Error AudioStreamPlaybackOpus::set_file(const String &p_file) {
file=p_file;
stream_valid=false;
Error err;
f=FileAccess::open(file,FileAccess::READ,&err);
if (err) {
ERR_FAIL_COND_V( err, err );
}
int _err;
opus_file = op_open_callbacks(f,&_op_callbacks,NULL,0,&_err);
switch (_err) {
case OP_EREAD: { // - Can't read the file.
memdelete(f); f=NULL;
ERR_FAIL_V( ERR_FILE_CANT_READ );
} break;
case OP_EVERSION: // - Unrecognized version number.
case OP_ENOTFORMAT: // - Stream is not Opus data.
case OP_EIMPL : { // - Stream used non-implemented feature.
memdelete(f); f=NULL;
ERR_FAIL_V( ERR_FILE_UNRECOGNIZED );
} break;
case OP_EBADLINK: // - Failed to find old data after seeking.
case OP_EBADTIMESTAMP: // - Timestamp failed the validity checks.
case OP_EBADHEADER: { // - Invalid or mising Opus bitstream header.
memdelete(f); f=NULL;
ERR_FAIL_V( ERR_FILE_CORRUPT );
} break;
case OP_EFAULT: { // - Internal logic fault; indicates a bug or heap/stack corruption.
memdelete(f); f=NULL;
ERR_FAIL_V( ERR_BUG );
} break;
}
const OpusHead *oinfo = op_head(opus_file,-1);
stream_channels=oinfo->channel_count;
pre_skip=oinfo->pre_skip;
frames_mixed=pre_skip;
ogg_int64_t len = op_pcm_total(opus_file,-1);
if(len < 0) {
length = 0;
} else {
length=(len/osrate);
}
op_free(opus_file);
memdelete(f);
f=NULL;
stream_valid=true;
return OK;
}
/*
* et_opus_read_file_info:
* @file: file to read info from
* @ETFileInfo: ET_File_Info to put information into
* @error: a GError or %NULL
*
* Read header information of an Opus file.
*
* Returns: %TRUE if successful otherwise %FALSE
*/
gboolean
et_opus_read_file_info (GFile *gfile, ET_File_Info *ETFileInfo,
GError **error)
{
OggOpusFile *file;
const OpusHead* head;
GFileInfo *info;
g_return_val_if_fail (gfile != NULL && ETFileInfo != NULL, FALSE);
g_return_val_if_fail (error == NULL || *error == NULL, FALSE);
file = et_opus_open_file (gfile, error);
if (!file)
{
g_assert (error == NULL || *error != NULL);
return FALSE;
}
/* FIXME: Improve error-checking. */
head = op_head (file, -1);
/* TODO: Read the vendor string from the Vorbis comment? */
ETFileInfo->version = head->version;
ETFileInfo->bitrate = op_bitrate (file, -1) / 1000;
ETFileInfo->mode = head->channel_count;
/* All Opus audio is encoded at 48 kHz, but the input sample rate can
* differ, and then input_sample_rate will be set. */
if (head->input_sample_rate != 0)
{
ETFileInfo->samplerate = head->input_sample_rate;
}
else
{
ETFileInfo->samplerate = 48000;
}
ETFileInfo->duration = op_pcm_total (file, -1) / 48000;
op_free (file);
info = g_file_query_info (gfile, G_FILE_ATTRIBUTE_STANDARD_SIZE,
G_FILE_QUERY_INFO_NONE, NULL, NULL);
if (info)
{
ETFileInfo->size = g_file_info_get_size (info);
g_object_unref (info);
}
else
{
ETFileInfo->size = 0;
}
g_assert (error == NULL || *error == NULL);
return TRUE;
}
static int
opusdec_read_metadata (DB_playItem_t *it) {
int res = -1;
DB_FILE *fp = NULL;
OggOpusFile *opusfile = NULL;
const OpusHead *head = NULL;
deadbeef->pl_lock ();
const char *uri = strdupa (deadbeef->pl_find_meta (it, ":URI"));
deadbeef->pl_unlock ();
fp = deadbeef->fopen (uri);
if (!fp) {
goto error;
}
if (fp->vfs->is_streaming ()) {
goto error;
}
opusfile = opus_file_open (fp);
if (!opusfile) {
goto error;
}
int tracknum = deadbeef->pl_find_meta_int (it, ":TRACKNUM", -1);
head = op_head (opusfile, tracknum);
if (!head) {
goto error;
}
if (update_vorbis_comments (it, opusfile, tracknum)) {
goto error;
}
res = 0;
error:
if (opusfile) {
op_free (opusfile);
}
