Lightningbeam/lightningbeam-ui/lightningbeam-editor/examples/ffmpeg_test.rs

414 lines
14 KiB
Rust

/// Minimal test program to validate FFmpeg audio encoding workflow
///
/// This program tests encoding raw PCM samples to MP3 using ffmpeg-next.
/// Run with: cargo run --example ffmpeg_test
use std::path::Path;
fn main() -> Result<(), String> {
println!("Testing FFmpeg audio encoding...");
// Initialize FFmpeg
ffmpeg_next::init().map_err(|e| format!("Failed to initialize FFmpeg: {}", e))?;
// Test 1: List available encoders
println!("\nAvailable MP3 encoders:");
if let Some(encoder) = ffmpeg_next::encoder::find(ffmpeg_next::codec::Id::MP3) {
println!(" - Found MP3 encoder: {}", encoder.name());
} else {
println!(" - No MP3 encoder found!");
}
println!("\nAvailable AAC encoders:");
if let Some(encoder) = ffmpeg_next::encoder::find(ffmpeg_next::codec::Id::AAC) {
println!(" - Found AAC encoder: {}", encoder.name());
} else {
println!(" - No AAC encoder found!");
}
// Test 2: Create a simple MP3 encoder and encode silence
test_mp3_encoding()?;
// Test 3: Create a simple AAC encoder and encode silence
test_aac_encoding()?;
println!("\n✅ All tests passed!");
Ok(())
}
fn test_mp3_encoding() -> Result<(), String> {
println!("\nTest: Encoding 1 second of silence to MP3...");
// Output file
let output_path = "/tmp/test_silence.mp3";
// Generate 1 second of stereo silence at 44.1 kHz
let sample_rate = 44100;
let channels = 2;
let duration_secs = 1.0;
let num_samples = (sample_rate as f64 * duration_secs * channels as f64) as usize;
let pcm_samples: Vec<f32> = vec![0.0; num_samples]; // Silence
println!(" Generated {} PCM samples ({}Hz, {} channels, {:.1}s)",
num_samples, sample_rate, channels, duration_secs);
// Encode to MP3
encode_pcm_to_mp3(&pcm_samples, sample_rate, channels, 320, output_path)?;
// Check output file exists
if Path::new(output_path).exists() {
let metadata = std::fs::metadata(output_path).unwrap();
println!(" ✅ Created MP3 file: {} ({} bytes)", output_path, metadata.len());
} else {
return Err("MP3 file was not created!".to_string());
}
Ok(())
}
fn test_aac_encoding() -> Result<(), String> {
println!("\nTest: Encoding 1 second of silence to AAC...");
// Output file
let output_path = "/tmp/test_silence.m4a";
// Generate 1 second of stereo silence at 44.1 kHz
let sample_rate = 44100;
let channels = 2;
let duration_secs = 1.0;
let num_samples = (sample_rate as f64 * duration_secs * channels as f64) as usize;
let pcm_samples: Vec<f32> = vec![0.0; num_samples]; // Silence
println!(" Generated {} PCM samples ({}Hz, {} channels, {:.1}s)",
num_samples, sample_rate, channels, duration_secs);
// Encode to AAC
encode_pcm_to_aac(&pcm_samples, sample_rate, channels, 192, output_path)?;
// Check output file exists
if Path::new(output_path).exists() {
let metadata = std::fs::metadata(output_path).unwrap();
println!(" ✅ Created AAC file: {} ({} bytes)", output_path, metadata.len());
} else {
return Err("AAC file was not created!".to_string());
}
Ok(())
}
/// Encode raw PCM samples to MP3 using ffmpeg-next
fn encode_pcm_to_mp3(
samples: &[f32],
sample_rate: u32,
channels: u32,
bitrate_kbps: u32,
output_path: &str,
) -> Result<(), String> {
use ffmpeg_next as ffmpeg;
// Find MP3 encoder
let encoder_codec = ffmpeg::encoder::find(ffmpeg::codec::Id::MP3)
.ok_or("MP3 encoder not found")?;
println!(" Using encoder: {}", encoder_codec.name());
// Create output format context FIRST (like transcode example)
let mut output = ffmpeg::format::output(&output_path)
.map_err(|e| format!("Failed to create output file: {}", e))?;
// Don't use stream parameters - create encoder directly
// The stream was just added but has no parameters set yet
let mut encoder = ffmpeg::codec::Context::new_with_codec(encoder_codec)
.encoder()
.audio()
.map_err(|e| format!("Failed to create encoder: {}", e))?;
println!(" Created encoder directly from codec");
// Determine channel layout first
let channel_layout = match channels {
1 => ffmpeg::channel_layout::ChannelLayout::MONO,
