diff --git a/examples/DSPE_examples.h b/examples/DSPE_examples.h
index 5bd1f3c5e2806f862dd60ed578223c15d34de1ec..bbaa0df1b52d6a90b66a56fb94fd35cb18e91e78 100644
--- a/examples/DSPE_examples.h
+++ b/examples/DSPE_examples.h
@@ -9,7 +9,7 @@
#define EXAMPLES_DSPE_EXAMPLES_H_
int test_hello(void);
-int test_sound_input(void);
+int test_sound_input(bool use_audio_output = true);
#endif /* EXAMPLES_DSPE_EXAMPLES_H_ */
diff --git a/examples/sound_input.cpp b/examples/sound_input.cpp
index e15d363e1afee5af3f9895d1ae5731608e7b50b1..da1364f234ba12f3b3ab5ae11461f2767f8d751e 100644
--- a/examples/sound_input.cpp
+++ b/examples/sound_input.cpp
@@ -7,11 +7,14 @@
#ifndef INCLUDE_DSPE_EXAMPLES
int main(void)
+{
+ bool use_audio_output = true;
#else
#include "DSPE_examples.h"
-int test_sound_input(void)
-#endif // INCLUDE_DSPE_EXAMPLES
+int test_sound_input(bool use_audio_output)
{
+#endif // INCLUDE_DSPE_EXAMPLES
+
DSP::Clock_ptr MasterClock, AudioInClock;
string tekst;
int temp;
@@ -25,17 +28,26 @@ int test_sound_input(void)
MasterClock=DSP::Clock::CreateMasterClock();
+#ifndef INCLUDE_DSPE_EXAMPLES
DSP::u::WaveInput WaveIn(MasterClock, "DSPElib.wav", ".");
+#else
+ DSP::u::WaveInput WaveIn(MasterClock, "DSPElib.wav", "../examples");
+#endif // INCLUDE_DSPE_EXAMPLES
+
Fp = WaveIn.GetSamplingRate();
- DSP::u::AudioOutput AudioOut(Fp);
+ std::shared_ptr<DSP::Block> AudioOut = nullptr;
+ if (use_audio_output == true)
+ AudioOut.reset(new DSP::u::AudioOutput(Fp));
+ else
+ AudioOut.reset(new DSP::u::Vacuum(1U));
- WaveIn.Output("out") >> AudioOut.Input("in");
+ WaveIn.Output("out") >> AudioOut->Input("in");
Fp2 = 8000;
long Fp_gcd = DSP::f::gcd(Fp, Fp2);
AudioInClock=DSP::Clock::GetClock(MasterClock, Fp2 / Fp_gcd, Fp / Fp_gcd);
- DSP::u::AudioInput AudioIn(AudioInClock, 8000, 1);
+ DSP::u::AudioInput AudioIn(AudioInClock, Fp2, 1);
DSP::u::FileOutput WaveOut("captured_sample.wav",DSP::e::SampleType::ST_short, 1, DSP::e::FileType::FT_wav, Fp2);
AudioIn.Output("out") >> WaveOut.Input("in");
@@ -51,8 +63,9 @@ int test_sound_input(void)
DSP::log << "MAIN" << DSP::e::LogMode::second << temp << endl;
temp++;
}
- while (WaveIn.GetBytesRead() != 0);
+ while ((temp <= 8) || (WaveIn.GetBytesRead() != 0));
+ AudioOut.reset();
DSP::Clock::FreeClocks();
DSP::log << DSP::e::LogMode::Error << "MAIN" << DSP::e::LogMode::second << "end" << endl;
diff --git a/src/Main.cpp b/src/Main.cpp
index eb68ee86d67e33bfc0599c522f4989e4bedfef81..333f2403aec0ea66342d53119a60e1c013dfc132 100644
--- a/src/Main.cpp
+++ b/src/Main.cpp
@@ -2977,7 +2977,8 @@ int main(int argc, char*argv[])
DSP::log << DSP::e::LogMode::Error << "Finished test_hello" << endl;
DSP::log << "Starting test_sound_input" << endl;
- test_sound_input();
+ bool use_audio_output = false;
+ test_sound_input(use_audio_output);
DSP::log << DSP::e::LogMode::Error << "Finished test_sound_input" << endl;
#endif // INCLUDE_DSPE_EXAMPLES
diff --git a/src/cpp/ALSA_support.cpp b/src/cpp/ALSA_support.cpp
index d7acc0c313d1621ec75cdafc66f7cceea1ec1607..8ecc999b4884bf913b21ff6454ec2742f82dec9d 100644
--- a/src/cpp/ALSA_support.cpp
+++ b/src/cpp/ALSA_support.cpp
@@ -43,7 +43,6 @@ DSP::ALSA_object_t::ALSA_object_t()
InDevNo = 1;
NextBufferOutInd = 0;
- NextBufferInInd = 0;
/* 44100 bits/second sampling rate (CD quality) */
sampling_rate_alsa = 44100;
@@ -51,6 +50,7 @@ DSP::ALSA_object_t::ALSA_object_t()
no_of_bytes_in_channel = 2;
audio_inbuffer_size_in_frames = 8000;
+ audio_outbuffer_size_in_frames = 8000;
size_b = 32;
}
@@ -68,6 +68,9 @@ DSP::ALSA_object_t::~ALSA_object_t()
buffers_32bit.clear();
buffers_32bit_f.clear();
buffers_64bit.clear();