if (fp) {
deadbeef->fclose (fp);
}
return res;
}
static bool
new_streaming_link(opusdec_info_t *info, const int new_link)
{
if (!info->info.file->vfs->is_streaming () || new_link < 0) {
return false;
}
update_vorbis_comments(info->it, info->opusfile, new_link);
send_event(info->it, DB_EV_SONGSTARTED);
send_event(info->it, DB_EV_TRACKINFOCHANGED);
deadbeef->sendmessage(DB_EV_PLAYLISTCHANGED, 0, DDB_PLAYLIST_CHANGE_CONTENT, 0);
info->cur_bit_stream = new_link;
const OpusHead *head = op_head (info->opusfile, new_link);
if (head && info->info.fmt.channels != head->channel_count) {
info->info.fmt.channels = head->channel_count;
deadbeef->pl_set_meta_int (info->it, ":CHANNELS", head->channel_count);
return true;
}
return false;
}
static void
xmms_opus_read_metadata (xmms_xform_t *xform, xmms_opus_data_t *data)
{
data->opushead = op_head (data->opusfile, -1);
data->opustags = op_tags (data->opusfile, -1);
data->channels = op_channel_count (data->opusfile, -1);
gint i;
if (!data->opustags)
return;
for (i = 0; i < data->opustags->comments; i++) {
const gchar *ptr, *entry;
gsize length;
gchar key[64];
entry = data->opustags->user_comments[i];
length = data->opustags->comment_lengths[i];
if (entry == NULL || *entry == '\0')
continue;
/* check whether it's a valid comment */
ptr = memchr (entry, '=', length);
if (ptr == NULL)
continue;
ptr++;
g_strlcpy (key, entry, MIN (ptr - entry, sizeof (key)));
if (!xmms_xform_metadata_mapper_match (xform, key, ptr, length - (ptr - entry))) {
XMMS_DBG ("Unhandled tag '%s'", entry);
}
}
}
/*
=================
S_OggOpus_CodecOpenStream
=================
*/
snd_stream_t *S_OggOpus_CodecOpenStream( const char *filename )
{
snd_stream_t *stream;
// Opus codec control structure
OggOpusFile *of;
// some variables used to get informations about the file
const OpusHead *opusInfo;
ogg_int64_t numSamples;
// check if input is valid
if ( !filename )
{
return NULL;
}
// Open the stream
stream = S_CodecUtilOpen( filename, &opus_codec );
if ( !stream )
{
return NULL;
}
// open the codec with our callbacks and stream as the generic pointer
of = op_open_callbacks( stream, &S_OggOpus_Callbacks, NULL, 0, NULL );
if ( !of )
{
S_CodecUtilClose( &stream );
return NULL;
}
// the stream must be seekable
if ( !op_seekable( of ) )
{
op_free( of );
S_CodecUtilClose( &stream );
return NULL;
}
// get the info about channels and rate
opusInfo = op_head( of, -1 );
if ( !opusInfo )
{
op_free( of );
S_CodecUtilClose( &stream );
return NULL;
}
if ( opusInfo->stream_count != 1 )
{
op_free( of );
S_CodecUtilClose( &stream );
Com_Printf( "Only Ogg Opus files with one stream are support\n" );
return NULL;
}
if ( opusInfo->channel_count != 1 && opusInfo->channel_count != 2 )
{
op_free( of );
S_CodecUtilClose( &stream );
Com_Printf( "Only mono and stereo Ogg Opus files are supported\n" );
return NULL;
}
// get the number of sample-frames in the file
numSamples = op_pcm_total( of, -1 );
// fill in the info-structure in the stream
stream->info.rate = 48000;
stream->info.width = OPUS_SAMPLEWIDTH;
stream->info.channels = opusInfo->channel_count;
stream->info.samples = numSamples;
stream->info.size = stream->info.samples * stream->info.channels * stream->info.width;
stream->info.dataofs = 0;
// We use stream->pos for the file pointer in the compressed ogg file
stream->pos = 0;
// We use the generic pointer in stream for the opus codec control structure
stream->ptr = of;
return stream;
}
AudioData LoadOpusCodec(std::string filename)
{
std::string audioFile;
try
