Mercurial > foo_out_sdl
diff foosdk/sdk/foobar2000/SDK/audio_chunk.h @ 1:20d02a178406 default tip
*: check in everything else
yay
| author | Paper <paper@tflc.us> |
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| date | Mon, 05 Jan 2026 02:15:46 -0500 |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/foosdk/sdk/foobar2000/SDK/audio_chunk.h Mon Jan 05 02:15:46 2026 -0500 @@ -0,0 +1,395 @@ +#pragma once + +#include <pfc/audio_sample.h> +#include <pfc/memalign.h> +#include "exception_io.h" + +#ifdef _WIN32 +#include <MMReg.h> +#endif +//! Thrown when audio_chunk sample rate or channel mapping changes in mid-stream and the code receiving audio_chunks can't deal with that scenario. +PFC_DECLARE_EXCEPTION(exception_unexpected_audio_format_change, exception_io_data, "Unexpected audio format change" ); + +//! Interface to container of a chunk of audio data. See audio_chunk_impl for an implementation. +class NOVTABLE audio_chunk { +public: + struct spec_t; // forward decl + + enum { + sample_rate_min = 1000, sample_rate_max = 20000000 + }; + static bool g_is_valid_sample_rate(t_uint32 p_val) {return p_val >= sample_rate_min && p_val <= sample_rate_max;} + + //! Channel map flag declarations. Note that order of interleaved channel data in the stream is same as order of these flags. + enum + { + channel_front_left = 1<<0, + channel_front_right = 1<<1, + channel_front_center = 1<<2, + channel_lfe = 1<<3, + channel_back_left = 1<<4, + channel_back_right = 1<<5, + channel_front_center_left = 1<<6, + channel_front_center_right = 1<<7, + channel_back_center = 1<<8, + channel_side_left = 1<<9, + channel_side_right = 1<<10, + channel_top_center = 1<<11, + channel_top_front_left = 1<<12, + channel_top_front_center = 1<<13, + channel_top_front_right = 1<<14, + channel_top_back_left = 1<<15, + channel_top_back_center = 1<<16, + channel_top_back_right = 1<<17, + + channels_back_left_right = channel_back_left | channel_back_right, + channels_side_left_right = channel_side_left | channel_side_right, + + channel_config_mono = channel_front_center, + channel_config_stereo = channel_front_left | channel_front_right, + channel_config_2point1 = channel_config_stereo | channel_lfe, + channel_config_3point0 = channel_config_stereo | channel_front_center, + channel_config_4point0 = channel_config_stereo | channels_back_left_right, + channel_config_4point0_side = channel_config_stereo | channels_side_left_right, + channel_config_4point1 = channel_config_4point0 | channel_lfe, + channel_config_5point0 = channel_config_4point0 | channel_front_center, + channel_config_6point0 = channel_config_4point0 | channels_side_left_right, + channel_config_5point1 = channel_config_4point0 | channel_front_center | channel_lfe, + channel_config_5point1_side = channel_config_4point0_side | channel_front_center | channel_lfe, + channel_config_7point1 = channel_config_5point1 | channels_side_left_right, + + + defined_channel_count = 18, + }; + + //! Helper function; guesses default channel map for the specified channel count. Returns 0 on failure. + static unsigned g_guess_channel_config(unsigned count); + //! Helper function; determines channel map for the specified channel count according to Xiph specs. Throws exception_io_data on failure. + static unsigned g_guess_channel_config_xiph(unsigned count); + + //! Helper function; translates audio_chunk channel map to WAVEFORMATEXTENSIBLE channel map. + static constexpr uint32_t g_channel_config_to_wfx(unsigned p_config) { return p_config;} + //! Helper function; translates WAVEFORMATEXTENSIBLE channel map to audio_chunk channel map. + static constexpr unsigned g_channel_config_from_wfx(uint32_t p_wfx) { return p_wfx;} + + //! Extracts flag describing Nth channel from specified map. Usable to figure what specific channel in a stream means. + static unsigned g_extract_channel_flag(unsigned p_config,unsigned p_index); + //! Counts channels specified by channel map. + static constexpr unsigned g_count_channels(unsigned p_config) { return pfc::countBits32(p_config); } + //! Calculates index of a channel specified by p_flag in a stream where channel map is described by p_config. + static unsigned g_channel_index_from_flag(unsigned p_config,unsigned p_flag); + + static const char * g_channel_name(unsigned p_flag); + static const char * g_channel_name_byidx(unsigned p_index); + static unsigned g_find_channel_idx(unsigned p_flag); + static void g_formatChannelMaskDesc(unsigned flags, pfc::string_base & out); + static pfc::string8 g_formatChannelMaskDesc(unsigned flags); + static const char* g_channelMaskName(unsigned flags); + + + + //! Retrieves audio data buffer pointer (non-const version). Returned pointer is for temporary use only; it is valid until next set_data_size call, or until the object is destroyed. \n + //! Size of returned buffer is equal to get_data_size() return