+
+ capture_buffer.clear();
+
pcm_buffer.clear();
pcm_buffer_size_in_frames.clear();
@@ -148,7 +151,7 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
//! Errors controller in set_snd_pcm_format() function
int errc;
- //! Local configuration space handler
+ //! Local configuration space handler
snd_pcm_hw_params_t *params;
//! For logging
@@ -165,7 +168,7 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
//! \TODO Test mode: Open mode (see SND_PCM_NONBLOCK, SND_PCM_ASYNC)
- if (stream_type == SND_PCM_STREAM_PLAYBACK)
+ if (stream_type == SND_PCM_STREAM_PLAYBACK)
DSP::log << "Opening PCM device for playback." << endl;
else
@@ -208,32 +211,69 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
snd_pcm_hw_params_set_rate_near(alsa_handle, params, &sampling_rate_alsa, &dir);
- if (audio_inbuffer_size_in_frames <= 0)
- audio_inbuffer_size_in_frames = 8000;
+ if (stream_type == SND_PCM_STREAM_PLAYBACK)
+ {
+ if (audio_outbuffer_size_in_frames <= 0)
+ audio_outbuffer_size_in_frames = 8000;
+
+ size_b = audio_outbuffer_size_in_frames * no_of_channels_alsa * no_of_bytes_in_channel;
- rc = snd_pcm_hw_params_set_buffer_size(alsa_handle, params, DSP::NoOfAudioOutputBuffers*audio_inbuffer_size_in_frames);
+ rc = snd_pcm_hw_params_set_buffer_size(alsa_handle, params, DSP::NoOfAudioOutputBuffers*audio_outbuffer_size_in_frames);
- snd_pcm_uframes_t requested_audio_inbuffer_size_in_frames = audio_inbuffer_size_in_frames;
+ snd_pcm_uframes_t requested_audio_outbuffer_size_in_frames = audio_outbuffer_size_in_frames;
- if (rc < 0)
- {
- DSP::log << "Unable to set a buffer size with error code: " << rc << endl;
- return -5;
+ if (rc < 0)
+ {
+ DSP::log << "Unable to set a buffer size with error code: " << rc << endl;
+ return -5;
+ }
+
+ /*! Set period size to desired number of frames. */
+ snd_pcm_hw_params_set_period_size_near(alsa_handle, params, &audio_outbuffer_size_in_frames, &dir);
+
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ if (audio_outbuffer_size_in_frames != requested_audio_outbuffer_size_in_frames)
+ {
+ DSP::log << "Current frames value should be equal: " << requested_audio_outbuffer_size_in_frames << endl;
+ DSP::log << "Frames is not equal to tmp_frames! Frames: " << audio_outbuffer_size_in_frames << endl;
+ }
+
+ else
+ DSP::log << "Frames has been set correctly." << endl;
+ #endif // AUDIO_DEBUG_MESSAGES_ON
}
- /*! Set period size to desired number of frames. */
- snd_pcm_hw_params_set_period_size_near(alsa_handle, params, &audio_inbuffer_size_in_frames, &dir);
+ else // stream_type == SND_PCM_STREAM_CAPTURE
+ {
+ if (audio_inbuffer_size_in_frames <= 0)
+ audio_inbuffer_size_in_frames = 8000;
+
+ size_b = audio_inbuffer_size_in_frames * no_of_channels_alsa * no_of_bytes_in_channel;
+
+ rc = snd_pcm_hw_params_set_buffer_size(alsa_handle, params, size_b);
+
+ snd_pcm_uframes_t requested_audio_inbuffer_size_in_frames = audio_inbuffer_size_in_frames;
- #ifdef AUDIO_DEBUG_MESSAGES_ON
- if (audio_inbuffer_size_in_frames != requested_audio_inbuffer_size_in_frames)
+ if (rc < 0)
{
- DSP::log << "Current frames value should be equal: " << requested_audio_inbuffer_size_in_frames << endl;
- DSP::log << "Frames is not equal to tmp_frames! Frames: " << audio_inbuffer_size_in_frames << endl;
+ DSP::log << "Unable to set a buffer size with error code: " << rc << endl;
+ return -5;
}
- else
- DSP::log << "Frames has been set correctly." << endl;
- #endif // AUDIO_DEBUG_MESSAGES_ON
+ /*! Set period size to desired number of frames. */
+ snd_pcm_hw_params_set_period_size_near(alsa_handle, params, &audio_inbuffer_size_in_frames, &dir);
