_unique<PlainExportOptionsEditor>(
OPUSOptions,
ExportOptionsEditor::SampleRateList { 8000, 12000, 16000, 24000, 48000 },
listener);
}
std::unique_ptr<ExportProcessor> ExportOpus::CreateProcessor(int) const
{
return std::make_unique<OpusExportProcessor>();
}
void OpusExportProcessor::WriteOpusHeader()
{
const auto headerSize =
// "OpusHead"
8 +
// Version number (always 1)
1 +
// Channels count
1 +
// Preskip
2 +
// Input sample rate
4 +
// Output gain (always 0)
2 +
// Channel mapping
1 +
(context.opus.channelMapping == 0 ? 0 :
(
// Stream count
1 +
// Two channel stream count
1 +
// Channel mapping
context.numChannels));
OggPacket headerPacket(0, headerSize, false);
// Header must have beginning-of-stream marker
headerPacket.MarkBOS();
headerPacket.Write("OpusHead", 8);
headerPacket.Write<uint8_t>(1);
headerPacket.Write<uint8_t>(context.numChannels);
headerPacket.Write(context.opus.preskip);
// Should we put the project sample rate here?
headerPacket.Write(context.sampleRate);
// Opus docs recommend encoders to use 0 as a gain
headerPacket.Write<uint16_t>(0);
headerPacket.Write(context.opus.channelMapping);
if (context.opus.channelMapping > 0)
{
headerPacket.Write(context.opus.nbStreams);
headerPacket.Write(context.opus.nbCoupled);
for (int i = 0; i < context.numChannels; ++i)
headerPacket.Write<uint8_t>(context.opus.streamMap[i]);
}
// This is guaranteed by the way we calculate the header size
assert(headerPacket.packet.bytes == headerSize);
context.ogg.PacketIn(headerPacket);
context.ogg.Flush(context.outFile);
}
void OpusExportProcessor::WriteTags()
{
OggPacket commentsPacket { 1, true };
commentsPacket.Write("OpusTags", 8);
const std::string_view vendor { opus_get_version_string() };
commentsPacket.Write<uint32_t>(vendor.size());
commentsPacket.Write(vendor.data(), vendor.size());
commentsPacket.Write<uint32_t>(context.metadata->Count());
for (const auto& pair : context.metadata->GetRange())
{
const auto key = pair.first == TAG_YEAR ? std::string("DATE") :
audacity::ToUTF8(pair.first);
const auto value = audacity::ToUTF8(pair.second);
commentsPacket.Write<uint32_t>(key.size() + value.size() + 1);
commentsPacket.Write(key.data(), key.size());
commentsPacket.Write("=", 1);
commentsPacket.Write(value.data(), value.size());
}
context.ogg.PacketIn(commentsPacket);
context.ogg.Flush(context.outFile);
}
OpusExportProcessor::~OpusExportProcessor()
{
}
bool OpusExportProcessor::Initialize(
AudacityProject& project, const Parameters& parameters,
const wxFileNameWrapper& fName, double t0, double t1, bool selectionOnly,
double sampleRate, unsigned numChannels, MixerOptions::Downmix* mixerSpec,
const Tags* metadata)
{
context.sampleRate = int32_t(sampleRate);
if (!IsValidSampleRate(context.sampleRate))
throw ExportException(XO("Unsupported sample rate").Translation());
context.t0 = t0;
context.t1 = t1;
context.numChannels = numChannels;
context.fName = fName;
// Internally the Opus is always in 48k, find out the multiplier for
// values, that expect 48k sample rate
context.opus.sampleRateFactor = 48000 / context.sampleRate;
const auto bitRate = ExportPluginHelpers::GetParameterValue<int>(
parameters, OPUSOptionIDBitRate, OPUS_AUTO);
const auto vbrMode = ExportPluginHelpers::GetParameterValue<int>(
