diff options
Diffstat (limited to 'scene/resources/audio_stream_wav.cpp')
-rw-r--r-- | scene/resources/audio_stream_wav.cpp | 73 |
1 files changed, 26 insertions, 47 deletions
diff --git a/scene/resources/audio_stream_wav.cpp b/scene/resources/audio_stream_wav.cpp index 08ebacc2b3..f9787dde2e 100644 --- a/scene/resources/audio_stream_wav.cpp +++ b/scene/resources/audio_stream_wav.cpp @@ -123,10 +123,8 @@ void AudioStreamPlaybackWAV::do_resample(const Depth *p_src, AudioFrame *p_dst, int16_t nibble, diff, step; p_ima_adpcm[i].last_nibble++; - const uint8_t *src_ptr = (const uint8_t *)base->data; - src_ptr += AudioStreamWAV::DATA_PAD; - uint8_t nbb = src_ptr[(p_ima_adpcm[i].last_nibble >> 1) * (is_stereo ? 2 : 1) + i]; + uint8_t nbb = p_src[(p_ima_adpcm[i].last_nibble >> 1) * (is_stereo ? 2 : 1) + i]; nibble = (p_ima_adpcm[i].last_nibble & 1) ? (nbb >> 4) : (nbb & 0xF); step = _ima_adpcm_step_table[p_ima_adpcm[i].step_index]; @@ -184,9 +182,8 @@ void AudioStreamPlaybackWAV::do_resample(const Depth *p_src, AudioFrame *p_dst, if (p_qoa->data_ofs != new_data_ofs) { p_qoa->data_ofs = new_data_ofs; - const uint8_t *src_ptr = (const uint8_t *)base->data; - src_ptr += p_qoa->data_ofs + AudioStreamWAV::DATA_PAD; - qoa_decode_frame(src_ptr, p_qoa->frame_len, &p_qoa->desc, p_qoa->dec, &p_qoa->dec_len); + const uint8_t *ofs_src = (uint8_t *)p_src + p_qoa->data_ofs; + qoa_decode_frame(ofs_src, p_qoa->frame_len, &p_qoa->desc, p_qoa->dec.ptr(), &p_qoa->dec_len); } uint32_t dec_idx = (interp_pos % QOA_FRAME_LEN) * p_qoa->desc.channels; @@ -267,7 +264,7 @@ void AudioStreamPlaybackWAV::do_resample(const Depth *p_src, AudioFrame *p_dst, } int AudioStreamPlaybackWAV::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) { - if (!base->data || !active) { + if (base->data.is_empty() || !active) { for (int i = 0; i < p_frames; i++) { p_buffer[i] = AudioFrame(0, 0); } @@ -300,7 +297,7 @@ int AudioStreamPlaybackWAV::mix(AudioFrame *p_buffer, float p_rate_scale, int p_ int64_t loop_end_fp = ((int64_t)base->loop_end << MIX_FRAC_BITS); int64_t length_fp = ((int64_t)len << MIX_FRAC_BITS); int64_t begin_limit = (base->loop_mode != AudioStreamWAV::LOOP_DISABLED) ? loop_begin_fp : 0; - int64_t end_limit = (base->loop_mode != AudioStreamWAV::LOOP_DISABLED) ? loop_end_fp : length_fp; + int64_t end_limit = (base->loop_mode != AudioStreamWAV::LOOP_DISABLED) ? loop_end_fp : length_fp - MIX_FRAC_LEN; bool is_stereo = base->stereo; int32_t todo = p_frames; @@ -324,8 +321,7 @@ int AudioStreamPlaybackWAV::mix(AudioFrame *p_buffer, float p_rate_scale, int p_ /* audio data */ - uint8_t *dataptr = (uint8_t *)base->data; - const void *data = dataptr + AudioStreamWAV::DATA_PAD; + const uint8_t *data = base->data.ptr() + AudioStreamWAV::DATA_PAD; AudioFrame *dst_buff = p_buffer; if (format == AudioStreamWAV::FORMAT_IMA_ADPCM) { @@ -479,15 +475,14 @@ Ref<AudioSamplePlayback> AudioStreamPlaybackWAV::get_sample_playback() const { void AudioStreamPlaybackWAV::set_sample_playback(const Ref<AudioSamplePlayback> &p_playback) { sample_playback = p_playback; + if (sample_playback.is_valid()) { + sample_playback->stream_playback = Ref<AudioStreamPlayback>(this); + } } AudioStreamPlaybackWAV::AudioStreamPlaybackWAV() {} -AudioStreamPlaybackWAV::~AudioStreamPlaybackWAV() { - if (qoa.dec) { - memfree(qoa.dec); - } -} +AudioStreamPlaybackWAV::~AudioStreamPlaybackWAV() {} ///////////////////// @@ -554,7 +549,7 @@ double AudioStreamWAV::get_length() const { break; case AudioStreamWAV::FORMAT_QOA: qoa_desc desc = {}; - qoa_decode_header((uint8_t *)data + DATA_PAD, data_bytes, &desc); + qoa_decode_header(data.ptr() + DATA_PAD, data_bytes, &desc); len = desc.samples * desc.channels; break; } @@ -572,22 +567,16 @@ bool AudioStreamWAV::is_monophonic() const { void AudioStreamWAV::set_data(const Vector<uint8_t> &p_data) { AudioServer::get_singleton()->lock(); - if (data) { - memfree(data); - data = nullptr; - data_bytes = 0; - } - int datalen = p_data.size(); - if (datalen) { - const uint8_t *r = p_data.ptr(); - int alloc_len = datalen + DATA_PAD * 2; - data = memalloc(alloc_len); //alloc with some padding for interpolation - memset(data, 0, alloc_len); - uint8_t *dataptr = (uint8_t *)data; - memcpy(dataptr + DATA_PAD, r, datalen); - data_bytes = datalen; - } + int src_data_len = p_data.size(); + + data.clear(); + + int alloc_len = src_data_len + DATA_PAD * 2; + data.resize(alloc_len); + memset(data.ptr(), 0, alloc_len); + memcpy(data.ptr() + DATA_PAD, p_data.ptr(), src_data_len); + data_bytes = src_data_len; AudioServer::get_singleton()->unlock(); } @@ -595,13 +584,9 @@ void AudioStreamWAV::set_data(const Vector<uint8_t> &p_data) { Vector<uint8_t> AudioStreamWAV::get_data() const { Vector<uint8_t> pv; - if (data) { + if (!data.is_empty()) { pv.resize(data_bytes); - { - uint8_t *w = pv.ptrw(); - uint8_t *dataptr = (uint8_t *)data; - memcpy(w, dataptr + DATA_PAD, data_bytes); - } + memcpy(pv.ptrw(), data.ptr() + DATA_PAD, data_bytes); } return pv; @@ -693,12 +678,12 @@ Ref<AudioStreamPlayback> AudioStreamWAV::instantiate_playback() { sample->base = Ref<AudioStreamWAV>(this); if (format == AudioStreamWAV::FORMAT_QOA) { - uint32_t ffp = qoa_decode_header((uint8_t *)data + DATA_PAD, data_bytes, &sample->qoa.desc); + uint32_t ffp = qoa_decode_header(data.ptr() + DATA_PAD, data_bytes, &sample->qoa.desc); ERR_FAIL_COND_V(ffp != 8, Ref<AudioStreamPlaybackWAV>()); sample->qoa.frame_len = qoa_max_frame_size(&sample->qoa.desc); int samples_len = (sample->qoa.desc.samples > QOA_FRAME_LEN ? QOA_FRAME_LEN : sample->qoa.desc.samples); - int alloc_len = sample->qoa.desc.channels * samples_len * sizeof(int16_t); - sample->qoa.dec = (int16_t *)memalloc(alloc_len); + int dec_len = sample->qoa.desc.channels * samples_len; + sample->qoa.dec.resize(dec_len); } return sample; @@ -780,10 +765,4 @@ void AudioStreamWAV::_bind_methods() { AudioStreamWAV::AudioStreamWAV() {} -AudioStreamWAV::~AudioStreamWAV() { - if (data) { - memfree(data); - data = nullptr; - data_bytes = 0; - } -} +AudioStreamWAV::~AudioStreamWAV() {} |