+
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ if (audio_inbuffer_size_in_frames != requested_audio_inbuffer_size_in_frames)
+ {
+ DSP::log << "Current frames value should be equal: " << requested_audio_inbuffer_size_in_frames << endl;
+ DSP::log << "Frames is not equal to tmp_frames! Frames: " << audio_inbuffer_size_in_frames << endl;
+ }
+
+ else
+ DSP::log << "Frames has been set correctly." << endl;
+ #endif // AUDIO_DEBUG_MESSAGES_ON
+ }
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(alsa_handle, params);
@@ -281,8 +321,8 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
snd_pcm_hw_params_get_period_time(hw_params, &val, &dir);
DSP::log << "period time = " << val << " us" << endl;
- snd_pcm_hw_params_get_period_size(hw_params, &audio_inbuffer_size_in_frames, &dir);
- DSP::log << "period size = " << (int) audio_inbuffer_size_in_frames << " frames" << endl;
+ snd_pcm_hw_params_get_period_size(hw_params, (snd_pcm_uframes_t *)&val, &dir);
+ DSP::log << "period size = " << val << " frames" << endl;
snd_pcm_hw_params_get_buffer_time(hw_params, &val, &dir);
DSP::log << "buffer time = " << val << " us" << endl;
@@ -353,13 +393,12 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
// Can be useful:
// snd_pcm_hw_params_get_period_size(hw_params, &frames, &dir);
-
- size_b = audio_inbuffer_size_in_frames * no_of_channels_alsa * no_of_bytes_in_channel;
if (stream_type == SND_PCM_STREAM_PLAYBACK)
{
pcm_buffer.resize(DSP::NoOfAudioOutputBuffers);
pcm_buffer_size_in_frames.resize(DSP::NoOfAudioOutputBuffers);
+
switch (no_of_bytes_in_channel)
{
case 1:
@@ -430,73 +469,45 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
}
else // (stream_type == SND_PCM_STREAM_CAPTURE)
{
- pcm_buffer.resize(DSP::NoOfAudioInputBuffers);
- pcm_buffer_size_in_frames.resize(DSP::NoOfAudioInputBuffers);
- switch (no_of_bytes_in_channel)
+ pcm_buffer.resize(1);
+ pcm_buffer_size_in_frames.resize(1);
+
+ capture_buffer.resize(size_b);
+
+ for(unsigned int ind = 0; ind < pcm_buffer.size(); ind++)
{
- case 1:
- buffers_8bit.resize(DSP::NoOfAudioInputBuffers);
+ pcm_buffer[ind] = (uint8_t *)(capture_buffer.data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) capture_buffer.size() / no_of_bytes_in_channel / no_of_channels_alsa;
+ }
- for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
- {
- buffers_8bit[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_8bit[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_8bit[ind].size() / no_of_channels_alsa;
- }
+ switch (no_of_bytes_in_channel)
+ {
+ case 1:
+ InSampleTypeALSA = DSP::e::SampleType::ST_uchar;
break;
case 2:
- buffers_16bit.resize(DSP::NoOfAudioInputBuffers);
-
- for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
- {
- buffers_16bit[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_16bit[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_16bit[ind].size() / no_of_channels_alsa;
- }
+ InSampleTypeALSA = DSP::e::SampleType::ST_short;
break;
case 3:
+ InSampleTypeALSA = DSP::e::SampleType::ST_int;
+ break;
+
case 4:
if (IsHigherQualityMode)
- {
- buffers_32bit.resize(DSP::NoOfAudioInputBuffers);
-
- for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
- {
- buffers_32bit[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_32bit[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_32bit[ind].size() / no_of_channels_alsa;
- }
- }
-
- else //! native mode
- {
- buffers_32bit_f.resize(DSP::NoOfAudioInputBuffers);
-
- for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
- {
