int qSpeexDecode(QSpeexCodecPtr handle, void* inputBytes, int inputSize, void* outputSamples, int outputSize)
{
int offset, remaining;
short *out = (short*)outputSamples;
/* If there is no input to read, we certainly can't read it. */
if (inputBytes != NULL) {
speex_bits_read_from(&handle->decBits, inputBytes, inputSize);
}
for (offset=0; offset<outputSize; offset+=handle->frameSize) {
if (inputBytes != NULL) {
if (!speex_bits_remaining(&handle->decBits)) {
// Ran out of input data
return 2;
}
speex_decode_int(handle->decState, &handle->decBits, out+offset);
}
else {
/* Extrapolate output-buffer based on current decoder state. */
speex_decode_int(handle->decState, NULL, out+offset);
}
}
remaining = speex_bits_remaining(&handle->decBits);
if (remaining >= 8) {
/* If less than a byte is left over, that's OK. */
fprintf(stderr, "qSpeexDecode(): %d bits left over\n", remaining);
return 1; // Still have encoded bits left over
}
else return 0; // A-OK!!
}
static int libspeex_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
LibSpeexContext *s = avctx->priv_data;
int16_t *output = data, *end;
int i, num_samples;
num_samples = s->frame_size * avctx->channels;
end = output + *data_size / sizeof(*output);
speex_bits_read_from(&s->bits, buf, buf_size);
for (i = 0; speex_bits_remaining(&s->bits) && output + num_samples < end; i++) {
int ret = speex_decode_int(s->dec_state, &s->bits, output);
if (ret <= -2) {
av_log(avctx, AV_LOG_ERROR, "Error decoding Speex frame.\n");
return -1;
} else if (ret == -1)
// end of stream
break;
if (avctx->channels == 2)
speex_decode_stereo_int(output, s->frame_size, &s->stereo);
output += num_samples;
}
avctx->frame_size = s->frame_size * i;
*data_size = avctx->channels * avctx->frame_size * sizeof(*output);
return buf_size;
}
/*****************************************************************************
* DecodePacket: decodes a Speex packet.
*****************************************************************************/
static block_t *DecodePacket( decoder_t *p_dec, ogg_packet *p_oggpacket )
{
decoder_sys_t *p_sys = p_dec->p_sys;
if( p_oggpacket->bytes )
{
/* Copy Ogg packet to Speex bitstream */
speex_bits_read_from( &p_sys->bits, (char *)p_oggpacket->packet,
p_oggpacket->bytes );
p_sys->i_frame_in_packet = 0;
}
/* Decode one frame at a time */
if( p_sys->i_frame_in_packet < p_sys->p_header->frames_per_packet )
{
block_t *p_aout_buffer;
if( p_sys->p_header->frame_size == 0 )
return NULL;
p_aout_buffer =
decoder_NewAudioBuffer( p_dec, p_sys->p_header->frame_size );
if( !p_aout_buffer )
{
return NULL;
}
switch( speex_decode_int( p_sys->p_state, &p_sys->bits,
(int16_t *)p_aout_buffer->p_buffer ) )
{
case -2:
msg_Err( p_dec, "decoding error: corrupted stream?" );
case -1: /* End of stream */
return NULL;
}
if( speex_bits_remaining( &p_sys->bits ) < 0 )
{
msg_Err( p_dec, "decoding overflow: corrupted stream?" );
}
if( p_sys->p_header->nb_channels == 2 )
speex_decode_stereo_int( (int16_t *)p_aout_buffer->p_buffer,
p_sys->p_header->frame_size,
&p_sys->stereo );
/* Date management */
p_aout_buffer->i_pts = date_Get( &p_sys->end_date );
p_aout_buffer->i_length =
date_Increment( &p_sys->end_date, p_sys->p_header->frame_size )
- p_aout_buffer->i_pts;
p_sys->i_frame_in_packet++;
return p_aout_buffer;
}
else
{
return NULL;
}
}
int Speex_2_Pcm16( unsigned char* out_buf, unsigned char* in_buf,
unsigned int size,
unsigned int channels, unsigned int rate, long h_codec )
{
SpeexState* ss;
short* pcm = (short*) out_buf;
int frames_out = 0;
ss = (SpeexState*) h_codec;
if (!ss || channels!=1)
return -1;
speex_bits_read_from(&ss->decoder.bits, (char*)in_buf, size);
/* We don't know where frame boundaries are,
but the minimum frame size is 43 */
while (speex_bits_remaining(&ss->decoder.bits)>40) {
int ret;
ret = speex_decode_int(ss->decoder.state, &ss->decoder.bits, pcm);
pcm+= ss->frame_size;
if (ret==-2) {
ERROR("while calling speex_decode\n");
return -1;
}
if (ret==-1) break;
frames_out++;
}
return frames_out*ss->frame_size*sizeof(short);
}
static int libspeex_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
LibSpeexContext *s = avctx->priv_data;
AVFrame *frame = data;
int16_t *output;
int ret, consumed = 0;
/* get output buffer */
frame->nb_samples = s->frame_size;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
output = (int16_t *)frame->data[0];
/* if there is not enough data left for the smallest possible frame or the
next 5 bits are a terminator code, reset the libspeex buffer using the
current packet, otherwise ignore the current packet and keep decoding
frames from the libspeex buffer. */
if (speex_bits_remaining(&s->bits) < 5 ||
speex_bits_peek_unsigned(&s->bits, 5) == 0xF) {
/* check for flush packet */
if (!buf || !buf_size) {
