Author: sayer
Date: 2008-08-25 22:55:09 +0200 (Mon, 25 Aug 2008)
New Revision: 1077
Modified:
branches/wb/core/AmAudio.cpp
branches/wb/core/AmAudio.h
branches/wb/core/AmConferenceChannel.cpp
branches/wb/core/AmConferenceChannel.h
branches/wb/core/AmMediaProcessor.cpp
branches/wb/core/AmPlaylist.cpp
branches/wb/core/AmPlaylist.h
branches/wb/core/AmPlayoutBuffer.cpp
branches/wb/core/AmPlayoutBuffer.h
branches/wb/core/AmPrecodedFile.cpp
branches/wb/core/AmRtpAudio.cpp
branches/wb/core/AmRtpAudio.h
branches/wb/core/AmRtpStream.cpp
branches/wb/core/amci/amci.h
branches/wb/core/plug-in/adpcm/adpcm.c
branches/wb/core/plug-in/g722/g722.c
branches/wb/core/plug-in/gsm/gsm.c
branches/wb/core/plug-in/ilbc/ilbc.c
branches/wb/core/plug-in/speex/speex.c
branches/wb/core/plug-in/wav/wav.c
Log:
wideband, take 2: correctly resample at the proper places.
- AmAudio::get() takes time in millisec instead of # samples
- AMCI_FMT_FRAME_LENGTH used (frame in millisec)
- note: on RTP->input, resampling is done before playout buffer
(i.e. playout buffer has SYSTEM_SAMPLERATE)
- AMCI_ENCODED_FRAME_SIZE removed (unused anyway)
Modified: branches/wb/core/AmAudio.cpp
===================================================================
--- branches/wb/core/AmAudio.cpp 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/AmAudio.cpp 2008-08-25 20:55:09 UTC (rev 1077)
@@ -45,7 +45,6 @@
amci_codec_t *codec;
int frame_size;
int frame_length;
- int frame_encoded_size;
long h_codec;
};
@@ -97,7 +96,6 @@
cc->codec = codec;
cc->frame_size = frame_size;
cc->frame_length = frame_length;
- cc->frame_encoded_size = frame_encoded_size;
cc->h_codec = h_codec;
m_codecContainerByPayload[payload] = cc;
}
@@ -107,7 +105,6 @@
codec = c->second->codec;
frame_size = c->second->frame_size;
frame_length = c->second->frame_length;
- frame_encoded_size = c->second->frame_encoded_size;
h_codec = c->second->h_codec;
}
if (m_currentPayloadP && codec) {
@@ -130,7 +127,7 @@
AmAudioFormat::AmAudioFormat()
: channels(-1), rate(-1), codec(NULL),
- frame_length(20), frame_size(20*SYSTEM_SAMPLERATE/1000),
frame_encoded_size(320)
+ frame_length(20), frame_size(20*SYSTEM_SAMPLERATE/1000)
{
}
@@ -152,11 +149,10 @@
unsigned int AmAudioFormat::calcBytesToRead(unsigned int needed_samples) const
{
if (codec && codec->samples2bytes)
- return codec->samples2bytes(h_codec, needed_samples) *
- rate * channels / SYSTEM_SAMPLERATE; // FIXME: channels, system_channels
+ return codec->samples2bytes(h_codec, needed_samples) * channels; // FIXME:
channels, system_channels
WARN("Cannot convert samples to bytes\n");
- return needed_samples * channels * rate / SYSTEM_SAMPLERATE;
+ return needed_samples * channels;
}
unsigned int AmAudioFormat::bytes2samples(unsigned int bytes) const
@@ -201,9 +197,6 @@
case AMCI_FMT_FRAME_SIZE: {
frame_size=fmt_i[i].value;
} break;
- case AMCI_FMT_ENCODED_FRAME_SIZE: {
- frame_encoded_size=fmt_i[i].value;
- } break;
default: {
DBG("Unknown codec format descriptor: %d\n", fmt_i[i].id);
} break;
@@ -230,7 +223,6 @@
amci_codec_t* AmAudioFormat::getCodec()
{
-
if(!codec){
int codec_id = getCodecId();
