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substantial work towards playing audio on IDEOS/G1 type MSM chipset

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android phones directly from C to dramatically reduce latency.
This commit is contained in:
gardners 2012-05-10 03:46:27 +09:30
parent d96d8ff8d6
commit 8d2792ce74
7 changed files with 770 additions and 46 deletions
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12
Android.mk
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@ -67,7 +67,10 @@ LOCAL_SRC_FILES:= \
serval-dna/vomp.c \
serval-dna/lsif.c \
serval-dna/monitor.c \
serval-dna/monitor-cli.c
serval-dna/monitor-cli.c \
serval-dna/codecs.c \
serval-dna/audiodevices.c \
serval-dna/audio_msm_g1.c
LOCAL_MODULE:= serval
@ -95,3 +98,10 @@ LOCAL_MODULE:= servald
LOCAL_SRC_FILES:= serval-dna/servalwrap.c
include $(BUILD_EXECUTABLE)
include $(CLEAR_VARS)
LOCAL_MODULE:= playwav
LOCAL_SRC_FILES:= serval-dna/playwav.c
include $(BUILD_EXECUTABLE)

5
Makefile.in
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@ -40,7 +40,10 @@ SRCS= batman.c \
vomp.c \
lsif.c \
monitor.c \
monitor-cli.c
monitor-cli.c \
codecs.c \
audiodevices.c \
audio_msm_g1.c
HAVE_VOIPTEST= @HAVE_VOIPTEST@
ifeq ($(HAVE_VOIPTEST), 1)

396
audio_msm_g1.c Normal file
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@ -0,0 +1,396 @@
/*
Copyright (C) 2012 Paul Gardner-Stephen
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
Contains code derived from playwav2.c, which has the following notice:
Copyright (C) 2008 The Android Open Source Project
*/
int setReason(char *fmt, ...);
#define WHY(X) setReason("%s:%d:%s() %s",__FILE__,__LINE__,__FUNCTION__,X)
#define WHYRETNULL(X) { setReason("%s:%d:%s() %s",__FILE__,__LINE__,__FUNCTION__,X); return NULL; }
#define WHYF(F, ...) setReason("%s:%d:%s() " F, __FILE__, __LINE__, __FUNCTION__, ##__VA_ARGS__)
#define WHY_perror(X) setReason("%s:%d:%s() %s: %s [errno=%d]", __FILE__, __LINE__, __FUNCTION__, X, strerror(errno), errno)
extern int playFd;
extern int recordFd;
extern int playBufferSize;
extern int recordBufferSize;
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <fcntl.h>
#include <stdint.h>
#include <sys/mman.h>
#include <sys/ioctl.h>
#include <errno.h>
#include <linux/ioctl.h>
#if 0
#include <linux/msm_audio.h>
#else
/* ---------- linux/msm_audio.h -------- */
#define AUDIO_IOCTL_MAGIC 'a'
#define AUDIO_START _IOW(AUDIO_IOCTL_MAGIC, 0, unsigned)
#define AUDIO_STOP _IOW(AUDIO_IOCTL_MAGIC, 1, unsigned)
#define AUDIO_FLUSH _IOW(AUDIO_IOCTL_MAGIC, 2, unsigned)
#define AUDIO_GET_CONFIG _IOR(AUDIO_IOCTL_MAGIC, 3, unsigned)
#define AUDIO_SET_CONFIG _IOW(AUDIO_IOCTL_MAGIC, 4, unsigned)
#define AUDIO_GET_STATS _IOR(AUDIO_IOCTL_MAGIC, 5, unsigned)
#define AUDIO_ENABLE_AUDPP _IOW(AUDIO_IOCTL_MAGIC, 6, unsigned)
#define AUDIO_SET_ADRC _IOW(AUDIO_IOCTL_MAGIC, 7, unsigned)
#define AUDIO_SET_EQ _IOW(AUDIO_IOCTL_MAGIC, 8, unsigned)
#define AUDIO_SET_RX_IIR _IOW(AUDIO_IOCTL_MAGIC, 9, unsigned)
#define EQ_MAX_BAND_NUM 12
#define ADRC_ENABLE 0x0001
#define ADRC_DISABLE 0x0000
#define EQ_ENABLE 0x0002
#define EQ_DISABLE 0x0000
#define IIR_ENABLE 0x0004
#define IIR_DISABLE 0x0000
struct eq_filter_type
{
int16_t gain;
uint16_t freq;
uint16_t type;
uint16_t qf;
};
struct eqalizer