*got_frame_ptr = 0;
return buf_size;
}
/* set new buffer */
speex_bits_read_from(&s->bits, buf, buf_size);
consumed = buf_size;
}
/* decode a single frame */
ret = speex_decode_int(s->dec_state, &s->bits, output);
if (ret <= -2) {
av_log(avctx, AV_LOG_ERROR, "Error decoding Speex frame.\n");
return AVERROR_INVALIDDATA;
}
if (avctx->channels == 2)
speex_decode_stereo_int(output, s->frame_size, &s->stereo);
*got_frame_ptr = 1;
return consumed;
}
static int decode(struct audec_state *st, int16_t *sampv,
size_t *sampc, const uint8_t *buf, size_t len)
{
const size_t n = st->channels * st->frame_size;
size_t i = 0;
/* Read into bit-stream */
speex_bits_read_from(&st->bits, (char *)buf, (int)len);
/* Handle multiple Speex frames in one RTP packet */
while (speex_bits_remaining(&st->bits) >= MIN_FRAME_SIZE) {
int ret;
if (*sampc < n)
return ENOMEM;
ret = speex_decode_int(st->dec, &st->bits,
(int16_t *)&sampv[i]);
if (ret < 0) {
if (-1 == ret) {
}
else if (-2 == ret) {
warning("speex: decode: corrupt stream\n");
}
else {
warning("speex: decode: speex_decode_int:"
" ret=%d\n", ret);
}
break;
}
/* Transforms a mono frame into a stereo frame
using intensity stereo info */
if (2 == st->channels) {
speex_decode_stereo_int((int16_t *)&sampv[i],
st->frame_size,
&st->stereo);
}
i += n;
*sampc -= n;
}
*sampc = i;
return 0;
}
tsk_size_t tdav_codec_speex_decode(tmedia_codec_t* self, const void* in_data, tsk_size_t in_size, void** out_data, tsk_size_t* out_max_size, const tsk_object_t* proto_hdr)
{
int ret;
tsk_size_t out_size = 0;
tdav_codec_speex_t* speex = (tdav_codec_speex_t*)self;
if(!self || !in_data || !in_size || !out_data){
TSK_DEBUG_ERROR("Invalid parameter");
return 0;
}
// initializes the bit-stream
speex_bits_read_from(&speex->decoder.bits, (char*)in_data, in_size);
do{
// performs decode()
if((ret = speex_decode_int(speex->decoder.state, &speex->decoder.bits, speex->decoder.buffer))){
TSK_DEBUG_ERROR("Failed to decode the buffer. retcode=%d", ret);
break;
}
if(*out_max_size <(out_size + speex->decoder.size)){
if((*out_data = tsk_realloc(*out_data, (out_size + speex->decoder.size)))){
*out_max_size = (out_size + speex->decoder.size);
}
else{
*out_max_size = 0;
return 0;
}
}
// copy output buffer
memcpy(&((uint8_t*)*out_data)[out_size], speex->decoder.buffer, speex->decoder.size);
out_size += speex->decoder.size;
}
while(speex_bits_remaining(&speex->decoder.bits) >= 5);
return out_size;
}
/*****************************************************************************
* ProcessPacket: processes a Speex packet.
*****************************************************************************/
static void *ProcessPacket( decoder_t *p_dec, ogg_packet *p_oggpacket,
block_t **pp_block )
{
decoder_sys_t *p_sys = p_dec->p_sys;
block_t *p_block = *pp_block;
if( p_block && p_block->i_flags & BLOCK_FLAG_DISCONTINUITY )
Flush( p_dec );
/* Date management */
if( p_block && p_block->i_pts > VLC_TS_INVALID &&
p_block->i_pts != date_Get( &p_sys->end_date ) )
{
date_Set( &p_sys->end_date, p_block->i_pts );
}
if( !date_Get( &p_sys->end_date ) )
{
/* We've just started the stream, wait for the first PTS. */
if( p_block ) block_Release( p_block );
return NULL;
}
*pp_block = NULL; /* To avoid being fed the same packet again */
if( p_sys->b_packetizer )
{
if ( p_sys->p_header->frames_per_packet > 1 )
{
short *p_frame_holder = NULL;
int i_bits_before = 0, i_bits_after = 0, i_bytes_in_speex_frame = 0,
i_pcm_output_size = 0, i_bits_in_speex_frame = 0;
block_t *p_new_block = NULL;
i_pcm_output_size = p_sys->p_header->frame_size;
p_frame_holder = (short*)xmalloc( sizeof(short)*i_pcm_output_size );
speex_bits_read_from( &p_sys->bits, (char*)p_oggpacket->packet,
p_oggpacket->bytes);
i_bits_before = speex_bits_remaining( &p_sys->bits );
speex_decode_int(p_sys->p_state, &p_sys->bits, p_frame_holder);
i_bits_after = speex_bits_remaining( &p_sys->bits );
i_bits_in_speex_frame = i_bits_before - i_bits_after;
i_bytes_in_speex_frame = ( i_bits_in_speex_frame +
(8 - (i_bits_in_speex_frame % 8)) )
/ 8;
p_new_block = block_Alloc( i_bytes_in_speex_frame );
memset( p_new_block->p_buffer, 0xff, i_bytes_in_speex_frame );
/*
* Copy the first frame in this packet to a new packet.
*/
speex_bits_rewind( &p_sys->bits );
speex_bits_write( &p_sys->bits,
(char*)p_new_block->p_buffer,
(int)i_bytes_in_speex_frame );
/*
* Move the remaining part of the original packet (subsequent
* frames, if there are any) into the beginning
* of the original packet so
* they are preserved following the realloc.
* Note: Any bits that
* remain in the initial packet
* are "filler" if they do not constitute
* an entire byte.
*/
if ( i_bits_after > 7 )
{
/* round-down since we rounded-up earlier (to include
* the speex terminator code.
