AgsWave and AgsBuffer
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AgsWave and AgsBuffer

The wave objects stores your buffers as a GList. There are analogous to notation functions to add or remove buffers.

AgsAudio holds a sorted list of AgsWave objects, gint ags_wave_sort_func(gconstpointer, gconstpointer) does the actual sorting. You can use it with GList* g_list_insert_sorted(GList*, gpointer, GCompareFunc).

AgsWave holds a sorted list of AgsBuffer objects, gint ags_buffer_sort_func (gconstpointer, gconstpointer) does the actual sorting. You can use it with GList* g_list_insert_sorted(GList*, gpointer, GCompareFunc). AgsWave:timestamp uses sample position with matching samplerate. As using void ags_timestamp_set_ags_offset (AgsTimestamp*, guint64) ags_offset equals 0 is your very first sample. You have to introduce after AGS_WAVE_DEFAULT_BUFFER_LENGTH * samplerate samples a new AgsWave object. The actual playback recall does bisect AgsWave and AgsBuffer in order to get current playing audio data.

AgsBuffer:data contains your actual audio data of AgsBuffer:format type. AgsBuffer:x is the actual sample position with matching samplerate.

Note audio effects are not applied to AgsWave but to AgsAudioSignal. The program flow is as following:

  1. ags-fx-playback does feed AgsWave to AgsAudioSignal of AgsInput.
  2. ags-fx-buffer does buffer AgsAudioSignal from AgsInput to AgsOutput.
  3. Another AgsAudio containing ags-fx-playback, then it plays it on your soundcard. Assumed you prior linked the the audio tree.

In this example, we first read audio data from 2 different files and concat the returned AgsWave objects. Note if you want to read multi-channel data, you have to modify the example with a for loop or such, to copy overlapping AgsBuffer. AgsBuffer:x shall be unique for specific audio channel.

Example 5.3. Concat AgsWave

#include <glib.h>
#include <glib-object.h>

#include <ags/libags.h>
#include <ags/libags-audio.h>

#define FILENAME_A "test_000.wav"
#define FILENAME_B "test_001.wav"

AgsAudio *audio;
AgsAudioFile *audio_file;

AgsTimestamp *timestamp_a, *timestamp_b;

AgsApplicationContext *application_context;

GObject *current_soundcard;

xmlNode *clipboard;

GList *start_soundcard;
GList *start_wave_a, *end_wave_a;
GList *start_wave_b;

guint64 file_a_frame_count;
guint audio_channels;
guint output_pads, input_pads;
guint i;

/* get application context and soundcard */
application_context = ags_application_context_get_instance();

start_soundcard = ags_sound_provider_get_soundcard(AGS_SOUND_PROVIDER(application_context));

current_soundcard = start_soundcard->data;

audio_channels = 1;

output_pads = 1;
input_pads = 1;

audio = ags_audio_new(current_soundcard);
ags_audio_set_flags(audio,
		    (AGS_AUDIO_SYNC |
		     AGS_AUDIO_OUTPUT_HAS_RECYCLING |
		     AGS_AUDIO_INPUT_HAS_RECYCLING));

ags_audio_set_audio_channels(audio,
                             audio_channels);

ags_audio_set_pads(audio,
                   AGS_TYPE_OUTPUT,
                   output_pads);
ags_audio_set_pads(audio,
                   AGS_TYPE_INPUT,
                   input_pads);

/* open first audio file */
audio_file = ags_audio_file_new(FILENAME_A,
				current_soundcard,
				-1);
ags_audio_file_open(audio_file);

ags_sound_resource_info(AGS_SOUND_RESOURCE(audio_file->sound_resource),
			&file_a_frame_count,
			NULL, NULL);

start_wave_a = ags_sound_resource_read_wave(AGS_SOUND_RESOURCE(audio_file->sound_resource),
					    current_soundcard,
					    0, // change to -1 for all audio channels
					    0,
					    0.0, 0);

/* open second audio file */
audio_file = ags_audio_file_new(FILENAME_B,
				current_soundcard,
				-1);
ags_audio_file_open(audio_file);

start_wave_b = ags_sound_resource_read_wave(AGS_SOUND_RESOURCE(audio_file->sound_resource),
					    current_soundcard,
					    0, // change to -1 for all audio channels
					    file_a_frame_count,
					    0.0, 0);

/* concat AgsWave */
audio->wave = start_wave_a;

end_wave_a = g_list_last(start_wave_a);

timestamp_a = ags_wave_get_timestamp(end_wave_a->data);

timestamp_b = ags_wave_get_timestamp(start_wave_b->data);

if(ags_timestamp_get_ags_offset(timestamp_a) == ags_timestamp_get_ags_offset(timestamp_b)){
  GList *start_buffer_a, *end_buffer_a;
  GList *start_buffer_b, *buffer_b;
  
  start_buffer_a = ags_wave_get_buffer(start_wave_a->data);

  end_buffer_a = g_list_last(start_buffer_a->data);
  
  buffer_b = 
    start_buffer_b = ags_wave_get_buffer(start_wave_b->data);

  if(ags_buffer_get_x(buffer_b->data) == ags_buffer_get_x(end_buffer_a->data)){
    AgsBuffer *current_mix_buffer_b;

    current_mix_buffer_b = start_buffer_b->data;

    start_buffer_b = start_buffer_b->next;
    
    ags_audio_buffer_util_copy_buffer_to_buffer(AGS_BUFFER(start_buffer_a->data)->data, 1, 0,
						current_mix_buffer_b->data, 1, 0,
						buffer_size, ags_audio_buffer_util_get_copy_mode(ags_audio_buffer_util_format_from_soundcard(ags_buffer_get_format(start_buffer_a->data)),
												 ags_audio_buffer_util_format_from_soundcard(ags_buffer_get_format(current_mix_buffer_b))));
    
    end_buffer_a->next = start_buffer_b;

    if(start_buffer_b != NULL){
      start_buffer_b->prev = end_buffer_a;
    }
  }else{
    end_buffer_a->next = start_buffer_b;
    start_buffer_b->prev = end_buffer_a;
  }
}else{
  end_wave_a->next = start_wave_b;
  start_wave_b->prev = end_wave_a;
}

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