{
uint16_t bands;
uint16_t params[132];
};
struct rx_iir_filter
{
uint16_t num_bands;
uint16_t iir_params[48];
};
struct msm_audio_config
{
uint32_t buffer_size;
uint32_t buffer_count;
uint32_t channel_count;
uint32_t sample_rate;
uint32_t codec_type;
uint32_t unused[3];
};
struct msm_audio_stats
{
uint32_t out_bytes;
uint32_t unused[3];
};
/* Audio routing */
#define SND_IOCTL_MAGIC 's'
#define SND_MUTE_UNMUTED 0
#define SND_MUTE_MUTED 1
struct msm_snd_device_config
{
uint32_t device;
uint32_t ear_mute;
uint32_t mic_mute;
};
#define SND_SET_DEVICE _IOW(SND_IOCTL_MAGIC, 2, struct msm_device_config *)
#define SND_METHOD_VOICE 0
#define SND_METHOD_VOICE_1 1
struct msm_snd_volume_config
{
uint32_t device;
uint32_t method;
uint32_t volume;
};
#define SND_SET_VOLUME _IOW(SND_IOCTL_MAGIC, 3, struct msm_snd_volume_config *)
/* Returns the number of SND endpoints supported. */
#define SND_GET_NUM_ENDPOINTS _IOR(SND_IOCTL_MAGIC, 4, unsigned *)
struct msm_snd_endpoint
{
int id; /* input and output */
char name[64]; /* output only */
};
/* Takes an index between 0 and one less than the number returned by
* SND_GET_NUM_ENDPOINTS, and returns the SND index and name of a
* SND endpoint. On input, the .id field contains the number of the
* endpoint, and on exit it contains the SND index, while .name contains
* the description of the endpoint.
*/
#define SND_GET_ENDPOINT _IOWR(SND_IOCTL_MAGIC, 5, struct msm_snd_endpoint *)
#endif
static int
do_route_audio_rpc (uint32_t device, int ear_mute, int mic_mute)
{
if (device == -1UL)
return 0;
int fd;
printf ("rpc_snd_set_device(%d, %d, %d)\n", device, ear_mute, mic_mute);
fd = open ("/dev/msm_snd", O_RDWR);
if (fd < 0)
{
perror ("Can not open snd device");
return -1;
}
// RPC call to switch audio path
/* rpc_snd_set_device(
* device, # Hardware device enum to use
* ear_mute, # Set mute for outgoing voice audio
* # this should only be unmuted when in-call
* mic_mute, # Set mute for incoming voice audio
* # this should only be unmuted when in-call or
* # recording.
* )
*/
struct msm_snd_device_config args;
args.device = device;
args.ear_mute = ear_mute ? SND_MUTE_MUTED : SND_MUTE_UNMUTED;
args.mic_mute = mic_mute ? SND_MUTE_MUTED : SND_MUTE_UNMUTED;
if (ioctl (fd, SND_SET_DEVICE, &args) < 0)
{
perror ("snd_set_device error.");
close (fd);
return -1;
}
close (fd);
return 0;
}
static int
set_volume_rpc (uint32_t device, uint32_t method, uint32_t volume)
{
int fd;
printf ("rpc_snd_set_volume(%d, %d, %d)\n", device, method, volume);
if (device == -1UL)
return 0;
fd = open ("/dev/msm_snd", O_RDWR);
if (fd < 0)
{
perror ("Can not open snd device");
return -1;
}
/* rpc_snd_set_volume(
* device, # Any hardware device enum, including
* # SND_DEVICE_CURRENT
* method, # must be SND_METHOD_VOICE to do anything useful
* volume, # integer volume level, in range [0,5].
* # note that 0 is audible (not quite muted)
* )
* rpc_snd_set_volume only works for in-call sound volume.
*/
struct msm_snd_volume_config args;
args.device = device;
args.method = method;
args.volume = volume;
if (ioctl (fd, SND_SET_VOLUME, &args) < 0)
{
perror ("snd_set_volume error.");
close (fd);
return -1;
}
close (fd);
return 0;
}
/* See if we can query end-points for this device.
If so, assume we have detected it.