{
audioFile = FS::PakPath::ReadFile(filename);
}
catch (std::system_error& err)
{
audioLogs.Warn("Failed to open %s: %s", filename, err.what());
return AudioData();
}
OpusDataSource dataSource = {&audioFile, 0};
OggOpusFile* opusFile = op_open_callbacks(&dataSource, &Opus_Callbacks, nullptr, 0, nullptr);
if (!opusFile) {
audioLogs.Warn("Error while reading %s", filename);
return AudioData();
}
const OpusHead* opusInfo = op_head(opusFile, -1);
if (!opusInfo) {
op_free(opusFile);
audioLogs.Warn("Could not read OpusHead in %s", filename);
return AudioData();
}
if (opusInfo->stream_count != 1) {
op_free(opusFile);
audioLogs.Warn("Only one stream is supported in Opus files: %s", filename);
return AudioData();
}
if (opusInfo->channel_count != 1 && opusInfo->channel_count != 2) {
op_free(opusFile);
audioLogs.Warn("Only mono and stereo Opus files are supported: %s", filename);
return AudioData();
}
const int sampleWidth = 2;
int sampleRate = 48000;
int numberOfChannels = opusInfo->channel_count;
// The buffer is big enough to hold 120ms worth of samples per channel
opus_int16* buffer = new opus_int16[numberOfChannels * 5760];
int samplesPerChannelRead = 0;
std::vector<opus_int16> samples;
while ((samplesPerChannelRead =
op_read(opusFile, buffer, sampleWidth * numberOfChannels * 5760, nullptr)) > 0) {
std::copy_n(buffer, samplesPerChannelRead * numberOfChannels, std::back_inserter(samples));
}
op_free(opusFile);
char* rawSamples = new char[sampleWidth * samples.size()];
std::copy_n(reinterpret_cast<char*>(samples.data()), sampleWidth * samples.size(), rawSamples);
return AudioData(sampleRate, sampleWidth, numberOfChannels, samples.size() * sampleWidth,
rawSamples);
}
static PyObject*
OpusDecoder_read(decoders_OpusDecoder* self, PyObject *args)
{
static opus_int16 pcm[BUF_SIZE];
int pcm_frames_read;
if (self->closed) {
PyErr_SetString(PyExc_ValueError, "stream is closed");
return NULL;
}
if ((pcm_frames_read = op_read(self->opus_file,
pcm,
BUF_SIZE,
NULL)) >= 0) {
const int channel_count = op_head(self->opus_file, -1)->channel_count;
int i;
pcm_FrameList *framelist = new_FrameList(self->audiotools_pcm,
channel_count,
BITS_PER_SAMPLE,
pcm_frames_read);
int *samples = framelist->samples;
for (i = 0; i < pcm_frames_read * channel_count; i++) {
samples[i] = pcm[i];
}
/*reorder channels to .wav order if necessary*/
switch (self->channel_count) {
case 1:
case 2:
default:
/*no change*/
break;
case 3:
/*fL fC fR -> fL fR fC*/
swap_channel_data(samples, 1, 2,
self->channel_count,
(unsigned)pcm_frames_read);
break;
case 4:
/*fL fR bL bR -> fL fR bL bR*/
/*no change*/
break;
case 5:
/*fL fC fR bL bR -> fL fR fC bL bR*/
swap_channel_data(samples, 1, 2,
self->channel_count,
(unsigned)pcm_frames_read);
break;
case 6:
/*fL fC fR bL bR LFE -> fL fR fC bL bR LFE*/
swap_channel_data(samples, 1, 2,
self->channel_count,
(unsigned)pcm_frames_read);
/*fL fR fC bL bR LFE -> fL fR fC LFE bR bL*/
swap_channel_data(samples, 3, 5,
self->channel_count,
(unsigned)pcm_frames_read);
/*fL fR fC LFE bR bL -> fL fR fC LFE bL bR*/
swap_channel_data(samples, 4, 5,
self->channel_count,
(unsigned)pcm_frames_read);
break;
case 7:
/*fL fC fR sL sR bC LFE -> fL fR fC sL sR bC LFE*/
swap_channel_data(samples, 1, 2,
self->channel_count,
(unsigned)pcm_frames_read);
/*fL fR fC sL sR bC LFE -> fL fR fC LFE sR bC sL*/
swap_channel_data(samples, 3, 6,
self->channel_count,
(unsigned)pcm_frames_read);
/*fL fR fC LFE sR bC sL -> fL fR fC LFE bC sR sL*/
swap_channel_data(samples, 4, 5,
