/*************************************************************************/ /* audio_stream_sample.cpp */ /*************************************************************************/ /* This file is part of: */ /* GODOT ENGINE */ /* https://godotengine.org */ /*************************************************************************/ /* Copyright (c) 2007-2020 Juan Linietsky, Ariel Manzur. */ /* Copyright (c) 2014-2020 Godot Engine contributors (cf. AUTHORS.md). */ /* */ /* Permission is hereby granted, free of charge, to any person obtaining */ /* a copy of this software and associated documentation files (the */ /* "Software"), to deal in the Software without restriction, including */ /* without limitation the rights to use, copy, modify, merge, publish, */ /* distribute, sublicense, and/or sell copies of the Software, and to */ /* permit persons to whom the Software is furnished to do so, subject to */ /* the following conditions: */ /* */ /* The above copyright notice and this permission notice shall be */ /* included in all copies or substantial portions of the Software. */ /* */ /* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */ /* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */ /* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/ /* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */ /* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */ /* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */ /* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */ /*************************************************************************/ #include "audio_stream_sample.h" #include "core/io/marshalls.h" #include "core/os/file_access.h" void AudioStreamPlaybackSample::start(float p_from_pos) { if (base->format == AudioStreamSample::FORMAT_IMA_ADPCM) { //no seeking in IMA_ADPCM for (int i = 0; i < 2; i++) { ima_adpcm[i].step_index = 0; ima_adpcm[i].predictor = 0; ima_adpcm[i].loop_step_index = 0; ima_adpcm[i].loop_predictor = 0; ima_adpcm[i].last_nibble = -1; ima_adpcm[i].loop_pos = 0x7FFFFFFF; ima_adpcm[i].window_ofs = 0; } offset = 0; } else { seek(p_from_pos); } sign = 1; active = true; } void AudioStreamPlaybackSample::stop() { active = false; } bool AudioStreamPlaybackSample::is_playing() const { return active; } int AudioStreamPlaybackSample::get_loop_count() const { return 0; } float AudioStreamPlaybackSample::get_playback_position() const { return float(offset >> MIX_FRAC_BITS) / base->mix_rate; } void AudioStreamPlaybackSample::seek(float p_time) { if (base->format == AudioStreamSample::FORMAT_IMA_ADPCM) return; //no seeking in ima-adpcm float max = base->get_length(); if (p_time < 0) { p_time = 0; } else if (p_time >= max) { p_time = max - 0.001; } offset = uint64_t(p_time * base->mix_rate) << MIX_FRAC_BITS; } template <class Depth, bool is_stereo, bool is_ima_adpcm> void AudioStreamPlaybackSample::do_resample(const Depth *p_src, AudioFrame *p_dst, int64_t &offset, int32_t &increment, uint32_t amount, IMA_ADPCM_State *ima_adpcm) { // this function will be compiled branchless by any decent compiler int32_t final, final_r, next, next_r; while (amount) { amount--; int64_t pos = offset >> MIX_FRAC_BITS; if (is_stereo && !is_ima_adpcm) pos <<= 1; if (is_ima_adpcm) { int64_t sample_pos = pos + ima_adpcm[0].window_ofs; while (sample_pos > ima_adpcm[0].last_nibble) { static const int16_t _ima_adpcm_step_table[89] = { 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 }; static const int8_t _ima_adpcm_index_table[16] = { -1, -1, -1, -1, 2, 4, 6, 8, -1, -1, -1, -1, 2, 4, 6, 8 }; for (int i = 0; i < (is_stereo ? 