*/
char *audio_msm_g1_detect()
{
int fd = open ("/dev/msm_snd", O_RDWR);
if (fd<0) return NULL;
int endpoints=0;
int rc =ioctl(fd,SND_GET_NUM_ENDPOINTS,&endpoints);
close(fd);
if (rc>0) return "G1/IDEOS style MSM audio";
else return NULL;
}
/* Prepare audio path, volume etc, and then open play and
record file descriptors.
*/
int audio_msm_g1_start_play()
{
/* Get audio control device */
int fd = open ("/dev/msm_snd", O_RDWR);
if (fd<0) return -1;
/* Look through endpoints for the regular in-call endpoint */
int endpoints=0;
int rc =ioctl(fd,SND_GET_NUM_ENDPOINTS,&endpoints);
int endpoint=-1;
int i;
for(i=0;i<endpoints;i++) {
struct msm_snd_endpoint ep;
ep.id=i;
ep.name[0]=0;
ioctl(fd,SND_GET_ENDPOINT,&ep);
if (!strcasecmp(ep.name,"HANDSET"))
/* should this be i, or ep.id ? */
endpoint=i;
}
close(fd);
/* Set the specified endpoint and unmute microphone and speaker */
do_route_audio_rpc(endpoint,SND_MUTE_UNMUTED,SND_MUTE_UNMUTED);
/* Set the volume (somewhat arbitrarily for now) */
int vol=5;
int dev=0xd; /* no one seems to know what this magic value means */
set_volume_rpc(dev,SND_METHOD_VOICE_1, vol);
playFd=open("/dev/msm_pcm_out",O_RDWR);
struct msm_audio_config config;
if (ioctl(playFd, AUDIO_GET_CONFIG,&config))
{
close(playFd);
return WHY("Could not read audio device configuration");
}
config.channel_count=1;
config.sample_rate=8000;
playBufferSize=config.buffer_size;
if (ioctl(playFd, AUDIO_SET_CONFIG,&config))
{
close(playFd);
return WHY("Could not set audio device configuration");
}
/* tell hardware to start playing */
ioctl(playFd,AUDIO_START,0);
WHY("G1/IDEOS style MSM audio device initialised and ready to play");
return 0;
}
int audio_msm_g1_stop_play()
{
if (playFd>-1) close(playFd);
playFd=-1;
return 0;
}
int audio_msm_g1_start_record()
{
recordFd=open("/dev/msm_pcm_out",O_RDWR);
struct msm_audio_config config;
if (ioctl(recordFd, AUDIO_GET_CONFIG,&config))
{
close(recordFd);
return WHY("Could not read audio device configuration");
}
config.channel_count=1;
config.sample_rate=8000;
if (ioctl(recordFd, AUDIO_SET_CONFIG,&config))
{
close(recordFd);
return WHY("Could not set audio device configuration");
}
/*
If recordBufferSize equates to too long an interval,
then try to reduce it in various ways.
*/
recordBufferSize=config.buffer_size;
float bufferTime=recordBufferSize/2*1.0/config.sample_rate;
if (bufferTime>0.02) {
WHYF("REC buf=%.3fsecs, which is too long. Trying to reduce it.");
/* 64 bytes = 32 samples = ~4ms */
config.buffer_size=64;
if (!ioctl(recordFd, AUDIO_SET_CONFIG,&config))
{
recordBufferSize=config.buffer_size;
bufferTime=recordBufferSize/2*1.0/config.sample_rate;
WHYF("Succeeded in reducing record buffer to %d bytes (%.3fsecs)",
recordBufferSize,bufferTime);
goto fixedBufferSize;
}
recordBufferSize=64;
#if 0
/* Ask for 2x speed and 2x channels, to divide effective buffer size by 4.