self->channel_count,
(unsigned)pcm_frames_read);
/*fL fR fC LFE bC sR sL -> fL fR fC LFE bC sL sR*/
swap_channel_data(samples, 5, 6,
self->channel_count,
(unsigned)pcm_frames_read);
break;
case 8:
/*fL fC fR sL sR bL bR LFE -> fL fR fC sL sR bL bR LFE*/
swap_channel_data(samples, 1, 2,
self->channel_count,
(unsigned)pcm_frames_read);
/*fL fR fC sL sR bL bR LFE -> fL fR fC LFE sR bL bR sL*/
swap_channel_data(samples, 3, 6,
self->channel_count,
(unsigned)pcm_frames_read);
/*fL fR fC LFE sR bL bR sL -> fL fR fC LFE bL sR bR sL*/
swap_channel_data(samples, 4, 5,
self->channel_count,
(unsigned)pcm_frames_read);
/*fL fR fC LFE bL sR bR sL -> fL fR fC LFE bL bR sR sL*/
swap_channel_data(samples, 5, 6,
self->channel_count,
(unsigned)pcm_frames_read);
/*fL fR fC LFE bL bR sR sL -> fL fR fC LFE bL bR sL sR*/
swap_channel_data(samples, 6, 7,
self->channel_count,
(unsigned)pcm_frames_read);
break;
}
return (PyObject*)framelist;
} else {
/*some sort of read error occurred*/
PyErr_SetString(PyExc_ValueError, "error reading from file");
return NULL;
}
}
static DB_playItem_t *
opusdec_insert (ddb_playlist_t *plt, DB_playItem_t *after, const char *fname) {
DB_FILE *fp = deadbeef->fopen (fname);
if (!fp) {
return NULL;
}
int64_t fsize = deadbeef->fgetlength (fp);
if (fp->vfs->is_streaming ()) {
DB_playItem_t *it = deadbeef->pl_item_alloc_init (fname, plugin.plugin.id);
deadbeef->plt_set_item_duration (plt, it, -1);
deadbeef->pl_add_meta (it, "title", NULL);
after = deadbeef->plt_insert_item (plt, after, it);
deadbeef->pl_item_unref (it);
deadbeef->fclose (fp);
return after;
}
OggOpusFile *opusfile = opus_file_open (fp);
if (!opusfile) {
deadbeef->fclose (fp);
return NULL;
}
long nstreams = op_link_count (opusfile);
int64_t currentsample = 0;
for (int stream = 0; stream < nstreams; stream++) {
const OpusHead *head = op_head (opusfile, stream);
if (!head) {
continue;
}
int64_t totalsamples = op_pcm_total (opusfile, stream);
const float duration = totalsamples / 48000.f;
DB_playItem_t *it = deadbeef->pl_item_alloc_init (fname, plugin.plugin.id);
deadbeef->pl_set_meta_int (it, ":TRACKNUM", stream);
deadbeef->plt_set_item_duration (plt, it, duration);
if (nstreams > 1) {
deadbeef->pl_item_set_startsample (it, currentsample);
deadbeef->pl_item_set_endsample (it, currentsample + totalsamples - 1);
deadbeef->pl_set_item_flags (it, DDB_IS_SUBTRACK);
}
if (update_vorbis_comments (it, opusfile, stream))
continue;
int samplerate = 48000;
int64_t startsample = deadbeef->pl_item_get_startsample (it);
int64_t endsample = deadbeef->pl_item_get_endsample (it);
const off_t start_offset = sample_offset(opusfile, startsample-1);
const off_t end_offset = sample_offset(opusfile, endsample);
char *filetype = NULL;
const off_t stream_size = oggedit_opus_stream_info(deadbeef->fopen(fname), start_offset, end_offset, &filetype);
if (filetype) {
deadbeef->pl_replace_meta(it, ":FILETYPE", filetype);
free(filetype);
}
if (stream_size > 0) {
set_meta_ll(it, ":OPUS_STREAM_SIZE", stream_size);
deadbeef->pl_set_meta_int(it, ":BITRATE", 8.f * samplerate * stream_size / totalsamples / 1000);
}
set_meta_ll (it, ":FILE_SIZE", fsize);
deadbeef->pl_set_meta_int (it, ":CHANNELS", head->channel_count);
deadbeef->pl_set_meta_int (it, ":SAMPLERATE", samplerate);
if (nstreams == 1) {
DB_playItem_t *cue = deadbeef->plt_process_cue (plt, after, it, totalsamples, samplerate);
if (cue) {
deadbeef->pl_item_unref (it);
op_free(opusfile);
deadbeef->fclose (fp);
return cue;
}
}
else {
currentsample += totalsamples;
