int opus_multistream_encode_float(
OpusMSEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
)
{
int i, ret;
VARDECL(opus_int16, in);
ALLOC_STACK;
ALLOC(in, frame_size*st->layout.nb_channels, opus_int16);
for (i=0;i<frame_size*st->layout.nb_channels;i++)
in[i] = FLOAT2INT16(pcm[i]);
ret = opus_multistream_encode(st, in, frame_size, data, max_data_bytes);
RESTORE_STACK;
return ret;
}
static GstFlowReturn
gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
{
guint8 *bdata, *data, *mdata = NULL;
gsize bsize, size;
gsize bytes;
gint ret = GST_FLOW_OK;
gint outsize;
GstBuffer *outbuf;
g_mutex_lock (enc->property_lock);
bytes = enc->frame_samples * enc->n_channels * 2;
if (G_LIKELY (buf)) {
bdata = GST_BUFFER_DATA (buf);
bsize = GST_BUFFER_SIZE (buf);
if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((bsize / bytes) + 1) * bytes;
mdata = g_malloc0 (size);
memcpy (mdata, bdata, bsize);
bdata = NULL;
data = mdata;
} else {
data = bdata;
size = bsize;
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
g_assert (size == bytes);
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
GST_BUFFER_OFFSET_NONE, enc->max_payload_size * enc->n_channels,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
if (GST_FLOW_OK != ret)
goto done;
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
enc->frame_samples, (int) bytes);
outsize =
opus_multistream_encode (enc->state, (const gint16 *) data,
enc->frame_samples, GST_BUFFER_DATA (outbuf),
enc->max_payload_size * enc->n_channels);
if (outsize < 0) {
GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
ret = GST_FLOW_ERROR;
goto done;
} else if (outsize > enc->max_payload_size) {
GST_WARNING_OBJECT (enc,
"Encoded size %d is higher than max payload size (%d bytes)",
outsize, enc->max_payload_size);
ret = GST_FLOW_ERROR;
goto done;
}
GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
GST_BUFFER_SIZE (outbuf) = outsize;
ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
enc->frame_samples);
done:
g_mutex_unlock (enc->property_lock);
if (mdata)
g_free (mdata);
return ret;
}
static result_t
encode_opus_file(char *filename,
struct PCMReader *pcmreader,
int quality,
unsigned original_sample_rate)
{
const int multichannel = (pcmreader->channels > 2);
const unsigned channel_mapping = (pcmreader->channels > 8 ? 255 :
pcmreader->channels > 2);
int stream_count;
int coupled_stream_count;
unsigned char stream_map[255];
int samples[BLOCK_SIZE * pcmreader->channels];
unsigned pcm_frames;
result_t result = ENCODE_OK;
FILE *output_file = NULL;
ogg_stream_state ogg_stream;
ogg_page ogg_page;
OpusEncoder *opus_encoder = NULL;
OpusMSEncoder *opus_ms_encoder = NULL;
int error;
opus_int16 *opus_samples = NULL;
unsigned char opus_frame[OPUS_FRAME_LEN];
ogg_int64_t granulepos = 0;
ogg_int64_t packetno = 0;
opus_int32 preskip;
/*open output file for writing*/
if ((output_file = fopen(filename, "w+b")) == NULL) {
return ERR_IOERROR;
}
if (!multichannel) {
if ((opus_encoder = opus_encoder_create(48000,
pcmreader->channels,
OPUS_APPLICATION_AUDIO,
&error)) == NULL) {
fclose(output_file);
return ERR_ENCODER_INIT;
}
opus_encoder_ctl(opus_encoder, OPUS_SET_COMPLEXITY(quality));
opus_encoder_ctl(opus_encoder, OPUS_GET_LOOKAHEAD(&preskip));