*/
config.sample_rate=16000;
config.channel_count=2;
#endif
}
fixedBufferSize:
/* tell hardware to start playing */
ioctl(recordFd,AUDIO_START,0);
WHY("G1/IDEOS style MSM audio device initialised and ready to record");
return 0;
}
int audio_msm_g1_stop_record()
{
if (recordFd>-1) close(recordFd);
recordFd=-1;
return 0;
}
int audio_msm_g1_stop()
{
audio_msm_g1_stop_play();
audio_msm_g1_stop_record();
}
int audio_msm_g1_start()
{
if (audio_msm_g1_start_play()) return -1;
if (audio_msm_g1_start_record()) {
audio_msm_g1_stop_play();
return -1;
}
return 0;
}

143
audiodevices.c Normal file
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@ -0,0 +1,143 @@
/*
Copyright (C) 2012 Paul Gardner-Stephen
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/
#include "serval.h"
#define AUDIO_MSM_G1_ETC 1
#define AUDIO_MSM_N1_ETC 2
int detectedAudioDevice=-1;
char *detectedAudioDeviceName=NULL;
int playFd=-1;
int recordFd=-1;
int playBufferSize=0;
int recordBufferSize=0;
int detectAudioDevice()
{
detectedAudioDeviceName=audio_msm_g1_detect();
if (detectedAudioDeviceName) {
detectedAudioDevice=AUDIO_MSM_G1_ETC;
WHYF("Detected audio device '%s'",detectedAudioDeviceName);
return 0;
}
return -1;
}
int getAudioPlayFd()
{
return playFd;
}
int getAudioRecordFd()
{
return recordFd;
}
/* read some audio, but not more than bufferSize-offset bytes.
*/
int getAudioBytes(unsigned char *buffer,
int offset,
int bufferSize)
{
switch(detectedAudioDevice) {
/* some devices require reading a whole buffer in one go */
case AUDIO_MSM_G1_ETC:
{
if (bufferSize-offset<recordBufferSize) {
return WHY("Supplied buffer has no space for new samples");
}
int b=read(recordFd,&buffer[offset],recordBufferSize);
if (b>0) offset+=b;
return offset;
}
break;
/* while others allow reading an arbitrary amount */
default:
{
int b=read(recordFd,&buffer[offset],bufferSize-offset);
if (b>0) offset+=b;
return offset;
}
break;
}
return WHYF("Reading audio for device class #%d not implemented",
detectedAudioDevice);
}
/* as with recording, some of the devices have a fixed buffer size that
we must completely fill.
*/
int playBufferBytes=0;
unsigned char playBuffer[65536];
int playAudio(unsigned char *data,int bytes)
{
switch(detectedAudioDevice) {
/* some devices require reading a whole buffer in one go */
case AUDIO_MSM_G1_ETC:
if (bytes+playBufferBytes>65536)
return WHY("Play marshalling buffer full");
bcopy(&data[0],&playBuffer[playBufferBytes],bytes);
playBufferBytes+=bytes;
int i;
for(i=0;i<playBufferBytes;i+=playBufferSize)
{
if (write(playFd,&playBuffer[i],playBufferSize)<
playBufferSize)
break;
}
bcopy(&playBuffer[i],&playBuffer[0],playBufferBytes-i);
playBufferBytes-=i;
break;
/* the rest we can just write() to */
default:
if (write(playFd,data,bytes)<bytes)
return WHY("short write() when playing audio");
return 0;
}
return WHYF("Playing audio for device class #%d not implemented",
detectedAudioDevice);
}
int stopAudio()
{
switch(detectedAudioDevice) {
case AUDIO_MSM_G1_ETC:
return audio_msm_g1_stop();
break;
default:
break;
}
return WHYF("Stopping audio for device class #%d not implemented",
detectedAudioDevice);
}
int startAudio()
{
switch(detectedAudioDevice) {
case AUDIO_MSM_G1_ETC:
return audio_msm_g1_start();
break;
default:
break;
}
return WHYF("Starting audio for device class #%d not implemented",
detectedAudioDevice);
}

42
codecs.c Normal file
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@ -0,0 +1,42 @@
/*
Copyright (C) 2012 Paul Gardner-Stephen
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/
