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fixed audio handling for the most part.

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aliasing noise and/or buffer underrun noise is present.
This commit is contained in:
gardners 2012-05-11 14:11:58 +09:30
parent 2ea218b567
commit c448eab720
1 changed files with 89 additions and 74 deletions
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163
audio_msm_g1.c
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@ -31,12 +31,16 @@
We may need to introduce a low-pass filter to prevent aliasing, assuming that
the microphone and ACD in these phones responds to requencies above 4KHz.
Added fun with this device is that we must read/write exactly one buffer full
at a time.
*/
#define DESIRED_BUFFER_SIZE 256
#define DESIRED_SAMPLE_RATE 32000
#define RESAMPLE_FACTOR (DESIRED_SAMPLE_RATE/8000)
int resamplingBufferSize=0;
unsigned char *resamplingBuffer=0;
unsigned char *playMarshallBuffer=0;
unsigned char *recordMarshallBuffer=0;
extern int playFd;
extern int recordFd;
@ -259,13 +263,6 @@ int audio_msm_g1_start_play()
{
if (playFd>-1) return 0;
/* Largest possible resampling buffer required for this chipset.
(we resample so that we can use 8KHz audio for transport,
but 32KHz sample rate on the audio hardware so that buffers
correspond to a shorter period of time */
resamplingBufferSize=2*RESAMPLE_FACTOR*8192;
resamplingBuffer=malloc(resamplingBufferSize);
/* Get audio control device */
int fd = open ("/dev/msm_snd", O_RDWR);
if (fd<0) return -1;
@ -324,6 +321,8 @@ int audio_msm_g1_start_play()
float bufferTime=playBufferSize/2*1.0/config.sample_rate;
WHYF("PLAY buf=%.3fsecs.",bufferTime);
playMarshallBuffer=malloc(playBufferSize);
/* tell hardware to start playing */
ioctl(playFd,AUDIO_START,0);
@ -336,7 +335,8 @@ int audio_msm_g1_stop_play()
{
WHY("stopping audio play");
if (playFd>-1) close(playFd);
playFd=-1;
if (playMarshallBuffer) free(playMarshallBuffer);
playFd=-1; playMarshallBuffer=NULL;
return 0;
}
@ -371,6 +371,9 @@ int audio_msm_g1_start_record()
float bufferTime=recordBufferSize/2*1.0/config.sample_rate;
WHYF("REC buf=%.3fsecs.",bufferTime);
if (!recordMarshallBuffer)
recordMarshallBuffer=malloc(recordBufferSize);
fcntl(recordFd,F_SETFL,
fcntl(recordFd, F_GETFL, NULL)|O_NONBLOCK);
@ -385,6 +388,8 @@ int audio_msm_g1_stop_record()
{
WHY("stopping recording");
if (recordFd>-1) close(recordFd);
if (recordMarshallBuffer) free(recordMarshallBuffer);
recordMarshallBuffer=NULL;
recordFd=-1;
return 0;
}
@ -417,50 +422,50 @@ int audio_msm_g1_poll_fds(struct pollfd *fds,int slots)
return count;
}
int recordMarshallBufferOffset=0;
int audio_msm_g1_read(unsigned char *buffer,int maximum_count)
{
if (recordFd==-1) return 0;
if (!resamplingBuffer) return 0;
if (!recordMarshallBuffer) return 0;
int maxRawBytes=maximum_count*RESAMPLE_FACTOR;
if (maxRawBytes>resamplingBufferSize)
maxRawBytes=resamplingBufferSize;
if (maxRawBytes>recordBufferSize)
maxRawBytes=recordBufferSize;
int supplied=0;
fcntl(recordFd,F_SETFL,fcntl(recordFd, F_GETFL, NULL)|O_NONBLOCK);
ioctl(recordFd,AUDIO_START,0);
WHY("calling read()");
/* read new samples if we don't have any lingering around */
if (!recordMarshallBufferOffset) {
fcntl(recordFd,F_SETFL,fcntl(recordFd, F_GETFL, NULL)|O_NONBLOCK);
ioctl(recordFd,AUDIO_START,0);
WHY("calling read()");
int b=read(recordFd,&recordMarshallBuffer[0],recordBufferSize);
if (b<1)
WHYF("read failed: b=%d, err=%s",b,strerror(errno));
if (errno==EBADF) recordFd=-1;
WHYF("read %d raw (upsampled) bytes",b);
recordMarshallBufferOffset=b;
}
/* read raw samples */
int b=read(recordFd,&resamplingBuffer[0],maxRawBytes);
if (b<1)
WHYF("read failed: b=%d, err=%s",b,strerror(errno));
if (errno==EBADF) recordFd=-1;
WHYF("read %d raw (upsampled) bytes",b);
/* downsample to output buffer */
{
int i;
/* copy every RESAMPLE_FACTOR-th sample (each being 16 bits)
to output buffer */
int outpos=0;
for(i=0;i<b;i+=(2*RESAMPLE_FACTOR))
{
buffer[outpos++]=resamplingBuffer[i];
buffer[outpos++]=resamplingBuffer[i+1];
}
b/=RESAMPLE_FACTOR;
/* supply audio from marshalling buffer if it has anything.
