X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=nageru%2Falsa_input.h;h=22420b9134c970442e7112020425f519faf161dd;hb=f9024d141398e69e7b4011becd3ebbe37eaa1776;hp=060b9212690fce317f9dfac701af37d597af908a;hpb=9b7d691b4cc5db7dbfc18c82e86c1207fcac4722;p=nageru

diff --git a/nageru/alsa_input.h b/nageru/alsa_input.h
index 060b921..22420b9 100644
--- a/nageru/alsa_input.h
+++ b/nageru/alsa_input.h
@@ -5,9 +5,7 @@
 // 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.
+// which is a set of samples, exactly one for each channel.
 
 #include <alsa/asoundlib.h>
 #include <stdint.h>
@@ -26,7 +24,7 @@ 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;
+	typedef std::function<bool(const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, 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();
@@ -50,8 +48,8 @@ public:
 	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:
+	bool done_init = false;
 	void capture_thread_func();
-	int64_t frames_to_pts(uint64_t n) const;
 
 	enum class CaptureEndReason {
 		REQUESTED_QUIT,