From d00dc201ea70b3e18bd369754d3f6ac7eedd52aa Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Damian=20Kara=C5=9B?= <s176030@student.pg.edu.pl>
Date: Tue, 16 Nov 2021 18:37:18 +0100
Subject: [PATCH] ALSA_support: improved open device for input
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Signed-off-by: Damian KaraĹ <s176030@student.pg.edu.pl>
---
src/cpp/ALSA_support.cpp | 165 +++++++++++++++++--------------------
src/include/ALSA_support.h | 8 +-
2 files changed, 82 insertions(+), 91 deletions(-)
diff --git a/src/cpp/ALSA_support.cpp b/src/cpp/ALSA_support.cpp
index d7726de..f067c58 100644
--- a/src/cpp/ALSA_support.cpp
+++ b/src/cpp/ALSA_support.cpp
@@ -51,6 +51,7 @@ 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;
}
@@ -208,32 +209,69 @@ 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 +319,17 @@ 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;
+ if (stream_type == SND_PCM_STREAM_PLAYBACK)
+ {
+ snd_pcm_hw_params_get_period_size(hw_params, &audio_outbuffer_size_in_frames, &dir);
+ DSP::log << "period size = " << (int) audio_outbuffer_size_in_frames << " frames" << endl;
+ }
+
+ else // stream_type == SND_PCM_STREAM_CAPTURE
+ {
+ 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_buffer_time(hw_params, &val, &dir);
DSP::log << "buffer time = " << val << " us" << endl;
@@ -353,8 +400,6 @@ 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)
{
@@ -430,74 +475,15 @@ 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)
- {
- case 1:
- buffers_8bit.resize(DSP::NoOfAudioInputBuffers);
-
- 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;
- }
- 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;
- }
- break;
-
- case 3:
- 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;
- }
- }
- 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;
- }
- break;
+ pcm_buffer.resize(1);
+ pcm_buffer_size_in_frames.resize(1);
+
+ capture_buffer.resize(size_b / no_of_bytes_in_channel);
- default:
- DSP::log << "Unsupported no of bytes in channel" << endl;
- return -6;
- break;
+ for(unsigned int ind = 0; ind < pcm_buffer.size(); ind++)
+ {
+ pcm_buffer[ind] = (uint8_t *)(capture_buffer.data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) capture_buffer.size() / no_of_channels_alsa;
}
switch (no_of_bytes_in_channel)
@@ -682,7 +668,7 @@ 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);
@@ -973,6 +959,7 @@ bool DSP::ALSA_object_t::get_wave_in_raw_buffer(DSP::e::SampleType &InSampleType
#endif // AUDIO_DEBUG_MESSAGES_ON
pcm_buffer_size_in_frames[ind] = 0;
+ return false;
}
break;
}
diff --git a/src/include/ALSA_support.h b/src/include/ALSA_support.h
index 99e7e65..0c7b59e 100644
--- a/src/include/ALSA_support.h
+++ b/src/include/ALSA_support.h
@@ -50,12 +50,15 @@ 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<int8_t> capture_buffer;
//! samples are integers rather than float values
bool IsHigherQualityMode;
@@ -75,7 +78,8 @@ 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;
--
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