git.sesse.net Git - nageru/blob - alsa_input.cpp

   1
   2 #include "alsa_input.h"
   3
   4 using namespace std;
   5
   6 ALSAInput::ALSAInput(const char *device, unsigned sample_rate, unsigned num_channels, audio_callback_t audio_callback)
   7         : device(device), sample_rate(sample_rate), num_channels(num_channels), audio_callback(audio_callback)
   8 {
   9         die_on_error(device, snd_pcm_open(&pcm_handle,device, SND_PCM_STREAM_CAPTURE, 0));
  10
  11         // Set format.
  12         snd_pcm_hw_params_t *hw_params;
  13         snd_pcm_hw_params_alloca(&hw_params);
  14         die_on_error("snd_pcm_hw_params_any()", snd_pcm_hw_params_any(pcm_handle, hw_params));
  15         die_on_error("snd_pcm_hw_params_set_access()", snd_pcm_hw_params_set_access(pcm_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED));
  16         snd_pcm_format_mask_t *format_mask;
  17         snd_pcm_format_mask_alloca(&format_mask);
  18         snd_pcm_format_mask_set(format_mask, SND_PCM_FORMAT_S16_LE);
  19         snd_pcm_format_mask_set(format_mask, SND_PCM_FORMAT_S24_LE);
  20         snd_pcm_format_mask_set(format_mask, SND_PCM_FORMAT_S32_LE);
  21         die_on_error("snd_pcm_hw_params_set_format()", snd_pcm_hw_params_set_format_mask(pcm_handle, hw_params, format_mask));
  22         die_on_error("snd_pcm_hw_params_set_rate_near()", snd_pcm_hw_params_set_rate_near(pcm_handle, hw_params, &sample_rate, 0));
  23         die_on_error("snd_pcm_hw_params_set_channels()", snd_pcm_hw_params_set_channels(pcm_handle, hw_params, num_channels));
  24
  25         die_on_error("snd_pcm_hw_params_set_channels()", snd_pcm_hw_params_set_channels(pcm_handle, hw_params, num_channels));
  26
  27         // Fragment size of 64 samples (about 1 ms at 48 kHz; a frame at 60
  28         // fps/48 kHz is 800 samples.) We ask for 64 such periods in our buffer
  29         // (~85 ms buffer); more than that, and our jitter is probably so high
  30         // that the resampling queue can't keep up anyway.
  31         // The entire thing with periods and such is a bit mysterious to me;
  32         // seemingly I can get 96 frames at a time with no problems even if
  33         // the period size is 64 frames. And if I set num_periods to e.g. 1,
  34         // I can't have a big buffer.
  35         num_periods = 16;
  36         int dir = 0;
  37         die_on_error("snd_pcm_hw_params_set_periods_near()", snd_pcm_hw_params_set_periods_near(pcm_handle, hw_params, &num_periods, &dir));
  38         period_size = 64;
  39         dir = 0;
  40         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));
  41         buffer_frames = 64 * 64;
  42         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));
  43         die_on_error("snd_pcm_hw_params()", snd_pcm_hw_params(pcm_handle, hw_params));
  44         //snd_pcm_hw_params_free(hw_params);
  45
  46         // Figure out which format the card actually chose.
