Re: [Alsa-user] C program for playing sound
Thank you very much, Gerhard! ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] C program for playing sound
Hello together!
On Wed, 24 Jun 2020 17:37:46 -0300
Sleep schrieb:
> // Insert samples to framebuffer.
> wav = fopen("pcm1611s.wav", "r");
> fseek(wav, 0L, SEEK_END);
> size = ftell(wav);
> fseek(wav, 0L, SEEK_SET);
> buffer = malloc(size);
> fread(buffer, sizeof(int16_t), size, wav);
> fclose(wav);
> size /= sizeof(int16_t);
>
> // Set ptrbuffer 46 bytes ahead to skip the header.
> for (int16_t *ptrbuffer = buffer + 46; size > ptrbuffer - buffer;
> ptrbuffer += FRAMEBUFFERSIZE * STEREO * BITS)
> {
I would:
avoid the #defines as far as possible and fetch the values out of the
RIFF-header;
first read the RIFF-header and with its contents calculate starting
point and length of data.
It is not guaranteed that the line with „for(“ (always) works – this as
a qick response …
Grüße, Gerhard
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Re: [Alsa-user] C program for playing sound
Nevermind, it is frames * STEREO here, this is where the error is. ptrbuffer += FRAMEBUFFERSIZE * STEREO * BITS ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] C program for playing sound
https://files.catbox.moe/5jvu18.wav
Here is the recording of the effect as requested, new code below.
#include
#include
#include
#define STEREO 2
#define BITS16 / 8
#define FRAMEBUFFERSIZE 512 // in samples
#define PERIODS 2
#define SAMPLERATE 11025
void snd_error_checker(int error, char *function_name)
{
if (error)
{
printf("Error (%s): %s\n", function_name, snd_strerror(error));
// exit(EXIT_FAILURE);
}
}
int main(void)
{
snd_pcm_t *handle;
uint32_t channels = STEREO;
uint32_t sample_size= BITS; // 16 bit
uint32_t frame_size = sample_size * channels;
snd_pcm_uframes_t frames= FRAMEBUFFERSIZE / frame_size;
snd_pcm_uframes_t frames_per_period = frames / PERIODS;
int32_t dir = 0; // No idea what this does.
snd_pcm_hw_params_t *params;
int16_t *buffer;
FILE *wav;
int32_t size;
int error;
error = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
snd_error_checker(error, "snd_pcm_open()");
snd_pcm_hw_params_alloca(¶ms);
error = snd_pcm_hw_params_any(handle, params);
snd_error_checker(error, "snd_pcm_hw_params_any()");
error = snd_pcm_hw_params_set_buffer_size(handle, params, frames);
snd_error_checker(error, "snd_pcm_hw_params_set_buffer_size()");
error = snd_pcm_hw_params_set_period_size(handle, params,
frames_per_period, dir);
snd_error_checker(error, "snd_pcm_hw_params_set_period_size()");
error = snd_pcm_hw_params_set_rate(handle, params, SAMPLERATE, dir);
snd_error_checker(error, "snd_pcm_hw_params_set_rate()");
error = snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_error_checker(error, "snd_pcm_hw_params_set_access()");
error = snd_pcm_hw_params_set_channels(handle, params, STEREO);
snd_error_checker(error, "snd_pcm_hw_params_set_channels()");
error = snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_S16_LE);
snd_error_checker(error, "snd_pcm_hw_params_set_format()");
error = snd_pcm_hw_params(handle, params);
snd_error_checker(error, "snd_pcm_hw_params()");
// Insert samples to framebuffer.
wav = fopen("pcm1611s.wav", "r");
fseek(wav, 0L, SEEK_END);
size = ftell(wav);
fseek(wav, 0L, SEEK_SET);
buffer = malloc(size);
fread(buffer, sizeof(int16_t), size, wav);
fclose(wav);
size /= sizeof(int16_t);
// Set ptrbuffer 46 bytes ahead to skip the header.
for (int16_t *ptrbuffer = buffer + 46; size > ptrbuffer - buffer;
ptrbuffer += FRAMEBUFFERSIZE * STEREO * BITS)
{
error = snd_pcm_writei(handle, ptrbuffer, frames);
if (error)
{
snd_pcm_recover(handle, error, 0);
}
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
exit(EXIT_SUCCESS);
}
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Re: [Alsa-user] C program for playing sound
snd_pcm_hw_params_set_buffer_size() and snd_pcm_hw_params_set_period_size() returns an error, a mere "Invalid argument." which doesn't help much, commenting them out doesn't appear to address the scratchy sound. ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] C program for playing sound
I've used snd_pcm_prepare() because underrun (-EPIPE) was occurring
because I had put the wrong value for frames, but despite adjusting the
value of the frames, I am still getting the scratchy effect, here is the
current code I have, now I am really stuck. Also yeah, "scratchy" would
be a better description of the effect I suppose.
I might post a recording of the result later, meanwhile here is the code
for now, I have removed snd_pcm_prepare() from it.