codec = AmPlugIn::instance()->codec(codec_id);
@@ -252,7 +244,6 @@
: fmt(new AmAudioSimpleFormat(CODEC_PCM16)),
max_rec_time(-1),
rec_time(0)
-
{
initSrc();
}
@@ -294,7 +285,8 @@
#endif
}
-void AmAudio::setFormat(AmAudioFormat* new_fmt) {
+void AmAudio::setFormat(AmAudioFormat* new_fmt)
+{
fmt.reset(new_fmt);
fmt->resetCodec();
}
@@ -304,21 +296,23 @@
}
// returns bytes read, else -1 if error (0 is OK)
-int AmAudio::get(unsigned int user_ts, unsigned char* buffer, unsigned int
nb_samples)
+int AmAudio::get(unsigned int user_ts, unsigned char* buffer, unsigned int
time_millisec)
{
+ unsigned int nb_samples = time_millisec * fmt->rate / 1000;
+
int size = calcBytesToRead(nb_samples);
-
size = read(user_ts,size);
- if(size <= 0) {
+ if (size <= 0)
return size;
- }
size = decode(size);
- if(size < 0) {
+
+ if (size < 0) {
DBG("decode returned %i\n",size);
return -1;
}
+
/* into internal format */
size = downMix(size);
@@ -335,7 +329,7 @@
return 0;
}
- if(max_rec_time > -1 && rec_time >= max_rec_time)
+ if((max_rec_time > -1) && (rec_time >= max_rec_time))
return -1;
@@ -343,8 +337,10 @@
/* from internal format */
size = upMix(size);
+
unsigned long wr_ts = user_ts;
- // unsigned int wr_ts =( (long)user_ts * (long)fmt->rate /
(long)SYSTEM_SAMPLERATE);
+
+ // wr_ts =( (long)user_ts * (long)fmt->rate / (long)SYSTEM_SAMPLERATE);
if (fmt->rate != SYSTEM_SAMPLERATE) {
if (fmt->rate > SYSTEM_SAMPLERATE) {
unsigned int f = fmt->rate / SYSTEM_SAMPLERATE;
@@ -354,15 +350,12 @@
wr_ts = wr_ts / f;
}
}
- // DBG("wr_ts = %ld\n", wr_ts);
int s = encode(size);
- if(s>0){
- //DBG("%s\n",typeid(this).name());
+ if (s>0) {
incRecordTime(bytes2samples(size));
return write(wr_ts,(unsigned int)s);
- }
- else{
+ } else {
return s;
}
}
@@ -373,7 +366,7 @@
short* end = (short*)(in_buf + size);
short* out = (short*)out_buf;
- while(in != end){
+ while (in != end) {
*(out++) = (*in + *(in+1)) / 2;
in += 2;
}
@@ -399,10 +392,10 @@
// return -1;
// }
- if(codec->decode){
+ if (codec->decode) {
s = (*codec->decode)(samples.back_buffer(),samples,s,
fmt->channels,fmt->rate,h_codec);
- if(s<0) return s;
+ if (s<0) return s;
samples.swap();
}
@@ -422,7 +415,7 @@
amci_codec_t* codec = fmt->getCodec();
long h_codec = fmt->getHCodec();
- if(codec->encode){
+ if (codec->encode) {
s = (*codec->encode)(samples.back_buffer(),samples,(unsigned int) size,
fmt->channels,fmt->rate,h_codec);
if(s<0) return s;
@@ -435,14 +428,14 @@
unsigned int AmAudio::downMix(unsigned int size)
{
unsigned int s = size;
- if(fmt->channels == 2){
+
+ if (fmt->channels == 2) {
stereo2mono(samples.back_buffer(),(unsigned char*)samples,s);
samples.swap();
}
#ifdef USE_LIBSAMPLERATE
if (fmt->rate != SYSTEM_SAMPLERATE) {
-// DBG("downmix %d->%d\n", SYSTEM_SAMPLERATE, fmt->rate);
s = resample(src_state_in, (double)SYSTEM_SAMPLERATE / (double)fmt->rate,
size);
}
#endif
@@ -453,14 +446,10 @@
unsigned int AmAudio::upMix(unsigned int size)
{
unsigned int s = size;
-// if(fmt->channels == 2){
-// stereo2mono(samples.back_buffer(),(unsigned char*)samples,s);
-// samples.swap();
-// }
+ // todo: system_channels, convert from/to stereo ?