value (in audio_samples). Amount of actual data may be smaller, depending on sample count and channel count. Conditions where sample count * channel count are greater than data size should not be possible. + virtual audio_sample * get_data() = 0; + //! Retrieves audio data buffer pointer (const version). Returned pointer is for temporary use only; it is valid until next set_data_size call, or until the object is destroyed. \n + //! Size of returned buffer is equal to get_data_size() return value (in audio_samples). Amount of actual data may be smaller, depending on sample count and channel count. Conditions where sample count * channel count are greater than data size should not be possible. + virtual const audio_sample * get_data() const = 0; + //! Retrieves size of allocated buffer space, in audio_samples. + virtual t_size get_data_size() const = 0; + //! Resizes audio data buffer to specified size. Throws std::bad_alloc on failure. + virtual void set_data_size(t_size p_new_size) = 0; + //! Sanity helper, same as set_data_size. + //! @param bQuicker Avoid memory allocation, permit up to 2x memory used + void allocate(size_t size, bool bQuicker = false); + + //! Retrieves sample rate of contained audio data. + virtual unsigned get_srate() const = 0; + //! Sets sample rate of contained audio data. + virtual void set_srate(unsigned val) = 0; + //! Retrieves channel count of contained audio data. + virtual unsigned get_channels() const = 0; + //! Helper - for consistency - same as get_channels(). + inline unsigned get_channel_count() const {return get_channels();} + //! Retrieves channel map of contained audio data. Conditions where number of channels specified by channel map don't match get_channels() return value should not be possible. + virtual unsigned get_channel_config() const = 0; + //! Sets channel count / channel map. + virtual void set_channels(unsigned p_count,unsigned p_config) = 0; + + //! Retrieves number of valid samples in the buffer. \n + //! Note that a "sample" means a unit of interleaved PCM data representing states of each channel at given point of time, not a single PCM value. \n + //! For an example, duration of contained audio data is equal to sample count / sample rate, while actual size of contained data is equal to sample count * channel count. + virtual t_size get_sample_count() const = 0; + + //! Sets number of valid samples in the buffer. WARNING: sample count * channel count should never be above allocated buffer size. + virtual void set_sample_count(t_size val) = 0; + + //! Helper, same as get_srate(). + inline unsigned get_sample_rate() const {return get_srate();} + //! Helper, same as set_srate(). + inline void set_sample_rate(unsigned val) {set_srate(val);} + + //! Helper; sets channel count to specified value and uses default channel map for this channel count. + void set_channels(unsigned val) {set_channels(val,g_guess_channel_config(val));} + + + //! Helper; resizes audio data buffer when its current size is smaller than requested. + inline void grow_data_size(t_size p_requested) {if (p_requested > get_data_size()) set_data_size(p_requested);} + + + //! Retrieves duration of contained audio data, in seconds. + inline double get_duration() const + { + double rv = 0; + t_size srate = get_srate (), samples = get_sample_count(); + if (srate>0 && samples>0) rv = (double)samples/(double)srate; + return rv; + } + + //! Returns whether the chunk is empty (contains no audio data). + inline bool is_empty() const {return get_channels()==0 || get_srate()==0 || get_sample_count()==0;} + + //! Returns whether the chunk contents are valid (for bug check purposes). + bool is_valid() const; + + void debugChunkSpec() const; + pfc::string8 formatChunkSpec() const; +#if PFC_DEBUG + void assert_valid(const char * ctx) const; +#else + void assert_valid(const char* ctx) const { (void)ctx; } +#endif + + + //! Returns whether the chunk contains valid sample rate & channel info (but allows an empty chunk). + bool is_spec_valid() const; + + //! Returns actual amount of audio data contained in the buffer (sample count * channel count). Must not be greater than data size (see get_data_size()). + size_t get_used_size() const {return get_sample_count() * get_channels();} + //! Same as get_used_size(); old confusingly named version. + size_t get_data_length() const {return get_sample_count() * get_channels();} + + //! Resets all audio_chunk data. + inline void reset() { + set_sample_count(0); + set_srate(0); + set_channels(0,0); + set_data_size(0); + } + + //! Helper, sets chunk data to contents of specified buffer, with specified number of channels / sample rate / channel map. + void set_data(const audio_sample * src,size_t samples,unsigned nch,unsigned srate,unsigned channel_config); + void set_data(const audio_sample* src, size_t samples, spec_t const& spec, bool