*/
i_bytes_in_speex_frame--;
speex_bits_write( &p_sys->bits,
(char*)p_block->p_buffer,
p_block->i_buffer - i_bytes_in_speex_frame );
p_block = block_Realloc( p_block,
0,
p_block->i_buffer-i_bytes_in_speex_frame );
*pp_block = p_block;
}
else
{
speex_bits_reset( &p_sys->bits );
}
free( p_frame_holder );
return SendPacket( p_dec, p_new_block);
}
else
{
return SendPacket( p_dec, p_block );
}
}
else
{
block_t *p_aout_buffer = DecodePacket( p_dec, p_oggpacket );
if( p_block )
block_Release( p_block );
return p_aout_buffer;
}
}
static GstFlowReturn
gst_speex_dec_parse_data (GstSpeexDec * dec, GstBuffer * buf)
{
GstFlowReturn res = GST_FLOW_OK;
gint i, fpp;
guint size;
guint8 *data;
SpeexBits *bits;
if (!dec->frame_duration)
goto not_negotiated;
if (G_LIKELY (GST_BUFFER_SIZE (buf))) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* send data to the bitstream */
speex_bits_read_from (&dec->bits, (char *) data, size);
fpp = dec->header->frames_per_packet;
bits = &dec->bits;
GST_DEBUG_OBJECT (dec, "received buffer of size %u, fpp %d, %d bits",
size, fpp, speex_bits_remaining (bits));
} else {
/* FIXME ? actually consider how much concealment is needed */
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
fpp = dec->header->frames_per_packet;
bits = NULL;
}
/* now decode each frame, catering for unknown number of them (e.g. rtp) */
for (i = 0; i < fpp; i++) {
GstBuffer *outbuf;
gint16 *out_data;
gint ret;
GST_LOG_OBJECT (dec, "decoding frame %d/%d, %d bits remaining", i, fpp,
bits ? speex_bits_remaining (bits) : -1);
res =
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
ret = speex_decode_int (dec->state, bits, out_data);
if (ret == -1) {
/* uh? end of stream */
if (fpp == 0 && speex_bits_remaining (bits) < 8) {
/* if we did not know how many frames to expect, then we get this
at the end if there are leftover bits to pad to the next byte */
GST_DEBUG_OBJECT (dec, "Discarding leftover bits");
} else {
GST_WARNING_OBJECT (dec, "Unexpected end of stream found");
}
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
} else if (ret == -2) {
GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?");
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
}
if (bits && speex_bits_remaining (bits) < 0) {
GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?");
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
}
if (dec->header->nb_channels == 2)
speex_decode_stereo_int (out_data, dec->frame_size, dec->stereo);
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
break;
}
}
return res;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
("decoder not initialized"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
boost::uint8_t*
AudioDecoderSpeex::decode(const EncodedAudioFrame& input,
boost::uint32_t& outputSize)
{
speex_bits_read_from(&_speex_bits, reinterpret_cast<char*>(input.data.get()),
input.dataSize);
std::vector<DecodedFrame*> decoded_frames;
boost::uint32_t total_size = 0;
while (speex_bits_remaining(&_speex_bits)) {
boost::scoped_array<short> output( new short[_speex_framesize] );
int rv = speex_decode_int(_speex_dec_state, &_speex_bits, output.get());
if (rv != 0) {
if (rv != -1) {
log_error(_("Corrupt Speex stream!"));
}
break;
}
boost::int16_t* conv_data = 0;
#ifdef RESAMPLING_SPEEX
spx_uint32_t conv_size = 0;
conv_data = new boost::int16_t[_target_frame_size];
memset(conv_data, 0, _target_frame_size * 2);
spx_uint32_t in_size = _speex_framesize;
// Our input format is mono and we want to expand to stereo. Speex
// won't do this for us, but we can ask it to skip a sample after
// writing one, so all we have to do is duplicate the samples.
speex_resampler_set_output_stride(_resampler, 2);
conv_size = _target_frame_size; // Assuming this hould be samples.
int err = speex_resampler_process_int(_resampler, 0 /* mono */, output.get(), &in_size, conv_data, &conv_size);
if (err != RESAMPLER_ERR_SUCCESS) {
log_error(_("Failed to resample Speex frame."));
delete [] conv_data;
continue;
}
// The returned size is the number of *mono* samples returned.
conv_size *= 2;
// Now, duplicate all the samples so we get a stereo sound.
for (boost::uint32_t i = 0; i < conv_size; i += 2) {
conv_data[i+1] = conv_data[i];
}
// Our interface requires returning the audio size in bytes.
conv_size *= sizeof(boost::int16_t);
#else
int outsize = 0;
AudioResampler::convert_raw_data(&conv_data, &outsize, output.get(),
_speex_framesize /* sample count*/, 2 /* sample size */,
16000, false /* stereo */, 44100 /* new rate */,
true /* convert to stereo */);
boost::uint32_t conv_size = outsize;
#endif
total_size += conv_size;
decoded_frames.push_back(new DecodedFrame(conv_data, conv_size));
}
outputSize = total_size;
// We have to jump through hoops because decode() requires as much
// data to be returned as possible.