#ifdef USE_LIBSAMPLERATE
if (fmt->rate != SYSTEM_SAMPLERATE) {
-// DBG("upmix %d->%d\n", fmt->rate,SYSTEM_SAMPLERATE);
s = resample(src_state_out, (double)fmt->rate / (double)SYSTEM_SAMPLERATE,
size);
}
#endif
@@ -470,7 +459,8 @@
#ifdef USE_LIBSAMPLERATE
-unsigned int AmAudio::resample(src_state* m_src_state, double factor, unsigned
int size) {
+unsigned int AmAudio::resample(src_state* m_src_state, double factor, unsigned
int size)
+{
if (!m_src_state)
return size;
@@ -526,13 +516,20 @@
unsigned int AmAudio::getFrameSize()
{
-
if (!fmt.get())
fmt.reset(new AmAudioSimpleFormat(CODEC_PCM16));
return fmt->frame_size;
}
+unsigned int AmAudio::getFrameLength()
+{
+ if (!fmt.get())
+ fmt.reset(new AmAudioSimpleFormat(CODEC_PCM16));
+
+ return fmt->frame_length;
+}
+
unsigned int AmAudio::calcBytesToRead(unsigned int nb_samples) const
{
return fmt->calcBytesToRead(nb_samples);
Modified: branches/wb/core/AmAudio.h
===================================================================
--- branches/wb/core/AmAudio.h 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/AmAudio.h 2008-08-25 20:55:09 UTC (rev 1077)
@@ -111,12 +111,10 @@
int channels;
/** Sampling rate. */
int rate;
- /* frame length in ms (frame based codecs) - unused */
+ /* frame length in ms (frame based codecs) */
int frame_length;
/* frame size in samples */
int frame_size;
- /* encoded frame size in bytes - unused */
- int frame_encoded_size;
string sdp_format_parameters;
@@ -300,13 +298,13 @@
virtual void close();
/**
- * Get some samples from input stream.
+ * Get some audio from input stream.
* @warning For packet based payloads / file formats, use:
* <pre> nb_sample = input buffer size / sample size of the
reference format
* </pre> whereby the format with/from which the codec works is
the reference one.
* @return # bytes read, else -1 if error (0 is OK)
*/
- virtual int get(unsigned int user_ts, unsigned char* buffer, unsigned int
nb_samples);
+ virtual int get(unsigned int user_ts, unsigned char* buffer, unsigned int
time_millisec);
/**
* Put some samples to the output stream.
@@ -317,8 +315,12 @@
*/
virtual int put(unsigned int user_ts, unsigned char* buffer, unsigned int
size);
+ /** get frame length in samples */
unsigned int getFrameSize();
+ /** get frame length in ms */
+ unsigned int getFrameLength();
+
void setRecordTime(unsigned int ms);
int incRecordTime(unsigned int samples);
Modified: branches/wb/core/AmConferenceChannel.cpp
===================================================================
--- branches/wb/core/AmConferenceChannel.cpp 2008-08-25 15:59:24 UTC (rev
1076)
+++ branches/wb/core/AmConferenceChannel.cpp 2008-08-25 20:55:09 UTC (rev
1077)
@@ -20,9 +20,10 @@
return size;
}
-int AmConferenceChannel::get(unsigned int user_ts, unsigned char* buffer,
unsigned int nb_samples)
+int AmConferenceChannel::get(unsigned int user_ts, unsigned char* buffer,
unsigned int time_millisec)
{
- unsigned int size = PCM16_S2B(nb_samples);
+ int size = PCM16_S2B(time_millisec * fmt->rate / 1000);
status->getMixer()->GetChannelPacket(channel_id,user_ts,buffer,size);
+
return size;
}
Modified: branches/wb/core/AmConferenceChannel.h
===================================================================
--- branches/wb/core/AmConferenceChannel.h 2008-08-25 15:59:24 UTC (rev
1076)
+++ branches/wb/core/AmConferenceChannel.h 2008-08-25 20:55:09 UTC (rev
1077)
@@ -51,7 +51,7 @@
int write(unsigned int user_ts, unsigned int size){ return -1; }
// override AmAudio
- int get(unsigned int user_ts, unsigned char* buffer, unsigned int
nb_samples);
+ int get(unsigned int user_ts, unsigned char* buffer, unsigned int
time_millisec);
int put(unsigned int user_ts, unsigned char* buffer, unsigned int size);
public:
Modified: branches/wb/core/AmMediaProcessor.cpp
===================================================================
--- branches/wb/core/AmMediaProcessor.cpp 2008-08-25 15:59:24 UTC (rev
1076)
+++ branches/wb/core/AmMediaProcessor.cpp 2008-08-25 20:55:09 UTC (rev
1077)
@@ -239,10 +239,10 @@
AmSession* s = (*it);
// todo: get frame size/checkInterval from local audio if local in+out (?)