#include "serval.h"
int encodeAndDispatchRecordedAudio(int fd,int callSessionToken,
int recordCodec,
unsigned char *sampleData,
int sampleBytes)
{
switch (recordCodec) {
case VOMP_CODEC_PCM:
/* leave data raw, so need to rewrite sampleData or sampleBytes */
break;
default:
WHYF("Codec not yet supported");
return -1;
}
char msg[128+MAX_AUDIO_BYTES];
snprintf(msg,128,"\n*%d:AUDIO:%x:%d\n",sampleBytes,callSessionToken,recordCodec);
int len=strlen(msg);
bcopy(&sampleData[0],&msg[len],sampleBytes);
len+=sampleBytes;
write(fd,msg,len);
return 0;
}

196
monitor-cli.c
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@ -1,3 +1,21 @@
/*
Copyright (C) 2012 Paul Gardner-Stephen
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/
#include <sys/socket.h>
#include <sys/un.h>
#include <stdio.h>
@ -28,6 +46,16 @@ int writeLine(char *msg)
int processChar(int c);
int autoAnswerP=1;
int pipeAudio=1;
int syntheticAudio=0;
int showReceived=1;
int interactiveP=1;
int recordCodec=VOMP_CODEC_PCM;
int recordCodecBlockSamples=320;
int recordCodecTimespan=20;
int callSessionToken=0;
int app_monitor_cli(int argc, const char *const *argv, struct command_line_option *o)
{
const char *sid=NULL;
@ -52,15 +80,26 @@ int app_monitor_cli(int argc, const char *const *argv, struct command_line_optio
exit(-1);
}
if (pipeAudio) {
detectAudioDevice();
char *name=detectedAudioDeviceName;
if (!name) {
WHY("Could not detect any audio device. Will not pipe audio.");
pipeAudio=0;
}
}
struct pollfd fds[128];
int fdcount=0;
fds[fdcount].fd=fd;
fds[fdcount].events=POLLIN;
fdcount++;
fds[fdcount].fd=STDIN_FILENO;
fds[fdcount].events=POLLIN;
fdcount++;
if (interactiveP) {
fds[fdcount].fd=STDIN_FILENO;
fds[fdcount].events=POLLIN;
fdcount++;
}
writeLine("monitor vomp\n");
writeLine("monitor rhizome\n");
@ -71,25 +110,76 @@ int app_monitor_cli(int argc, const char *const *argv, struct command_line_optio
writeLine(msg);
}
char line[1024];
/* Allow for up to one second of audio read from the microphone
to be buffered. This is probably more than we will ever need.
The primary purpose of the buffer is in fact to handle the fact
that we are unlikely to ever read exaclty the number of samples
we need, so we need to keep any left over ones from the previous
read. */
int audioRecordBufferBytes=0;
unsigned char audioRecordBuffer[8000*2];
int base_fd_count=fdcount;
while(1) {
fdcount=base_fd_count;
int audio_fd = getAudioRecordFd();
if (audio_fd>-1) {
fds[fdcount].fd=STDIN_FILENO;
fds[fdcount].events=POLLIN;
fdcount++;
}
poll(fds,fdcount,1000);
fcntl(fd,F_SETFL,
fcntl(fd, F_GETFL, NULL)|O_NONBLOCK);
fcntl(STDIN_FILENO,F_SETFL,
fcntl(STDIN_FILENO, F_GETFL, NULL)|O_NONBLOCK);
if (interactiveP)
fcntl(STDIN_FILENO,F_SETFL,
fcntl(STDIN_FILENO, F_GETFL, NULL)|O_NONBLOCK);
if (audio_fd>-1)
fcntl(audio_fd,F_SETFL,
fcntl(audio_fd, F_GETFL, NULL)|O_NONBLOCK);
char line[1024];
int bytes;
int i;
line[0]=0;
bytes=read(fd,line,1024);
if (bytes>0)
for(i=0;i<bytes;i++) processChar(line[i]);
bytes=read(STDIN_FILENO,line,1024);
if (bytes>0) {
line[bytes]=0;
printf("< %s",line);
write(fd,line,bytes);
if (interactiveP) {
bytes=read(STDIN_FILENO,line,1024);
if (bytes>0) {
line[bytes]=0;
printf("< %s",line);
write(fd,line,bytes);
}
if (audio_fd>-1) {
/* attempt to read next audio buffer.
Some audio devices that we are using require an entire buffer
to be read at a time. */
audioRecordBufferBytes
=getAudioBytes(audioRecordBuffer,audioRecordBufferBytes,sizeof(audioRecordBuffer));
/* 8KHz 16 bit samples = 16000 bytes per second.