} else {
if ((opus_ms_encoder =
opus_multistream_surround_encoder_create(
48000,
pcmreader->channels,
channel_mapping,
&stream_count,
&coupled_stream_count,
stream_map,
OPUS_APPLICATION_AUDIO,
&error)) == NULL) {
fclose(output_file);
return ERR_ENCODER_INIT;
}
opus_multistream_encoder_ctl(opus_ms_encoder,
OPUS_SET_COMPLEXITY(quality));
opus_multistream_encoder_ctl(opus_ms_encoder,
OPUS_GET_LOOKAHEAD(&preskip));
}
srand((unsigned)time(NULL));
ogg_stream_init(&ogg_stream, rand());
/*write header and comment packets to Ogg stream*/
{
BitstreamRecorder *header =
bw_open_bytes_recorder(BS_LITTLE_ENDIAN);
BitstreamWriter *header_w =(BitstreamWriter*)header;
BitstreamRecorder *comment =
bw_open_bytes_recorder(BS_LITTLE_ENDIAN);
BitstreamWriter *comment_w = (BitstreamWriter*)comment;
int i;
/*write header packet to Ogg stream*/
const char opushead[] = "OpusHead";
const char opuscomment[] = "OpusTags";
const char *vendor_string = opus_get_version_string();
const size_t vendor_string_len = strlen(vendor_string);
ogg_packet packet_head;
ogg_packet packet_tags;
header_w->write_bytes(header_w,
(uint8_t*)opushead,
(unsigned)strlen(opushead));
header_w->write(header_w, 8, 1); /*version*/
header_w->write(header_w, 8, pcmreader->channels);
header_w->write(header_w, 16, preskip);
header_w->write(header_w, 32, original_sample_rate);
header_w->write(header_w, 16, 0); /*output gain*/
header_w->write(header_w, 8, channel_mapping);
if (channel_mapping != 0) {
header_w->write(header_w, 8, stream_count);
header_w->write(header_w, 8, coupled_stream_count);
for (i = 0; i < pcmreader->channels; i++) {
header_w->write(header_w, 8, stream_map[i]);
}
}
packet_head.packet = malloc(header->bytes_written(header));
header->data(header, (uint8_t*)packet_head.packet);
packet_head.bytes = header->bytes_written(header);
packet_head.b_o_s = 1;
packet_head.e_o_s = 0;
packet_head.granulepos = 0;
packet_head.packetno = packetno++;
header->close(header);
ogg_stream_packetin(&ogg_stream, &packet_head);
for (i = ogg_stream_flush(&ogg_stream, &ogg_page);
i != 0;
i = ogg_stream_flush(&ogg_stream, &ogg_page)) {
fwrite(ogg_page.header, 1, ogg_page.header_len, output_file);
fwrite(ogg_page.body, 1, ogg_page.body_len, output_file);
}
free(packet_head.packet);
/*write comment packet to Ogg stream*/
comment_w->write_bytes(comment_w,
(uint8_t*)opuscomment,
(unsigned)strlen(opuscomment));
comment_w->write(comment_w, 32, (unsigned)vendor_string_len);
comment_w->write_bytes(comment_w,
(uint8_t*)vendor_string,
(unsigned)vendor_string_len);
comment_w->write(comment_w, 32, 0);
packet_tags.packet = malloc(comment->bytes_written(comment));
comment->data(comment, (uint8_t*)packet_tags.packet);
packet_tags.bytes = comment->bytes_written(comment);
packet_tags.b_o_s = 0;
packet_tags.e_o_s = 0;
packet_tags.granulepos = 0;
packet_tags.packetno = packetno++;
comment->close(comment);
ogg_stream_packetin(&ogg_stream, &packet_tags);
for (i = ogg_stream_flush(&ogg_stream, &ogg_page);
i != 0;
i = ogg_stream_flush(&ogg_stream, &ogg_page)) {
fwrite(ogg_page.header, 1, ogg_page.header_len, output_file);
fwrite(ogg_page.body, 1, ogg_page.body_len, output_file);
}
free(packet_tags.packet);
}
opus_samples = malloc(sizeof(opus_int16) *