boost::uint8_t* rv = new boost::uint8_t[total_size];
boost::uint8_t* ptr = rv;
for (std::vector<DecodedFrame*>::iterator it = decoded_frames.begin(),
end = decoded_frames.end(); it != end; ++it) {
DecodedFrame* frame = *it;
memcpy(ptr, frame->data.get(), frame->size);
ptr += frame->size;
delete frame;
}
outputSize = total_size;
return rv;
}
static GstFlowReturn
speex_dec_chain_parse_data (GstSpeexDec * dec, GstBuffer * buf,
GstClockTime timestamp, GstClockTime duration)
{
GstFlowReturn res = GST_FLOW_OK;
gint i, fpp;
guint size;
guint8 *data;
SpeexBits *bits;
if (!dec->frame_duration)
goto not_negotiated;
if (timestamp != -1) {
dec->segment.last_stop = timestamp;
} else {
timestamp = dec->segment.last_stop;
}
if (buf) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* send data to the bitstream */
speex_bits_read_from (&dec->bits, (char *) data, size);
fpp = 0;
bits = &dec->bits;
GST_DEBUG_OBJECT (dec, "received buffer of size %u, fpp %d", size, fpp);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
fpp = dec->header->frames_per_packet;
bits = NULL;
}
/* now decode each frame, catering for unknown number of them (e.g. rtp) */
for (i = 0; (!fpp || i < fpp) && (!bits || speex_bits_remaining (bits) > 0);
i++) {
GstBuffer *outbuf;
gint16 *out_data;
gint ret;
GST_LOG_OBJECT (dec, "decoding frame %d/%d", i, fpp);
res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad,
GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2,
GST_PAD_CAPS (dec->srcpad), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
ret = speex_decode_int (dec->state, bits, out_data);
if (ret == -1) {
/* uh? end of stream */
GST_WARNING_OBJECT (dec, "Unexpected end of stream found");
gst_buffer_unref (outbuf);
outbuf = NULL;
break;
} else if (ret == -2) {
GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?");
gst_buffer_unref (outbuf);
outbuf = NULL;
break;
}
if (bits && speex_bits_remaining (bits) < 0) {
GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?");
gst_buffer_unref (outbuf);
outbuf = NULL;
break;
}
if (dec->header->nb_channels == 2)
speex_decode_stereo_int (out_data, dec->frame_size, dec->stereo);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = dec->frame_duration;
dec->segment.last_stop += dec->frame_duration;
timestamp = dec->segment.last_stop;
GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (dec->frame_duration));
res = gst_pad_push (dec->srcpad, outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
break;
}
}
return res;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
("decoder not initialized"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
static GstFlowReturn
gst_speex_dec_parse_data (GstSpeexDec * dec, GstBuffer * buf)
{
GstFlowReturn res = GST_FLOW_OK;
gint i, fpp;
SpeexBits *bits;
GstMapInfo map;
if (!dec->frame_duration)
goto not_negotiated;
if (G_LIKELY (gst_buffer_get_size (buf))) {
/* send data to the bitstream */
gst_buffer_map (buf, &map, GST_MAP_READ);
speex_bits_read_from (&dec->bits, (gchar *) map.data, map.size);
gst_buffer_unmap (buf, &map);
fpp = dec->header->frames_per_packet;
bits = &dec->bits;
GST_DEBUG_OBJECT (dec, "received buffer of size %" G_GSIZE_FORMAT
", fpp %d, %d bits", map.size, fpp, speex_bits_remaining (bits));
} else {
/* FIXME ? actually consider how much concealment is needed */
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
fpp = dec->header->frames_per_packet;
bits = NULL;
}
/* now decode each frame, catering for unknown number of them (e.g. rtp) */
for (i = 0; i < fpp; i++) {
GstBuffer *outbuf;
gboolean corrupted = FALSE;
gint ret;
GST_LOG_OBJECT (dec, "decoding frame %d/%d, %d bits remaining", i, fpp,
bits ? speex_bits_remaining (bits) : -1);
#if 0
res =
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
#endif
/* FIXME, we can use a bufferpool because we have fixed size buffers. We
* could also use an allocator */
outbuf =
gst_buffer_new_allocate (NULL,
dec->frame_size * dec->header->nb_channels * 2, NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
ret = speex_decode_int (dec->state, bits, (spx_int16_t *) map.data);
if (ret == -1) {
/* uh? end of stream */
GST_WARNING_OBJECT (dec, "Unexpected end of stream found");
corrupted = TRUE;
} else if (ret == -2) {
GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?");
corrupted = TRUE;
}
if (bits && speex_bits_remaining (bits) < 0) {
GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?");
corrupted = TRUE;
}
if (dec->header->nb_channels == 2)
speex_decode_stereo_int ((spx_int16_t *) map.data, dec->frame_size,
dec->stereo);
gst_buffer_unmap (outbuf, &map);
if (!corrupted) {
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
} else {
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
gst_buffer_unref (outbuf);
}
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
break;
}
}
return res;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
("decoder not initialized"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
int AudioStreamPlaybackSpeex::mix(int16_t *p_buffer, int p_frames) {
//printf("update, loops %i, read ofs %i\n", (int)loops, read_ofs);
//printf("playing %i, paused %i\n", (int)playing, (int)paused);
if (!active || !playing || !data.size())
return 0;
/*
if (read_ofs >= data.size()) {
if (loops) {
reload();
++loop_count;
} else {
