Move everything into a separate futatabi/ subdir, for the upcoming merge with Futatabi.

[nageru] / nageru / alsa_input.h

diff --git a/nageru/alsa_input.h b/nageru/alsa_input.h
new file mode 100644 (file)
index 0000000..060b921
--- /dev/null
+++ b/nageru/alsa_input.h
@@ -0,0 +1,78 @@
+#ifndef _ALSA_INPUT_H
+#define _ALSA_INPUT_H 1
+
+// ALSA sound input, running in a separate thread and sending audio back
+// in callbacks.
+//
+// Note: “frame” here generally refers to the ALSA definition of frame,
+// which is a set of samples, exactly one for each channel. The only exception
+// is in frame_length, where it means the TIMEBASE length of the buffer
+// as a whole, since that's what AudioMixer::add_audio() wants.
+
+#include <alsa/asoundlib.h>
+#include <stdint.h>
+#include <sys/types.h>
+#include <atomic>
+#include <chrono>
+#include <functional>
+#include <memory>
+#include <string>
+#include <thread>
+
+#include "bmusb/bmusb.h"
+#include "quittable_sleeper.h"
+
+class ALSAPool;
+
+class ALSAInput {
+public:
+       typedef std::function<bool(const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length, std::chrono::steady_clock::time_point ts)> audio_callback_t;
+
+       ALSAInput(const char *device, unsigned sample_rate, unsigned num_channels, audio_callback_t audio_callback, ALSAPool *parent_pool, unsigned internal_dev_index);
+       ~ALSAInput();
+
+       // If not called before start_capture_thread(), the capture thread
+       // will call it until it succeeds.
+       bool open_device();
+
+       // Not valid before the device has been successfully opened.
+       // NOTE: Might very well be different from the sample rate given to the
+       // constructor, since the card might not support the one you wanted.
+       unsigned get_sample_rate() const { return sample_rate; }
+
+       void start_capture_thread();
+       void stop_capture_thread();
+
+       // Set access, sample rate and format parameters on the given ALSA PCM handle.
+       // Returns the computed parameter set and the chosen sample rate. Note that
+       // sample_rate is an in/out parameter; you send in the desired rate,
+       // and ALSA picks one as close to that as possible.
+       static bool set_base_params(const char *device_name, snd_pcm_t *pcm_handle, snd_pcm_hw_params_t *hw_params, unsigned *sample_rate);
+
+private:
+       void capture_thread_func();
+       int64_t frames_to_pts(uint64_t n) const;
+
+       enum class CaptureEndReason {
+               REQUESTED_QUIT,
+               DEVICE_GONE,
+               OTHER_ERROR
+       };
+       CaptureEndReason do_capture();
+
+       std::string device;
+       unsigned sample_rate, num_channels, num_periods;
+       snd_pcm_uframes_t period_size;
+       snd_pcm_uframes_t buffer_frames;
+       bmusb::AudioFormat audio_format;
+       audio_callback_t audio_callback;
+
+       snd_pcm_t *pcm_handle = nullptr;
+       std::thread capture_thread;
+       QuittableSleeper should_quit;
+       std::unique_ptr<uint8_t[]> buffer;
+       ALSAPool *parent_pool;
+       unsigned internal_dev_index;
+};
+
+#endif  // !defined(_ALSA_INPUT_H)