pcmreader->channels *
BLOCK_SIZE);
pcm_frames = pcmreader->read(pcmreader, BLOCK_SIZE, samples);
if (!pcm_frames && (pcmreader->status != PCM_OK)) {
result = ERR_PCMREADER;
goto cleanup;
}
/*for each non-empty FrameList from PCMReader, encode Opus frame*/
while (pcm_frames) {
const int short_framelist = (pcm_frames < BLOCK_SIZE);
unsigned i;
opus_int32 encoded_size;
ogg_packet packet;
granulepos += pcm_frames;
/*pad FrameList with additional null samples if necessary*/
memset(samples + pcm_frames * pcmreader->channels,
0,
sizeof(int) * (BLOCK_SIZE - pcm_frames) * pcmreader->channels);
/*rearrange channels to Vorbis order if necessary*/
reorder_channels(pcmreader->channel_mask, BLOCK_SIZE, samples);
/*place samples in interleaved buffer*/
for (i = 0; i < (pcm_frames * pcmreader->channels); i++) {
opus_samples[i] = (opus_int16)samples[i];
}
/*call opus_encode on interleaved buffer to get next packet*/
if (!multichannel) {
encoded_size = opus_encode(opus_encoder,
opus_samples,
BLOCK_SIZE,
opus_frame,
OPUS_FRAME_LEN);
} else {
encoded_size = opus_multistream_encode(opus_ms_encoder,
opus_samples,
BLOCK_SIZE,
opus_frame,
OPUS_FRAME_LEN);
}
/*get next FrameList to encode*/
pcm_frames = pcmreader->read(pcmreader, BLOCK_SIZE, samples);
if (!pcm_frames && (pcmreader->status != PCM_OK)) {
result = ERR_PCMREADER;
goto cleanup;
}
/*dump Opus packet to Ogg stream*/
/*do this *after* reading the next FrameList in order to detect
the end of stream properly based on whether the FrameList
has no frames*/
packet.packet = (unsigned char *)opus_frame;
packet.bytes = encoded_size;
packet.b_o_s = 0;
packet.e_o_s = (short_framelist || (pcm_frames == 0));
packet.granulepos = granulepos;
packet.packetno = packetno;
ogg_stream_packetin(&ogg_stream, &packet);
while (ogg_stream_pageout(&ogg_stream, &ogg_page)) {
fwrite(ogg_page.header, 1, ogg_page.header_len, output_file);
fwrite(ogg_page.body, 1, ogg_page.body_len, output_file);
}
}
/*flush any remaining Ogg pages to disk*/
while (ogg_stream_flush(&ogg_stream, &ogg_page)) {
fwrite(ogg_page.header, 1, ogg_page.header_len, output_file);
fwrite(ogg_page.body, 1, ogg_page.body_len, output_file);
}
cleanup:
fclose(output_file);
ogg_stream_clear(&ogg_stream);
if (!multichannel) {
opus_encoder_destroy(opus_encoder);
} else {
opus_multistream_encoder_destroy(opus_ms_encoder);
}
free(opus_samples);
return result;
}
static GstFlowReturn
gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
{
guint8 *bdata = NULL, *data, *mdata = NULL;
gsize bsize, size;
gsize bytes = enc->frame_samples * enc->n_channels * 2;
gint ret = GST_FLOW_OK;
GstMapInfo map;
GstMapInfo omap;
gint outsize;
GstBuffer *outbuf;
g_mutex_lock (enc->property_lock);
if (G_LIKELY (buf)) {
gst_buffer_map (buf, &map, GST_MAP_READ);
bdata = map.data;
bsize = map.size;
if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((bsize / bytes) + 1) * bytes;
mdata = g_malloc0 (size);
memcpy (mdata, bdata, bsize);
data = mdata;
} else {
data = bdata;
size = bsize;
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
g_assert (size == bytes);
outbuf = gst_buffer_new_and_alloc (enc->max_payload_size * enc->n_channels);
if (!outbuf)
goto done;
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