2 : 1); i++) { int16_t nibble, diff, step; ima_adpcm[i].last_nibble++; const uint8_t *src_ptr = (const uint8_t *)base->data; src_ptr += AudioStreamSample::DATA_PAD; uint8_t nbb = src_ptr[(ima_adpcm[i].last_nibble >> 1) * (is_stereo ? 2 : 1) + i]; nibble = (ima_adpcm[i].last_nibble & 1) ? (nbb >> 4) : (nbb & 0xF); step = _ima_adpcm_step_table[ima_adpcm[i].step_index]; ima_adpcm[i].step_index += _ima_adpcm_index_table[nibble]; if (ima_adpcm[i].step_index < 0) ima_adpcm[i].step_index = 0; if (ima_adpcm[i].step_index > 88) ima_adpcm[i].step_index = 88; diff = step >> 3; if (nibble & 1) diff += step >> 2; if (nibble & 2) diff += step >> 1; if (nibble & 4) diff += step; if (nibble & 8) diff = -diff; ima_adpcm[i].predictor += diff; if (ima_adpcm[i].predictor < -0x8000) ima_adpcm[i].predictor = -0x8000; else if (ima_adpcm[i].predictor > 0x7FFF) ima_adpcm[i].predictor = 0x7FFF; /* store loop if there */ if (ima_adpcm[i].last_nibble == ima_adpcm[i].loop_pos) { ima_adpcm[i].loop_step_index = ima_adpcm[i].step_index; ima_adpcm[i].loop_predictor = ima_adpcm[i].predictor; } //printf("%i - %i - pred %i\n",int(ima_adpcm[i].last_nibble),int(nibble),int(ima_adpcm[i].predictor)); } } final = ima_adpcm[0].predictor; if (is_stereo) { final_r = ima_adpcm[1].predictor; } } else { final = p_src[pos]; if (is_stereo) final_r = p_src[pos + 1]; if (sizeof(Depth) == 1) { /* conditions will not exist anymore when compiled! */ final <<= 8; if (is_stereo) final_r <<= 8; } if (is_stereo) { next = p_src[pos + 2]; next_r = p_src[pos + 3]; } else { next = p_src[pos + 1]; } if (sizeof(Depth) == 1) { next <<= 8; if (is_stereo) next_r <<= 8; } int32_t frac = int64_t(offset & MIX_FRAC_MASK); final = final + ((next - final) * frac >> MIX_FRAC_BITS); if (is_stereo) final_r = final_r + ((next_r - final_r) * frac >> MIX_FRAC_BITS); } if (!is_stereo) { final_r = final; //copy to right channel if stereo } p_dst->l = final / 32767.0; p_dst->r = final_r / 32767.0; p_dst++; offset += increment; } } void AudioStreamPlaybackSample::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) { if (!base->data || !active) { for (int i = 0; i < p_frames; i++) { p_buffer[i] = AudioFrame(0, 0); } return; } int len = base->data_bytes; switch (base->format) { case AudioStreamSample::FORMAT_8_BITS: len /= 1; break; case AudioStreamSample::FORMAT_16_BITS: len /= 2; break; case AudioStreamSample::FORMAT_IMA_ADPCM: len *= 2; break; } if (base->stereo) { len /= 2; } /* some 64-bit fixed point precaches */ int64_t loop_begin_fp = ((int64_t)base->loop_begin << MIX_FRAC_BITS); 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 != AudioStreamSample::LOOP_DISABLED) ? loop_begin_fp : 0; int64_t end_limit = (base->loop_mode != AudioStreamSample::LOOP_DISABLED) ? loop_end_fp : length_fp; bool is_stereo = base->stereo; int32_t todo = p_frames; if (base->loop_mode == AudioStreamSample::LOOP_BACKWARD) { sign = -1; } float base_rate = AudioServer::get_singleton()->get_mix_rate(); float srate = base->mix_rate; srate *= p_rate_scale; float fincrement = srate / base_rate; int32_t increment = int32_t(fincrement * MIX_FRAC_LEN); increment *= sign; //looping AudioStreamSample::LoopMode loop_format = base->loop_mode; AudioStreamSample::Format format = base->format; /* audio data */ uint8_t *dataptr = (uint8_t *)base->data; const void *data = dataptr + AudioStreamSample::DATA_PAD; AudioFrame *dst_buff = p_buffer; if (format == AudioStreamSample::FORMAT_IMA_ADPCM) { if (loop_format != AudioStreamSample::LOOP_DISABLED) { ima_adpcm[0].loop_pos = loop_begin_fp >> MIX_FRAC_BITS; ima_adpcm[1].loop_pos = loop_begin_fp >> MIX_FRAC_BITS; loop_format = AudioStreamSample::LOOP_FORWARD; } } while (todo > 0) { int64_t limit = 0; int32_t target = 0, aux = 0; /** LOOP CHECKING **/ if (increment < 0) { /* going backwards */ if (loop_format != AudioStreamSample::LOOP_DISABLED && offset < loop_begin_fp) { /* loopstart reached */ if (loop_format == AudioStreamSample::LOOP_PING_PONG) { /* bounce ping pong */ offset = loop_begin_fp + (loop_begin_fp - offset); increment = -increment; sign *= -1; } else { /* go to loop-end */ offset = loop_end_fp - (loop_begin_fp - offset); } } else { /* check for sample not reaching beginning */ if (offset < 0) { active = false; break; } } } else { /* going forward */ if (loop_format != AudioStreamSample::LOOP_DISABLED && offset >= loop_end_fp) { /* loopend reached */ if (loop_format == AudioStreamSample::LOOP_PING_PONG) { /* bounce ping pong */ offset = loop_end_fp - (offset - loop_end_fp); increment = -increment; sign *= -1; } else { /* go to loop-begin */ if (format == AudioStreamSample::FORMAT_IMA_ADPCM) { for (int i = 0; i < 2; i++) { ima_adpcm[i].step_index = ima_adpcm[i].loop_step_index; ima_adpcm[i].predictor = ima_adpcm[i].loop_predictor; ima_adpcm[i].last_nibble = loop_begin_fp >> MIX_FRAC_BITS; } offset = loop_begin_fp; } else { offset = loop_begin_fp + (offset - loop_end_fp); } } } else { /* no loop, check for end of sample */ if (offset >= length_fp) { active = false; break; } } } /** MIXCOUNT COMPUTING **/ /* next possible limit (looppoints or sample begin/end */ limit = (increment < 0) ? begin_limit : end_limit; /* compute what is shorter, the todo or the limit? */ aux = (limit - offset) / increment + 1; target = (aux < todo) ? aux : todo; /* mix target is the shorter buffer */ /* check just in case */ if (target <= 0) { active = false; break; } todo -= target; switch (base->format) { case AudioStreamSample::FORMAT_8_BITS: { if (is_stereo) do_resample<int8_t, true, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm); else do_resample<int8_t, false, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm); } break; case AudioStreamSample::FORMAT_16_BITS: { if (is_stereo) do_resample<int16_t, true, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm); else do_resample<int16_t, false, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm); } break; case AudioStreamSample::FORMAT_IMA_ADPCM: { if (is_stereo) do_resample<int8_t, true, true>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm); else do_resample<int8_t, false, true>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm); } break; } dst_buff += target; } if (todo) { //bit was missing from mix int todo_ofs = p_frames - todo; for (int i = todo_ofs; i < p_frames; i++) { p_buffer[i] = AudioFrame(0, 0); } } } AudioStreamPlaybackSample::AudioStreamPlaybackSample() { active = false; offset = 0; sign = 1; } ///////////////////// void AudioStreamSample::set_format(Format p_format) { format = p_format; } AudioStreamSample::Format AudioStreamSample::get_format() const { return format; } void AudioStreamSample::set_loop_mode(LoopMode p_loop_mode) { loop_mode = p_loop_mode; } AudioStreamSample::LoopMode AudioStreamSample::get_loop_mode() const { return loop_mode; } void AudioStreamSample::set_loop_begin(int