return;
};
};
*/
int todo = p_frames;
if (todo < page_size) {
return 0;
};
int eos = 0;
while (todo > page_size) {
int ret = 0;
while ((todo > page_size && packets_available && !eos) || (ret = ogg_sync_pageout(&oy, &og)) == 1) {
if (!packets_available) {
/*Add page to the bitstream*/
ogg_stream_pagein(&os, &og);
page_granule = ogg_page_granulepos(&og);
page_nb_packets = ogg_page_packets(&og);
packet_no = 0;
if (page_granule > 0 && frame_size) {
skip_samples = page_nb_packets * frame_size * nframes - (page_granule - last_granule);
if (ogg_page_eos(&og))
skip_samples = -skip_samples;
/*else if (!ogg_page_bos(&og))
skip_samples = 0;*/
} else {
skip_samples = 0;
}
last_granule = page_granule;
packets_available = true;
}
/*Extract all available packets*/
while (todo > page_size && !eos) {
if (ogg_stream_packetout(&os, &op) != 1) {
packets_available = false;
break;
}
packet_no++;
/*End of stream condition*/
if (op.e_o_s)
eos = 1;
/*Copy Ogg packet to Speex bitstream*/
speex_bits_read_from(&bits, (char *)op.packet, op.bytes);
for (int j = 0; j != nframes; j++) {
int16_t *out = p_buffer;
int ret;
/*Decode frame*/
ret = speex_decode_int(st, &bits, out);
/*for (i=0;i<frame_size*channels;i++)
printf ("%d\n", (int)output[i]);*/
if (ret == -1) {
printf("decode returned -1\n");
break;
};
if (ret == -2) {
OS::get_singleton()->printerr("Decoding error: corrupted stream?\n");
break;
}
if (speex_bits_remaining(&bits) < 0) {
OS::get_singleton()->printerr("Decoding overflow: corrupted stream?\n");
break;
}
//if (channels==2)
// speex_decode_stereo_int(output, frame_size, &stereo);
/*Convert to short and save to output file*/
for (int i = 0; i < frame_size * stream_channels; i++) {
out[i] = le_short(out[i]);
}
{
int new_frame_size = frame_size;
/*printf ("packet %d %d\n", packet_no, skip_samples);*/
if (packet_no == 1 && j == 0 && skip_samples > 0) {
/*printf ("chopping first packet\n");*/
new_frame_size -= skip_samples;
}
if (packet_no == page_nb_packets && skip_samples < 0) {
int packet_length = nframes * frame_size + skip_samples;
new_frame_size = packet_length - j * frame_size;
if (new_frame_size < 0)
new_frame_size = 0;
if (new_frame_size > frame_size)
new_frame_size = frame_size;
/*printf ("chopping end: %d %d %d\n", new_frame_size, packet_length, packet_no);*/
}
p_buffer += new_frame_size * stream_channels;
todo -= new_frame_size;
}
}
};
};
//todo = get_todo();
//todo is still greater than page size, can write more
if (todo > page_size || eos) {
if (read_ofs < data.size()) {
//char *buf;
int nb_read = MIN(data.size() - read_ofs, READ_CHUNK);
/*Get the ogg buffer for writing*/
char *ogg_dst = ogg_sync_buffer(&oy, nb_read);
/*Read bitstream from input file*/
copymem(ogg_dst, &data[read_ofs], nb_read);
read_ofs += nb_read;
ogg_sync_wrote(&oy, nb_read);
} else {
if (loops) {
reload();
++loop_count;
//break;
} else {
playing = false;
unload();
break;
};
}
};
};
return p_frames - todo;
};
int universal_speex_decode(void* handle, const void* pCodedData,
unsigned cbCodedPacketSize, void* pAudioBuffer,
unsigned cbBufferSize, unsigned *pcbDecodedSize,
const struct RtpHeader* pRtpHeader)
{
SpeexBits bits;
struct speex_codec_data_decoder *mpSpeexDec =
(struct speex_codec_data_decoder *)handle;
assert(handle != NULL);
if (mpSpeexDec->mNumSamplesPerFrame > cbBufferSize)
{
return RPLG_INVALID_ARGUMENT;
}
/* Reset number of decoded samples */
*pcbDecodedSize = 0;
/* Decode incoming packet if present. Do PLC if no packet. */
if (pCodedData)
{
/* Prepare data for Speex decoder */
speex_bits_set_bit_buffer(&bits,(char*)pCodedData, cbCodedPacketSize);
/* Decode while there are something to decode and enough space
* for decoded data. */
while (cbBufferSize >= mpSpeexDec->mNumSamplesPerFrame &&
(speex_bits_remaining(&bits) > 0))
{
int res;
/* Decode frame */
res = speex_decode_int(mpSpeexDec->mpDecoderState, &bits,
((spx_int16_t*)pAudioBuffer)+(*pcbDecodedSize));
if (res == 0)
{
/* Update number of decoded and available samples on success */
*pcbDecodedSize += mpSpeexDec->mNumSamplesPerFrame;
cbBufferSize -= mpSpeexDec->mNumSamplesPerFrame;
}
else
{
/* If it's the end of the stream or corrupted stream just return */
break;
}
}
}
else
{
int res;
/* Do PLC */
res = speex_decode_int(mpSpeexDec->mpDecoderState, NULL,
((spx_int16_t*)pAudioBuffer)+(*pcbDecodedSize));
if (res == 0)
{
/* Update number of decoded and available samples on success */
*pcbDecodedSize += mpSpeexDec->mNumSamplesPerFrame;
}
else
{
/* This mustn't happen for lost packet, but who knows... */
return RPLG_FAILED;
}
}
return RPLG_SUCCESS;
}
JNIEXPORT int JNICALL FUNCSPEEX(decodeRemaining)(JNIEnv* env, jobject obj) {
return speex_bits_remaining(&dbits);
}
UInt32 OggSpeexDecoder::ReadAudio(AudioBufferList *bufferList, UInt32 frameCount)
{
if(!IsOpen() || NULL == bufferList || bufferList->mNumberBuffers != mFormat.mChannelsPerFrame || 0 == frameCount)
return 0;
UInt32 framesRead = 0;
// Reset output buffer data size
for(UInt32 i = 0; i < bufferList->mNumberBuffers; ++i)
bufferList->mBuffers[i].mDataByteSize = 0;
for(;;) {
UInt32 framesRemaining = frameCount - framesRead;
UInt32 framesToSkip = static_cast<UInt32>(bufferList->mBuffers[0].mDataByteSize / sizeof(float));
UInt32 framesInBuffer = static_cast<UInt32>(mBufferList->mBuffers[0].mDataByteSize / sizeof(float));