enc->frame_samples, (int) bytes);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
enc->frame_samples, (int) bytes);
outsize =
opus_multistream_encode (enc->state, (const gint16 *) data,
enc->frame_samples, omap.data, enc->max_payload_size * enc->n_channels);
gst_buffer_unmap (outbuf, &omap);
if (outsize < 0) {
GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
ret = GST_FLOW_ERROR;
goto done;
} else if (outsize > enc->max_payload_size) {
GST_WARNING_OBJECT (enc,
"Encoded size %d is higher than max payload size (%d bytes)",
outsize, enc->max_payload_size);
ret = GST_FLOW_ERROR;
goto done;
}
GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
gst_buffer_set_size (outbuf, outsize);
ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
enc->frame_samples);
done:
if (bdata)
gst_buffer_unmap (buf, &map);
g_mutex_unlock (enc->property_lock);
if (mdata)
g_free (mdata);
return ret;
}
virtual bool CookSurround(FName Format, const TArray<TArray<uint8> >& SrcBuffers, FSoundQualityInfo& QualityInfo, TArray<uint8>& CompressedDataStore) const override
{
check(Format == NAME_OPUS);
// Get best compatible sample rate
const uint16 kOpusSampleRate = GetBestOutputSampleRate(QualityInfo.SampleRate);
// Frame size must be one of 2.5, 5, 10, 20, 40 or 60 ms
const int32 kOpusFrameSizeMs = 60;
// Calculate frame size required by Opus
const int32 kOpusFrameSizeSamples = (kOpusSampleRate * kOpusFrameSizeMs) / 1000;
const uint32 kSampleStride = SAMPLE_SIZE * QualityInfo.NumChannels;
const int32 kBytesPerFrame = kOpusFrameSizeSamples * kSampleStride;
// Check whether source has compatible sample rate
TArray<TArray<uint8>> SrcBufferCopies;
if (QualityInfo.SampleRate != kOpusSampleRate)
{
for (int32 Index = 0; Index < SrcBuffers.Num(); Index++)
{
TArray<uint8>& NewCopy = *new (SrcBufferCopies) TArray<uint8>;
if (!ResamplePCM(1, SrcBuffers[Index], QualityInfo.SampleRate, NewCopy, kOpusSampleRate))
{
return false;
}
}
}
else
{
// Take a copy of the source regardless
for (int32 Index = 0; Index < SrcBuffers.Num(); Index++)
{
SrcBufferCopies[Index] = SrcBuffers[Index];
}
}
// Ensure that all channels are the same length
int32 SourceSize = -1;
for (int32 Index = 0; Index < SrcBufferCopies.Num(); Index++)
{
if (!Index)
{
SourceSize = SrcBufferCopies[Index].Num();
}
else
{
if (SourceSize != SrcBufferCopies[Index].Num())
{
return false;
}
}
}
if (SourceSize <= 0)
{
return false;
}
// Initialise the Opus multistream encoder
OpusMSEncoder* Encoder = NULL;
int32 EncError = 0;
int32 streams = 0;
int32 coupled_streams = 0;
// mapping_family not documented but figured out: 0 = 1 or 2 channels, 1 = 1 to 8 channel surround sound, 255 = up to 255 channels with no surround processing
int32 mapping_family = 1;
TArray<uint8> mapping;
mapping.AddUninitialized(QualityInfo.NumChannels);
#if USE_UE4_MEM_ALLOC
int32 EncSize = opus_multistream_surround_encoder_get_size(QualityInfo.NumChannels, mapping_family);
Encoder = (OpusMSEncoder*)FMemory::Malloc(EncSize);
EncError = opus_multistream_surround_encoder_init(Encoder, kOpusSampleRate, QualityInfo.NumChannels, mapping_family, &streams, &coupled_streams, mapping.GetData(), OPUS_APPLICATION_AUDIO);
#else