- buffers_32bit_f[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_32bit_f[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_32bit_f[ind].size() / no_of_channels_alsa;
- }
- }
+ InSampleTypeALSA = DSP::e::SampleType::ST_int;
+ else
+ InSampleTypeALSA = DSP::e::SampleType::ST_float;
break;
case 8:
- buffers_64bit.resize(DSP::NoOfAudioInputBuffers);
-
- for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
- {
- buffers_64bit[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_64bit[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_64bit[ind].size() / no_of_channels_alsa;
- }
+ InSampleTypeALSA = DSP::e::SampleType::ST_double;
break;
default:
DSP::log << "Unsupported no of bytes in channel" << endl;
- return -6;
+ return -9;
break;
}
}
@@ -651,7 +662,7 @@ long DSP::ALSA_object_t::open_PCM_device_4_output(const int &no_of_channels, int
no_of_channels_alsa = (unsigned int) no_of_channels;
no_of_bytes_in_channel = (unsigned int) no_of_bits / 8;
sampling_rate_alsa = (unsigned int) sampling_rate;
- audio_inbuffer_size_in_frames = (snd_pcm_uframes_t) audio_outbuffer_size;
+ audio_outbuffer_size_in_frames = (snd_pcm_uframes_t) audio_outbuffer_size;
rc = open_alsa_device(SND_PCM_STREAM_PLAYBACK);
@@ -883,19 +894,85 @@ bool DSP::ALSA_object_t::stop_recording(void)
bool DSP::ALSA_object_t::start_recording(void)
{
- assert(!"DSP::ALSA_object_t::stop_recording not implemented yet");
+ IsRecordingNow = true;
return true;
}
bool DSP::ALSA_object_t::get_wave_in_raw_buffer(DSP::e::SampleType &InSampleType, std::vector<char> &wave_in_raw_buffer)
{
- assert(!"DSP::ALSA_object_t::get_wave_in_raw_buffer not implemented yet");
-
- InSampleTypeALSA = InSampleType;
+ InSampleType = this->InSampleTypeALSA;
snd_pcm_sframes_t rc;
- return true;
+ for (unsigned int ind = 0; ind < pcm_buffer.size(); ind++)
+ {
+ while (pcm_buffer_size_in_frames[ind])
+ {
+ rc = snd_pcm_readi(alsa_handle, pcm_buffer[ind], pcm_buffer_size_in_frames[ind]);
+
+ switch (-rc)
+ {
+ case EPIPE:
+
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ // EPIPE means underrun
+ DSP::log << "Underrun occurred" << endl;
+ #endif // AUDIO_DEBUG_MESSAGES_ON
+
+ snd_pcm_prepare(alsa_handle);
+ break;
+
+ case EAGAIN:
+
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ DSP::log << "EAGAIN occurred. Waiting for a free buffer." << endl;
+ #endif // AUDIO_DEBUG_MESSAGES_ON
+
+ DSP::f::Sleep(0);
+ break;
+
+ default:
+ if (rc > 0)
+ {
+ pcm_buffer_size_in_frames[ind] -= rc;
+ pcm_buffer[ind] += rc * no_of_channels_alsa * no_of_bytes_in_channel;
+
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ DSP::log << "Short read. Current rc = " << rc << "." << endl;
+ #endif // AUDIO_DEBUG_MESSAGES_ON
+
+ if (pcm_buffer_size_in_frames[ind] == 0)
+ {
+ wave_in_raw_buffer.resize(capture_buffer.size());
+ std::swap(wave_in_raw_buffer, capture_buffer);
+
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ DSP::log << "Inbuffer is full." << endl;
+ #endif // AUDIO_DEBUG_MESSAGES_ON
+
+ pcm_buffer[ind] = (uint8_t *)(capture_buffer.data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) capture_buffer.size() / no_of_bytes_in_channel / no_of_channels_alsa;
+
+ return true;
+ }
+ }
+
+ else
+ {
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ // EPIPE means underrun
+ DSP::log << "Unsupported error." << endl;
+ DSP::log << "Error from readi: " << snd_strerror(rc) << endl;
+ #endif // AUDIO_DEBUG_MESSAGES_ON
+
+ pcm_buffer_size_in_frames[ind] = 0;
+ return false;
+ }
+ break;
+ }
+ }
+ }
+ return false;
}
snd_pcm_sframes_t DSP::ALSA_object_t::pcm_writei(const void *buffer, const snd_pcm_uframes_t &frames)
diff --git a/src/cpp/DSP_IO.cpp b/src/cpp/DSP_IO.cpp
