[SKYPEKIT][DM368][IPNC] Audio PCM/ALSA Experiment on DM368-IPNC,

The previous trial on skypekit audio ([SKYPEKIT][DM368][IPNC] Audio PCM Experiment on DM368-IPNC) was using OSS (/dev/dsp) for Audio PCM host input and output voice. The major cons are that it's lack of powerful helper functions, i.e. API, which might lead to delay.

After trying to implement Voice RTP host (a nightmare since DM368 is lack of hardware G.729 encode/decode support. Even if using G.711, the CPU usage is not significant less compared to deploying Audio PCM host), we decided to use ALSA instead of OSS.


The code to start with is still PCMHost loopback. The steps are:

1. Remove buffer manipulation codes

Similar to what we did in OSS, we don't need loopback_buf anymore, as well as it's Pull() and Push() methods. It's safe to remove those codes in Run()@PCMHostLoopback.c. It can save you some CPU time.

2. Open ALSA device for voice input and output

I decided to put the initialization in Start():PCMLoopback. If the deviceType is INPUT_DEVICE, the code would be something like:

int rc, dir = 0;
unsigned int val;
snd_pcm_hw_params_t *hw_params = NULL;
snd_pcm_uframes_t uframe;
if(capture_handle == 0)
{
rc = snd_pcm_open(&capture_handle, "default", SND_PCM_STREAM_CAPTURE, 0);
if (rc < 0)
{
SID_ERROR("unable to open pcm device: %s\n", snd_strerror(rc));
return PCMIF_ERROR;
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(&hw_params);
/* Fill it in with default values. */
snd_pcm_hw_params_any(capture_handle, hw_params);
/* Sampling rate*/
val = input_sampleRate;
snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &val, 0);
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(capture_handle, hw_params, SND_PCM_FORMAT_S16_LE);
/* Interleaved mode */
snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
/* Num channels */
snd_pcm_hw_params_set_channels(capture_handle, hw_params, 1);
uframe = (output_sampleRate / 100);
rc = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params, &uframe, 0);
if (rc < 0)
{
SID_ERROR("snd_pcm_hw_params_set_period_size_near %s\n", snd_strerror(rc));
return PCMIF_ERROR;
}
rc = snd_pcm_hw_params(capture_handle, hw_params);
if (rc < 0)
{
SID_ERROR("snd_pcm_hw_params %s\n", snd_strerror(rc));
return PCMIF_ERROR;
}

Similarly, if deviceType == OUTPUT_DEVICE, the code would be

int rc, dir = 0;
unsigned int val;
snd_pcm_hw_params_t *hw_params = NULL;
snd_pcm_uframes_t uframe;
if(playback_handle == 0)
{
rc = snd_pcm_open(&playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (rc < 0)
{
SID_ERROR("unable to open pcm device: %s\n", snd_strerror(rc));
return PCMIF_ERROR;
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(&hw_params);
/* Fill it in with default values. */
snd_pcm_hw_params_any(playback_handle, hw_params);
/* Sampling rate*/
val = output_sampleRate;
snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &val, 0);
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_S16_LE);
/* Interleaved mode */
snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
/* Num channels */
snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1);
uframe = (output_sampleRate / 100);
rc = snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &uframe, 0);
if (rc < 0)
{
SID_ERROR("snd_pcm_hw_params_set_period_size_near %s\n", snd_strerror(rc));
return PCMIF_ERROR;
}
rc = snd_pcm_hw_params(playback_handle, hw_params);
if (rc < 0)
{
SID_ERROR("snd_pcm_hw_params %s\n", snd_strerror(rc));
return PCMIF_ERROR;
}

3. Close ALSA device when necessary
I put them in Stop()::PCMLoopback, maybe there is good idea to put it at other place like ~PCMLoopback() or Uninit().

snd_pcm_close(playback_handle);
snd_pcm_close(capture_handle);

4. Feed data to ALSA playback and Deliver ALSA captured data to Skypekit Runtime

in Run()::PCMLoopback

if output_started is true, using 

len = snd_pcm_writei(playback_handle, output_buf.m_data.data(), output_buf.m_data.size()>>1);
to send voice out.

if input_started is true, using

len = snd_pcm_readi(capture_handle, input_buf.m_data.data(), input_sampleRate / 100); //input_buf.m_data.size());

to get captured data and send it to Skypekit Runtime by invoking InputDeviceReady().

Do remember to call snd_pcm_recover() if snd_pcm_readi or snd_pcm_writei returns error.

The CPU usage of using ALSA is almost the same as we are using OSS, but we can get more control on  audio I/O.

Easy job, right?

[SKYPEKIT][DM368][IPNC] Audio PCM Experiment on DM368-IPNC

The very beginning experiment of porting Skypekit to DM368-IPNC is enabling Audio PCM to test voice conversation. With basic DM368-IPNC developing experience, you shall be able to make it work within few days.

Before starting, please make sure you have scanned Audio PCM Tutorial from Skype Developer Network. (You have to join in and log in first to read the article)

Steps:

1. Prepare Skypekit environment

Skypekit runtime - linux-armv5-skypekit-voicepcm-novideo (version 4.1.2)
Skypekit client - skypekitclient (version 4.1.2), which can be build at directory skypekit-sdk_sdk-4.1.2/interfaces/skype/cpp_embedded

If you have no idea of those, please check the other article for startup [SKYPEKIT][DM368][IPNC] Starting Skypekit on DM368-IPNC - setup environment and build the 1st tutorial example

2. Build reference audio PCMHost loopback

Following Audio PCM Quick Start, you can get PCMHost loopback "voicepcmhost-loopback" and try it with the UI skypekitclient.

3. Modify voicepcmhost-loopback

By tracing voicepcmhost-loopback reference code (precisely, PCMLoopback.cpp), you can see the voice data loopback ( is actually done by Run() method of the class PCMLoopback. The thing we are going to do is quite simple -  get and put the voice data from/to physical device, instead of using loopback_buf.

The audio device we can use is /dev/dsp. According to Skypekit requirement, we need 16bit PCM, and at least 8k sample rate. Thus, the code would be something like:

fd = open("/dev/dsp", O_RDWR);
status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg);

in the if statement "if(output_started)", add something like
write(fd, output_buf.m_data.data(),  output_buf.m_data.size()); // for incoming voice

in the if statement "if(input_started)", add something like
read(fd, input_buf.m_data.data(), input_buf.m_data.size()); // for outgoing voice

4. Kill av_server.out

av_server.out consumes MIC input data (see DRV_audioOpen() @ drv_audio.c) which we shall avoid. To simplify this experiment, we just kill av_server.out.

After all, we can repeat Step 2 to try our new voicepcmhost-loopback, which provides 2-way skype voice communication. Easy work, isn't it?