Don't forget to downsample first. */
int marshall_offset=0;
while(marshall_offset<recordMarshallBufferOffset
&&supplied<maximum_count) {
buffer[supplied+0]=recordMarshallBuffer[marshall_offset];
buffer[supplied+1]=recordMarshallBuffer[marshall_offset+1];
supplied+=2;
marshall_offset+=2*RESAMPLE_FACTOR;
}
WHYF("Read %d samples.",b/2);
bcopy(&recordMarshallBuffer[marshall_offset],
&recordMarshallBuffer[0],
recordMarshallBufferOffset-marshall_offset);
recordMarshallBufferOffset-=marshall_offset;
/* Else we read exactly one buffer full into the marshalling buffer */
#warning for debug
WHYF("Echoing %d bytes of 8KHz audio",b);
audio_msm_g1_write(buffer,b);
WHYF("Read %d samples.",supplied/2);
return b;
return supplied;
}
int playMarshallBufferOffset=0;
int audio_msm_g1_write(unsigned char *data,int bytes)
{
if (playFd==-1) return 0;
@ -468,41 +473,51 @@ int audio_msm_g1_write(unsigned char *data,int bytes)
WHYF("Writing %d bytes of 8KHz audio",bytes);
int maxBytes=bytes;
if (maxBytes*RESAMPLE_FACTOR>resamplingBufferSize)
maxBytes=resamplingBufferSize/RESAMPLE_FACTOR;
int i,played=0;
/* upsample ready for play back.
XXX This is a really crude approach, and it could be done much better. */
int i,j,outpos=0;
for(i=0;i<bytes;i+=2)
for(j=0;j<RESAMPLE_FACTOR;j++) {
resamplingBuffer[outpos++]=data[i];
resamplingBuffer[outpos++]=data[i+1];
}
WHYF("Writing %d bytes of upsampled audio",outpos);
struct msm_audio_stats stats;
if (ioctl (playFd, AUDIO_GET_STATS, &stats) == 0)
WHYF("stats.out_bytes = %10d", stats.out_bytes);
/* even if set non-blocking the following write can block
if we don't call this ioctl first */
ioctl(playFd,AUDIO_START,0);
int w=write(playFd,&resamplingBuffer[0],outpos);
w/=RESAMPLE_FACTOR;
if (w<1)
while(played<bytes)
{
WHYF("Failed to write, returned %d (errno=%s)",
w,strerror(errno));
if (errno==EBADF) playFd=-1;
} else {
WHYF("Wrote %d bytes of audio",w);
i+=w;
}
if (playMarshallBufferOffset==playBufferSize) {
/* we have a buffer full of samples, so play it */
struct msm_audio_stats stats;
if (ioctl (playFd, AUDIO_GET_STATS, &stats) == 0)
WHYF("stats.out_bytes = %10d", stats.out_bytes);
/* even if set non-blocking the following write can block
if we don't call this ioctl first */
ioctl(playFd,AUDIO_START,0);
int w=write(playFd,&playMarshallBuffer[0],playBufferSize);
if (w<1)
{
WHYF("Failed to write, returned %d (errno=%s)",
w,strerror(errno));
if (errno==EBADF) playFd=-1;
} else {
if (w<=playBufferSize) {
/* short write, so update buffer status and inform caller */
bcopy(&playMarshallBuffer[w],&playMarshallBuffer[0],
playBufferSize-w);
playMarshallBufferOffset-=w;
WHYF("short write: %d of %d raw bytes written",
w,playBufferSize);
return w/RESAMPLE_FACTOR;
}
}
playMarshallBufferOffset=0;
}
WHY("done writing");
return bytes;
/* upsample for playing back */
for(i=0;i<RESAMPLE_FACTOR;i++) {
playMarshallBuffer[playMarshallBufferOffset++]
=data[played];
playMarshallBuffer[playMarshallBufferOffset++]
=data[played+1];
}
played+=2;
}
WHYF("done writing %d audio bytes",played);
return played;
}
/* See if we can query end-points for this device.