#include
#include
#include
#define STEREO 2
#define BITS 16 / 8
#define FRAMEBUFFERSIZE 512 // in samples
#define PERIODS 2
#define SAMPLERATE 11025
int main(void)
{
snd_pcm_t *handle;
uint32_t channels = STEREO;
uint32_t sample_size = BITS; // 16 bit
uint32_t frame_size = sample_size * channels;
snd_pcm_uframes_t frames = FRAMEBUFFERSIZE / frame_size;
snd_pcm_uframes_t frames_per_period = frames / PERIODS;
int32_t dir = 0; // No idea what this does.
snd_pcm_hw_params_t *params;
int16_t framebuffer[FRAMEBUFFERSIZE * STEREO * BITS] = {0};
FILE *wav;
int32_t size;
snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_buffer_size(handle, params, frames);
snd_pcm_hw_params_set_period_size(handle, params,
frames_per_period, dir);
snd_pcm_hw_params_set_rate(handle, params, SAMPLERATE, dir);
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_channels(handle, params, STEREO);
snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params(handle, params);
// Insert samples to framebuffer.
wav = fopen("pcm1611s.wav", "r");
fseek(wav, 0L, SEEK_END);
size = ftell(wav);
fseek(wav, 0L, SEEK_SET);
size /= sizeof(int16_t);
while (size > 0)
{
fread(framebuffer, sizeof(int16_t), FRAMEBUFFERSIZE * STEREO *
BITS, wav);
snd_pcm_writei(handle, framebuffer, frames);
size -= FRAMEBUFFERSIZE * BITS * STEREO;
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
exit(EXIT_SUCCESS);
}
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Re: [Alsa-user] C program for playing sound
now the problem is that the sound is skipping.
Can you provide a recording of what it sounds like?
In alsa-lib there is an example test/latency.c which just moves samples from
one device to another.
When I run it, the sound that comes out of my speakers is what I would call
"scratchy".
I submitted a bug report here, which has recordings of what I hear when I run
./latency:
https://github.com/alsa-project/alsa-lib/issues/53
It was closed by someone named Jaroslav Kysela who is the author of ALSA,
according to Wikipedia. But I'm not sure if he read the actual bug report.
I don't know where you got the idea to call snd_pcm_prepare in your loop, or
why that fixed the problem.
I hope this is helpful.
Frederick
On Tue, Jun 23, 2020 at 04:54:09PM -0300, Sleep wrote:
Update, forgot snd_pcm_prepare(), now the problem is that the sound is
skipping.
#include
#include
#include
#define STEREO 2
#define BITS 16 / 8
#define FRAMEBUFFERSIZE 512 // in samples
#define PERIODS 2
#define SAMPLERATE 11025
int main(void)
{
snd_pcm_t *handle;
uint32_t channels = STEREO;
uint32_t sample_size = BITS; // 16 bit
uint32_t frame_size = sample_size * channels;
snd_pcm_uframes_t frames = FRAMEBUFFERSIZE;
snd_pcm_uframes_t frames_per_period = frames / PERIODS;
int dir = 0; // No idea what this does.
snd_pcm_hw_params_t *params;
int16_t framebuffer[FRAMEBUFFERSIZE * STEREO * BITS] = {0};
FILE *wav;
int size;
int status;
snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0); //
SND_PCM_NONBLOCK);
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_buffer_size(handle, params, frames);
snd_pcm_hw_params_set_period_size(handle, params,
frames_per_period, dir);
snd_pcm_hw_params_set_rate(handle, params, SAMPLERATE, dir);
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_channels(handle, params, STEREO);
snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params(handle, params);
// Insert samples to framebuffer.
wav = fopen("pcm1611s.wav", "r");
fseek(wav, 0L, SEEK_END);
size = ftell(wav);
fseek(wav, 0L, SEEK_SET);
size /= sizeof(signed short);
while (size > 0)
{
fread(framebuffer, sizeof(int16_t), FRAMEBUFFERSIZE * STEREO *
BITS, wav);
status = snd_pcm_writei(handle, framebuffer, frames);
if (status == -EPIPE)
{
snd_pcm_prepare(handle);
}
size -= FRAMEBUFFERSIZE * STEREO * BITS;
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
exit(EXIT_SUCCESS);
}
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Re: [Alsa-user] C program for playing sound
Update, forgot snd_pcm_prepare(), now the problem is that the sound is
skipping.
#include
#include
#include
#define STEREO 2
#define BITS 16 / 8
#define FRAMEBUFFERSIZE 512 // in samples
#define PERIODS 2
#define SAMPLERATE 11025
int main(void)
{
snd_pcm_t *handle;
uint32_t channels = STEREO;
uint32_t sample_size = BITS; // 16 bit
uint32_t frame_size = sample_size * channels;
snd_pcm_uframes_t frames = FRAMEBUFFERSIZE;
snd_pcm_uframes_t frames_per_period = frames / PERIODS;
int dir = 0; // No idea what this does.
snd_pcm_hw_params_t *params;
int16_t framebuffer[FRAMEBUFFERSIZE * STEREO * BITS] = {0};
FILE *wav;
int size;
int status;
snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0); //
SND_PCM_NONBLOCK);
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_buffer_size(handle, params, frames);
snd_pcm_hw_params_set_period_size(handle, params,
frames_per_period, dir);
snd_pcm_hw_params_set_rate(handle, params, SAMPLERATE, dir);
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_channels(handle, params, STEREO);
snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params(handle, params);
// Insert samples to framebuffer.
wav = fopen("pcm1611s.wav", "r");
fseek(wav, 0L, SEEK_END);
size = ftell(wav);
fseek(wav, 0L, SEEK_SET);
size /= sizeof(signed short);
while (size > 0)
{
fread(framebuffer, sizeof(int16_t), FRAMEBUFFERSIZE * STEREO *
BITS, wav);
status = snd_pcm_writei(handle, framebuffer, frames);
if (status == -EPIPE)
{
snd_pcm_prepare(handle);
}
size -= FRAMEBUFFERSIZE * STEREO * BITS;
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
exit(EXIT_SUCCESS);
}
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