Thus one 1ms of audio = 16 bytes. */
int audioRecordBufferOffset=0;
while ((audioRecordBufferBytes-audioRecordBufferOffset)
>recordCodecTimespan*16) {
/* encode and deliver audio block to servald via monitor interface */
encodeAndDispatchRecordedAudio(fd,callSessionToken,recordCodec,
&audioRecordBuffer[audioRecordBufferOffset],
recordCodecTimespan*16);
/* skip over the samples we have already processed */
audioRecordBufferOffset+=recordCodecTimespan*16;
}
/* copy the remaining buffered bytes down and correct buffer length */
if (audioRecordBufferOffset<0) audioRecordBufferOffset=0;
if (audioRecordBufferOffset>audioRecordBufferBytes)
audioRecordBufferOffset=audioRecordBufferBytes;
bcopy(&audioRecordBuffer[audioRecordBufferOffset],
&audioRecordBuffer[0],
audioRecordBufferBytes-audioRecordBufferOffset);
audioRecordBufferBytes-=audioRecordBufferOffset;
}
}
fcntl(fd,F_SETFL,
@ -106,52 +196,70 @@ int counter=0;
int callState=0;
int processLine(char *cmd,unsigned char *data,int dataLen)
{
int l_id,r_id,l_state,r_state;
printf("> %s\n",cmd);
if (data) {
int i,j;
for(i=0;i<dataLen;i+=16) {
printf(" %04x :",i);
for(j=0;j<16;j++)
if (i+j<dataLen) printf(" %02x",data[i+j]); else printf(" ");
printf(" ");
for(j=0;j<16;j++)
if (i+j<dataLen) {
if (data[i+j]>=0x20&&data[i+j]<0x7e)
printf("%c",data[i+j]); else printf(".");
}
printf("\n");
int l_id,r_id,l_state,r_state,codec;
long long start_time,end_time;
if (showReceived) {
printf("> %s\n",cmd);
if (data) {
int i,j;
for(i=0;i<dataLen;i+=16) {
printf(" %04x :",i);
for(j=0;j<16;j++)
if (i+j<dataLen) printf(" %02x",data[i+j]); else printf(" ");
printf(" ");
for(j=0;j<16;j++)
if (i+j<dataLen) {
if (data[i+j]>=0x20&&data[i+j]<0x7e)
printf("%c",data[i+j]); else printf(".");
}
printf("\n");
}
}
}
if (sscanf(cmd,"AUDIOPACKET:%x:%x:%d:%d:%d:%lld:%lld",
&l_id,&r_id,&l_state,&r_state,
&codec,&start_time,&end_time)==7)
{
if (pipeAudio) {
playAudio(data,dataLen);
}
}
if (sscanf(cmd,"CALLSTATUS:%x:%x:%d:%d",
&l_id,&r_id,&l_state,&r_state)==4)
{
if (l_state==4) {
if (l_state==4&&autoAnswerP) {
// We are ringing, so pickup
char msg[1024];
sprintf(msg,"pickup %x\n",l_id);
writeLine(msg);
}
if (l_state==5) {
startAudio();
callSessionToken=l_id;
} else {
stopAudio();
callSessionToken=0;
}
callState=l_state;
}
if (sscanf(cmd,"KEEPALIVE:%x",&l_id)==1) {
if (callState==5) {
/* Send synthetic audio packet */
char buffer[1024];
sprintf(buffer,"*320:AUDIO:%x:8\n"
"%08d pasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456",l_id,counter++);
writeLine(buffer);
printf("< *320:AUDIO:%x:8\\n<320 bytes>\n",l_id);
}
if (callState==5&&syntheticAudio) {
/* Send synthetic audio packet */
char buffer[1024];
sprintf(buffer,"*320:AUDIO:%x:8\n"
"%08d pasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456"
"qwertyuiopasdfghjklzxcvbnm123456",l_id,counter++);
writeLine(buffer);
printf("< *320:AUDIO:%x:8\\n<320 bytes>\n",l_id);
}
}
cmd[0]=0;
cmdLen=0;

22
serval.h
View File

@ -1407,3 +1407,25 @@ int monitor_get_fds(struct pollfd *fds,int *fdcount,int fdmax);
int monitor_call_status(vomp_call_state *call);
int monitor_send_audio(vomp_call_state *call,overlay_mdp_frame *audio);
extern int monitor_socket_count;
#define AUDIO_MSM_G1_ETC 1
#define AUDIO_MSM_N1_ETC 2
extern int detectedAudioDevice;
extern char *detectedAudioDeviceName;
int detectAudioDevice();
int getAudioPlayFd();
int getAudioRecordFd();
int getAudioFd();
int getAudioBytes(unsigned char *buffer,
int offset,
int bufferSize);
int playAudio(unsigned char *data,int bytes);
int stopAudio();
int startAudio();
int encodeAndDispatchRecordedAudio(int fd,int callSessionToken,
int recordCodec,
unsigned char *sampleData,
int sampleBytes);
char *audio_msm_g1_detect();
int audio_msm_g1_start();
int audio_msm_g1_stop();