initial commit

include/encoder.h

@@ -0,0 +1,67 @@
+#ifndef ENCODER_H
+#define ENCODER_H
+
+#include <stdint.h>
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libswscale/swscale.h>
+#include <libswresample/swresample.h>
+
+// Default Audio Bitrate for AAC/Opus (lossy codecs)
+#define DEFAULT_AUDIO_BIT_RATE 64000
+
+/* Video quality enumeration */
+typedef enum {
+    QUALITY_LOW,
+    QUALITY_MEDIUM,
+    QUALITY_HIGH
+} Quality;
+
+/* Audio codec enumeration */
+typedef enum {
+    AUDIO_CODEC_AAC,
+    AUDIO_CODEC_PCM,
+    AUDIO_CODEC_OPUS
+} AudioCodec;
+
+typedef struct {
+    AVFormatContext *fmt_ctx;
+    AVCodecContext *video_enc_ctx;
+    AVStream *video_stream;
+    AVCodecContext *audio_enc_ctx;
+    AVStream *audio_stream;
+    struct SwsContext *sws_ctx;
+    struct SwrContext *swr_ctx;    // audio resampling context
+    int frame_index;
+    int64_t audio_pts;           // running PTS (in samples) for audio
+    Quality quality;
+    char filename[512]; // The output filename
+} EncoderContext;
+
+/*
+ * Initializes the encoder.
+ * 'width' and 'height' are the recording dimensions (which may differ from native screen size).
+ * 'fps' is the capture framerate, and 'sample_rate' and 'channels' are audio parameters.
+ * 'audio_codec' selects the audio codec: AAC (lossy), PCM (lossless), or Opus (modern lossy).
+ * 'audio_bitrate' specifies the desired audio bitrate (e.g., DEFAULT_AUDIO_BIT_RATE for AAC/Opus).
+ * The output file is initially created in ~/Videos/Screenrecords/ with a generated name.
+ */
+EncoderContext* encoder_init(Quality quality, int width, int height, int fps, int sample_rate, int channels, AudioCodec audio_codec, int audio_bitrate);
+
+/* Encode one video frame (input data in RGB24 format) */
+int encoder_encode_video_frame(EncoderContext* ctx, uint8_t* data);
+
+/* Encode one audio frame with PCM data.
+   The input data is expected to be S16 interleaved.
+   Internally, the data is converted to the encoder’s sample format.
+*/
+int encoder_encode_audio_frame(EncoderContext* ctx, uint8_t* data, int size);
+
+/* Finalize the output file */
+int encoder_finalize(EncoderContext* ctx);
+
+/* Cleanup the encoder resources */
+void encoder_cleanup(EncoderContext* ctx);
+
+#endif // ENCODER_H
+