- unsigned int f_size = s->rtp_str.getFrameSize();
+ unsigned int frame_length = s->rtp_str.getFrameLength(); // ms
// complete frame time reached?
- if (s->rtp_str.checkInterval(ts, f_size)) {
+ if (s->rtp_str.checkInterval(ts)) {
s->lockAudio();
int rcvd_audio_len = -1;
@@ -272,7 +272,7 @@
break;
}
} else {
- rcvd_audio_len = s->rtp_str.get(ts,buffer,f_size);
+ rcvd_audio_len = s->rtp_str.get(ts,buffer,frame_length);
if (s->isDtmfDetectionEnabled() && rcvd_audio_len > 0)
s->putDtmfAudio(buffer, rcvd_audio_len, ts);
@@ -282,7 +282,7 @@
// input is local - get audio from local_in
AmAudio* local_input = s->getLocalInput();
if (local_input) {
- rcvd_audio_len = local_input->get(ts,buffer,f_size);
+ rcvd_audio_len = local_input->get(ts,buffer,frame_length);
}
}
@@ -311,9 +311,9 @@
AmAudio* output = s->getOutput();
if(output && s->rtp_str.sendIntReached()){
-
- int size = output->get(ts,buffer,s->rtp_str.getFrameSize());
-
+ unsigned int frame_length = s->rtp_str.getFrameLength();
+ int size = output->get(ts,buffer,frame_length);
+
if(size <= 0){
DBG("output->get() returned: %i\n",size);
postRequest(new SchedRequest(AmMediaProcessor::ClearSession,s));
Modified: branches/wb/core/AmPlaylist.cpp
===================================================================
--- branches/wb/core/AmPlaylist.cpp 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/AmPlaylist.cpp 2008-08-25 20:55:09 UTC (rev 1077)
@@ -64,7 +64,7 @@
}
}
-int AmPlaylist::get(unsigned int user_ts, unsigned char* buffer, unsigned int
nb_samples)
+int AmPlaylist::get(unsigned int user_ts, unsigned char* buffer, unsigned int
time_millisec)
{
int ret = -1;
@@ -73,13 +73,14 @@
while(cur_item &&
cur_item->play &&
- (ret = cur_item->play->get(user_ts,buffer,nb_samples)) <= 0){
+ (ret = cur_item->play->get(user_ts,buffer,time_millisec)) <= 0){
DBG("get: gotoNextItem\n");
gotoNextItem();
}
if(!cur_item || !cur_item->play) {
+ unsigned int nb_samples = time_millisec * fmt->rate / 1000;
ret = calcBytesToRead(nb_samples);
memset(buffer,0,ret);
}
Modified: branches/wb/core/AmPlaylist.h
===================================================================
--- branches/wb/core/AmPlaylist.h 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/AmPlaylist.h 2008-08-25 20:55:09 UTC (rev 1077)
@@ -76,7 +76,7 @@
int write(unsigned int user_ts, unsigned int size){ return -1; }
// override AmAudio
- int get(unsigned int user_ts, unsigned char* buffer, unsigned int
nb_samples);
+ int get(unsigned int user_ts, unsigned char* buffer, unsigned int
time_millisec);
int put(unsigned int user_ts, unsigned char* buffer, unsigned int size);
public:
Modified: branches/wb/core/AmPlayoutBuffer.cpp
===================================================================
--- branches/wb/core/AmPlayoutBuffer.cpp 2008-08-25 15:59:24 UTC (rev
1076)
+++ branches/wb/core/AmPlayoutBuffer.cpp 2008-08-25 20:55:09 UTC (rev
1077)
@@ -59,7 +59,6 @@
void AmPlayoutBuffer::write(u_int32_t ref_ts, u_int32_t rtp_ts,
int16_t* buf, u_int32_t len, bool begin_talk)
{
-
unsigned int mapped_ts;
if(!recv_offset_i)
{
@@ -94,6 +93,7 @@
&& (mapped_ts - last_ts <= PLC_MAX_SAMPLES))
{
unsigned char tmp[AUDIO_BUFFER_SIZE * 2];
+ DBG("loss - conceal from last_ts %u to mapped_ts %u\n", last_ts,
mapped_ts);
