From 30f41c02aec763d32e62351452da9ef582bc3472 Mon Sep 17 00:00:00 2001 From: 3gg <3gg@shellblade.net> Date: Fri, 6 Mar 2026 13:30:59 -0800 Subject: Move contrib libraries to contrib repo --- contrib/SDL-3.2.8/src/audio/SDL_audiocvt.c | 1381 ---------------------------- 1 file changed, 1381 deletions(-) delete mode 100644 contrib/SDL-3.2.8/src/audio/SDL_audiocvt.c (limited to 'contrib/SDL-3.2.8/src/audio/SDL_audiocvt.c') diff --git a/contrib/SDL-3.2.8/src/audio/SDL_audiocvt.c b/contrib/SDL-3.2.8/src/audio/SDL_audiocvt.c deleted file mode 100644 index f751b0e..0000000 --- a/contrib/SDL-3.2.8/src/audio/SDL_audiocvt.c +++ /dev/null @@ -1,1381 +0,0 @@ -/* - Simple DirectMedia Layer - Copyright (C) 1997-2025 Sam Lantinga - - This software is provided 'as-is', without any express or implied - warranty. In no event will the authors be held liable for any damages - arising from the use of this software. - - Permission is granted to anyone to use this software for any purpose, - including commercial applications, and to alter it and redistribute it - freely, subject to the following restrictions: - - 1. The origin of this software must not be misrepresented; you must not - claim that you wrote the original software. If you use this software - in a product, an acknowledgment in the product documentation would be - appreciated but is not required. - 2. Altered source versions must be plainly marked as such, and must not be - misrepresented as being the original software. - 3. This notice may not be removed or altered from any source distribution. -*/ -#include "SDL_internal.h" - -#include "SDL_sysaudio.h" - -#include "SDL_audioqueue.h" -#include "SDL_audioresample.h" - -#ifndef SDL_INT_MAX -#define SDL_INT_MAX ((int)(~0u>>1)) -#endif - -#ifdef SDL_SSE3_INTRINSICS -// Convert from stereo to mono. Average left and right. -static void SDL_TARGETING("sse3") SDL_ConvertStereoToMono_SSE3(float *dst, const float *src, int num_frames) -{ - LOG_DEBUG_AUDIO_CONVERT("stereo", "mono (using SSE3)"); - - const __m128 divby2 = _mm_set1_ps(0.5f); - int i = num_frames; - - /* Do SSE blocks as long as we have 16 bytes available. - Just use unaligned load/stores, if the memory at runtime is - aligned it'll be just as fast on modern processors */ - while (i >= 4) { // 4 * float32 - _mm_storeu_ps(dst, _mm_mul_ps(_mm_hadd_ps(_mm_loadu_ps(src), _mm_loadu_ps(src + 4)), divby2)); - i -= 4; - src += 8; - dst += 4; - } - - // Finish off any leftovers with scalar operations. - while (i) { - *dst = (src[0] + src[1]) * 0.5f; - dst++; - i--; - src += 2; - } -} -#endif - -#ifdef SDL_SSE_INTRINSICS -// Convert from mono to stereo. Duplicate to stereo left and right. -static void SDL_TARGETING("sse") SDL_ConvertMonoToStereo_SSE(float *dst, const float *src, int num_frames) -{ - LOG_DEBUG_AUDIO_CONVERT("mono", "stereo (using SSE)"); - - // convert backwards, since output is growing in-place. - src += (num_frames-4) * 1; - dst += (num_frames-4) * 2; - - /* Do SSE blocks as long as we have 16 bytes available. - Just use unaligned load/stores, if the memory at runtime is - aligned it'll be just as fast on modern processors */ - // convert backwards, since output is growing in-place. - int i = num_frames; - while (i >= 4) { // 4 * float32 - const __m128 input = _mm_loadu_ps(src); // A B C D - _mm_storeu_ps(dst, _mm_unpacklo_ps(input, input)); // A A B B - _mm_storeu_ps(dst + 4, _mm_unpackhi_ps(input, input)); // C C D D - i -= 4; - src -= 4; - dst -= 8; - } - - // Finish off any leftovers with scalar operations. - src += 3; - dst += 6; // adjust for smaller buffers. - while (i) { // convert backwards, since output is growing in-place. - const float srcFC = src[0]; - dst[1] /* FR */ = srcFC; - dst[0] /* FL */ = srcFC; - i--; - src--; - dst -= 2; - } -} -#endif - -// Include the autogenerated channel converters... -#include "SDL_audio_channel_converters.h" - -static bool SDL_IsSupportedAudioFormat(const SDL_AudioFormat fmt) -{ - switch (fmt) { - case SDL_AUDIO_U8: - case SDL_AUDIO_S8: - case SDL_AUDIO_S16LE: - case SDL_AUDIO_S16BE: - case SDL_AUDIO_S32LE: - case SDL_AUDIO_S32BE: - case SDL_AUDIO_F32LE: - case SDL_AUDIO_F32BE: - return true; // supported. - - default: - break; - } - - return false; // unsupported. -} - -static bool SDL_IsSupportedChannelCount(const int channels) -{ - return ((channels >= 1) && (channels <= 8)); -} - -bool SDL_ChannelMapIsBogus(const int *chmap, int channels) -{ - if (chmap) { - for (int i = 0; i < channels; i++) { - const int mapping = chmap[i]; - if ((mapping < -1) || (mapping >= channels)) { - return true; - } - } - } - return false; -} - -bool SDL_ChannelMapIsDefault(const int *chmap, int channels) -{ - if (chmap) { - for (int i = 0; i < channels; i++) { - if (chmap[i] != i) { - return false; - } - } - } - return true; -} - -// Swizzle audio channels. src and dst can be the same pointer. It does not change the buffer size. -static void SwizzleAudio(const int num_frames, void *dst, const void *src, int channels, const int *map, SDL_AudioFormat fmt) -{ - const int bitsize = (int) SDL_AUDIO_BITSIZE(fmt); - - bool has_null_mappings = false; // !!! FIXME: calculate this when setting the channel map instead. - for (int i = 0; i < channels; i++) { - if (map[i] == -1) { - has_null_mappings = true; - break; - } - } - - #define CHANNEL_SWIZZLE(bits) { \ - Uint##bits *tdst = (Uint##bits *) dst; /* treat as UintX; we only care about moving bits and not the type here. */ \ - const Uint##bits *tsrc = (const Uint##bits *) src; \ - if (src != dst) { /* don't need to copy to a temporary frame first. */ \ - if (has_null_mappings) { \ - const Uint##bits silence = (Uint##bits) SDL_GetSilenceValueForFormat(fmt); \ - for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \ - for (int ch = 0; ch < channels; ch++) { \ - const int m = map[ch]; \ - tdst[ch] = (m == -1) ? silence : tsrc[m]; \ - } \ - } \ - } else { \ - for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \ - for (int ch = 0; ch < channels; ch++) { \ - tdst[ch] = tsrc[map[ch]]; \ - } \ - } \ - } \ - } else { \ - bool isstack; \ - Uint##bits *tmp = (Uint##bits *) SDL_small_alloc(int, channels, &isstack); /* !!! FIXME: allocate this when setting the channel map instead. */ \ - if (tmp) { \ - if (has_null_mappings) { \ - const Uint##bits silence = (Uint##bits) SDL_GetSilenceValueForFormat(fmt); \ - for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \ - for (int ch = 0; ch < channels; ch++) { \ - const int m = map[ch]; \ - tmp[ch] = (m == -1) ? silence : tsrc[m]; \ - } \ - for (int ch = 0; ch < channels; ch++) { \ - tdst[ch] = tmp[ch]; \ - } \ - } \ - } else { \ - for (int i = 0; i < num_frames; i++, tsrc += channels, tdst += channels) { \ - for (int ch = 0; ch < channels; ch++) { \ - tmp[ch] = tsrc[map[ch]]; \ - } \ - for (int ch = 0; ch < channels; ch++) { \ - tdst[ch] = tmp[ch]; \ - } \ - } \ - } \ - SDL_small_free(tmp, isstack); \ - } \ - } \ - } - - switch (bitsize) { - case 8: CHANNEL_SWIZZLE(8); break; - case 16: CHANNEL_SWIZZLE(16); break; - case 32: CHANNEL_SWIZZLE(32); break; - // we don't currently have int64 or double audio datatypes, so no `case 64` for now. - default: SDL_assert(!"Unsupported audio datatype size"); break; - } - - #undef CHANNEL_SWIZZLE -} - - -// This does type and channel conversions _but not resampling_ (resampling happens in SDL_AudioStream). -// This does not check parameter validity, (beyond asserts), it expects you did that already! -// All of this has to function as if src==dst==scratch (conversion in-place), but as a convenience -// if you're just going to copy the final output elsewhere, you can specify a different output pointer. -// -// The scratch buffer must be able to store `num_frames * CalculateMaxSampleFrameSize(src_format, src_channels, dst_format, dst_channels)` bytes. -// If the scratch buffer is NULL, this restriction applies to the output buffer instead. -// -// Since this is a convenient point that audio goes through even if it doesn't need format conversion, -// we also handle gain adjustment here, so we don't have to make another pass over the data later. -// Strictly speaking, this is also a "conversion". :) -void ConvertAudio(int num_frames, - const void *src, SDL_AudioFormat src_format, int src_channels, const int *src_map, - void *dst, SDL_AudioFormat dst_format, int dst_channels, const int *dst_map, - void *scratch, float gain) -{ - SDL_assert(src != NULL); - SDL_assert(dst != NULL); - SDL_assert(SDL_IsSupportedAudioFormat(src_format)); - SDL_assert(SDL_IsSupportedAudioFormat(dst_format)); - SDL_assert(SDL_IsSupportedChannelCount(src_channels)); - SDL_assert(SDL_IsSupportedChannelCount(dst_channels)); - - if (!num_frames) { - return; // no data to convert, quit. - } - -#if DEBUG_AUDIO_CONVERT - SDL_Log("SDL_AUDIO_CONVERT: Convert format %04x->%04x, channels %u->%u", src_format, dst_format, src_channels, dst_channels); -#endif - - const int dst_bitsize = (int) SDL_AUDIO_BITSIZE(dst_format); - const int dst_sample_frame_size = (dst_bitsize / 8) * dst_channels; - - const bool chmaps_match = (src_channels == dst_channels) && SDL_AudioChannelMapsEqual(src_channels, src_map, dst_map); - if (chmaps_match) { - src_map = dst_map = NULL; // NULL both these out so we don't do any unnecessary swizzling. - } - - /* Type conversion goes like this now: - - swizzle through source channel map to "standard" layout. - - byteswap to CPU native format first if necessary. - - convert to native Float32 if necessary. - - change channel count if necessary. - - convert to final data format. - - byteswap back to foreign format if necessary. - - swizzle through dest channel map from "standard" layout. - - The expectation is we can process data faster in float32 - (possibly with SIMD), and making several passes over the same - buffer is likely to be CPU cache-friendly, avoiding the - biggest performance hit in modern times. Previously we had - (script-generated) custom converters for every data type and - it was a bloat on SDL compile times and final library size. */ - - // swizzle input to "standard" format if necessary. - if (src_map) { - void* buf = scratch ? scratch : dst; // use scratch if available, since it has to be big enough to hold src, unless it's NULL, then dst has to be. - SwizzleAudio(num_frames, buf, src, src_channels, src_map, src_format); - src = buf; - } - - // see if we can skip float conversion entirely. - if ((src_channels == dst_channels) && (gain == 1.0f)) { - if (src_format == dst_format) { - // nothing to do, we're already in the right format, just copy it over if necessary. - if (dst_map) { - SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_format); - } else if (src != dst) { - SDL_memcpy(dst, src, num_frames * dst_sample_frame_size); - } - return; - } - - // just a byteswap needed? - if ((src_format ^ dst_format) == SDL_AUDIO_MASK_BIG_ENDIAN) { - if (dst_map) { // do this first, in case we duplicate channels, we can avoid an extra copy if src != dst. - SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_format); - src = dst; - } - ConvertAudioSwapEndian(dst, src, num_frames * dst_channels, dst_bitsize); - return; // all done. - } - } - - if (!scratch) { - scratch = dst; - } - - const bool srcconvert = src_format != SDL_AUDIO_F32; - const bool channelconvert = src_channels != dst_channels; - const bool dstconvert = dst_format != SDL_AUDIO_F32; - - // get us to float format. - if (srcconvert) { - void* buf = (channelconvert || dstconvert) ? scratch : dst; - ConvertAudioToFloat((float *) buf, src, num_frames * src_channels, src_format); - src = buf; - } - - // Gain adjustment - if (gain != 1.0f) { - float *buf = (float *)((channelconvert || dstconvert) ? scratch : dst); - const int total_samples = num_frames * src_channels; - if (src == buf) { - for (int i = 0; i < total_samples; i++) { - buf[i] *= gain; - } - } else { - float *fsrc = (float *)src; - for (int i = 0; i < total_samples; i++) { - buf[i] = fsrc[i] * gain; - } - } - src = buf; - } - - // Channel conversion - - if (channelconvert) { - SDL_AudioChannelConverter channel_converter; - SDL_AudioChannelConverter override = NULL; - - // SDL_IsSupportedChannelCount should have caught these asserts, or we added a new format and forgot to update the table. - SDL_assert(src_channels <= SDL_arraysize(channel_converters)); - SDL_assert(dst_channels <= SDL_arraysize(channel_converters[0])); - - channel_converter = channel_converters[src_channels - 1][dst_channels - 1]; - SDL_assert(channel_converter != NULL); - - // swap in some SIMD versions for a few of these. - if (channel_converter == SDL_ConvertStereoToMono) { - #ifdef SDL_SSE3_INTRINSICS - if (!override && SDL_HasSSE3()) { override = SDL_ConvertStereoToMono_SSE3; } - #endif - } else if (channel_converter == SDL_ConvertMonoToStereo) { - #ifdef SDL_SSE_INTRINSICS - if (!override && SDL_HasSSE()) { override = SDL_ConvertMonoToStereo_SSE; } - #endif - } - - if (override) { - channel_converter = override; - } - - void* buf = dstconvert ? scratch : dst; - channel_converter((float *) buf, (const float *) src, num_frames); - src = buf; - } - - // Resampling is not done in here. SDL_AudioStream handles that. - - // Move to final data type. - if (dstconvert) { - ConvertAudioFromFloat(dst, (const float *) src, num_frames * dst_channels, dst_format); - src = dst; - } - - SDL_assert(src == dst); // if we got here, we _had_ to have done _something_. Otherwise, we should have memcpy'd! - - if (dst_map) { - SwizzleAudio(num_frames, dst, src, dst_channels, dst_map, dst_format); - } -} - -// Calculate the largest frame size needed to convert between the two formats. -static int CalculateMaxFrameSize(SDL_AudioFormat src_format, int src_channels, SDL_AudioFormat dst_format, int dst_channels) -{ - const int src_format_size = SDL_AUDIO_BYTESIZE(src_format); - const int dst_format_size = SDL_AUDIO_BYTESIZE(dst_format); - const int max_app_format_size = SDL_max(src_format_size, dst_format_size); - const int max_format_size = SDL_max(max_app_format_size, sizeof (float)); // ConvertAudio and ResampleAudio use floats. - const int max_channels = SDL_max(src_channels, dst_channels); - return max_format_size * max_channels; -} - -static Sint64 GetAudioStreamResampleRate(SDL_AudioStream* stream, int src_freq, Sint64 resample_offset) -{ - src_freq = (int)((float)src_freq * stream->freq_ratio); - - Sint64 resample_rate = SDL_GetResampleRate(src_freq, stream->dst_spec.freq); - - // If src_freq == dst_freq, and we aren't between frames, don't resample - if ((resample_rate == 0x100000000) && (resample_offset == 0)) { - resample_rate = 0; - } - - return resample_rate; -} - -static bool UpdateAudioStreamInputSpec(SDL_AudioStream *stream, const SDL_AudioSpec *spec, const int *chmap) -{ - if (SDL_AudioSpecsEqual(&stream->input_spec, spec, stream->input_chmap, chmap)) { - return true; - } - - if (!SDL_ResetAudioQueueHistory(stream->queue, SDL_GetResamplerHistoryFrames())) { - return false; - } - - if (!chmap) { - stream->input_chmap = NULL; - } else { - const size_t chmaplen = sizeof (*chmap) * spec->channels; - stream->input_chmap = stream->input_chmap_storage; - SDL_memcpy(stream->input_chmap, chmap, chmaplen); - } - - SDL_copyp(&stream->input_spec, spec); - - return true; -} - -SDL_AudioStream *SDL_CreateAudioStream(const SDL_AudioSpec *src_spec, const SDL_AudioSpec *dst_spec) -{ - SDL_ChooseAudioConverters(); - SDL_SetupAudioResampler(); - - SDL_AudioStream *result = (SDL_AudioStream *)SDL_calloc(1, sizeof(SDL_AudioStream)); - if (!result) { - return NULL; - } - - result->freq_ratio = 1.0f; - result->gain = 1.0f; - result->queue = SDL_CreateAudioQueue(8192); - - if (!result->queue) { - SDL_free(result); - return NULL; - } - - result->lock = SDL_CreateMutex(); - if (!result->lock) { - SDL_free(result->queue); - SDL_free(result); - return NULL; - } - - OnAudioStreamCreated(result); - - if (!SDL_SetAudioStreamFormat(result, src_spec, dst_spec)) { - SDL_DestroyAudioStream(result); - return NULL; - } - - return result; -} - -SDL_PropertiesID SDL_GetAudioStreamProperties(SDL_AudioStream *stream) -{ - if (!stream) { - SDL_InvalidParamError("stream"); - return 0; - } - SDL_LockMutex(stream->lock); - if (stream->props == 0) { - stream->props = SDL_CreateProperties(); - } - SDL_UnlockMutex(stream->lock); - return stream->props; -} - -bool SDL_SetAudioStreamGetCallback(SDL_AudioStream *stream, SDL_AudioStreamCallback callback, void *userdata) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - SDL_LockMutex(stream->lock); - stream->get_callback = callback; - stream->get_callback_userdata = userdata; - SDL_UnlockMutex(stream->lock); - return true; -} - -bool SDL_SetAudioStreamPutCallback(SDL_AudioStream *stream, SDL_AudioStreamCallback callback, void *userdata) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - SDL_LockMutex(stream->lock); - stream->put_callback = callback; - stream->put_callback_userdata = userdata; - SDL_UnlockMutex(stream->lock); - return true; -} - -bool SDL_LockAudioStream(SDL_AudioStream *stream) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - SDL_LockMutex(stream->lock); - return true; -} - -bool SDL_UnlockAudioStream(SDL_AudioStream *stream) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - SDL_UnlockMutex(stream->lock); - return true; -} - -bool SDL_GetAudioStreamFormat(SDL_AudioStream *stream, SDL_AudioSpec *src_spec, SDL_AudioSpec *dst_spec) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - - SDL_LockMutex(stream->lock); - if (src_spec) { - SDL_copyp(src_spec, &stream->src_spec); - } - if (dst_spec) { - SDL_copyp(dst_spec, &stream->dst_spec); - } - SDL_UnlockMutex(stream->lock); - - if (src_spec && src_spec->format == 0) { - return SDL_SetError("Stream has no source format"); - } else if (dst_spec && dst_spec->format == 0) { - return SDL_SetError("Stream has no destination format"); - } - - return true; -} - -bool SDL_SetAudioStreamFormat(SDL_AudioStream *stream, const SDL_AudioSpec *src_spec, const SDL_AudioSpec *dst_spec) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - - // note that while we've removed the maximum frequency checks, SDL _will_ - // fail to resample to extremely high sample rates correctly. Really high, - // like 196608000Hz. File a bug. :P - - if (src_spec) { - if (!SDL_IsSupportedAudioFormat(src_spec->format)) { - return SDL_InvalidParamError("src_spec->format"); - } else if (!SDL_IsSupportedChannelCount(src_spec->channels)) { - return SDL_InvalidParamError("src_spec->channels"); - } else if (src_spec->freq <= 0) { - return SDL_InvalidParamError("src_spec->freq"); - } - } - - if (dst_spec) { - if (!SDL_IsSupportedAudioFormat(dst_spec->format)) { - return SDL_InvalidParamError("dst_spec->format"); - } else if (!SDL_IsSupportedChannelCount(dst_spec->channels)) { - return SDL_InvalidParamError("dst_spec->channels"); - } else if (dst_spec->freq <= 0) { - return SDL_InvalidParamError("dst_spec->freq"); - } - } - - SDL_LockMutex(stream->lock); - - // quietly refuse to change the format of the end currently bound to a device. - if (stream->bound_device) { - if (stream->bound_device->physical_device->recording) { - src_spec = NULL; - } else { - dst_spec = NULL; - } - } - - if (src_spec) { - if (src_spec->channels != stream->src_spec.channels) { - SDL_free(stream->src_chmap); - stream->src_chmap = NULL; - } - SDL_copyp(&stream->src_spec, src_spec); - } - - if (dst_spec) { - if (dst_spec->channels != stream->dst_spec.channels) { - SDL_free(stream->dst_chmap); - stream->dst_chmap = NULL; - } - SDL_copyp(&stream->dst_spec, dst_spec); - } - - SDL_UnlockMutex(stream->lock); - - return true; -} - -bool SetAudioStreamChannelMap(SDL_AudioStream *stream, const SDL_AudioSpec *spec, int **stream_chmap, const int *chmap, int channels, int isinput) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - - bool result = true; - - SDL_LockMutex(stream->lock); - - if (channels != spec->channels) { - result = SDL_SetError("Wrong number of channels"); - } else if (!*stream_chmap && !chmap) { - // already at default, we're good. - } else if (*stream_chmap && chmap && (SDL_memcmp(*stream_chmap, chmap, sizeof (*chmap) * channels) == 0)) { - // already have this map, don't allocate/copy it again. - } else if (SDL_ChannelMapIsBogus(chmap, channels)) { - result = SDL_SetError("Invalid channel mapping"); - } else { - if (SDL_ChannelMapIsDefault(chmap, channels)) { - chmap = NULL; // just apply a default mapping. - } - if (chmap) { - int *dupmap = SDL_ChannelMapDup(chmap, channels); - if (!dupmap) { - result = SDL_SetError("Invalid channel mapping"); - } else { - SDL_free(*stream_chmap); - *stream_chmap = dupmap; - } - } else { - SDL_free(*stream_chmap); - *stream_chmap = NULL; - } - } - - SDL_UnlockMutex(stream->lock); - return result; -} - -bool SDL_SetAudioStreamInputChannelMap(SDL_AudioStream *stream, const int *chmap, int channels) -{ - return SetAudioStreamChannelMap(stream, &stream->src_spec, &stream->src_chmap, chmap, channels, 1); -} - -bool