parameters, OPUSOptionIDVBRMode, VBRMode::VBR);
const int complexity = ExportPluginHelpers::GetParameterValue<int>(
parameters, OPUSOptionIDQuality, 10);
const int frameMultiplier = ExportPluginHelpers::GetParameterValue<int>(
parameters, OPUSOptionIDFrameDuration, 200);
const int application = ExportPluginHelpers::GetParameterValue<int>(
parameters, OPUSOptionIDApplication, OPUS_APPLICATION_AUDIO);
const int cutoff = ExportPluginHelpers::GetParameterValue<int>(
parameters, OPUSOptionIDCutoff, OPUS_AUTO);
// Number of samples per frame per channel
context.opus.frameSize = frameMultiplier * context.sampleRate / 10000;
context.status = selectionOnly ? XO("Exporting selected audio as Opus") :
XO("Exporting the audio as Opus");
// Create opus encoder
int error;
if (numChannels <= 2)
{
context.opus.channelMapping = 0;
context.opus.nbStreams = 1;
context.opus.nbCoupled = numChannels - 1;
context.opus.streamMap[0] = 0;
context.opus.streamMap[1] = 1;
context.opus.encoder = opus_multistream_encoder_create(
sampleRate, numChannels, context.opus.nbStreams,
context.opus.nbCoupled, context.opus.streamMap, application, &error);
}
else
{
context.opus.channelMapping = numChannels <= 8 ? 1 : 255;
int nbStreams {}, nbCoupled {};
context.opus.encoder = opus_multistream_surround_encoder_create(
sampleRate, numChannels, context.opus.channelMapping,
&nbStreams, &nbCoupled,
context.opus.streamMap, application, &error);
// opus_multistream_surround_encoder_create is expected to fill
// stream count with values in [0, 255]
context.opus.nbStreams = uint8_t(nbStreams);
context.opus.nbCoupled = uint8_t(nbCoupled);
}
if (error != OPUS_OK)
FailExport(XO("Unable to create Opus encoder"), error);
error = opus_multistream_encoder_ctl(
context.opus.encoder, OPUS_SET_BITRATE(bitRate));
if (error != OPUS_OK)
FailExport(XO("Unable to set bitrate"), error);
error = opus_multistream_encoder_ctl(
context.opus.encoder, OPUS_SET_COMPLEXITY(complexity));
if (error != OPUS_OK)
FailExport(XO("Unable to set complexity"), error);
error = opus_multistream_encoder_ctl(
context.opus.encoder, OPUS_SET_BANDWIDTH(cutoff));
if (error != OPUS_OK)
FailExport(XO("Unable to set bandwidth"), error);
error = opus_multistream_encoder_ctl(
context.opus.encoder, OPUS_SET_VBR(vbrMode == VBRMode::CBR ? 0 : 1));
if (error != OPUS_OK)
FailExport(XO("Unable to set VBR mode"), error);
if (vbrMode == VBRMode::CVBR)
{
error = opus_multistream_encoder_ctl(
context.opus.encoder, OPUS_SET_VBR_CONSTRAINT(1));
if (error != OPUS_OK)
FailExport(XO("Unable to set CVBR mode"), error);
}
// Calculate the encoder latency. This value is needed in header
// and to flush the encoder
int lookahead {};
error = opus_multistream_encoder_ctl(
context.opus.encoder, OPUS_GET_LOOKAHEAD(&lookahead));
if (error != OPUS_OK)
FailExport(XO("Unable to get lookahead"), error);
// Latency is always in 48k encoded samples
const auto calculatedPreskip = lookahead * context.opus.sampleRateFactor;
if (
calculatedPreskip < 0 ||
calculatedPreskip >= std::numeric_limits<uint16_t>::max())
FailExport(XO("Failed to calculate correct preskip"), OPUS_BAD_ARG);
// It is safe to cast to uint16_t here
context.opus.preskip = uint16_t(calculatedPreskip);
// Resize the audio packet so it can contain all the raw data.