2 => ffmpeg::channel_layout::ChannelLayout::STEREO,
_ => return Err(format!("Unsupported channel count: {}", channels)),
};
// Configure encoder with explicit format (required in ffmpeg-next 8.0)
encoder.set_rate(sample_rate as i32);
encoder.set_channel_layout(channel_layout);
// Set format to S16 Planar (s16p) which libmp3lame supports
use ffmpeg_next::format::sample::Type;
use ffmpeg_next::format::Sample;
encoder.set_format(Sample::I16(Type::Planar));
encoder.set_bit_rate((bitrate_kbps * 1000) as usize);
encoder.set_time_base(ffmpeg::Rational(1, sample_rate as i32));
println!(" Encoder configured: {}Hz, {} channels, {} kbps",
sample_rate, channels, bitrate_kbps);
println!(" Format before open: {:?}", encoder.format());
// Open encoder (like transcode-audio example)
let mut encoder = encoder.open_as(encoder_codec)
.map_err(|e| format!("Failed to open encoder: {}", e))?;
println!(" ✅ Encoder opened successfully!");
println!(" Opened encoder format: {:?}", encoder.format());
// Now add stream and set its parameters from the opened encoder
let mut stream = output.add_stream(encoder_codec)
.map_err(|e| format!("Failed to add stream: {}", e))?;
stream.set_parameters(&encoder);
// Write header
output.write_header()
.map_err(|e| format!("Failed to write header: {}", e))?;
println!(" Encoding {} samples...", samples.len());
// Convert interleaved f32 to planar i16
let num_frames = samples.len() / channels as usize;
let planar_samples = convert_to_planar_i16(samples, channels);
// Get encoder frame size
let frame_size = encoder.frame_size();
let samples_per_frame = if frame_size > 0 {
frame_size as usize
} else {
1152 // Default MP3 frame size
};
println!(" Frame size: {} samples", samples_per_frame);
// Encode in chunks
let mut samples_encoded = 0;
while samples_encoded < num_frames {
let samples_remaining = num_frames - samples_encoded;
let chunk_size = samples_remaining.min(samples_per_frame);
// Create audio frame
let mut frame = ffmpeg::frame::Audio::new(
ffmpeg::format::Sample::I16(ffmpeg::format::sample::Type::Planar),
chunk_size,
channel_layout,
);
frame.set_rate(sample_rate);
// Copy planar samples to frame
unsafe {
for ch in 0..channels as usize {
let plane = frame.data_mut(ch);
let offset = samples_encoded;
let src = &planar_samples[ch][offset..offset + chunk_size];
std::ptr::copy_nonoverlapping(
src.as_ptr() as *const u8,
plane.as_mut_ptr(),
chunk_size * std::mem::size_of::<i16>(),
);
}
}
// Send frame to encoder
encoder.send_frame(&frame)
.map_err(|e| format!("Failed to send frame: {}", e))?;
// Receive and write packets
receive_and_write_packets(&mut encoder, &mut output)?;
samples_encoded += chunk_size;
}
// Flush encoder
encoder.send_eof()
.map_err(|e| format!("Failed to send EOF: {}", e))?;
receive_and_write_packets(&mut encoder, &mut output)?;
// Write trailer
output.write_trailer()
.map_err(|e| format!("Failed to write trailer: {}", e))?;
println!(" Encoding complete - {} frames encoded", num_frames);
Ok(())
}
/// Convert interleaved f32 samples to planar i16 format
fn convert_to_planar_i16(interleaved: &[f32], channels: u32) -> Vec<Vec<i16>> {
let num_frames = interleaved.len() / channels as usize;
let mut planar = vec![vec![0i16; num_frames]; channels as usize];
for (i, chunk) in interleaved.chunks(channels as usize).enumerate() {
for (ch, &sample) in chunk.iter().enumerate() {
// Clamp and convert f32 (-1.0 to 1.0) to i16
let clamped = sample.max(-1.0).min(1.0);
planar[ch][i] = (clamped * 32767.0) as i16;
}
}
planar
}
/// Receive encoded packets and write to output
fn receive_and_write_packets(
encoder: &mut ffmpeg_next::encoder::Audio,
output: &mut ffmpeg_next::format::context::Output,
) -> Result<(), String> {
let mut encoded = ffmpeg_next::Packet::empty();
while encoder.receive_packet(&mut encoded).is_ok() {
encoded.set_stream(0);
encoded.write_interleaved(output)
.map_err(|e| format!("Failed to write packet: {}", e))?;
}
Ok(())
}
/// Encode raw PCM samples to AAC using ffmpeg-next
fn encode_pcm_to_aac(
samples: &[f32],