  47         die_on_error("snd_pcm_hw_params_current()", snd_pcm_hw_params_current(pcm_handle, hw_params));
  48         snd_pcm_format_t chosen_format;
  49         die_on_error("snd_pcm_hw_params_get_format()", snd_pcm_hw_params_get_format(hw_params, &chosen_format));
  50
  51         audio_format.num_channels = num_channels;
  52         audio_format.bits_per_sample = 0;
  53         switch (chosen_format) {
  54         case SND_PCM_FORMAT_S16_LE:
  55                 audio_format.bits_per_sample = 16;
  56                 break;
  57         case SND_PCM_FORMAT_S24_LE:
  58                 audio_format.bits_per_sample = 24;
  59                 break;
  60         case SND_PCM_FORMAT_S32_LE:
  61                 audio_format.bits_per_sample = 32;
  62                 break;
  63         default:
  64                 assert(false);
  65         }
  66         //printf("num_periods=%u period_size=%u buffer_frames=%u sample_rate=%u bits_per_sample=%d\n",
  67         //      num_periods, unsigned(period_size), unsigned(buffer_frames), sample_rate, audio_format.bits_per_sample);
  68
  69         buffer.reset(new uint8_t[buffer_frames * num_channels * audio_format.bits_per_sample / 8]);
  70
  71         snd_pcm_sw_params_t *sw_params;
  72         snd_pcm_sw_params_alloca(&sw_params);
  73         die_on_error("snd_pcm_sw_params_current()", snd_pcm_sw_params_current(pcm_handle, sw_params));
  74         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));
  75         die_on_error("snd_pcm_sw_params()", snd_pcm_sw_params(pcm_handle, sw_params));
  76
  77         die_on_error("snd_pcm_nonblock()", snd_pcm_nonblock(pcm_handle, 1));
  78         die_on_error("snd_pcm_prepare()", snd_pcm_prepare(pcm_handle));
  79
  80 }
  81
  82 ALSAInput::~ALSAInput()
  83 {
  84         die_on_error("snd_pcm_close()", snd_pcm_close(pcm_handle));
  85 }
  86
  87 void ALSAInput::start_capture_thread()
  88 {
  89         should_quit = false;
  90         capture_thread = thread(&ALSAInput::capture_thread_func, this);
  91 }
  92
  93 void ALSAInput::stop_capture_thread()
  94 {
  95         should_quit = true;
  96         capture_thread.join();
  97 }
  98
  99 void ALSAInput::capture_thread_func()
 100 {
 101         die_on_error("snd_pcm_start()", snd_pcm_start(pcm_handle));
 102         uint64_t num_frames_output = 0;
 103         while (!should_quit) {
 104                 int ret = snd_pcm_wait(pcm_handle, /*timeout=*/100);
 105                 if (ret == 0) continue;  // Timeout.
 106                 if (ret == -EPIPE) {
 107                         fprintf(stderr, "[%s] ALSA overrun\n", device.c_str());
 108                         snd_pcm_prepare(pcm_handle);
 109                         snd_pcm_start(pcm_handle);
 110                         continue;
 111                 }
 112                 die_on_error("snd_pcm_wait()", ret);
 113
 114                 snd_pcm_sframes_t frames = snd_pcm_readi(pcm_handle, buffer.get(), buffer_frames);
 115                 if (frames == -EPIPE) {
 116                         fprintf(stderr, "[%s] ALSA overrun\n", device.c_str());
 117                         snd_pcm_prepare(pcm_handle);
 118                         snd_pcm_start(pcm_handle);
 119                         continue;
 120                 }
 121                 if (frames == 0) {
 122                         fprintf(stderr, "snd_pcm_readi() returned 0\n");
 123                         break;
 124                 }
 125                 die_on_error("snd_pcm_readi()", frames);
 126
 127                 const int64_t prev_pts = frames_to_pts(num_frames_output);
 128                 const int64_t pts = frames_to_pts(num_frames_output + frames);
 129                 audio_callback(buffer.get(), frames, audio_format, pts - prev_pts);
 130                 num_frames_output += frames;
 131         }
 132 }
 133
 134 int64_t ALSAInput::frames_to_pts(uint64_t n) const
 135 {
 136         return (n * TIMEBASE) / sample_rate;
 137 }
 138
 139 void ALSAInput::die_on_error(const char *func_name, int err)
 140 {
 141         if (err < 0) {
 142                 fprintf(stderr, "[%s] %s: %s\n", device.c_str(), func_name, snd_strerror(err));
 143                 exit(1);
 144         }
 145 }
 146