Encoder = opus_multistream_surround_encoder_create(kOpusSampleRate, QualityInfo.NumChannels, mapping_family, &streams, &coupled_streams, mapping.GetData(), OPUS_APPLICATION_AUDIO, &EncError);
#endif
if (EncError != OPUS_OK)
{
Destroy(Encoder);
return false;
}
int32 BitRate = GetBitRateFromQuality(QualityInfo);
opus_multistream_encoder_ctl(Encoder, OPUS_SET_BITRATE(BitRate));
// Create a buffer to store compressed data
CompressedDataStore.Empty();
FMemoryWriter CompressedData(CompressedDataStore);
int32 SrcBufferOffset = 0;
// Calc frame and sample count
int32 FramesToEncode = SourceSize / (kOpusFrameSizeSamples * SAMPLE_SIZE);
uint32 TrueSampleCount = SourceSize / SAMPLE_SIZE;
// Add another frame if Source does not divide into an equal number of frames
if (SourceSize % (kOpusFrameSizeSamples * SAMPLE_SIZE) != 0)
{
FramesToEncode++;
}
check(QualityInfo.NumChannels <= MAX_uint8);
check(FramesToEncode <= MAX_uint16);
SerializeHeaderData(CompressedData, kOpusSampleRate, TrueSampleCount, QualityInfo.NumChannels, FramesToEncode);
// Temporary storage for source data in an interleaved format
TArray<uint8> TempInterleavedSrc;
TempInterleavedSrc.AddUninitialized(kBytesPerFrame);
// Temporary storage with more than enough to store any compressed frame
TArray<uint8> TempCompressedData;
TempCompressedData.AddUninitialized(kBytesPerFrame);
while (SrcBufferOffset < SourceSize)
{
// Read a frames worth of data from the source and pack it into interleaved temporary storage
for (int32 SampleIndex = 0; SampleIndex < kOpusFrameSizeSamples; ++SampleIndex)
{
int32 CurrSrcOffset = SrcBufferOffset + SampleIndex*SAMPLE_SIZE;
int32 CurrInterleavedOffset = SampleIndex*kSampleStride;
if (CurrSrcOffset < SourceSize)
{
for (uint32 ChannelIndex = 0; ChannelIndex < QualityInfo.NumChannels; ++ChannelIndex)
{
// Interleave the channels in the Vorbis format, so that the correct channel is used for LFE
int32 OrderedChannelIndex = VorbisChannelInfo::Order[QualityInfo.NumChannels - 1][ChannelIndex];
int32 CurrInterleavedIndex = CurrInterleavedOffset + ChannelIndex*SAMPLE_SIZE;
// Copy both bytes that make up a single sample
TempInterleavedSrc[CurrInterleavedIndex] = SrcBufferCopies[OrderedChannelIndex][CurrSrcOffset];
TempInterleavedSrc[CurrInterleavedIndex + 1] = SrcBufferCopies[OrderedChannelIndex][CurrSrcOffset + 1];
}
}
else
{
// Zero the rest of the temp buffer to make it an exact frame
FMemory::Memzero(TempInterleavedSrc.GetData() + CurrInterleavedOffset, kBytesPerFrame - CurrInterleavedOffset);
SampleIndex = kOpusFrameSizeSamples;
}
}
int32 CompressedLength = opus_multistream_encode(Encoder, (const opus_int16*)(TempInterleavedSrc.GetData()), kOpusFrameSizeSamples, TempCompressedData.GetData(), TempCompressedData.Num());
if (CompressedLength < 0)
{
const char* ErrorStr = opus_strerror(CompressedLength);
UE_LOG(LogAudio, Warning, TEXT("Failed to encode: [%d] %s"), CompressedLength, ANSI_TO_TCHAR(ErrorStr));
Destroy(Encoder);
CompressedDataStore.Empty();
return false;
}
else
{
// Store frame length and copy compressed data before incrementing pointers
check(CompressedLength < MAX_uint16);
SerialiseFrameData(CompressedData, TempCompressedData.GetData(), CompressedLength);
SrcBufferOffset += kOpusFrameSizeSamples * SAMPLE_SIZE;
