--- /dev/null
+
+#include "alsa_input.h"
+
+using namespace std;
+
+ALSAInput::ALSAInput(const char *device, unsigned sample_rate, unsigned num_channels, audio_callback_t audio_callback)
+ : device(device), sample_rate(sample_rate), num_channels(num_channels), audio_callback(audio_callback)
+{
+ die_on_error(device, snd_pcm_open(&pcm_handle,device, SND_PCM_STREAM_CAPTURE, 0));
+
+ // Set format.
+ snd_pcm_hw_params_t *hw_params;
+ snd_pcm_hw_params_alloca(&hw_params);
+ die_on_error("snd_pcm_hw_params_any()", snd_pcm_hw_params_any(pcm_handle, hw_params));
+ die_on_error("snd_pcm_hw_params_set_access()", snd_pcm_hw_params_set_access(pcm_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED));
+ snd_pcm_format_mask_t *format_mask;
+ snd_pcm_format_mask_alloca(&format_mask);
+ snd_pcm_format_mask_set(format_mask, SND_PCM_FORMAT_S16_LE);
+ snd_pcm_format_mask_set(format_mask, SND_PCM_FORMAT_S24_LE);
+ snd_pcm_format_mask_set(format_mask, SND_PCM_FORMAT_S32_LE);
+ die_on_error("snd_pcm_hw_params_set_format()", snd_pcm_hw_params_set_format_mask(pcm_handle, hw_params, format_mask));
+ die_on_error("snd_pcm_hw_params_set_rate_near()", snd_pcm_hw_params_set_rate_near(pcm_handle, hw_params, &sample_rate, 0));
+ die_on_error("snd_pcm_hw_params_set_channels()", snd_pcm_hw_params_set_channels(pcm_handle, hw_params, num_channels));
+
+ die_on_error("snd_pcm_hw_params_set_channels()", snd_pcm_hw_params_set_channels(pcm_handle, hw_params, num_channels));
+
+ // Fragment size of 64 samples (about 1 ms at 48 kHz; a frame at 60
+ // fps/48 kHz is 800 samples.) We ask for 64 such periods in our buffer
+ // (~85 ms buffer); more than that, and our jitter is probably so high
+ // that the resampling queue can't keep up anyway.
+ // The entire thing with periods and such is a bit mysterious to me;
+ // seemingly I can get 96 frames at a time with no problems even if
+ // the period size is 64 frames. And if I set num_periods to e.g. 1,
+ // I can't have a big buffer.
+ num_periods = 16;
+ int dir = 0;
+ die_on_error("snd_pcm_hw_params_set_periods_near()", snd_pcm_hw_params_set_periods_near(pcm_handle, hw_params, &num_periods, &dir));
+ period_size = 64;
+ dir = 0;
+ die_on_error("snd_pcm_hw_params_set_period_size_near()", snd_pcm_hw_params_set_period_size_near(pcm_handle, hw_params, &period_size, &dir));
+ buffer_frames = 64 * 64;
+ die_on_error("snd_pcm_hw_params_set_buffer_size_near()", snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hw_params, &buffer_frames));
+ die_on_error("snd_pcm_hw_params()", snd_pcm_hw_params(pcm_handle, hw_params));
+ //snd_pcm_hw_params_free(hw_params);
+
+ // Figure out which format the card actually chose.
+ die_on_error("snd_pcm_hw_params_current()", snd_pcm_hw_params_current(pcm_handle, hw_params));
+ snd_pcm_format_t chosen_format;
+ die_on_error("snd_pcm_hw_params_get_format()", snd_pcm_hw_params_get_format(hw_params, &chosen_format));
+
+ audio_format.num_channels = num_channels;
+ audio_format.bits_per_sample = 0;
+ switch (chosen_format) {
+ case SND_PCM_FORMAT_S16_LE:
+ audio_format.bits_per_sample = 16;
+ break;
+ case SND_PCM_FORMAT_S24_LE:
+ audio_format.bits_per_sample = 24;
+ break;
+ case SND_PCM_FORMAT_S32_LE:
+ audio_format.bits_per_sample = 32;
+ break;
+ default:
+ assert(false);
+ }
+ //printf("num_periods=%u period_size=%u buffer_frames=%u sample_rate=%u bits_per_sample=%d\n",
+ // num_periods, unsigned(period_size), unsigned(buffer_frames), sample_rate, audio_format.bits_per_sample);
+
+ buffer.reset(new uint8_t[buffer_frames * num_channels * audio_format.bits_per_sample / 8]);
+
+ snd_pcm_sw_params_t *sw_params;
+ snd_pcm_sw_params_alloca(&sw_params);
+ die_on_error("snd_pcm_sw_params_current()", snd_pcm_sw_params_current(pcm_handle, sw_params));
+ die_on_error("snd_pcm_sw_params_set_start_threshold", snd_pcm_sw_params_set_start_threshold(pcm_handle, sw_params, num_periods * period_size / 2));
+ die_on_error("snd_pcm_sw_params()", snd_pcm_sw_params(pcm_handle, sw_params));
+
+ die_on_error("snd_pcm_nonblock()", snd_pcm_nonblock(pcm_handle, 1));
+ die_on_error("snd_pcm_prepare()", snd_pcm_prepare(pcm_handle));
+
+}
+
+ALSAInput::~ALSAInput()
+{
+ die_on_error("snd_pcm_close()", snd_pcm_close(pcm_handle));
+}
+
+void ALSAInput::start_capture_thread()
+{
+ should_quit = false;
+ capture_thread = thread(&ALSAInput::capture_thread_func, this);
+}
+
+void ALSAInput::stop_capture_thread()
+{
+ should_quit = true;
+ capture_thread.join();
+}
+
+void ALSAInput::capture_thread_func()
+{
+ die_on_error("snd_pcm_start()", snd_pcm_start(pcm_handle));
+ uint64_t num_frames_output = 0;
+ while (!should_quit) {
+ int ret = snd_pcm_wait(pcm_handle, /*timeout=*/100);
+ if (ret == 0) continue; // Timeout.
+ if (ret == -EPIPE) {
+ fprintf(stderr, "[%s] ALSA overrun\n", device.c_str());
+ snd_pcm_prepare(pcm_handle);
+ snd_pcm_start(pcm_handle);
+ continue;
+ }
+ die_on_error("snd_pcm_wait()", ret);
+
+ snd_pcm_sframes_t frames = snd_pcm_readi(pcm_handle, buffer.get(), buffer_frames);
+ if (frames == -EPIPE) {
+ fprintf(stderr, "[%s] ALSA overrun\n", device.c_str());
+ snd_pcm_prepare(pcm_handle);
+ snd_pcm_start(pcm_handle);
+ continue;
+ }
+ if (frames == 0) {
+ fprintf(stderr, "snd_pcm_readi() returned 0\n");
+ break;
+ }
+ die_on_error("snd_pcm_readi()", frames);
+
+ const int64_t prev_pts = frames_to_pts(num_frames_output);
+ const int64_t pts = frames_to_pts(num_frames_output + frames);
+ audio_callback(buffer.get(), frames, audio_format, pts - prev_pts);
+ num_frames_output += frames;
+ }
+}
+
+int64_t ALSAInput::frames_to_pts(uint64_t n) const
+{
+ return (n * TIMEBASE) / sample_rate;
+}
+
+void ALSAInput::die_on_error(const char *func_name, int err)
+{
+ if (err < 0) {
+ fprintf(stderr, "[%s] %s: %s\n", device.c_str(), func_name, snd_strerror(err));
+ exit(1);
+ }
+}
+
projects / nageru / blobdiff