p_frame) { loop_begin = p_frame; } int AudioStreamSample::get_loop_begin() const { return loop_begin; } void AudioStreamSample::set_loop_end(int p_frame) { loop_end = p_frame; } int AudioStreamSample::get_loop_end() const { return loop_end; } void AudioStreamSample::set_mix_rate(int p_hz) { ERR_FAIL_COND(p_hz == 0); mix_rate = p_hz; } int AudioStreamSample::get_mix_rate() const { return mix_rate; } void AudioStreamSample::set_stereo(bool p_enable) { stereo = p_enable; } bool AudioStreamSample::is_stereo() const { return stereo; } float AudioStreamSample::get_length() const { int len = data_bytes; switch (format) { case AudioStreamSample::FORMAT_8_BITS: len /= 1; break; case AudioStreamSample::FORMAT_16_BITS: len /= 2; break; case AudioStreamSample::FORMAT_IMA_ADPCM: len *= 2; break; } if (stereo) { len /= 2; } return float(len) / mix_rate; } void AudioStreamSample::set_data(const PoolVector<uint8_t> &p_data) { AudioServer::get_singleton()->lock(); if (data) { AudioServer::get_singleton()->audio_data_free(data); data = NULL; data_bytes = 0; } int datalen = p_data.size(); if (datalen) { PoolVector<uint8_t>::Read r = p_data.read(); int alloc_len = datalen + DATA_PAD * 2; data = AudioServer::get_singleton()->audio_data_alloc(alloc_len); //alloc with some padding for interpolation zeromem(data, alloc_len); uint8_t *dataptr = (uint8_t *)data; copymem(dataptr + DATA_PAD, r.ptr(), datalen); data_bytes = datalen; } AudioServer::get_singleton()->unlock(); } PoolVector<uint8_t> AudioStreamSample::get_data() const { PoolVector<uint8_t> pv; if (data) { pv.resize(data_bytes); { PoolVector<uint8_t>::Write w = pv.write(); uint8_t *dataptr = (uint8_t *)data; copymem(w.ptr(), dataptr + DATA_PAD, data_bytes); } } return pv; } Error AudioStreamSample::save_to_wav(const String &p_path) { if (format == AudioStreamSample::FORMAT_IMA_ADPCM) { WARN_PRINTS("Saving IMA_ADPC samples are not supported yet"); return ERR_UNAVAILABLE; } int sub_chunk_2_size = data_bytes; //Subchunk2Size = Size of data in bytes // Format code // 1:PCM format (for 8 or 16 bit) // 3:IEEE float format int format_code = (format == FORMAT_IMA_ADPCM) ? 3 : 1; int n_channels = stereo ? 2 : 1; long sample_rate = mix_rate; int byte_pr_sample = 0; switch (format) { case AudioStreamSample::FORMAT_8_BITS: byte_pr_sample = 1; break; case AudioStreamSample::FORMAT_16_BITS: byte_pr_sample = 2; break; case AudioStreamSample::FORMAT_IMA_ADPCM: byte_pr_sample = 4; break; } String file_path = p_path; if (!(file_path.substr(file_path.length() - 4, 4) == ".wav")) { file_path += ".wav"; } FileAccessRef file = FileAccess::open(file_path, FileAccess::WRITE); //Overrides existing file if present ERR_FAIL_COND_V(!file, ERR_FILE_CANT_WRITE); // Create WAV Header file->store_string("RIFF"); //ChunkID file->store_32(sub_chunk_2_size + 36); //ChunkSize = 36 + SubChunk2Size (size of entire file minus the 8 bits for this and previous header) file->store_string("WAVE"); //Format file->store_string("fmt "); //Subchunk1ID file->store_32(16); //Subchunk1Size = 16 file->store_16(format_code); //AudioFormat file->store_16(n_channels); //Number of Channels file->store_32(sample_rate); //SampleRate file->store_32(sample_rate * n_channels * byte_pr_sample); //ByteRate file->store_16(n_channels * byte_pr_sample); //BlockAlign = NumChannels * BytePrSample file->store_16(byte_pr_sample * 8); //BitsPerSample file->store_string("data"); //Subchunk2ID file->store_32(sub_chunk_2_size); //Subchunk2Size // Add data