}
after = deadbeef->plt_insert_item (plt, after, it);
deadbeef->pl_item_unref (it);
}
op_free(opusfile);
deadbeef->fclose (fp);
return after;
}
static int
opusdec_init (DB_fileinfo_t *_info, DB_playItem_t *it) {
opusdec_info_t *info = (opusdec_info_t *)_info;
if (!info->info.file) {
deadbeef->pl_lock ();
const char *uri = strdupa (deadbeef->pl_find_meta (it, ":URI"));
deadbeef->pl_unlock ();
DB_FILE *fp = deadbeef->fopen (uri);
if (!fp) {
return -1;
}
info->info.file = fp;
info->it = it;
deadbeef->pl_item_ref (it);
}
info->opusfile = opus_file_open (info->info.file);
if (!info->opusfile) {
return -1;
}
const OpusHead *head = op_head (info->opusfile, 0);
if (head->channel_count > 8) {
trace ("opus: the track has %d channels, but 8 is max supported.\n");
return -1;
}
// take this parameters from your input file
// we set constants for clarity sake
_info->fmt.bps = 32;
_info->fmt.is_float = 1;
_info->fmt.channels = head->channel_count;
_info->fmt.samplerate = 48000;
if (head->mapping_family == 1) {
info->channelmap = oggedit_vorbis_channel_map (head->channel_count);
}
for (int i = 0; i < (_info->fmt.channels&0x1f); i++) {
_info->fmt.channelmask |= 1 << i;
}
_info->readpos = 0;
_info->plugin = &plugin;
// set all gain adjustment to 0, because deadbeef is performing that.
op_set_gain_offset (info->opusfile, OP_ABSOLUTE_GAIN, 0);
if (info->info.file->vfs->is_streaming ()) {
deadbeef->pl_item_set_startsample (it, 0);
if (deadbeef->pl_get_item_duration (it) < 0) {
deadbeef->pl_item_set_endsample (it, -1);
}
else {
deadbeef->pl_item_set_endsample (it, op_pcm_total (info->opusfile, -1) - 1);
}
if (update_vorbis_comments (it, info->opusfile, -1))
return -1;
deadbeef->pl_set_meta_int(it, ":TRACKNUM", 0);
}
else {
opusdec_seek_sample (_info, 0);
}
deadbeef->pl_replace_meta (it, "!FILETYPE", "Ogg Opus");
deadbeef->pl_set_meta_int (it, ":CHANNELS", head->channel_count);
info->cur_bit_stream = -1;
return 0;
}
static int
update_vorbis_comments (DB_playItem_t *it, OggOpusFile *opusfile, const int tracknum) {
const OpusTags *vc = op_tags (opusfile, tracknum);
if (!vc) {
return -1;
}
deadbeef->pl_delete_all_meta (it);
for (int i = 0; i < vc->comments; i++) {
char *tag = strdup(vc->user_comments[i]);
char *value;
if (tag && (value = strchr(tag, '='))
#ifdef ANDROID
&& strlen (value) < 4000
#endif
) {
// skip the ignored RG fields, and the picture
if (_is_replaygain_tag (it, tag) || !strcasecmp (tag, "METADATA_BLOCK_PICTURE")) {
free (tag);
continue;
}
*value++ = '\0';
deadbeef->pl_append_meta(it, oggedit_map_tag(tag, "tag2meta"), value);
}
if (tag) {
free(tag);
}
}
const char *r128_trackgain = deadbeef->pl_find_meta (it, "R128_TRACK_GAIN");
if (r128_trackgain) {
int trackgain = atoi (r128_trackgain) + op_head (opusfile, tracknum)->output_gain;
if (trackgain != 0) {
deadbeef->pl_set_item_replaygain (it, DDB_REPLAYGAIN_TRACKGAIN, trackgain / 256.0f + 5.0f);
deadbeef->pl_delete_meta (it, "R128_TRACK_GAIN");
}
}
int albumgain = op_head (opusfile, tracknum)->output_gain;
const char *r128_albumgain = deadbeef->pl_find_meta (it, "R128_ALBUM_GAIN");
if (r128_albumgain) {
albumgain += atoi (r128_albumgain);
deadbeef->pl_delete_meta (it, "R128_ALBUM_GAIN");
}
if (albumgain != 0) {
deadbeef->pl_set_item_replaygain (it, DDB_REPLAYGAIN_ALBUMGAIN, albumgain / 256.0f + 5.0f);
}
char s[100];
int output_gain = op_head (opusfile, tracknum)->output_gain;
snprintf (s, sizeof (s), "%0.2f dB", output_gain / 256.0f + 5.0f);
deadbeef->pl_replace_meta (it, ":OPUS_HEADER_GAIN", s);