UInt32 framesToCopy = std::min(framesInBuffer, framesRemaining);
// Copy data from the buffer to output
for(UInt32 i = 0; i < mBufferList->mNumberBuffers; ++i) {
float *floatBuffer = static_cast<float *>(bufferList->mBuffers[i].mData);
memcpy(floatBuffer + framesToSkip, mBufferList->mBuffers[i].mData, framesToCopy * sizeof(float));
bufferList->mBuffers[i].mDataByteSize += static_cast<UInt32>(framesToCopy * sizeof(float));
// Move remaining data in buffer to beginning
if(framesToCopy != framesInBuffer) {
floatBuffer = static_cast<float *>(mBufferList->mBuffers[i].mData);
memmove(floatBuffer, floatBuffer + framesToCopy, (framesInBuffer - framesToCopy) * sizeof(float));
}
mBufferList->mBuffers[i].mDataByteSize -= static_cast<UInt32>(framesToCopy * sizeof(float));
}
framesRead += framesToCopy;
// All requested frames were read
if(framesRead == frameCount)
break;
// EOS reached
if(mSpeexEOSReached)
break;
// Attempt to process the desired number of packets
unsigned packetsDesired = 1;
while(0 < packetsDesired && !mSpeexEOSReached) {
// Process any packets in the current page
while(0 < packetsDesired && !mSpeexEOSReached) {
// Grab a packet from the streaming layer
ogg_packet oggPacket;
int result = ogg_stream_packetout(&mOggStreamState, &oggPacket);
if(-1 == result) {
LOGGER_ERR("org.sbooth.AudioEngine.AudioDecoder.OggSpeex", "Ogg Speex decoding error: Ogg loss of streaming");
break;
}
// If result is 0, there is insufficient data to assemble a packet
if(0 == result)
break;
// Otherwise, we got a valid packet for processing
if(1 == result) {
if(5 <= oggPacket.bytes && !memcmp(oggPacket.packet, "Speex", 5))
mSpeexSerialNumber = mOggStreamState.serialno;
if(-1 == mSpeexSerialNumber || mOggStreamState.serialno != mSpeexSerialNumber)
break;
// Ignore the following:
// - Speex comments in packet #2
// - Extra headers (optionally) in packets 3+
if(1 != mOggPacketCount && 1 + mExtraSpeexHeaderCount <= mOggPacketCount) {
// Detect Speex EOS
if(oggPacket.e_o_s && mOggStreamState.serialno == mSpeexSerialNumber)
mSpeexEOSReached = true;
// SPEEX_GET_FRAME_SIZE is in samples
spx_int32_t speexFrameSize;
speex_decoder_ctl(mSpeexDecoder, SPEEX_GET_FRAME_SIZE, &speexFrameSize);
float buffer [(2 == mFormat.mChannelsPerFrame) ? 2 * speexFrameSize : speexFrameSize];
// Copy the Ogg packet to the Speex bitstream
speex_bits_read_from(&mSpeexBits, (char *)oggPacket.packet, static_cast<int>(oggPacket.bytes));
// Decode each frame in the Speex packet
for(spx_int32_t i = 0; i < mSpeexFramesPerOggPacket; ++i) {
result = speex_decode(mSpeexDecoder, &mSpeexBits, buffer);
// -1 indicates EOS
if(-1 == result)
break;
else if(-2 == result) {
LOGGER_ERR("org.sbooth.AudioEngine.AudioDecoder.OggSpeex", "Ogg Speex decoding error: possible corrupted stream");
break;
}
if(0 > speex_bits_remaining(&mSpeexBits)) {
LOGGER_ERR("org.sbooth.AudioEngine.AudioDecoder.OggSpeex", "Ogg Speex decoding overflow: possible corrupted stream");
break;
}
// Normalize the values
float maxSampleValue = 1u << 15;
vDSP_vsdiv(buffer, 1, &maxSampleValue, buffer, 1, speexFrameSize);
// Copy the frames from the decoding buffer to the output buffer, skipping over any frames already decoded
framesInBuffer = static_cast<UInt32>(mBufferList->mBuffers[0].mDataByteSize / sizeof(float));
memcpy(static_cast<float *>(mBufferList->mBuffers[0].mData) + framesInBuffer, buffer, speexFrameSize * sizeof(float));
mBufferList->mBuffers[0].mDataByteSize += static_cast<UInt32>(speexFrameSize * sizeof(float));
// Process stereo channel, if present
if(2 == mFormat.mChannelsPerFrame) {
speex_decode_stereo(buffer, speexFrameSize, mSpeexStereoState);
vDSP_vsdiv(buffer + speexFrameSize, 1, &maxSampleValue, buffer + speexFrameSize, 1, speexFrameSize);
memcpy(static_cast<float *>(mBufferList->mBuffers[1].mData) + framesInBuffer, buffer + speexFrameSize, speexFrameSize * sizeof(float));
mBufferList->mBuffers[1].mDataByteSize += static_cast<UInt32>(speexFrameSize * sizeof(float));
}
// Packet processing finished
--packetsDesired;
}
}
++mOggPacketCount;
}
}
// Grab a new Ogg page for processing, if necessary
if(!mSpeexEOSReached && 0 < packetsDesired) {
while(1 != ogg_sync_pageout(&mOggSyncState, &mOggPage)) {
// Get the ogg buffer for writing
char *data = ogg_sync_buffer(&mOggSyncState, READ_SIZE_BYTES);
// Read bitstream from input file
ssize_t bytesRead = GetInputSource()->Read(data, READ_SIZE_BYTES);
if(-1 == bytesRead) {
LOGGER_ERR("org.sbooth.AudioEngine.AudioDecoder.OggSpeex", "Unable to read from the input file");
break;
}
ogg_sync_wrote(&mOggSyncState, bytesRead);
// No more data available from input file
if(0 == bytesRead)
break;
}
// Ensure all Ogg streams are read
if(ogg_page_serialno(&mOggPage) != mOggStreamState.serialno)
ogg_stream_reset_serialno(&mOggStreamState, ogg_page_serialno(&mOggPage));
// Get the resultant Ogg page
int result = ogg_stream_pagein(&mOggStreamState, &mOggPage);
if(0 != result) {
LOGGER_ERR("org.sbooth.AudioEngine.AudioDecoder.OggSpeex", "Error reading Ogg page");
break;
}
}
}
}
mCurrentFrame += framesRead;
if(0 == framesRead && mSpeexEOSReached)
mTotalFrames = mCurrentFrame;
return framesRead;
}
/* This function will iterate frames & submodes in the Speex bits.
* Returns 0 if a frame found, otherwise returns -1.