PoolVector<uint8_t> data = get_data(); PoolVector<uint8_t>::Read read_data = data.read(); switch (format) { case AudioStreamSample::FORMAT_8_BITS: for (unsigned int i = 0; i < data_bytes; i++) { uint8_t data_point = (read_data[i] + 128); file->store_8(data_point); } break; case AudioStreamSample::FORMAT_16_BITS: for (unsigned int i = 0; i < data_bytes / 2; i++) { uint16_t data_point = decode_uint16(&read_data[i * 2]); file->store_16(data_point); } break; case AudioStreamSample::FORMAT_IMA_ADPCM: //Unimplemented break; } file->close(); return OK; } Ref<AudioStreamPlayback> AudioStreamSample::instance_playback() { Ref<AudioStreamPlaybackSample> sample; sample.instance(); sample->base = Ref<AudioStreamSample>(this); return sample; } String AudioStreamSample::get_stream_name() const { return ""; } void AudioStreamSample::_bind_methods() { ClassDB::bind_method(D_METHOD("set_data", "data"), &AudioStreamSample::set_data); ClassDB::bind_method(D_METHOD("get_data"), &AudioStreamSample::get_data); ClassDB::bind_method(D_METHOD("set_format", "format"), &AudioStreamSample::set_format); ClassDB::bind_method(D_METHOD("get_format"), &AudioStreamSample::get_format); ClassDB::bind_method(D_METHOD("set_loop_mode", "loop_mode"), &AudioStreamSample::set_loop_mode); ClassDB::bind_method(D_METHOD("get_loop_mode"), &AudioStreamSample::get_loop_mode); ClassDB::bind_method(D_METHOD("set_loop_begin", "loop_begin"), &AudioStreamSample::set_loop_begin); ClassDB::bind_method(D_METHOD("get_loop_begin"), &AudioStreamSample::get_loop_begin); ClassDB::bind_method(D_METHOD("set_loop_end", "loop_end"), &AudioStreamSample::set_loop_end); ClassDB::bind_method(D_METHOD("get_loop_end"), &AudioStreamSample::get_loop_end); ClassDB::bind_method(D_METHOD("set_mix_rate", "mix_rate"), &AudioStreamSample::set_mix_rate); ClassDB::bind_method(D_METHOD("get_mix_rate"), &AudioStreamSample::get_mix_rate); ClassDB::bind_method(D_METHOD("set_stereo", "stereo"), &AudioStreamSample::set_stereo); ClassDB::bind_method(D_METHOD("is_stereo"), &AudioStreamSample::is_stereo); ClassDB::bind_method(D_METHOD("save_to_wav", "path"), &AudioStreamSample::save_to_wav); ADD_PROPERTY(PropertyInfo(Variant::POOL_BYTE_ARRAY, "data", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_NOEDITOR), "set_data", "get_data"); ADD_PROPERTY(PropertyInfo(Variant::INT, "format", PROPERTY_HINT_ENUM, "8-Bit,16-Bit,IMA-ADPCM"), "set_format", "get_format"); ADD_PROPERTY(PropertyInfo(Variant::INT, "loop_mode", PROPERTY_HINT_ENUM, "Disabled,Forward,Ping-Pong,Backward"), "set_loop_mode", "get_loop_mode"); ADD_PROPERTY(PropertyInfo(Variant::INT, "loop_begin"), "set_loop_begin", "get_loop_begin"); ADD_PROPERTY(PropertyInfo(Variant::INT, "loop_end"), "set_loop_end", "get_loop_end"); ADD_PROPERTY(PropertyInfo(Variant::INT, "mix_rate"), "set_mix_rate", "get_mix_rate"); ADD_PROPERTY(PropertyInfo(Variant::BOOL, "stereo"), "set_stereo", "is_stereo"); BIND_ENUM_CONSTANT(FORMAT_8_BITS); BIND_ENUM_CONSTANT(FORMAT_16_BITS); BIND_ENUM_CONSTANT(FORMAT_IMA_ADPCM); BIND_ENUM_CONSTANT(LOOP_DISABLED); BIND_ENUM_CONSTANT(LOOP_FORWARD); BIND_ENUM_CONSTANT(LOOP_PING_PONG); BIND_ENUM_CONSTANT(LOOP_BACKWARD); } AudioStreamSample::AudioStreamSample() { format = FORMAT_8_BITS; loop_mode = LOOP_DISABLED; stereo = false; loop_begin = 0; loop_end = 0; mix_rate = 44100; data = NULL; data_bytes = 0; } AudioStreamSample::~AudioStreamSample() { if (data) { AudioServer::get_singleton()->audio_data_free(data); data = NULL; data_bytes = 0; } }