// This is overkill, but should be enough to hold all the data from
// the encode float
context.ogg.audioStreamPacket.Resize(
context.opus.frameSize * sizeof(float) * numChannels);
// Try to open the file for writing
if (
!context.outFile.Create(fName.GetFullPath(), true) ||
!context.outFile.IsOpened())
{
throw ExportException(_("Unable to open target file for writing"));
}
WriteOpusHeader();
context.metadata = std::make_unique<Tags>(
metadata == nullptr ? Tags::Get(project) : *metadata);
WriteTags();
context.mixer = ExportPluginHelpers::CreateMixer(
project, selectionOnly, t0, t1, numChannels, context.opus.frameSize, true,
sampleRate, floatSample, mixerSpec);
return true;
}
ExportResult OpusExportProcessor::Process(ExportProcessorDelegate& delegate)
{
delegate.SetStatusString(context.status);
auto exportResult = ExportResult::Success;
int64_t granulePos = 0;
int32_t latencyLeft = context.opus.preskip;
while (exportResult == ExportResult::Success)
{
auto samplesThisRun = context.mixer->Process();
if (samplesThisRun == 0)
break;
auto mixedAudioBuffer =
reinterpret_cast<const float*>(context.mixer->GetBuffer());
// bestFrameSize <= context.opus.frameSize by design
auto bestFrameSize = GetBestFrameSize(samplesThisRun);
if (samplesThisRun < bestFrameSize)
{
// Opus expects that the full frame is passed to the encoder, fill missing data with zeroes
context.encodeBuffer.resize(bestFrameSize * context.numChannels);
std::copy(
mixedAudioBuffer, mixedAudioBuffer + samplesThisRun * context.numChannels,
context.encodeBuffer.begin());
std::fill(
context.encodeBuffer.begin() + samplesThisRun * context.numChannels,
context.encodeBuffer.begin() + bestFrameSize * context.numChannels,
0);
mixedAudioBuffer = context.encodeBuffer.data();
auto zeroesCount = bestFrameSize - int32_t(samplesThisRun);
if (zeroesCount < latencyLeft)
samplesThisRun += zeroesCount;
else
samplesThisRun += latencyLeft;
// Reduce the latency by the number of zeroes pushed (potentially
// removing the need to flush the encoder)
latencyLeft = std::max(0, latencyLeft - zeroesCount);
}
auto result = opus_multistream_encode_float(
context.opus.encoder, mixedAudioBuffer, bestFrameSize,
context.ogg.audioStreamPacket.GetBuffer(),
context.ogg.audioStreamPacket.GetBufferSize());
if (result < 0)
FailExport(XO("Failed to encode input buffer"), result);
// granulePos is the index of the last real sample in the packet at 48k rate
granulePos += samplesThisRun * context.opus.sampleRateFactor;
context.ogg.audioStreamPacket.packet.bytes = result;
context.ogg.audioStreamPacket.packet.granulepos = granulePos;
if (latencyLeft == 0)
context.ogg.audioStreamPacket.MarkEOS();
context.ogg.PacketIn(context.ogg.audioStreamPacket);
context.ogg.WriteOut(context.outFile);
context.ogg.audioStreamPacket.packet.packetno++;
exportResult = ExportPluginHelpers::UpdateProgress(
delegate, *context.mixer, context.t0, context.t1);
}
// Flush the encoder
while (latencyLeft > 0)
{
auto frameSize = GetBestFrameSize(latencyLeft);
context.encodeBuffer.resize(frameSize * context.numChannels);
std::fill(
context.encodeBuffer.begin(),
context.encodeBuffer.begin() + frameSize * context.numChannels, 0);
auto samplesOut = std::min(latencyLeft, frameSize);
auto result = opus_multistream_encode_float(
context.opus.encoder, context.encodeBuffer.data(),
frameSize, context.ogg.audioStreamPacket.GetBuffer(),
context.ogg.audioStreamPacket.GetBufferSize());
if (result < 0)
FailExport(XO("Failed to encode input buffer"), result);
granulePos += samplesOut * context.opus.sampleRateFactor;
context.ogg.audioStreamPacket.packet.bytes = result;
context.ogg.audioStreamPacket.packet.granulepos = granulePos;
if (latencyLeft == samplesOut)
context.ogg.audioStreamPacket.MarkEOS();
context.ogg.PacketIn(context.ogg.audioStreamPacket);
context.ogg.WriteOut(context.outFile);
context.ogg.audioStreamPacket.packet.packetno++;
latencyLeft -= samplesOut;
}
context.ogg.Flush(context.outFile);
if (!context.outFile.Close())
return ExportResult::Error;
return exportResult;
}
static ExportPluginRegistry::RegisteredPlugin sRegisteredPlugin{ "Opus",
[]{ return std::make_unique< ExportOpus >(); }
};
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