*/
int speex_get_next_frame(SpeexBits *bits)
{
static const int inband_skip_table[NB_SUBMODES] =
{1, 1, 4, 4, 4, 4, 4, 4, 8, 8, 16, 16, 32, 32, 64, 64 };
static const int wb_skip_table[SB_SUBMODES] =
{SB_SUBMODE_BITS+1, 36, 112, 192, 352, -1, -1, -1};
unsigned submode;
unsigned nb_count = 0;
while (speex_bits_remaining(bits) >= 5) {
unsigned wb_count = 0;
unsigned bit_ptr = bits->bitPtr;
unsigned char_ptr = bits->charPtr;
/* WB frame */
while ((speex_bits_remaining(bits) >= 4)
&& speex_bits_unpack_unsigned(bits, 1))
{
int advance;
submode = speex_bits_unpack_unsigned(bits, 3);
advance = wb_skip_table[submode];
if (advance < 0) {
TRACE__((THIS_FUNC, "Invalid mode encountered. "
"The stream is corrupted."));
return -1;
}
TRACE__((THIS_FUNC, "WB layer skipped: %d bits", advance));
advance -= (SB_SUBMODE_BITS+1);
speex_bits_advance(bits, advance);
bit_ptr = bits->bitPtr;
char_ptr = bits->charPtr;
/* Consecutive subband frames may not exceed 2 frames */
if (++wb_count > 2)
return -1;
}
/* End of bits, return the frame */
if (speex_bits_remaining(bits) < 4) {
TRACE__((THIS_FUNC, "End of stream"));
return 0;
}
/* Stop iteration, return the frame */
if (nb_count > 0) {
bits->bitPtr = bit_ptr;
bits->charPtr = char_ptr;
return 0;
}
/* Get control bits */
submode = speex_bits_unpack_unsigned(bits, 4);
TRACE__((THIS_FUNC, "Control bits: %d at %d",
submode, bits->charPtr*8+bits->bitPtr));
if (submode == 15) {
TRACE__((THIS_FUNC, "Found submode: terminator"));
return -1;
} else if (submode == 14) {
/* in-band signal; next 4 bits contain signal id */
submode = speex_bits_unpack_unsigned(bits, 4);
TRACE__((THIS_FUNC, "Found submode: in-band %d bits",
inband_skip_table[submode]));
speex_bits_advance(bits, inband_skip_table[submode]);
} else if (submode == 13) {
/* user in-band; next 5 bits contain msg len */
submode = speex_bits_unpack_unsigned(bits, 5);
TRACE__((THIS_FUNC, "Found submode: user-band %d bytes", submode));
speex_bits_advance(bits, submode * 8);
} else if (submode > 8) {
TRACE__((THIS_FUNC, "Unknown sub-mode %d", submode));
return -1;
} else {
/* NB frame */
unsigned int advance = submode;
speex_mode_query(&speex_nb_mode, SPEEX_SUBMODE_BITS_PER_FRAME, &advance);
if (advance < 0) {
TRACE__((THIS_FUNC, "Invalid mode encountered. "
"The stream is corrupted."));
return -1;
}
TRACE__((THIS_FUNC, "Submode %d: %d bits", submode, advance));
advance -= (NB_SUBMODE_BITS+1);
speex_bits_advance(bits, advance);
++nb_count;
}
}
return 0;
}
static int speex_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
uint8_t* buf, int buf_size)
{
SpeexContext *s = avctx->priv_data;
int ret, bits_remaining, used = 0, bits_remaining_before;
av_log(avctx,AV_LOG_DEBUG,"speexdec: out: %p %p(%d), in: %p, %d\n",
data, data_size, data_size?*data_size:0,buf,buf_size);
if (data)
{
/* we'll save this info to log later */
bits_remaining_before = bits_remaining = speex_bits_remaining(&s->bits);
*data_size = 0;
/* We are not interpolating a frame */
if (buf)
{
if (bits_remaining > 0)
{
speex_bits_read_whole_bytes(&s->bits, (char*) buf, buf_size);
}
else
{
/*Copy buf to Speex bitstream*/
speex_bits_read_from(&s->bits, (char*) buf, buf_size);
}
} // end if we are not interpolating a frame
ret = speex_decode_int(s->st, buf ? &s->bits : NULL, data);
switch (ret)
{
case -2:
av_log(avctx, AV_LOG_ERROR, "Error: corrupted speex stream\n");
speex_bits_reset(&s->bits);
return -1;
case -1:
/* no frame has been decoded, return bytes used */
//av_log(avctx, AV_LOG_INFO, "Warning: no samples have been decoded (end of stream)\n");
speex_bits_rewind(&s->bits);
return buf_size;
default:
if (buf)
{
/* Calculate how many bytes we've used */
bits_remaining = speex_bits_remaining(&s->bits);
if (bits_remaining < 0)
{
//av_log(avctx, AV_LOG_DEBUG, "Warning: decoding overflow, need more data (%d used)\n", buf_size);
speex_bits_rewind(&s->bits);
return buf_size;
}
used = buf_size - (bits_remaining / 8);
}
else
{
av_log(avctx, AV_LOG_DEBUG, "Info: interpolating a lost frame\n");
}
*data_size = avctx->frame_size * avctx->channels * sizeof(short);
if (avctx->channels == 2)
speex_decode_stereo_int(data, avctx->frame_size, &s->stereo);
if (avctx->debug & FF_DEBUG_BITSTREAM)
{
int bitrate;
speex_decoder_ctl(s->st, SPEEX_GET_BITRATE, &bitrate);
av_log(avctx, AV_LOG_DEBUG, "Speex: decoder used: %d/%d bytes, output: %d bytes, bitrate: %d bps (remaining in stream: %.1f bytes after %d bits)\n",
used, buf_size, *data_size, bitrate, (float) bits_remaining / 8, bits_remaining_before);
}
speex_bits_reset(&s->bits);
break;
}
}
else
{
speex_bits_reset(&s->bits);
}
return used;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
int error = CODEC_ERROR;
SpeexBits bits;
int eof = 0;
spx_ogg_sync_state oy;
spx_ogg_page og;
spx_ogg_packet op;
spx_ogg_stream_state os;
spx_int64_t page_granule = 0;
spx_int64_t cur_granule = 0;
int enh_enabled = 1;
int nframes = 2;
int eos = 0;
SpeexStereoState *stereo;
int channels = -1;
int samplerate = ci->id3->frequency;
int extra_headers = 0;
int stream_init = 0;
/* rockbox: comment 'set but unused' variables
int page_nb_packets;