deadbeef->pl_set_meta_int (it, ":SAMPLERATE_ORIGINAL", op_head (opusfile, tracknum)->input_sample_rate);
deadbeef->pl_add_meta (it, "title", NULL);
uint32_t f = deadbeef->pl_get_item_flags (it);
f &= ~DDB_TAG_MASK;
f |= DDB_TAG_VORBISCOMMENTS;
deadbeef->pl_set_item_flags (it, f);
ddb_playlist_t *plt = deadbeef->plt_get_curr ();
if (plt) {
deadbeef->plt_modified (plt);
deadbeef->plt_unref (plt);
}
return 0;
}
void VoiceMessagesLoader::onLoad(AudioData *audio) {
bool started = false;
int32 audioindex = -1;
Loader *l = 0;
Loaders::iterator j = _loaders.end();
{
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
VoiceMessages::Msg &m(voice->_data[i]);
if (m.audio != audio || !m.loading) continue;
audioindex = i;
j = _loaders.find(audio);
if (j != _loaders.end() && (j.value()->fname != m.fname || j.value()->data.size() != m.data.size())) {
delete j.value();
_loaders.erase(j);
j = _loaders.end();
}
if (j == _loaders.end()) {
l = (j = _loaders.insert(audio, new Loader())).value();
l->fname = m.fname;
l->data = m.data;
int ret;
if (m.data.isEmpty()) {
l->file = op_open_file(m.fname.toUtf8().constData(), &ret);
} else {
l->file = op_open_memory((const unsigned char*)m.data.constData(), m.data.size(), &ret);
}
if (!l->file) {
LOG(("Audio Error: op_open_file failed for '%1', data size '%2', error code %3").arg(m.fname).arg(m.data.size()).arg(ret));
m.state = VoiceMessageStopped;
return loadError(j);
}
ogg_int64_t duration = op_pcm_total(l->file, -1);
if (duration < 0) {
LOG(("Audio Error: op_pcm_total failed to get full duration for '%1', data size '%2', error code %3").arg(m.fname).arg(m.data.size()).arg(duration));
m.state = VoiceMessageStopped;
return loadError(j);
}
m.duration = duration;
m.skipStart = 0;
m.skipEnd = duration;
m.position = 0;
m.started = 0;
started = true;
} else {
if (!m.skipEnd) continue;
l = j.value();
}
break;
}
}
if (j == _loaders.end()) {
LOG(("Audio Error: trying to load part of audio, that is not playing at the moment"));
emit error(audio);
return;
}
if (started) {
l->pcm_offset = op_pcm_tell(l->file);
l->pcm_print_offset = l->pcm_offset - AudioVoiceMsgFrequency;
}
bool finished = false;
DEBUG_LOG(("Audio Info: reading buffer for file '%1', data size '%2', current pcm_offset %3").arg(l->fname).arg(l->data.size()).arg(l->pcm_offset));
QByteArray result;
int64 samplesAdded = 0;
while (result.size() < AudioVoiceMsgBufferSize) {
opus_int16 pcm[AudioVoiceMsgFrequency * AudioVoiceMsgChannels];
int ret = op_read_stereo(l->file, pcm, sizeof(pcm) / sizeof(*pcm));
if (ret < 0) {
{
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (voice) {
VoiceMessages::Msg &m(voice->_data[audioindex]);
if (m.audio == audio) {
m.state = VoiceMessageStopped;
}
}
}
LOG(("Audio Error: op_read_stereo failed, error code %1").arg(ret));
return loadError(j);
}
int li = op_current_link(l->file);
if (li != l->prev_li) {
const OpusHead *head = op_head(l->file, li);
const OpusTags *tags = op_tags(l->file, li);
for (int32 ci = 0; ci < tags->comments; ++ci) {
const char *comment = tags->user_comments[ci];
if (opus_tagncompare("METADATA_BLOCK_PICTURE", 22, comment) == 0) {
OpusPictureTag pic;
int err = opus_picture_tag_parse(&pic, comment);
if (err >= 0) {
opus_picture_tag_clear(&pic);
}
}
}
if (!op_seekable(l->file)) {
l->pcm_offset = op_pcm_tell(l->file) - ret;
}
}
if (li != l->prev_li || l->pcm_offset >= l->pcm_print_offset + AudioVoiceMsgFrequency) {
l->pcm_print_offset = l->pcm_offset;
}
l->pcm_offset = op_pcm_tell(l->file);
if (!ret) {
