audio handling

This commit is contained in:
minjaesong
2025-09-16 22:23:31 +09:00
parent 47f93194a7
commit 9e8aeeb112

View File

@@ -188,6 +188,7 @@ typedef struct {
int mp2_packet_size;
int mp2_rate_index;
int target_audio_buffer_size;
double audio_frames_in_buffer;
// Subtitle processing
subtitle_entry_t *subtitles;
@@ -1244,7 +1245,6 @@ static int start_video_conversion(tav_encoder_t *enc) {
// Start audio conversion
static int start_audio_conversion(tav_encoder_t *enc) {
return 1;
if (!enc->has_audio) return 1;
char command[2048];
@@ -1563,16 +1563,23 @@ static int process_audio(tav_encoder_t *enc, int frame_num, FILE *output) {
int is_mono = (header[3] >> 6) == 3;
enc->mp2_rate_index = mp2_packet_size_to_rate_index(enc->mp2_packet_size, is_mono);
enc->target_audio_buffer_size = 4; // 4 audio packets in buffer
enc->audio_frames_in_buffer = 0.0;
}
// Calculate how much audio we need for this frame
double frame_duration = 1.0 / enc->fps;
double samples_per_frame = 32000.0 * frame_duration; // 32kHz sample rate
int target_buffer_samples = (int)(samples_per_frame * enc->target_audio_buffer_size);
int target_buffer_bytes = (target_buffer_samples * enc->mp2_packet_size) / 1152; // 1152 samples per MP2 frame
// Calculate how much audio time each frame represents (in seconds)
double frame_audio_time = 1.0 / enc->fps;
// Calculate how much audio time each MP2 packet represents
// MP2 frame contains 1152 samples at 32kHz = 0.036 seconds
#define MP2_SAMPLE_RATE 32000
double packet_audio_time = 1152.0 / MP2_SAMPLE_RATE;
// Estimate how many packets we consume per video frame
double packets_per_frame = frame_audio_time / packet_audio_time;
// Allocate MP2 buffer if needed
if (!enc->mp2_buffer) {
enc->mp2_buffer_size = target_buffer_bytes * 2; // Extra buffer space
enc->mp2_buffer_size = enc->mp2_packet_size * 2; // Space for multiple packets
enc->mp2_buffer = malloc(enc->mp2_buffer_size);
if (!enc->mp2_buffer) {
fprintf(stderr, "Failed to allocate audio buffer\n");
@@ -1580,34 +1587,71 @@ static int process_audio(tav_encoder_t *enc, int frame_num, FILE *output) {
}
}
// Read audio data
size_t bytes_to_read = target_buffer_bytes;
if (bytes_to_read > enc->audio_remaining) {
bytes_to_read = enc->audio_remaining;
}
if (bytes_to_read > enc->mp2_buffer_size) {
bytes_to_read = enc->mp2_buffer_size;
// Audio buffering strategy: maintain target buffer level
int packets_to_insert = 0;
if (frame_num == 0) {
// Prime buffer to target level initially
packets_to_insert = enc->target_audio_buffer_size;
enc->audio_frames_in_buffer = 0; // count starts from 0
if (enc->verbose) {
printf("Frame %d: Priming audio buffer with %d packets\n", frame_num, packets_to_insert);
}
} else {
// Simulate buffer consumption (fractional consumption per frame)
double old_buffer = enc->audio_frames_in_buffer;
enc->audio_frames_in_buffer -= packets_per_frame;
// Calculate how many packets we need to maintain target buffer level
// Only insert when buffer drops below target, and only insert enough to restore target
double target_level = (double)enc->target_audio_buffer_size;
if (enc->audio_frames_in_buffer < target_level) {
double deficit = target_level - enc->audio_frames_in_buffer;
// Insert packets to cover the deficit, but at least maintain minimum flow
packets_to_insert = (int)ceil(deficit);
// Cap at reasonable maximum to prevent excessive insertion
if (packets_to_insert > enc->target_audio_buffer_size) {
packets_to_insert = enc->target_audio_buffer_size;
}
if (enc->verbose) {
printf("Frame %d: Buffer low (%.2f->%.2f), deficit %.2f, inserting %d packets\n",
frame_num, old_buffer, enc->audio_frames_in_buffer, deficit, packets_to_insert);
}
} else if (enc->verbose && old_buffer != enc->audio_frames_in_buffer) {
printf("Frame %d: Buffer sufficient (%.2f->%.2f), no packets\n",
frame_num, old_buffer, enc->audio_frames_in_buffer);
}
}
size_t bytes_read = fread(enc->mp2_buffer, 1, bytes_to_read, enc->mp2_file);
if (bytes_read == 0) {
return 1; // No more audio
}
// Insert the calculated number of audio packets
for (int q = 0; q < packets_to_insert; q++) {
size_t bytes_to_read = enc->mp2_packet_size;
if (bytes_to_read > enc->audio_remaining) {
bytes_to_read = enc->audio_remaining;
}
// Write audio packet
uint8_t audio_packet_type = TAV_PACKET_AUDIO_MP2;
uint32_t audio_len = (uint32_t)bytes_read;
fwrite(&audio_packet_type, 1, 1, output);
fwrite(&audio_len, 4, 1, output);
fwrite(enc->mp2_buffer, 1, bytes_read, output);
size_t bytes_read = fread(enc->mp2_buffer, 1, bytes_to_read, enc->mp2_file);
if (bytes_read == 0) break;
// Track audio bytes written
enc->audio_remaining -= bytes_read;
// Write TAV MP2 audio packet
uint8_t audio_packet_type = TAV_PACKET_AUDIO_MP2;
uint32_t audio_len = (uint32_t)bytes_read;
fwrite(&audio_packet_type, 1, 1, output);
fwrite(&audio_len, 4, 1, output);
fwrite(enc->mp2_buffer, 1, bytes_read, output);
if (enc->verbose) {
printf("Frame %d: Audio packet %zu bytes (remaining: %zu)\n",
frame_num, bytes_read, enc->audio_remaining);
// Track audio bytes written
enc->audio_remaining -= bytes_read;
enc->audio_frames_in_buffer++;
if (frame_num == 0) {
enc->audio_frames_in_buffer = enc->target_audio_buffer_size / 2; // trick the buffer simulator so that it doesn't count the frame 0 priming
}
if (enc->verbose) {
printf("Audio packet %d: %zu bytes (buffer: %.2f packets)\n",
q, bytes_read, enc->audio_frames_in_buffer);
}
}
return 1;