sample_rate: u32,
channels: u32,
bitrate_kbps: u32,
output_path: &str,
) -> Result<(), String> {
use ffmpeg_next as ffmpeg;
// Find AAC encoder
let encoder_codec = ffmpeg::encoder::find(ffmpeg::codec::Id::AAC)
.ok_or("AAC encoder not found")?;
println!(" Using encoder: {}", encoder_codec.name());
// Create output format context
let mut output = ffmpeg::format::output(&output_path)
.map_err(|e| format!("Failed to create output file: {}", e))?;
// Create encoder directly from codec
let mut encoder = ffmpeg::codec::Context::new_with_codec(encoder_codec)
.encoder()
.audio()
.map_err(|e| format!("Failed to create encoder: {}", e))?;
println!(" Created encoder directly from codec");
// Determine channel layout
let channel_layout = match channels {
1 => ffmpeg::channel_layout::ChannelLayout::MONO,
2 => ffmpeg::channel_layout::ChannelLayout::STEREO,
_ => return Err(format!("Unsupported channel count: {}", channels)),
};
// Configure encoder - AAC supports F32 Planar (fltp)
encoder.set_rate(sample_rate as i32);
encoder.set_channel_layout(channel_layout);
encoder.set_format(ffmpeg::format::Sample::F32(ffmpeg::format::sample::Type::Planar));
encoder.set_bit_rate((bitrate_kbps * 1000) as usize);
encoder.set_time_base(ffmpeg::Rational(1, sample_rate as i32));
println!(" Encoder configured: {}Hz, {} channels, {} kbps",
sample_rate, channels, bitrate_kbps);
println!(" Format before open: {:?}", encoder.format());
// Open encoder
let mut encoder = encoder.open_as(encoder_codec)
.map_err(|e| format!("Failed to open encoder: {}", e))?;
println!(" ✅ Encoder opened successfully!");
println!(" Opened encoder format: {:?}", encoder.format());
// Add stream and set parameters
{
let mut stream = output.add_stream(encoder_codec)
.map_err(|e| format!("Failed to add stream: {}", e))?;
stream.set_parameters(&encoder);
}
// Write header
output.write_header()
.map_err(|e| format!("Failed to write header: {}", e))?;
println!(" Encoding {} samples...", samples.len());
// Convert interleaved f32 to planar f32
let num_frames = samples.len() / channels as usize;
let planar_samples = convert_to_planar_f32(samples, channels);
// Get encoder frame size
let frame_size = encoder.frame_size();
let samples_per_frame = if frame_size > 0 {
frame_size as usize
} else {
1024 // Default AAC frame size
};
println!(" Frame size: {} samples", samples_per_frame);
// Encode in chunks
let mut samples_encoded = 0;
while samples_encoded < num_frames {
let samples_remaining = num_frames - samples_encoded;
let chunk_size = samples_remaining.min(samples_per_frame);
// Create audio frame
let mut frame = ffmpeg::frame::Audio::new(
ffmpeg::format::Sample::F32(ffmpeg::format::sample::Type::Planar),
chunk_size,
channel_layout,
);
frame.set_rate(sample_rate);
// Copy planar samples to frame
unsafe {
for ch in 0..channels as usize {
let plane = frame.data_mut(ch);
let offset = samples_encoded;
let src = &planar_samples[ch][offset..offset + chunk_size];
std::ptr::copy_nonoverlapping(
src.as_ptr() as *const u8,
plane.as_mut_ptr(),
chunk_size * std::mem::size_of::<f32>(),
);
}
}
// Send frame to encoder
encoder.send_frame(&frame)
.map_err(|e| format!("Failed to send frame: {}", e))?;
// Receive and write packets
receive_and_write_packets(&mut encoder, &mut output)?;
samples_encoded += chunk_size;
}
// Flush encoder
encoder.send_eof()
.map_err(|e| format!("Failed to send EOF: {}", e))?;
receive_and_write_packets(&mut encoder, &mut output)?;
// Write trailer
output.write_trailer()
.map_err(|e| format!("Failed to write trailer: {}", e))?;
println!(" Encoding complete - {} frames encoded", num_frames);
Ok(())
}
/// Convert interleaved f32 samples to planar f32 format
fn convert_to_planar_f32(interleaved: &[f32], channels: u32) -> Vec<Vec<f32>> {
let num_frames = interleaved.len() / channels as usize;
let mut planar = vec![vec![0.0f32; num_frames]; channels as usize];
for (i, chunk) in interleaved.chunks(channels as usize).enumerate() {
for (ch, &sample) in chunk.iter().enumerate() {
planar[ch][i] = sample;
}
}
planar
}