/* * Copyright (c) 2012 Jan Vesely * All rights reserved. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * * - Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * - Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer in the * documentation and/or other materials provided with the distribution. * - The name of the author may not be used to endorse or promote products * derived from this software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ /** @addtogroup audio * @brief HelenOS sound server * @{ */ /** @file */ #include #include #include #include #include #include "format.h" #define uint8_t_le2host(x) (x) #define host2uint8_t_le(x) (x) #define uint8_t_be2host(x) (x) #define host2uint8_t_be(x) (x) #define int8_t_le2host(x) (x) #define host2int8_t_le(x) (x) #define int16_t_le2host(x) uint16_t_le2host(x) #define host2int16_t_le(x) host2uint16_t_le(x) #define int32_t_le2host(x) uint32_t_le2host(x) #define host2int32_t_le(x) host2uint32_t_le(x) #define int8_t_be2host(x) (x) #define host2int8_t_be(x) (x) #define int16_t_be2host(x) uint16_t_be2host(x) #define host2int16_t_be(x) host2uint16_t_be(x) #define int32_t_be2host(x) uint32_t_be2host(x) #define host2int32_t_be(x) host2uint32_t_be(x) // TODO float endian? #define float_le2host(x) (x) #define float_be2host(x) (x) #define host2float_le(x) (x) #define host2float_be(x) (x) #define from(x, type, endian) (float)(type ## _ ## endian ## 2host(x)) #define to(x, type, endian) (float)(host2 ## type ## _ ## endian(x)) const pcm_format_t AUDIO_FORMAT_DEFAULT = { .channels = 2, .sampling_rate = 44100, .sample_format = PCM_SAMPLE_SINT16_LE, }; const pcm_format_t AUDIO_FORMAT_ANY = { .channels = 0, .sampling_rate = 0, .sample_format = 0, }; static float get_normalized_sample(const void *buffer, size_t size, unsigned frame, unsigned channel, const pcm_format_t *f); /** * Compare PCM format attribtues. * @param a Format description. * @param b Format description. * @return True if a and b describe the same format, false otherwise. */ bool pcm_format_same(const pcm_format_t *a, const pcm_format_t* b) { assert(a); assert(b); return a->sampling_rate == b->sampling_rate && a->channels == b->channels && a->sample_format == b->sample_format; } /** * Fill audio buffer with silence in the specified format. * @param dst Destination audio buffer. * @param size Size of the destination audio buffer. * @param f Pointer to the format description. */ void pcm_format_silence(void *dst, size_t size, const pcm_format_t *f) { #define SET_NULL(type, endian, nullv) \ do { \ type *buffer = dst; \ const size_t sample_count = size / sizeof(type); \ for (unsigned i = 0; i < sample_count; ++i) { \ buffer[i] = to((type)nullv, type, endian); \ } \ } while (0) switch (f->sample_format) { case PCM_SAMPLE_UINT8: SET_NULL(uint8_t, le, INT8_MIN); break; case PCM_SAMPLE_SINT8: SET_NULL(int8_t, le, 0); break; case PCM_SAMPLE_UINT16_LE: SET_NULL(uint16_t, le, INT16_MIN); break; case PCM_SAMPLE_SINT16_LE: SET_NULL(int16_t, le, 0); break; case PCM_SAMPLE_UINT16_BE: SET_NULL(uint16_t, be, INT16_MIN); break; case PCM_SAMPLE_SINT16_BE: SET_NULL(int16_t, be, 0); break; case PCM_SAMPLE_UINT32_LE: SET_NULL(uint32_t, le, INT32_MIN); break; case PCM_SAMPLE_SINT32_LE: SET_NULL(int32_t, le, 0); break; case PCM_SAMPLE_UINT32_BE: SET_NULL(uint32_t, be, INT32_MIN); break; case PCM_SAMPLE_SINT32_BE: SET_NULL(int32_t, le, 0); break; case PCM_SAMPLE_UINT24_32_LE: case PCM_SAMPLE_SINT24_32_LE: case PCM_SAMPLE_UINT24_32_BE: case PCM_SAMPLE_SINT24_32_BE: case PCM_SAMPLE_UINT24_LE: case PCM_SAMPLE_SINT24_LE: case PCM_SAMPLE_UINT24_BE: case PCM_SAMPLE_SINT24_BE: case PCM_SAMPLE_FLOAT32: default: ; } #undef SET_NULL } /** * Mix audio data of the same format and size. * @param dst Destination buffer * @param src Source buffer * @param size Size of both the destination and the source buffer * @param f Pointer to the format descriptor. * @return Error code. */ int pcm_format_mix(void *dst, const void *src, size_t size, const pcm_format_t *f) { return pcm_format_convert_and_mix(dst, size, src, size, f, f); } /** * Add and mix audio data. * @param dst Destination audio buffer * @param dst_size Size of the destination buffer * @param src Source audio buffer * @param src_size Size of the source buffer. * @param sf Pointer to the source format descriptor. * @param df Pointer to the destination format descriptor. * @return Error code. * * Buffers must contain entire frames. Destination