*/
int frame_size;
int packet_count = 0;
int lookahead;
int headerssize = 0;
unsigned long strtoffset = ci->id3->offset;
void *st = NULL;
int j = 0;
intptr_t param;
memset(&bits, 0, sizeof(bits));
memset(&oy, 0, sizeof(oy));
/* Ogg handling still uses mallocs, so reset the malloc buffer per track */
if (codec_init()) {
goto exit;
}
ci->seek_buffer(0);
ci->set_elapsed(0);
stereo = speex_stereo_state_init();
spx_ogg_sync_init(&oy);
spx_ogg_alloc_buffer(&oy,2*CHUNKSIZE);
codec_set_replaygain(ci->id3);
eof = 0;
while (!eof) {
enum codec_command_action action = ci->get_command(¶m);
if (action == CODEC_ACTION_HALT)
break;
/*seek (seeks to the page before the position) */
if (action == CODEC_ACTION_SEEK_TIME) {
if(samplerate!=0&&packet_count>1){
LOGF("Speex seek page:%lld,%lld,%ld,%lld,%d\n",
((spx_int64_t)param/1000) *
(spx_int64_t)samplerate,
page_granule, param,
(page_granule/samplerate)*1000, samplerate);
speex_seek_page_granule(((spx_int64_t)param/1000) *
(spx_int64_t)samplerate,
page_granule, &oy, headerssize);
}
ci->set_elapsed(param);
ci->seek_complete();
}
next_page:
/*Get the ogg buffer for writing*/
if(get_more_data(&oy)<1){/*read error*/
goto done;
}
/* Loop for all complete pages we got (most likely only one) */
while (spx_ogg_sync_pageout(&oy, &og) == 1) {
int packet_no;
if (stream_init == 0) {
spx_ogg_stream_init(&os, spx_ogg_page_serialno(&og));
stream_init = 1;
}
/* Add page to the bitstream */
spx_ogg_stream_pagein(&os, &og);
page_granule = spx_ogg_page_granulepos(&og);
/* page_nb_packets = spx_ogg_page_packets(&og); */
cur_granule = page_granule;
/* Extract all available packets */
packet_no=0;
while (!eos && spx_ogg_stream_packetout(&os, &op)==1){
/* If first packet, process as Speex header */
if (packet_count==0){
st = process_header(&op, enh_enabled, &frame_size,
&samplerate, &nframes, &channels,
stereo, &extra_headers);
speex_decoder_ctl(st, SPEEX_GET_LOOKAHEAD, &lookahead);
if (!nframes)
nframes=1;
if (!st){
goto done;
}
ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
if (channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
} else if (channels == 1) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
}
/* Speex header in its own page, add the whole page
headersize */
headerssize += og.header_len+og.body_len;
} else if (packet_count<=1+extra_headers){
/* add packet to headersize */
headerssize += op.bytes;
/* Ignore extra headers */
} else {
if (packet_count <= 2+extra_headers) {
if (strtoffset) {
ci->seek_buffer(strtoffset);
spx_ogg_sync_reset(&oy);
packet_count++;
goto next_page;
}
}
packet_no++;
if (op.e_o_s) /* End of stream condition */
eos=1;
/* Set Speex bitstream to point to Ogg packet */
speex_bits_set_bit_buffer(&bits, (char *)op.packet,
op.bytes);
for (j = 0; j != nframes; j++){
int ret;
/* Decode frame */
ret = speex_decode_int(st, &bits, output);
if (ret == -1)
break;
if (ret == -2)
break;
if (speex_bits_remaining(&bits) < 0)
break;
if (channels == 2)
speex_decode_stereo_int(output, frame_size, stereo);
if (frame_size > 0) {
spx_int16_t *frame_start = output + lookahead;
if (channels == 2)
frame_start += lookahead;
ci->pcmbuf_insert(frame_start, NULL,
frame_size - lookahead);
lookahead = 0;
/* 2 bytes/sample */
cur_granule += frame_size / 2;
ci->set_offset((long) ci->curpos);
ci->set_elapsed((samplerate == 0) ? 0 :
cur_granule * 1000 / samplerate);
}
}
}
packet_count++;
}
}
}
error = CODEC_OK;
done:
/* Clean things up for the next track */
speex_bits_destroy(&bits);
if (st)
speex_decoder_destroy(st);
if (stream_init)
spx_ogg_stream_destroy(&os);
spx_ogg_sync_destroy(&oy);
exit:
return error;
}
static void dec_process(MSFilter *f){
DecState *s=(DecState*)f->data;
mblk_t *im;
mblk_t *om;
int err=-2;
SpeexBits bits;
int bytes=s->frsz*2;
bool_t bits_initd=FALSE;
while((im=ms_queue_get(f->inputs[0]))!=NULL){
int rem_bits=(im->b_wptr-im->b_rptr)*8;
if (!bits_initd) {
speex_bits_init(&bits);
bits_initd=TRUE;
}else speex_bits_reset(&bits);
speex_bits_read_from(&bits,(char*)im->b_rptr,im->b_wptr-im->b_rptr);
/* support for multiple frame in one RTP packet */
do{
om=allocb(bytes,0);
mblk_meta_copy(im, om);
err=speex_decode_int(s->state,&bits,(int16_t*)om->b_wptr);
om->b_wptr+=bytes;
if (err==0){
ms_queue_put(f->outputs[0],om);
if (s->sample_time==0) s->sample_time=f->ticker->time;
s->sample_time+=20;
if (s->plc_count>0){
// ms_warning("Did speex packet loss concealment during %i ms",s->plc_count*20);
s->plc_count=0;
}
}else {
if (err==-1)
ms_warning("speex end of stream");
else if (err==-2)
ms_warning("speex corrupted stream");
freemsg(om);
}
}while((rem_bits= speex_bits_remaining(&bits))>10);
freemsg(im);
}
if (s->plc && s->sample_time!=0 && f->ticker->time>=s->sample_time){
/* we should output a frame but no packet were decoded
thus do packet loss concealment*/
om=allocb(bytes,0);
err=speex_decode_int(s->state,NULL,(int16_t*)om->b_wptr);
om->b_wptr+=bytes;
mblk_set_plc_flag(om, 1);
ms_queue_put(f->outputs[0],om);
s->sample_time+=20;
s->plc_count++;
if (s->plc_count>=plc_max){
s->sample_time=0;
}
}
if (bits_initd) speex_bits_destroy(&bits);
}