SDL_SetAudioStreamOutputChannelMap(SDL_AudioStream *stream, const int *chmap, int channels) -{ - return SetAudioStreamChannelMap(stream, &stream->dst_spec, &stream->dst_chmap, chmap, channels, 0); -} - -int *SDL_GetAudioStreamInputChannelMap(SDL_AudioStream *stream, int *count) -{ - int *result = NULL; - int channels = 0; - if (stream) { - SDL_LockMutex(stream->lock); - channels = stream->src_spec.channels; - result = SDL_ChannelMapDup(stream->src_chmap, channels); - SDL_UnlockMutex(stream->lock); - } - - if (count) { - *count = channels; - } - - return result; -} - -int *SDL_GetAudioStreamOutputChannelMap(SDL_AudioStream *stream, int *count) -{ - int *result = NULL; - int channels = 0; - if (stream) { - SDL_LockMutex(stream->lock); - channels = stream->dst_spec.channels; - result = SDL_ChannelMapDup(stream->dst_chmap, channels); - SDL_UnlockMutex(stream->lock); - } - - if (count) { - *count = channels; - } - - return result; -} - -float SDL_GetAudioStreamFrequencyRatio(SDL_AudioStream *stream) -{ - if (!stream) { - SDL_InvalidParamError("stream"); - return 0.0f; - } - - SDL_LockMutex(stream->lock); - const float freq_ratio = stream->freq_ratio; - SDL_UnlockMutex(stream->lock); - - return freq_ratio; -} - -bool SDL_SetAudioStreamFrequencyRatio(SDL_AudioStream *stream, float freq_ratio) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - - // Picked mostly arbitrarily. - const float min_freq_ratio = 0.01f; - const float max_freq_ratio = 100.0f; - - if (freq_ratio < min_freq_ratio) { - return SDL_SetError("Frequency ratio is too low"); - } else if (freq_ratio > max_freq_ratio) { - return SDL_SetError("Frequency ratio is too high"); - } - - SDL_LockMutex(stream->lock); - stream->freq_ratio = freq_ratio; - SDL_UnlockMutex(stream->lock); - - return true; -} - -float SDL_GetAudioStreamGain(SDL_AudioStream *stream) -{ - if (!stream) { - SDL_InvalidParamError("stream"); - return -1.0f; - } - - SDL_LockMutex(stream->lock); - const float gain = stream->gain; - SDL_UnlockMutex(stream->lock); - - return gain; -} - -bool SDL_SetAudioStreamGain(SDL_AudioStream *stream, float gain) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } else if (gain < 0.0f) { - return SDL_InvalidParamError("gain"); - } - - SDL_LockMutex(stream->lock); - stream->gain = gain; - SDL_UnlockMutex(stream->lock); - - return true; -} - -static bool CheckAudioStreamIsFullySetup(SDL_AudioStream *stream) -{ - if (stream->src_spec.format == 0) { - return SDL_SetError("Stream has no source format"); - } else if (stream->dst_spec.format == 0) { - return SDL_SetError("Stream has no destination format"); - } - - return true; -} - -static bool PutAudioStreamBuffer(SDL_AudioStream *stream, const void *buf, int len, SDL_ReleaseAudioBufferCallback callback, void* userdata) -{ -#if DEBUG_AUDIOSTREAM - SDL_Log("AUDIOSTREAM: wants to put %d bytes", len); -#endif - - SDL_LockMutex(stream->lock); - - if (!CheckAudioStreamIsFullySetup(stream)) { - SDL_UnlockMutex(stream->lock); - return false; - } - - if ((len % SDL_AUDIO_FRAMESIZE(stream->src_spec)) != 0) { - SDL_UnlockMutex(stream->lock); - return SDL_SetError("Can't add partial sample frames"); - } - - SDL_AudioTrack* track = NULL; - - if (callback) { - track = SDL_CreateAudioTrack(stream->queue, &stream->src_spec, stream->src_chmap, (Uint8 *)buf, len, len, callback, userdata); - - if (!track) { - SDL_UnlockMutex(stream->lock); - return false; - } - } - - const int prev_available = stream->put_callback ? SDL_GetAudioStreamAvailable(stream) : 0; - - bool result = true; - - if (track) { - SDL_AddTrackToAudioQueue(stream->queue, track); - } else { - result = SDL_WriteToAudioQueue(stream->queue, &stream->src_spec, stream->src_chmap, (const Uint8 *)buf, len); - } - - if (result) { - if (stream->put_callback) { - const int newavail = SDL_GetAudioStreamAvailable(stream) - prev_available; - stream->put_callback(stream->put_callback_userdata, stream, newavail, newavail); - } - } - - SDL_UnlockMutex(stream->lock); - - return result; -} - -static void SDLCALL FreeAllocatedAudioBuffer(void *userdata, const void *buf, int len) -{ - SDL_free((void*) buf); -} - -bool SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } else if (!buf) { - return SDL_InvalidParamError("buf"); - } else if (len < 0) { - return SDL_InvalidParamError("len"); - } else if (len == 0) { - return true; // nothing to do. - } - - // When copying in large amounts of data, try and do as much work as possible - // outside of the stream lock, otherwise the output device is likely to be starved. - const int large_input_thresh = 64 * 1024; - - if (len >= large_input_thresh) { - void *data = SDL_malloc(len); - - if (!data) { - return false; - } - - SDL_memcpy(data, buf, len); - buf = data; - - bool ret = PutAudioStreamBuffer(stream, buf, len, FreeAllocatedAudioBuffer, NULL); - if (!ret) { - SDL_free(data); - } - return ret; - } - - return PutAudioStreamBuffer(stream, buf, len, NULL, NULL); -} - -bool SDL_FlushAudioStream(SDL_AudioStream *stream) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - - SDL_LockMutex(stream->lock); - SDL_FlushAudioQueue(stream->queue); - SDL_UnlockMutex(stream->lock); - - return true; -} - -/* this does not save the previous contents of stream->work_buffer. It's a