int l_size = m_plcbuffer->conceal_loss(mapped_ts - last_ts, tmp);
if (l_size>0)
{
@@ -123,7 +123,7 @@
u_int32_t rlen=0;
if(ts_less()(r_ts+PCM16_B2S(AUDIO_BUFFER_SIZE),w_ts))
rlen = PCM16_B2S(AUDIO_BUFFER_SIZE);
- else
+ else
rlen = w_ts - r_ts;
buffer_get(r_ts,buf,rlen);
@@ -140,6 +140,7 @@
if(ts_less()(w_ts,ts+len))
w_ts = ts + len;
+// DBG("@%u put %u\n", ts, len);
}
void AmPlayoutBuffer::buffer_get(unsigned int ts, ShortSample* buf, unsigned
int len)
@@ -148,6 +149,7 @@
if(ts_less()(r_ts,ts+len))
r_ts = ts + len;
+// DBG("@%u get %u\n", ts, len);
}
//
Modified: branches/wb/core/AmPlayoutBuffer.h
===================================================================
--- branches/wb/core/AmPlayoutBuffer.h 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/AmPlayoutBuffer.h 2008-08-25 20:55:09 UTC (rev 1077)
@@ -42,13 +42,13 @@
#define EXP_THRESHOLD 20
#define SHR_THRESHOLD 180
-#define WSOLA_START_OFF 10 * SYSTEM_SAMPLERATE / 1000
+#define WSOLA_START_OFF (10 * SYSTEM_SAMPLERATE / 1000)
#define WSOLA_SCALED_WIN 50
// the maximum packet size that will be processed (80ms)
-#define MAX_PACKET_SAMPLES 80 * SYSTEM_SAMPLERATE / 1000
+#define MAX_PACKET_SAMPLES (80 * SYSTEM_SAMPLERATE / 1000)
// search segments of size TEMPLATE_SEG samples (10 ms)
-#define TEMPLATE_SEG 10 * SYSTEM_SAMPLERATE / 1000
+#define TEMPLATE_SEG (10 * SYSTEM_SAMPLERATE / 1000)
// Maximum value: AUDIO_BUFFER_SIZE / 2
// Note: plc result get stored in our back buffer
Modified: branches/wb/core/AmPrecodedFile.cpp
===================================================================
--- branches/wb/core/AmPrecodedFile.cpp 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/AmPrecodedFile.cpp 2008-08-25 20:55:09 UTC (rev 1077)
@@ -64,7 +64,6 @@
frame_size = precoded_payload.frame_ms * precoded_payload.sample_rate / 1000;
// fill unused stuff
frame_length = precoded_payload.frame_ms;
- frame_encoded_size = precoded_payload.frame_bytes;
}
AmPrecodedRtpFormat::~AmPrecodedRtpFormat() {
Modified: branches/wb/core/AmRtpAudio.cpp
===================================================================
--- branches/wb/core/AmRtpAudio.cpp 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/AmRtpAudio.cpp 2008-08-25 20:55:09 UTC (rev 1077)
@@ -33,7 +33,7 @@
AmRtpAudio::AmRtpAudio(AmSession* _s)
: AmRtpStream(_s), AmAudio(0),
- /*last_ts_i(false),*/ use_default_plc(true),
+ use_default_plc(true),
send_only(false), playout_buffer(new AmPlayoutBuffer(this)),
last_check(0),last_check_i(false),send_int(false)
{
@@ -48,17 +48,31 @@
#endif // USE_SPANDSP_PLC
}
-bool AmRtpAudio::checkInterval(unsigned int ts, unsigned int frame_size)
+bool AmRtpAudio::checkInterval(unsigned int ts)
{
+ unsigned int frame_size = getFrameSize();
+
+ unsigned int used_ts = ts;
+
+ if (fmt->rate != SYSTEM_SAMPLERATE) {
+ if (fmt->rate > SYSTEM_SAMPLERATE) {
+ unsigned int f = fmt->rate / SYSTEM_SAMPLERATE;
+ used_ts = used_ts * f;
+ } else {
+ unsigned int f = SYSTEM_SAMPLERATE / fmt->rate ;
+ used_ts = used_ts / f;
+ }
+ }
+
if(!last_check_i){
send_int = true;
last_check_i = true;
- last_check = ts;
+ last_check = used_ts;
}
else {
- if((ts - last_check) >= frame_size){
+ if((used_ts - last_check) >= frame_size){
send_int = true;
- last_check = ts;
+ last_check = used_ts;
}
else {
