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Realmedia related codecs fixes and enhancements

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* More tolerance to the file format variations.
 * AC3 coded files in realaudio format are now playable

Full credit to Igor Poretsky

Change-Id: Id24e94bc00623e89fb8c80403efa92f69ab1e5d7
This commit is contained in:
Solomon Peachy 2018-12-22 20:04:28 -05:00
parent eee3f0ce79
commit 9b9b30bd54
8 changed files with 309 additions and 124 deletions
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106
lib/rbcodec/codecs/cook.c
View file

@ -21,7 +21,6 @@
#include <string.h>
#include "logf.h"
#include "codeclib.h"
#include "inttypes.h"
#include "libcook/cook.h"
@ -38,6 +37,22 @@ static void init_rm(RMContext *rmctx)
memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext));
}
static int request_packet(int size)
{
int consumed = 0;
while (1)
{
uint8_t *buffer = ci->request_buffer((size_t *)(&consumed), size);
if (!consumed)
break;
consumed = rm_get_packet(&buffer, &rmctx, &pkt);
if (consumed < 0 || consumed == size)
break;
ci->advance_buffer(size);
}
return consumed;
}
/* this is the codec entry point */
enum codec_status codec_main(enum codec_entry_call_reason reason)
{
@ -49,12 +64,10 @@ enum codec_status codec_main(enum codec_entry_call_reason reason)
/* this is called for each file to process */
enum codec_status codec_run(void)
{
static size_t buff_size;
int datasize, res, consumed, i, time_offset;
uint8_t *bit_buffer;
uint16_t fs,sps,h;
uint32_t packet_count;
int scrambling_unit_size, num_units;
int spn, packet_header_size, scrambling_unit_size, num_units;
size_t resume_offset;
intptr_t param;
long action;
@ -84,14 +97,17 @@ enum codec_status codec_run(void)
ci->configure(DSP_SET_STEREO_MODE, rmctx.nb_channels == 1 ?
STEREO_MONO : STEREO_NONINTERLEAVED);
packet_header_size = PACKET_HEADER_SIZE +
((rmctx.flags & RM_PKT_V1) ? 1 : 0);
packet_count = rmctx.nb_packets;
rmctx.audio_framesize = rmctx.block_align;
rmctx.block_align = rmctx.sub_packet_size;
fs = rmctx.audio_framesize;
sps= rmctx.block_align;
h = rmctx.sub_packet_h;
scrambling_unit_size = h * (fs + PACKET_HEADER_SIZE);
scrambling_unit_size = h * (fs + packet_header_size);
spn = h * fs / sps;
res =cook_decode_init(&rmctx, &q);
if(res < 0) {
DEBUGF("failed to initialize cook decoder\n");
@ -101,10 +117,10 @@ enum codec_status codec_run(void)
/* check for a mid-track resume and force a seek time accordingly */
if(resume_offset) {
resume_offset -= MIN(resume_offset, rmctx.data_offset + DATA_HEADER_SIZE);
num_units = (int)resume_offset / scrambling_unit_size;
/* put number of subpackets to skip in resume_offset */
resume_offset /= (sps + PACKET_HEADER_SIZE);
param = (int)resume_offset * ((sps * 8 * 1000)/rmctx.bit_rate);
num_units = (int)resume_offset / scrambling_unit_size;
/* put number of packets to skip in resume_offset */
resume_offset = num_units * h;
param = (int)resume_offset * ((8000LL * fs)/rmctx.bit_rate);
}
if (param) {
@ -120,14 +136,15 @@ enum codec_status codec_run(void)
seek_start :
while(packet_count)
{
bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
consumed = request_packet(scrambling_unit_size);
if (!consumed)
break;
if(consumed < 0) {
DEBUGF("rm_get_packet failed\n");
return CODEC_ERROR;
}
for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
for (i = 0; i < spn; i++)
{
if (action == CODEC_ACTION_NULL)
action = ci->get_command(&param);
@ -155,52 +172,65 @@ seek_start :
action = CODEC_ACTION_NULL;
goto seek_start;
}
num_units = (param/(sps*1000*8/rmctx.bit_rate))/(h*(fs/sps));
ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * num_units);
bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
num_units = (param/(sps*1000*8/rmctx.bit_rate))/spn;
ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + scrambling_unit_size * num_units);
consumed = request_packet(scrambling_unit_size);
if (!consumed) {
ci->seek_complete();
return CODEC_OK;
}
if(consumed < 0) {
DEBUGF("rm_get_packet failed\n");
DEBUGF("rm_get_packet failed\n");
ci->seek_complete();
return CODEC_ERROR;
}
packet_count = rmctx.nb_packets - rmctx.audio_pkt_cnt * num_units;
packet_count = rmctx.nb_packets - h * num_units;
rmctx.frame_number = (param/(sps*1000*8/rmctx.bit_rate));
while(rmctx.audiotimestamp > (unsigned) param) {
while(rmctx.audiotimestamp > (unsigned)param && num_units-- > 0) {
rmctx.audio_pkt_cnt = 0;
ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * (num_units-1));
bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
packet_count += rmctx.audio_pkt_cnt;
num_units--;
ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + scrambling_unit_size * num_units);
consumed = request_packet(scrambling_unit_size);
if (!consumed) {
ci->seek_complete();
return CODEC_OK;
}
if(consumed < 0) {
ci->seek_complete();
DEBUGF("rm_get_packet failed\n");
return CODEC_ERROR;
}
packet_count += h;
}
if (num_units < 0)
rmctx.audiotimestamp = 0;
time_offset = param - rmctx.audiotimestamp;
i = (time_offset/((sps * 8 * 1000)/rmctx.bit_rate));
ci->set_elapsed(rmctx.audiotimestamp+(1000*8*sps/rmctx.bit_rate)*i);
ci->set_elapsed(param);
ci->seek_complete();
}
action = CODEC_ACTION_NULL;
res = cook_decode_frame(&rmctx,&q, rm_outbuf, &datasize, pkt.frames[i], rmctx.block_align);
rmctx.frame_number++;
res = cook_decode_frame(&rmctx,&q, rm_outbuf, &datasize, pkt.frames[i], sps);
/* skip the first two frames; no valid audio */
if(rmctx.frame_number < 3) continue;
if(res != rmctx.block_align) {
if (res != sps) {
DEBUGF("codec error\n");
return CODEC_ERROR;
}
ci->pcmbuf_insert(rm_outbuf,
rm_outbuf+q.samples_per_channel,
q.samples_per_channel);
if(datasize)
ci->pcmbuf_insert(rm_outbuf,
rm_outbuf+q.samples_per_channel,
q.samples_per_channel);
ci->set_elapsed(rmctx.audiotimestamp+(1000*8*sps/rmctx.bit_rate)*i);
rmctx.frame_number++;
}
packet_count -= rmctx.audio_pkt_cnt;
packet_count -= h;
rmctx.audio_pkt_cnt = 0;
ci->advance_buffer(consumed);
ci->advance_buffer(scrambling_unit_size);
}
return CODEC_OK;