work buffer!! - The returned buffer is aligned/padded for use with SIMD instructions. */ -static Uint8 *EnsureAudioStreamWorkBufferSize(SDL_AudioStream *stream, size_t newlen) -{ - if (stream->work_buffer_allocation >= newlen) { - return stream->work_buffer; - } - - Uint8 *ptr = (Uint8 *) SDL_aligned_alloc(SDL_GetSIMDAlignment(), newlen); - if (!ptr) { - return NULL; // previous work buffer is still valid! - } - - SDL_aligned_free(stream->work_buffer); - stream->work_buffer = ptr; - stream->work_buffer_allocation = newlen; - return ptr; -} - -static Sint64 NextAudioStreamIter(SDL_AudioStream* stream, void** inout_iter, - Sint64* inout_resample_offset, SDL_AudioSpec* out_spec, int **out_chmap, bool* out_flushed) -{ - SDL_AudioSpec spec; - bool flushed; - int *chmap; - size_t queued_bytes = SDL_NextAudioQueueIter(stream->queue, inout_iter, &spec, &chmap, &flushed); - - if (out_spec) { - SDL_copyp(out_spec, &spec); - } - - if (out_chmap) { - *out_chmap = chmap; - } - - // There is infinite audio available, whether or not we are resampling - if (queued_bytes == SDL_SIZE_MAX) { - *inout_resample_offset = 0; - - if (out_flushed) { - *out_flushed = false; - } - - return SDL_MAX_SINT32; - } - - Sint64 resample_offset = *inout_resample_offset; - Sint64 resample_rate = GetAudioStreamResampleRate(stream, spec.freq, resample_offset); - Sint64 output_frames = (Sint64)(queued_bytes / SDL_AUDIO_FRAMESIZE(spec)); - - if (resample_rate) { - // Resampling requires padding frames to the left and right of the current position. - // Past the end of the track, the right padding is filled with silence. - // But we only want to do that if the track is actually finished (flushed). - if (!flushed) { - output_frames -= SDL_GetResamplerPaddingFrames(resample_rate); - } - - output_frames = SDL_GetResamplerOutputFrames(output_frames, resample_rate, &resample_offset); - } - - if (flushed) { - resample_offset = 0; - } - - *inout_resample_offset = resample_offset; - - if (out_flushed) { - *out_flushed = flushed; - } - - return output_frames; -} - -static Sint64 GetAudioStreamAvailableFrames(SDL_AudioStream* stream, Sint64* out_resample_offset) -{ - void* iter = SDL_BeginAudioQueueIter(stream->queue); - - Sint64 resample_offset = stream->resample_offset; - Sint64 output_frames = 0; - - while (iter) { - output_frames += NextAudioStreamIter(stream, &iter, &resample_offset, NULL, NULL, NULL); - - // Already got loads of frames. Just clamp it to something reasonable - if (output_frames >= SDL_MAX_SINT32) { - output_frames = SDL_MAX_SINT32; - break; - } - } - - if (out_resample_offset) { - *out_resample_offset = resample_offset; - } - - return output_frames; -} - -static Sint64 GetAudioStreamHead(SDL_AudioStream* stream, SDL_AudioSpec* out_spec, int **out_chmap, bool* out_flushed) -{ - void* iter = SDL_BeginAudioQueueIter(stream->queue); - - if (!iter) { - SDL_zerop(out_spec); - *out_flushed = false; - return 0; - } - - Sint64 resample_offset = stream->resample_offset; - return NextAudioStreamIter(stream, &iter, &resample_offset, out_spec, out_chmap, out_flushed); -} - -// You must hold stream->lock and validate your parameters before calling this! -// Enough input data MUST be available! -static bool GetAudioStreamDataInternal(SDL_AudioStream *stream, void *buf, int output_frames, float gain) -{ - const SDL_AudioSpec* src_spec = &stream->input_spec; - const SDL_AudioSpec* dst_spec = &stream->dst_spec; - - const SDL_AudioFormat src_format = src_spec->format; - const int src_channels = src_spec->channels; - - const SDL_AudioFormat dst_format = dst_spec->format; - const int dst_channels = dst_spec->channels; - const int *dst_map = stream->dst_chmap; - - const int max_frame_size = CalculateMaxFrameSize(src_format, src_channels, dst_format, dst_channels); - const Sint64 resample_rate = GetAudioStreamResampleRate(stream, src_spec->freq, stream->resample_offset); - -#if DEBUG_AUDIOSTREAM - SDL_Log("AUDIOSTREAM: asking for %d frames.", output_frames); -#endif - - SDL_assert(output_frames > 0); - - // Not resampling? It's an easy conversion (and maybe not even that!) - if (resample_rate == 0) { - Uint8* work_buffer = NULL; - - // Ensure we have enough scratch space for any conversions - if ((src_format != dst_format) || (src_channels != dst_channels) || (gain != 1.0f)) { - work_buffer = EnsureAudioStreamWorkBufferSize(stream, output_frames * max_frame_size); - - if (!work_buffer) { - return false; - } - } - - if (SDL_ReadFromAudioQueue(stream->queue, (Uint8 *)buf, dst_format, dst_channels, dst_map, 0, output_frames, 0, work_buffer, gain) != buf) { - return SDL_SetError("Not enough data in queue"); - } - - return true; - } - - // Time to do some resampling! - // Calculate the number of input frames necessary for this request. - // Because resampling happens "between" frames, The same number of output_frames - // can require a different number of input_frames, depending on the resample_offset. - // In fact, input_frames can sometimes even be zero when upsampling. - const int input_frames = (int) SDL_GetResamplerInputFrames(output_frames, resample_rate, stream->resample_offset); - - const int padding_frames = SDL_GetResamplerPaddingFrames(resample_rate); - - const SDL_AudioFormat resample_format = SDL_AUDIO_F32; - - // If increasing channels, do it after resampling, since we'd just - // do more work to resample duplicate channels. If we're decreasing, do - // it first so we resample the interpolated data instead of interpolating - // the resampled data. - const int resample_channels = SDL_min(src_channels, dst_channels); - - // The size of the frame used when resampling - const int resample_frame_size = SDL_AUDIO_BYTESIZE(resample_format) * resample_channels; - - // The main portion of the work_buffer can be used to store 3 things: - // src_sample_frame_size * (left_padding+input_buffer+right_padding) - // resample_frame_size * (left_padding+input_buffer+right_padding) - // dst_sample_frame_size * output_frames - // - // ResampleAudio also requires an additional buffer if it can't write straight to the output: - // resample_frame_size * output_frames - // - // Note, ConvertAudio requires (num_frames * max_sample_frame_size) of scratch space - const int work_buffer_frames = input_frames + (padding_frames * 2); - int work_buffer_capacity = work_buffer_frames * max_frame_size; - int resample_buffer_offset = -1; - - // Check if we can resample directly into the output buffer. - // Note, this is just to avoid extra copies. - // Some other formats may fit directly into the output buffer, but i'd rather process data in a SIMD-aligned buffer. - if ((dst_format != resample_format) || (dst_channels != resample_channels)) { - // Allocate space for converting the resampled output to the destination format - int resample_convert_bytes = output_frames * max_frame_size; - work_buffer_capacity = SDL_max(work_buffer_capacity, resample_convert_bytes); - - // SIMD-align the buffer - int simd_alignment = (int) SDL_GetSIMDAlignment(); - work_buffer_capacity += simd_alignment - 1; - work_buffer_capacity -= work_buffer_capacity % simd_alignment; - - // Allocate space for the resampled output - int resample_bytes = output_frames * resample_frame_size; - resample_buffer_offset = work_buffer_capacity; - work_buffer_capacity += resample_bytes; - } - - Uint8* work_buffer = EnsureAudioStreamWorkBufferSize(stream, work_buffer_capacity); - - if (!work_buffer) { - return false; - } - - // adjust gain either before resampling or after, depending on which point has less - // samples to process. - const float preresample_gain = (input_frames > output_frames) ? 1.0f : gain; - const float postresample_gain = (input_frames > output_frames) ? gain : 1.0f; - - // (dst channel map is NULL because we'll do the final swizzle on ConvertAudio after resample.) - const Uint8* input_buffer = SDL_ReadFromAudioQueue(stream->queue, - NULL, resample_format, resample_channels, NULL, - padding_frames, input_frames, padding_frames, work_buffer, preresample_gain); - - if (!input_buffer) { - return SDL_SetError("Not enough data in queue (resample)"); - } - - input_buffer += padding_frames * resample_frame_size; - - // Decide where the resampled output goes - void* resample_buffer = (resample_buffer_offset != -1) ? (work_buffer + resample_buffer_offset) : buf; - - SDL_ResampleAudio(resample_channels, - (const float *) input_buffer, input_frames, - (float*) resample_buffer, output_frames, - resample_rate, &stream->resample_offset); - - // Convert to the final format, if necessary (src channel map is NULL because SDL_ReadFromAudioQueue already handled this). - ConvertAudio(output_frames, resample_buffer, resample_format, resample_channels, NULL, buf, dst_format, dst_channels, dst_map, work_buffer, postresample_gain); - - return true; -} - -// get converted/resampled data from the stream -int SDL_GetAudioStreamDataAdjustGain(SDL_AudioStream *stream, void *voidbuf, int len, float extra_gain) -{ - Uint8 *buf = (Uint8 *) voidbuf; - -#if DEBUG_AUDIOSTREAM - SDL_Log("AUDIOSTREAM: want to get %d converted bytes", len); -#endif - - if (!stream) { - SDL_InvalidParamError("stream"); - return -1; - } else if (!buf) { - SDL_InvalidParamError("buf"); - return -1; - } else if (len < 0) { - SDL_InvalidParamError("len"); - return -1; - } else if (len == 0) { - return 0; // nothing to do. - } - - SDL_LockMutex(stream->lock); - - if (!CheckAudioStreamIsFullySetup(stream)) { - SDL_UnlockMutex(stream->lock); - return -1; - } - - const float gain = stream->gain * extra_gain; - const int dst_frame_size = SDL_AUDIO_FRAMESIZE(stream->dst_spec); - - len -= len % dst_frame_size; // chop off any fractional sample frame. - - // give the callback a chance to fill in more stream data if it wants. - if (stream->get_callback) { - Sint64 total_request = len / dst_frame_size; // start with sample frames desired - Sint64 additional_request = total_request; - - Sint64 resample_offset = 0; - Sint64 available_frames = GetAudioStreamAvailableFrames(stream, &resample_offset); - - additional_request -= SDL_min(additional_request, available_frames); - - Sint64 resample_rate = GetAudioStreamResampleRate(stream, stream->src_spec.freq, resample_offset); - - if (resample_rate) { - total_request = SDL_GetResamplerInputFrames(total_request, resample_rate, resample_offset); - additional_request = SDL_GetResamplerInputFrames(additional_request, resample_rate, resample_offset); - } - - total_request *= SDL_AUDIO_FRAMESIZE(stream->src_spec); // convert