}
}
Destroy(Encoder);
return CompressedDataStore.Num() > 0;
}
void OggOpusFile::EncodeChunks(void* pcmBuf, int numSamples, bool lastChunk)
{
int numFrames = numSamples/PCM_SAMPLES_IN_FRAME;
bool flush = false;
for(int j = 0; j < numFrames; j++)
{
if(j == (numFrames -1) && (lastChunk == true))
{
flush =true;
}
nb_samples=PCM_SAMPLES_IN_FRAME;//make it 160
id++;
int size_segments,cur_frame_size;
////frame_size=160 for 8k
cur_frame_size=frame_size;
/*Encode current frame*/
//Stereo: each pcm samples will have 2 short samples(L/R interlaced) so the num of bytes for each frame will be double to mono
nbBytes=opus_multistream_encode (st, (short*)pcmBuf + (j*PCM_SAMPLES_IN_FRAME*chan), cur_frame_size, m_outBuf, max_frame_bytes);
if(nbBytes<0){
fprintf(stderr, "Encoding failed: %s. Aborting.\n", opus_strerror(nbBytes));
return;
}
nb_encoded+=cur_frame_size;
enc_granulepos+=cur_frame_size*48000/coding_rate;
total_bytes+=nbBytes;
size_segments=(nbBytes+255)/255;
peak_bytes=IMAX(nbBytes,peak_bytes);
min_bytes=IMIN(nbBytes,min_bytes);
//printf("TotalByes:%d\n", total_bytes);
/*Flush early if adding this packet would make us end up with a
continued page which we wouldn't have otherwise.*/
while((((size_segments<=255)&&(last_segments+size_segments>255))|| (enc_granulepos-last_granulepos>max_ogg_delay))
&& ogg_stream_flush_fill(&os, &og,255*255)){
if(ogg_page_packets(&og)!=0) last_granulepos=ogg_page_granulepos(&og);
last_segments-=og.header[26];
ret=oe_write_page(&og);
if(ret!=og.header_len+og.body_len){
fprintf(stderr,"Error: failed writing data to output stream\n");
exit(1);
}
bytes_written+=ret;
pages_out++;
}
/*The downside of early reading is if the input is an exact
multiple of the frame_size you'll get an extra frame that needs
to get cropped off. The downside of late reading is added delay.
If your ogg_delay is 120ms or less we'll assume you want the
low delay behavior.*/
// if(max_ogg_delay>5760){
// nb_samples = inopt.read_samples(inopt.readdata,input,frame_size);
// total_samples+=nb_samples;
// if(nb_samples==0)op.e_o_s=1;
// } else nb_samples=-1;
op.packet=(unsigned char *)m_outBuf;
op.bytes=nbBytes;
op.b_o_s=0;
op.granulepos=enc_granulepos;
if(flush == true){
/*We compute the final GP as ceil(len*48k/input_rate)+preskip. When a
resampling decoder does the matching floor((len-preskip)*input_rate/48k)
conversion, the resulting output length will exactly equal the original
input length when 0<input_rate<=48000.*/
op.granulepos=((original_samples*48000+rate-1)/rate)+header.preskip;
}
op.packetno=2+id;
ogg_stream_packetin(&os, &op);
last_segments+=size_segments;
/*If the stream is over or we're sure that the delayed flush will fire,
go ahead and flush now to avoid adding delay.*/
while(((flush == true) || (enc_granulepos+(frame_size*48000/coding_rate)-last_granulepos>max_ogg_delay)||
(last_segments>=255))?
ogg_stream_flush_fill(&os, &og,255*255):
ogg_stream_pageout_fill(&os, &og,255*255)){
if(ogg_page_packets(&og)!=0)last_granulepos=ogg_page_granulepos(&og);
last_segments-=og.header[26];
ret=oe_write_page(&og);
if(ret!=og.header_len+og.body_len){
fprintf(stderr,"Error: failed writing data to output stream\n");
exit(1);
}
bytes_written+=ret;
pages_out++;
}
}
}
