web: Implement ADPCM stream sounds on web
This commit is contained in:
Mike Welsh
parent
eaea6aaf20
commit
f07dfb4eb6
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@ -28,6 +28,7 @@ struct StreamData {
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audio_data: Vec<u8>,
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num_sample_frames: u32,
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samples_per_block: u32,
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adpcm_block_offsets: Vec<usize>,
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}
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type AudioBufferPtr = Rc<RefCell<web_sys::AudioBuffer>>;
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@ -214,31 +215,12 @@ impl WebAudioBackend {
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format: &swf::SoundFormat,
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audio_data: &[u8],
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num_sample_frames: u32,
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adpcm_block_offsets: Option<&[usize]>,
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) -> AudioBufferPtr {
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if format.compression == AudioCompression::Mp3 {
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return self.decompress_mp3_to_audio_buffer(format, audio_data, num_sample_frames);
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}
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// This sucks. Firefox doesn't like 5512Hz sample rate, so manually double up the samples.
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// 5512Hz should be relatively rare.
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let audio_buffer = if format.sample_rate > 5512 {
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self.context
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.create_buffer(
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if format.is_stereo { 2 } else { 1 },
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num_sample_frames,
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f32::from(format.sample_rate),
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)
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.unwrap()
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} else {
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self.context
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.create_buffer(
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if format.is_stereo { 2 } else { 1 },
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num_sample_frames * 2,
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11025.0,
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)
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.unwrap()
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};
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match format.compression {
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AudioCompression::Uncompressed => {
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// TODO: Check for is_16_bit.
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@ -259,16 +241,31 @@ impl WebAudioBackend {
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}
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}
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AudioCompression::Adpcm => {
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let mut decoder =
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AdpcmDecoder::new(audio_data, format.is_stereo, format.sample_rate);
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if format.is_stereo {
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while let Some(frame) = decoder.next() {
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let (l, r) = (frame[0], frame[1]);
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self.left_samples.push(f32::from(l) / 32767.0);
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self.right_samples.push(f32::from(r) / 32767.0);
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// For stream sounds, the ADPCM header is included in each block,
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// so we must recreate the decoder for each block.
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// Event sounds don't have this issue.
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let full = [0, audio_data.len()];
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let adpcm_block_offsets = adpcm_block_offsets.unwrap_or(&full);
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self.left_samples.clear();
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self.right_samples.clear();
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for block in adpcm_block_offsets.windows(2) {
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let start = block[0];
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let end = block[1];
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let mut decoder = AdpcmDecoder::new(
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&audio_data[start..end],
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format.is_stereo,
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format.sample_rate,
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);
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if format.is_stereo {
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while let Some(frame) = decoder.next() {
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let (l, r) = (frame[0], frame[1]);
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self.left_samples.push(f32::from(l) / 32767.0);
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self.right_samples.push(f32::from(r) / 32767.0);
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}
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} else {
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self.left_samples
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.extend(decoder.map(|n| f32::from(n[0]) / 32767.0));
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}
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} else {
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self.left_samples = decoder.map(|n| f32::from(n[0]) / 32767.0).collect();
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}
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}
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_ => unimplemented!(),
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@ -293,6 +290,27 @@ impl WebAudioBackend {
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}
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}
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// This sucks. Firefox doesn't like 5512Hz sample rate, so manually double up the samples.
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// 5512Hz should be relatively rare.
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let num_sample_frames = self.left_samples.len() as u32;
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let audio_buffer = if format.sample_rate > 5512 {
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self.context
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.create_buffer(
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if format.is_stereo { 2 } else { 1 },
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num_sample_frames,
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f32::from(format.sample_rate),
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)
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.unwrap()
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} else {
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self.context
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.create_buffer(
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if format.is_stereo { 2 } else { 1 },
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num_sample_frames * 2,
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11025.0,
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)
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.unwrap()
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};
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audio_buffer
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.copy_to_channel(&mut self.left_samples, 0)
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.unwrap();
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@ -406,6 +424,7 @@ impl AudioBackend for WebAudioBackend {
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&sound.format,
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data,
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sound.num_samples,
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None,
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)),
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};
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Ok(self.sounds.insert(sound))
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@ -423,6 +442,7 @@ impl AudioBackend for WebAudioBackend {
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audio_data: vec![],
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num_sample_frames: 0,
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samples_per_block: stream_info.num_samples_per_block.into(),
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adpcm_block_offsets: vec![],
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});
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}
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@ -444,6 +464,13 @@ impl AudioBackend for WebAudioBackend {
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// Second two bytes = 'latency seek' (amount to skip when seeking to this frame)
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stream.audio_data.extend_from_slice(&audio_data[4..]);
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}
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AudioCompression::Adpcm => {
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// For ADPCM data, we must keep track of where each block starts,
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// so that we read the header in each block.
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stream.num_sample_frames += stream.samples_per_block;
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stream.adpcm_block_offsets.push(stream.audio_data.len());
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stream.audio_data.extend_from_slice(audio_data);
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}
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_ => {
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// TODO: This is a guess and will vary slightly from block to block!
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stream.num_sample_frames += stream.samples_per_block;
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@ -453,12 +480,18 @@ impl AudioBackend for WebAudioBackend {
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}
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fn preload_sound_stream_end(&mut self, clip_id: swf::CharacterId) {
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if let Some(stream) = self.stream_data.remove(&clip_id) {
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if let Some(mut stream) = self.stream_data.remove(&clip_id) {
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if !stream.audio_data.is_empty() {
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let audio_buffer = self.decompress_to_audio_buffer(
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&stream.format,
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&stream.audio_data[..],
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stream.num_sample_frames,
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if stream.format.compression == AudioCompression::Adpcm {
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stream.adpcm_block_offsets.push(stream.audio_data.len());
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Some(&stream.adpcm_block_offsets[..])
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} else {
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None
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},
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);
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let handle = self.sounds.insert(Sound {
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format: stream.format,
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