sample frames to bytes. - additional_request *= SDL_AUDIO_FRAMESIZE(stream->src_spec); // convert sample frames to bytes. - stream->get_callback(stream->get_callback_userdata, stream, (int) SDL_min(additional_request, SDL_INT_MAX), (int) SDL_min(total_request, SDL_INT_MAX)); - } - - // Process the data in chunks to avoid allocating too much memory (and potential integer overflows) - const int chunk_size = 4096; - - int total = 0; - - while (total < len) { - // Audio is processed a track at a time. - SDL_AudioSpec input_spec; - int *input_chmap; - bool flushed; - const Sint64 available_frames = GetAudioStreamHead(stream, &input_spec, &input_chmap, &flushed); - - if (available_frames == 0) { - if (flushed) { - SDL_PopAudioQueueHead(stream->queue); - SDL_zero(stream->input_spec); - stream->resample_offset = 0; - stream->input_chmap = NULL; - continue; - } - // There are no frames available, but the track hasn't been flushed, so more might be added later. - break; - } - - if (!UpdateAudioStreamInputSpec(stream, &input_spec, input_chmap)) { - total = total ? total : -1; - break; - } - - // Clamp the output length to the maximum currently available. - // GetAudioStreamDataInternal requires enough input data is available. - int output_frames = (len - total) / dst_frame_size; - output_frames = SDL_min(output_frames, chunk_size); - output_frames = (int) SDL_min(output_frames, available_frames); - - if (!GetAudioStreamDataInternal(stream, &buf[total], output_frames, gain)) { - total = total ? total : -1; - break; - } - - total += output_frames * dst_frame_size; - } - - SDL_UnlockMutex(stream->lock); - -#if DEBUG_AUDIOSTREAM - SDL_Log("AUDIOSTREAM: Final result was %d", total); -#endif - - return total; -} - -int SDL_GetAudioStreamData(SDL_AudioStream *stream, void *voidbuf, int len) -{ - return SDL_GetAudioStreamDataAdjustGain(stream, voidbuf, len, 1.0f); -} - -// number of converted/resampled bytes available for output -int SDL_GetAudioStreamAvailable(SDL_AudioStream *stream) -{ - if (!stream) { - SDL_InvalidParamError("stream"); - return -1; - } - - SDL_LockMutex(stream->lock); - - if (!CheckAudioStreamIsFullySetup(stream)) { - SDL_UnlockMutex(stream->lock); - return 0; - } - - Sint64 count = GetAudioStreamAvailableFrames(stream, NULL); - - // convert from sample frames to bytes in destination format. - count *= SDL_AUDIO_FRAMESIZE(stream->dst_spec); - - SDL_UnlockMutex(stream->lock); - - // if this overflows an int, just clamp it to a maximum. - return (int) SDL_min(count, SDL_INT_MAX); -} - -// number of sample frames that are currently queued as input. -int SDL_GetAudioStreamQueued(SDL_AudioStream *stream) -{ - if (!stream) { - SDL_InvalidParamError("stream"); - return -1; - } - - SDL_LockMutex(stream->lock); - - size_t total = SDL_GetAudioQueueQueued(stream->queue); - - SDL_UnlockMutex(stream->lock); - - // if this overflows an int, just clamp it to a maximum. - return (int) SDL_min(total, SDL_INT_MAX); -} - -bool SDL_ClearAudioStream(SDL_AudioStream *stream) -{ - if (!stream) { - return SDL_InvalidParamError("stream"); - } - - SDL_LockMutex(stream->lock); - - SDL_ClearAudioQueue(stream->queue); - SDL_zero(stream->input_spec); - stream->input_chmap = NULL; - stream->resample_offset = 0; - - SDL_UnlockMutex(stream->lock); - return true; -} - -void SDL_DestroyAudioStream(SDL_AudioStream *stream) -{ - if (!stream) { - return; - } - - SDL_DestroyProperties(stream->props); - - OnAudioStreamDestroy(stream); - - const bool simplified = stream->simplified; - if (simplified) { - if (stream->bound_device) { - SDL_assert(stream->bound_device->simplified); - SDL_CloseAudioDevice(stream->bound_device->instance_id); // this will unbind the stream. - } - } else { - SDL_UnbindAudioStream(stream); - } - - SDL_aligned_free(stream->work_buffer); - SDL_DestroyAudioQueue(stream->queue); - SDL_DestroyMutex(stream->lock); - - SDL_free(stream); -} - -static void SDLCALL DontFreeThisAudioBuffer(void *userdata, const void *buf, int len) -{ - // We don't own the buffer, but know it will outlive the stream -} - -bool SDL_ConvertAudioSamples(const SDL_AudioSpec *src_spec, const Uint8 *src_data, int src_len, const SDL_AudioSpec *dst_spec, Uint8 **dst_data, int *dst_len) -{ - if (dst_data) { - *dst_data = NULL; - } - - if (dst_len) { - *dst_len = 0; - } - - if (!src_data) { - return SDL_InvalidParamError("src_data"); - } else if (src_len < 0) { - return SDL_InvalidParamError("src_len"); - } else if (!dst_data) { - return SDL_InvalidParamError("dst_data"); - } else if (!dst_len) { - return SDL_InvalidParamError("dst_len"); - } - - bool result = false; - Uint8 *dst = NULL; - int dstlen = 0; - - SDL_AudioStream *stream = SDL_CreateAudioStream(src_spec, dst_spec); - if (stream) { - if (PutAudioStreamBuffer(stream, src_data, src_len, DontFreeThisAudioBuffer, NULL) && - SDL_FlushAudioStream(stream)) { - dstlen = SDL_GetAudioStreamAvailable(stream); - if (dstlen >= 0) { - dst = (Uint8 *)SDL_malloc(dstlen); - if (dst) { - result = (SDL_GetAudioStreamData(stream, dst, dstlen) == dstlen); - } - } - } - } - - if (result) { - *dst_data = dst; - *dst_len = dstlen; - } else { - SDL_free(dst); - } - - SDL_DestroyAudioStream(stream); - return result; -} -- cgit v1.2.3