send_int = false;
@@ -90,7 +104,7 @@
size = AmRtpStream::receive((unsigned char*)samples,
(unsigned int)AUDIO_BUFFER_SIZE, rtp_ts,
payload);
- // DBG("rtp_ts recvd %u\n", rtp_ts);
+
if(size <= 0)
break;
@@ -114,7 +128,6 @@
/* into internal format */
size = downMix(size);
-
// rtp_ts = SYSTEM_SAMPLERATE * rtp_ts / fmt->rate;
if (fmt->rate && fmt->rate != SYSTEM_SAMPLERATE) {
if (fmt->rate > SYSTEM_SAMPLERATE) {
@@ -132,9 +145,22 @@
return size;
}
-int AmRtpAudio::get(unsigned int ref_ts, unsigned char* buffer, unsigned int
nb_samples)
+int AmRtpAudio::get(unsigned int ref_ts, unsigned char* buffer, unsigned int
time_millisec)
{
+ // for resampling before playout buffer (buffer has SYSTEM_SAMPLERATE),
+ // otherwise fmt->rate
+ unsigned int nb_samples = time_millisec * SYSTEM_SAMPLERATE / 1000;
+
int size = read(ref_ts,PCM16_S2B(nb_samples));
+
+ if (size < 0)
+ return size;
+
+ // /* convert here for sampling in internal format after playout buffer */
+ // size = downMix(size);
+ // if (size < 0)
+ // return size;
+
memcpy(buffer,(unsigned char*)samples,size);
return size;
}
@@ -146,7 +172,9 @@
->read(ref_ts,
(ShortSample*)((unsigned char*)samples),
PCM16_B2S(size));
+// DBG("read ref/@%u want %u got %u samples\n", ref_ts, PCM16_B2S(size),
rlen);
+
return PCM16_S2B(rlen);
}
Modified: branches/wb/core/AmRtpAudio.h
===================================================================
--- branches/wb/core/AmRtpAudio.h 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/AmRtpAudio.h 2008-08-25 20:55:09 UTC (rev 1077)
@@ -96,7 +96,7 @@
AmRtpAudio(AmSession* _s=0);
~AmRtpAudio();
- bool checkInterval(unsigned int ts, unsigned int frame_size);
+ bool checkInterval(unsigned int ts);
bool sendIntReached();
int setCurrentPayload(int payload);
@@ -113,7 +113,7 @@
int write(unsigned int user_ts, unsigned int size);
int get(unsigned int user_ts, unsigned char* buffer,
- unsigned int nb_samples);
+ unsigned int time_millisec);
// AmRtpStream interface
void init(const vector<SdpPayload*>& sdp_payloads);
Modified: branches/wb/core/AmRtpStream.cpp
===================================================================
--- branches/wb/core/AmRtpStream.cpp 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/AmRtpStream.cpp 2008-08-25 20:55:09 UTC (rev 1077)
@@ -335,8 +335,11 @@
memcpy(buffer,rp->getData(),rp->getDataSize());
ts = rp->timestamp;
+
+#if SYSTEM_SAMPLERATE >=16000
if (rp->payload == PAYLOAD_ID_G722)
ts*=2; // G722 rate registered as 8khz due to historical error
+#endif
out_payload = rp->payload;
Modified: branches/wb/core/amci/amci.h
===================================================================
--- branches/wb/core/amci/amci.h 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/amci/amci.h 2008-08-25 20:55:09 UTC (rev 1077)
@@ -67,12 +67,11 @@
/** @def AMCI_WRONLY Write only mode. */
#define AMCI_WRONLY 2
+ /* todo: change in ptr to structure owned by codec */
/** @def AMCI_FMT_FRAME_LENGTH frame length in ms (for framed codecs; must
be multiple of 10) see codec_init */
#define AMCI_FMT_FRAME_LENGTH 1
/** @def AMCI_FMT_FRAME_SIZE frame size in samples */
#define AMCI_FMT_FRAME_SIZE 2
- /** @def AMCI_FMT_ENCODED_FRAME_SIZE encoded frame size in bytes */
-#define AMCI_FMT_ENCODED_FRAME_SIZE 3
struct amci_codec_t;