buffer is always filled. * If there are not enough data in the source buffer silent data is assumed. */ int pcm_format_convert_and_mix(void *dst, size_t dst_size, const void *src, size_t src_size, const pcm_format_t *sf, const pcm_format_t *df) { if (!dst || !src || !sf || !df) return EINVAL; const size_t src_frame_size = pcm_format_frame_size(sf); if ((src_size % src_frame_size) != 0) return EINVAL; const size_t dst_frame_size = pcm_format_frame_size(df); if ((dst_size % dst_frame_size) != 0) return EINVAL; /* This is so ugly it eats kittens, and puppies, and ducklings, * and all little fluffy things... */ #define LOOP_ADD(type, endian, low, high) \ do { \ const unsigned frame_count = dst_size / dst_frame_size; \ for (size_t i = 0; i < frame_count; ++i) { \ for (unsigned j = 0; j < df->channels; ++j) { \ const float a = \ get_normalized_sample(dst, dst_size, i, j, df);\ const float b = \ get_normalized_sample(src, src_size, i, j, sf);\ float c = (a + b); \ if (c < -1.0) c = -1.0; \ if (c > 1.0) c = 1.0; \ c += 1.0; \ c *= ((float)(type)high - (float)(type)low) / 2; \ c += (float)(type)low; \ type *dst_buf = dst; \ const unsigned pos = i * df->channels + j; \ if (pos < (dst_size / sizeof(type))) \ dst_buf[pos] = to((type)c, type, endian); \ } \ } \ } while (0) switch (df->sample_format) { case PCM_SAMPLE_UINT8: LOOP_ADD(uint8_t, le, UINT8_MIN, UINT8_MAX); break; case PCM_SAMPLE_SINT8: LOOP_ADD(uint8_t, le, INT8_MIN, INT8_MAX); break; case PCM_SAMPLE_UINT16_LE: LOOP_ADD(uint16_t, le, UINT16_MIN, UINT16_MAX); break; case PCM_SAMPLE_SINT16_LE: LOOP_ADD(int16_t, le, INT16_MIN, INT16_MAX); break; case PCM_SAMPLE_UINT16_BE: LOOP_ADD(uint16_t, be, UINT16_MIN, UINT16_MAX); break; case PCM_SAMPLE_SINT16_BE: LOOP_ADD(int16_t, be, INT16_MIN, INT16_MAX); break; case PCM_SAMPLE_UINT24_32_LE: case PCM_SAMPLE_UINT32_LE: // TODO this are not right for 24bit LOOP_ADD(uint32_t, le, UINT32_MIN, UINT32_MAX); break; case PCM_SAMPLE_SINT24_32_LE: case PCM_SAMPLE_SINT32_LE: LOOP_ADD(int32_t, le, INT32_MIN, INT32_MAX); break; case PCM_SAMPLE_UINT24_32_BE: case PCM_SAMPLE_UINT32_BE: LOOP_ADD(uint32_t, be, UINT32_MIN, UINT32_MAX); break; case PCM_SAMPLE_SINT24_32_BE: case PCM_SAMPLE_SINT32_BE: LOOP_ADD(int32_t, be, INT32_MIN, INT32_MAX); break; case PCM_SAMPLE_UINT24_LE: case PCM_SAMPLE_SINT24_LE: case PCM_SAMPLE_UINT24_BE: case PCM_SAMPLE_SINT24_BE: case PCM_SAMPLE_FLOAT32: default: return ENOTSUP; } return EOK; #undef LOOP_ADD } /** * Converts all sample formats to float <-1,1> * @param buffer Audio data * @param size Size of the buffer * @param frame Index of the frame to read * @param channel Channel within the frame * @param f Pointer to a format descriptor * @return Normalized sample <-1,1>, 0.0 if the data could not be read */ static float get_normalized_sample(const void *buffer, size_t size, unsigned frame, unsigned channel, const pcm_format_t *f) { assert(f); assert(buffer); if (channel >= f->channels) return 0.0f; #define GET(type, endian, low, high) \ do { \ const type *src = buffer; \ const size_t sample_count = size / sizeof(type); \ const size_t sample_pos = frame * f->channels + channel; \ if (sample_pos >= sample_count) {\ return 0.0f; \ } \ float sample = from(src[sample_pos], type, endian); \ /* This makes it positive */ \ sample -= (float)(type)low; \ /* This makes it <0,2> */ \ sample /= (((float)(type)high - (float)(type)low) / 2.0f); \ return sample - 1.0f; \ } while (0) switch (f->sample_format) { case PCM_SAMPLE_UINT8: GET(uint8_t, le, UINT8_MIN, UINT8_MAX); case PCM_SAMPLE_SINT8: GET(int8_t, le, INT8_MIN, INT8_MAX); case PCM_SAMPLE_UINT16_LE: GET(uint16_t, le, UINT16_MIN, UINT16_MAX); case PCM_SAMPLE_SINT16_LE: GET(int16_t, le, INT16_MIN, INT16_MAX); case PCM_SAMPLE_UINT16_BE: GET(uint16_t, be, UINT16_MIN, UINT16_MAX); case PCM_SAMPLE_SINT16_BE: GET(int16_t, be, INT16_MIN, INT16_MAX); case PCM_SAMPLE_UINT24_32_LE: case PCM_SAMPLE_UINT32_LE: GET(uint32_t, le, UINT32_MIN, UINT32_MAX); case PCM_SAMPLE_SINT24_32_LE: case PCM_SAMPLE_SINT32_LE: GET(int32_t, le, INT32_MIN, INT32_MAX); case PCM_SAMPLE_UINT24_32_BE: case PCM_SAMPLE_UINT32_BE: GET(uint32_t, be, UINT32_MIN, UINT32_MAX); case PCM_SAMPLE_SINT24_32_BE: case PCM_SAMPLE_SINT32_BE: GET(int32_t, le, INT32_MIN, INT32_MAX); case PCM_SAMPLE_UINT24_LE: case PCM_SAMPLE_SINT24_LE: case PCM_SAMPLE_UINT24_BE: case PCM_SAMPLE_SINT24_BE: case PCM_SAMPLE_FLOAT32: default: ; } return 0; #undef GET } /** * @} */