index c5a112a7dbd448ff1eea546df12ba3f9647396ad..99fbbee523a308a831774fdc9508561db1da7764 100644
--- a/src/cpp/DSP_IO.cpp
+++ b/src/cpp/DSP_IO.cpp
@@ -3995,6 +3995,28 @@ bool DSP::u::AudioInput::SOUND_object_callback(const DSP::e::SampleType &InSampl
DSP::Float_ptr Sample;
unsigned int ind;
+#ifdef __DEBUG__
+ switch (InSampleType) {
+ case DSP::e::SampleType::ST_short: {
+ if (InBufferLen != wave_in_raw_buffer.size() / sizeof(short)) {
+ DSP::log << "DSP::u::AudioInput::SOUND_object_callback" << DSP::e::LogMode::second << "incorrect wave_in_raw_buffer size: " <<
+ wave_in_raw_buffer.size() << " instead of " << InBufferLen * sizeof(short) << endl;
+ }
+ }
+ break;
+ case DSP::e::SampleType::ST_uchar: {
+ if (InBufferLen != wave_in_raw_buffer.size() / sizeof(uint8_t)) {
+ DSP::log << "DSP::u::AudioInput::SOUND_object_callback" << DSP::e::LogMode::second << "incorrect wave_in_raw_buffer size: " <<
+ wave_in_raw_buffer.size() << " instead of " << InBufferLen * sizeof(uint8_t) << endl;
+ }
+ }
+ break;
+ default:
+ DSP::log << "DSP::u::AudioInput::SOUND_object_callback" << DSP::e::LogMode::second << "Unsupported Current->InSampleType" << endl;
+ break;
+ }
+#endif // __DEBUG__
+
// reading sound card input buffer
Sample=InBuffers[EmptyBufferIndex].data();
// ************************************************** //
@@ -4028,7 +4050,7 @@ bool DSP::u::AudioInput::SOUND_object_callback(const DSP::e::SampleType &InSampl
default:
#ifdef __DEBUG__
- DSP::log << "DSP::u::AudioInput::waveInProc_short" << DSP::e::LogMode::second << "Unsupported Current->InSampleType" << endl;
+ DSP::log << "DSP::u::AudioInput::SOUND_object_callback" << DSP::e::LogMode::second << "Unsupported Current->InSampleType" << endl;
#endif
break;
}
diff --git a/src/include/ALSA_support.h b/src/include/ALSA_support.h
index 0bd6078495c83a692f724e833de484cd1f365e46..e06cc1e469561570fdd11d470847843da517bea8 100644
--- a/src/include/ALSA_support.h
+++ b/src/include/ALSA_support.h
@@ -37,8 +37,6 @@ namespace DSP {
//! keeping track of which outbuffer is currently being filled
unsigned int NextBufferOutInd;
- //! keeping track of which inbuffer is currently being filled
- unsigned int NextBufferInInd;
//! sampling rate
unsigned int sampling_rate_alsa;
@@ -50,12 +48,15 @@ namespace DSP {
/*! It is better to use STD containers - they are more convenient,
and they mean fewer problems with memory leaks.
*/
- //! buffers depending on samples type
+ //! outbuffers depending on samples type
std::vector<std::vector<uint8_t>> buffers_8bit;
std::vector<std::vector<int16_t>> buffers_16bit;
std::vector<std::vector<int32_t>> buffers_32bit;
std::vector<std::vector<float>> buffers_32bit_f;
std::vector<std::vector<double>> buffers_64bit;
+
+ //! inbuffer
+ std::vector<char> capture_buffer;
//! samples are integers rather than float values
bool IsHigherQualityMode;
@@ -75,7 +76,8 @@ namespace DSP {
bool IsRecordingNow;
//! just samples
- snd_pcm_uframes_t audio_inbuffer_size_in_frames; // M.B. more meaningful variable name
+ snd_pcm_uframes_t audio_inbuffer_size_in_frames;
+ snd_pcm_uframes_t audio_outbuffer_size_in_frames;
//! Type of samples in WaveInBuffers
DSP::e::SampleType InSampleTypeALSA;
@@ -97,6 +99,9 @@ namespace DSP {
//! playback
snd_pcm_sframes_t pcm_writei(const void *buffer, const snd_pcm_uframes_t &frames); // M.B. this will be more transparent
+
+ //! recording
+ //snd_pcm_sframes_t DSP::ALSA_object_t::pcm_readi(const void *buffer, const snd_pcm_uframes_t &frames)
//! Set SND PCM format depending on no of bytes in channel and CPU endianness
int set_snd_pcm_format(snd_pcm_hw_params_t *params);