Modified: branches/wb/core/plug-in/adpcm/adpcm.c
===================================================================
--- branches/wb/core/plug-in/adpcm/adpcm.c 2008-08-25 15:59:24 UTC (rev
1076)
+++ branches/wb/core/plug-in/adpcm/adpcm.c 2008-08-25 20:55:09 UTC (rev
1077)
@@ -116,7 +116,14 @@
g72x_init_state(&cinst->to_g726);
g72x_init_state(&cinst->from_g726);
- format_description[0].id = 0;
+ format_description[0].id = AMCI_FMT_FRAME_LENGTH;
+ format_description[0].value = 20;
+ format_description[1].id = AMCI_FMT_FRAME_SIZE;
+ format_description[1].value = 160;
+/* format_description[2].id = AMCI_FMT_ENCODED_FRAME_SIZE; */
+/* format_description[2].value = 160; */
+ format_description[2].id = 0;
+
return (long) cinst;
}
Modified: branches/wb/core/plug-in/g722/g722.c
===================================================================
--- branches/wb/core/plug-in/g722/g722.c 2008-08-25 15:59:24 UTC (rev
1076)
+++ branches/wb/core/plug-in/g722/g722.c 2008-08-25 20:55:09 UTC (rev
1077)
@@ -1,7 +1,7 @@
/*
This is a simple interface to the spandsp's g722 implementation.
- This uses the 8khz compatibility mode - audio is encodec and decoded
- in 8khz.
+ In narrowband this uses the 8khz compatibility mode - audio is
+ encoded and decoded in 8khz.
Copyright (C) 2008 iptego GmbH
@@ -65,7 +65,7 @@
END_FILE_FORMATS
END_EXPORTS
-
+
typedef struct {
g722_encode_state_t encode_state;
g722_decode_state_t decode_state;
@@ -108,6 +108,16 @@
return 0;
}
+ format_description[0].id = AMCI_FMT_FRAME_LENGTH;
+ format_description[0].value = 20;
+ format_description[1].id = AMCI_FMT_FRAME_SIZE;
+#if SYSTEM_SAMPLERATE >=16000
+ format_description[1].value = 320; /* 20 ms at 16khz */
+#else
+ format_description[1].value = 160;
+#endif
+ format_description[2].id = 0;
+
return (long)gs;
}
Modified: branches/wb/core/plug-in/gsm/gsm.c
===================================================================
--- branches/wb/core/plug-in/gsm/gsm.c 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/plug-in/gsm/gsm.c 2008-08-25 20:55:09 UTC (rev 1077)
@@ -83,8 +83,8 @@
blocks = div(size,320);
if(blocks.rem){
- ERROR("pcm16_2_gsm: number of blocks should be integral (block size =
320)\n");
- return -1;
+ ERROR("pcm16_2_gsm: number of blocks should be integral (block size = 320,
got %u)\n", size);
+ // return -1;
}
for (i=0;i<blocks.quot;i++)
@@ -143,10 +143,9 @@
format_description[0].value = 20;
format_description[1].id = AMCI_FMT_FRAME_SIZE;
format_description[1].value = 160;
- format_description[2].id = AMCI_FMT_ENCODED_FRAME_SIZE;
- format_description[2].value = 33;
- format_description[3].id = 0;
-
+/* format_description[2].id = AMCI_FMT_ENCODED_FRAME_SIZE; */
+/* format_description[2].value = 33; */
+ format_description[2].id = 0;
return (long)h_codec;
}
Modified: branches/wb/core/plug-in/ilbc/ilbc.c
===================================================================
--- branches/wb/core/plug-in/ilbc/ilbc.c 2008-08-25 15:59:24 UTC (rev
1076)
+++ branches/wb/core/plug-in/ilbc/ilbc.c 2008-08-25 20:55:09 UTC (rev
1077)
@@ -153,9 +153,9 @@
format_description[0].value = mode;
format_description[1].id = AMCI_FMT_FRAME_SIZE;
format_description[1].value = mode==30 ? 240 : 160;
- format_description[2].id = AMCI_FMT_ENCODED_FRAME_SIZE;