DEBUG_LOG(("Audio Info: read completed"));
finished = true;
break;
}
result.append((const char*)pcm, sizeof(*pcm) * ret * AudioVoiceMsgChannels);
l->prev_li = li;
samplesAdded += ret;
{
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
VoiceMessages::Msg &m(voice->_data[audioindex]);
if (m.audio != audio || !m.loading || m.fname != l->fname || m.data.size() != l->data.size()) {
LOG(("Audio Error: playing changed while loading"));
m.state = VoiceMessageStopped;
return loadError(j);
}
}
}
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
VoiceMessages::Msg &m(voice->_data[audioindex]);
if (m.audio != audio || !m.loading || m.fname != l->fname || m.data.size() != l->data.size()) {
LOG(("Audio Error: playing changed while loading"));
m.state = VoiceMessageStopped;
return loadError(j);
}
if (started) {
if (m.source) {
alSourceStop(m.source);
for (int32 i = 0; i < 3; ++i) {
if (m.samplesCount[i]) {
alSourceUnqueueBuffers(m.source, 1, m.buffers + i);
m.samplesCount[i] = 0;
}
}
m.nextBuffer = 0;
}
}
if (samplesAdded) {
if (!m.source) {
alGenSources(1, &m.source);
alSourcef(m.source, AL_PITCH, 1.f);
alSourcef(m.source, AL_GAIN, 1.f);
alSource3f(m.source, AL_POSITION, 0, 0, 0);
alSource3f(m.source, AL_VELOCITY, 0, 0, 0);
alSourcei(m.source, AL_LOOPING, 0);
}
if (!m.buffers[m.nextBuffer]) alGenBuffers(3, m.buffers);
if (!_checkALError()) {
m.state = VoiceMessageStopped;
return loadError(j);
}
if (m.samplesCount[m.nextBuffer]) {
alSourceUnqueueBuffers(m.source, 1, m.buffers + m.nextBuffer);
m.skipStart += m.samplesCount[m.nextBuffer];
}
m.samplesCount[m.nextBuffer] = samplesAdded;
alBufferData(m.buffers[m.nextBuffer], AL_FORMAT_STEREO16, result.constData(), result.size(), AudioVoiceMsgFrequency);
alSourceQueueBuffers(m.source, 1, m.buffers + m.nextBuffer);
m.skipEnd -= samplesAdded;
m.nextBuffer = (m.nextBuffer + 1) % 3;
if (!_checkALError()) {
m.state = VoiceMessageStopped;
return loadError(j);
}
} else {
finished = true;
}
if (finished) {
m.skipEnd = 0;
m.duration = m.skipStart + m.samplesCount[0] + m.samplesCount[1] + m.samplesCount[2];
}
m.loading = false;
if (m.state == VoiceMessageResuming || m.state == VoiceMessagePlaying || m.state == VoiceMessageStarting) {
ALint state = AL_INITIAL;
alGetSourcei(m.source, AL_SOURCE_STATE, &state);
if (_checkALError()) {
if (state != AL_PLAYING) {
alSourcePlay(m.source);
emit needToCheck();
}
}
}
}
static qboolean S_OPUS_CodecOpenStream (snd_stream_t *stream)
{
OggOpusFile *opFile;
const OpusHead *op_info;
long numstreams;
int res;
opFile = op_open_callbacks(&stream->fh, &opc_qfs, NULL, 0, &res);
if (!opFile)
{
Con_Printf("%s is not a valid Opus file (error %i).\n",
stream->name, res);
goto _fail;
}
stream->priv = opFile;
if (!op_seekable(opFile))
{
Con_Printf("Opus stream %s not seekable.\n", stream->name);
goto _fail;
}
op_info = op_head(opFile, -1);
if (!op_info)
{
Con_Printf("Unable to get stream information for %s.\n", stream->name);
goto _fail;
}
/* FIXME: handle section changes */
numstreams = op_info->stream_count;
if (numstreams != 1)
{
Con_Printf("More than one (%ld) stream in %s\n",
(long)op_info->stream_count, stream->name);
goto _fail;
}
if (op_info->channel_count != 1 && op_info->channel_count != 2)
{
Con_Printf("Unsupported number of channels %d in %s\n",
op_info->channel_count, stream->name);
goto _fail;
}
/* All Opus audio is coded at 48 kHz, and should also be decoded
* at 48 kHz for playback: info->input_sample_rate only tells us
* the sampling rate of the original input before opus encoding.
