Remove the unused frame_length parameter to AudioMixer::add_audio().

[nageru] / nageru / alsa_input.h

diff --git a/nageru/alsa_input.h b/nageru/alsa_input.h
index 060b9212690fce317f9dfac701af37d597af908a..825d4b686d05c11a6ffddcb17510f61a9f361dbe 100644 (file)
--- 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();
@@ -51,7 +49,6 @@ public:
 
 private:
        void capture_thread_func();
-       int64_t frames_to_pts(uint64_t n) const;
 
        enum class CaptureEndReason {
                REQUESTED_QUIT,