- format_description[2].value = mode==30 ? 50 : 38;
- format_description[3].id = 0;
+/* format_description[2].id = AMCI_FMT_ENCODED_FRAME_SIZE; */
+/* format_description[2].value = mode==30 ? 50 : 38; */
+ format_description[2].id = 0;
if (format_parameters) {
DBG("ilbc with format parameters : '%s', mode=%d.\n", format_parameters,
mode);
Modified: branches/wb/core/plug-in/speex/speex.c
===================================================================
--- branches/wb/core/plug-in/speex/speex.c 2008-08-25 15:59:24 UTC (rev
1076)
+++ branches/wb/core/plug-in/speex/speex.c 2008-08-25 20:55:09 UTC (rev
1077)
@@ -233,16 +233,17 @@
format_description[1].id = AMCI_FMT_FRAME_SIZE;
format_description[1].value = SPEEX_NB_SAMPLES_PER_FRAME *
ss->frames_per_packet;
- format_description[2].id = AMCI_FMT_ENCODED_FRAME_SIZE;
- format_description[2].value = bits/8 + (bits % 8 ? 1 : 0) + 1;
+/* format_description[2].id = AMCI_FMT_ENCODED_FRAME_SIZE; */
+/* format_description[2].value = bits/8 + (bits % 8 ? 1 : 0) + 1; */
DBG("set AMCI_FMT_FRAME_LENGTH to %d\n", format_description[0].value);
DBG("set AMCI_FMT_FRAME_SIZE to %d\n", format_description[1].value);
- DBG("set AMCI_FMT_ENCODED_FRAME_SIZE to %d\n", format_description[2].value);
+/* DBG("set AMCI_FMT_ENCODED_FRAME_SIZE to %d\n",
format_description[2].value); */
- format_description[3].id = 0;
+ format_description[2].id = 0;
- DBG("SpeexState %p inserted with mode %d and %d frames per packet,\n", ss,
ss->mode, ss->frames_per_packet);
+ DBG("SpeexState %p inserted with mode %d and %d frames per packet,\n",
+ ss, ss->mode, ss->frames_per_packet);
return (long)ss;
}
Modified: branches/wb/core/plug-in/wav/wav.c
===================================================================
--- branches/wb/core/plug-in/wav/wav.c 2008-08-25 15:59:24 UTC (rev 1076)
+++ branches/wb/core/plug-in/wav/wav.c 2008-08-25 20:55:09 UTC (rev 1077)
@@ -77,6 +77,8 @@
/** @def WAV_ULAW subtype declaration. */
#define WAV_ULAW 7
+static long g711_create(const char* format_parameters, amci_codec_fmt_info_t*
format_description);
+
static int ULaw_2_Pcm16( unsigned char* out_buf, unsigned char* in_buf,
unsigned int size,
unsigned int channels, unsigned int rate, long h_codec
);
@@ -96,10 +98,10 @@
BEGIN_CODECS
CODEC( CODEC_ULAW, Pcm16_2_ULaw, ULaw_2_Pcm16,
- AMCI_NO_CODEC_PLC, AMCI_NO_CODECCREATE, AMCI_NO_CODECDESTROY,
+ AMCI_NO_CODEC_PLC, g711_create, AMCI_NO_CODECDESTROY,
g711_bytes2samples, g711_samples2bytes )
CODEC( CODEC_ALAW, Pcm16_2_ALaw, ALaw_2_Pcm16,
- AMCI_NO_CODEC_PLC, AMCI_NO_CODECCREATE, AMCI_NO_CODECDESTROY,
+ AMCI_NO_CODEC_PLC, g711_create, AMCI_NO_CODECDESTROY,
g711_bytes2samples, g711_samples2bytes )
END_CODECS
@@ -120,6 +122,18 @@
END_EXPORTS
+ /* to set frame size 160 */
+static long g711_create(const char* format_parameters, amci_codec_fmt_info_t*
format_description)
+{
+ format_description[0].id = AMCI_FMT_FRAME_LENGTH;
+ format_description[0].value = 20;
+ format_description[1].id = AMCI_FMT_FRAME_SIZE;
+ format_description[1].value = 160;
+ format_description[2].id = 0;
+
+ return 1;
+}
+
static unsigned int g711_bytes2samples(long h_codec, unsigned int num_bytes)
{
/* ALAW and ULAW formats has one sample per byte */
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