* S_RawSamples() shall already downsample this, as necessary. */
stream->info.rate = 48000;
stream->info.channels = op_info->channel_count;
/* op_read() yields 16-bit output using native endian ordering: */
stream->info.bits = 16;
stream->info.width = 2;
return true;
_fail:
if (opFile)
op_free(opFile);
return false;
}
/*the OpusDecoder.read() method*/
static PyObject*
OpusDecoder_read(decoders_OpusDecoder* self, PyObject *args)
{
static opus_int16 pcm[BUF_SIZE];
int pcm_frames_read;
if (self->closed) {
PyErr_SetString(PyExc_ValueError, "stream is closed");
return NULL;
}
if ((pcm_frames_read = op_read(self->opus_file,
pcm,
BUF_SIZE,
NULL)) >= 0) {
const int channel_count = op_head(self->opus_file, -1)->channel_count;
aa_int *channels = self->channels;
int c;
channels->reset(channels);
/*split buffer by channel*/
for (c = 0; c < channel_count; c++) {
a_int *channel = channels->append(channels);
int i;
channel->resize(channel, pcm_frames_read);
for (i = 0; i < pcm_frames_read; i++) {
a_append(channel, pcm[c + (i * channel_count)]);
}
}
/*reorder channels to .wav order if necessary*/
switch (channel_count) {
case 1:
case 2:
default:
/*no change*/
break;
case 3:
/*fL fC fR -> fL fR fC*/
a_int_swap(channels->_[1], channels->_[2]);
break;
case 4:
/*fL fR bL bR -> fL fR bL bR*/
/*no change*/
break;
case 5:
/*fL fC fR bL bR -> fL fR fC bL bR*/
a_int_swap(channels->_[1], channels->_[2]);
break;
case 6:
/*fL fC fR bL bR LFE -> fL fR fC bL bR LFE*/
a_int_swap(channels->_[1], channels->_[2]);
/*fL fR fC bL bR LFE -> fL fR fC LFE bR bL*/
a_int_swap(channels->_[3], channels->_[5]);
/*fL fR fC LFE bR bL -> fL fR fC LFE bL bR*/
a_int_swap(channels->_[4], channels->_[5]);
break;
case 7:
/*fL fC fR sL sR bC LFE -> fL fR fC sL sR bC LFE*/
a_int_swap(channels->_[1], channels->_[2]);
/*fL fR fC sL sR bC LFE -> fL fR fC LFE sR bC sL*/
a_int_swap(channels->_[3], channels->_[6]);
/*fL fR fC LFE sR bC sL -> fL fR fC LFE bC sR sL*/
a_int_swap(channels->_[4], channels->_[5]);
/*fL fR fC LFE bC sR sL -> fL fR fC LFE bC sL sR*/
a_int_swap(channels->_[5], channels->_[6]);
break;
case 8:
/*fL fC fR sL sR bL bR LFE -> fL fR fC sL sR bL bR LFE*/
a_int_swap(channels->_[1], channels->_[2]);
/*fL fR fC sL sR bL bR LFE -> fL fR fC LFE sR bL bR sL*/
a_int_swap(channels->_[3], channels->_[6]);
/*fL fR fC LFE sR bL bR sL -> fL fR fC LFE bL sR bR sL*/
a_int_swap(channels->_[4], channels->_[5]);
/*fL fR fC LFE bL sR bR sL -> fL fR fC LFE bL bR sR sL*/
a_int_swap(channels->_[5], channels->_[6]);
/*fL fR fC LFE bL bR sR sL -> fL fR fC LFE bL bR sL sR*/
a_int_swap(channels->_[6], channels->_[7]);
break;
}
/*return new FrameList object*/
return aa_int_to_FrameList(self->audiotools_pcm,
channels,
16);
} else {
/*some sort of read error occurred*/
PyErr_SetString(PyExc_ValueError, "error reading from file");
return NULL;
}
}