bQucker = false); + void set_data(const audio_sample* src, t_size samples, unsigned nch, unsigned srate); + + void set_data_int16(const int16_t * src,t_size samples,unsigned nch,unsigned srate,unsigned channel_config); + + //! Helper, sets chunk data to contents of specified buffer, using default win32/wav conventions for signed/unsigned switch. + inline void set_data_fixedpoint(const void * ptr,t_size bytes,unsigned srate,unsigned nch,unsigned bps,unsigned channel_config) { + this->set_data_fixedpoint_ms(ptr, bytes, srate, nch, bps, channel_config); + } + + void set_data_fixedpoint_signed(const void * ptr,t_size bytes,unsigned srate,unsigned nch,unsigned bps,unsigned channel_config); + + enum + { + FLAG_LITTLE_ENDIAN = 1, + FLAG_BIG_ENDIAN = 2, + FLAG_SIGNED = 4, + FLAG_UNSIGNED = 8, + }; + + inline static unsigned flags_autoendian() { + return pfc::byte_order_is_big_endian ? FLAG_BIG_ENDIAN : FLAG_LITTLE_ENDIAN; + } + + void set_data_fixedpoint_ex(const void * ptr,t_size bytes,unsigned p_sample_rate,unsigned p_channels,unsigned p_bits_per_sample,unsigned p_flags,unsigned p_channel_config);//p_flags - see FLAG_* above + + void set_data_fixedpoint_ms(const void * ptr, size_t bytes, unsigned sampleRate, unsigned channels, unsigned bps, unsigned channelConfig); + + void set_data_floatingpoint_ex(const void * ptr,t_size bytes,unsigned p_sample_rate,unsigned p_channels,unsigned p_bits_per_sample,unsigned p_flags,unsigned p_channel_config);//signed/unsigned flags dont apply + static bool is_supported_floatingpoint(unsigned bps) { return bps == 32 || bps == 64 || bps == 16 || bps == 24; } + + void set_data_32(const float* src, t_size samples, unsigned nch, unsigned srate); + void set_data_32(const float* src, t_size samples, spec_t const & spec ); + + //! Appends silent samples at the end of the chunk. \n + //! The chunk may be empty prior to this call, its sample rate & channel count will be set to the specified values then. \n + //! The chunk may have different sample rate than requested; silent sample count will be recalculated to the used sample rate retaining actual duration. + //! @param samples Number of silent samples to append. + //! @param hint_nch If no channel count is set on this chunk, it will be set to this value. + //! @param hint_srate The sample rate of silent samples being inserted. If no sampler ate is set on this chunk, it will be set to this value.\n + //! Otherwise if chunk's sample rate doesn't match hint_srate, sample count will be recalculated to chunk's actual sample rate. + void pad_with_silence_ex(t_size samples,unsigned hint_nch,unsigned hint_srate); + //! Appends silent samples at the end of the chunk. \n + //! The chunk must have valid sample rate & channel count prior to this call. + //! @param samples Number of silent samples to append. + void pad_with_silence(t_size samples); + //! Inserts silence at the beginning of the audio chunk. + //! @param samples Number of silent samples to insert. + void insert_silence_fromstart(t_size samples); + //! Helper; removes N first samples from the chunk. \n + //! If the chunk contains fewer samples than requested, it becomes empty. + //! @returns Number of samples actually removed. + t_size skip_first_samples(t_size samples); + //! Produces a chunk of silence, with the specified duration. \n + //! Any existing audio sdata will be discarded. \n + //! Expects sample rate and channel count to be set first. \n + //! Also allocates memory for the requested amount of data see: set_data_size(). + //! @param samples Desired number of samples. + void set_silence(t_size samples); + //! Produces a chunk of silence, with the specified duration. \n + //! Any existing audio sdata will be discarded. \n + //! Expects sample rate and channel count to be set first. \n + //! Also allocates memory for the requested amount of data see: set_data_size(). + //! @param seconds Desired duration in seconds. + void set_silence_seconds( double seconds ); + + //! Helper; skips first samples of the chunk updating a remaining to-skip counter. + //! @param skipDuration Reference to the duration of audio remining to be skipped, in seconds. Updated by each call. + //! @returns False if the chunk became empty, true otherwise. + bool process_skip(double & skipDuration); + + //! Simple function to get original PCM stream back. Assumes host's endianness, integers are signed - including the 8bit mode; 32bit mode assumed to be float. + //! @returns false when the conversion could not be performed because of unsupported bit depth etc. + bool to_raw_data(class mem_block_container & out, t_uint32 bps, bool useUpperBits = true, audio_sample scale = 1.0) const; + + //! Convert audio_chunk contents to fixed-point PCM format. + //! @param useUpperBits relevant if bps != bpsValid, signals whether upper or lower bits of each sample should be used. + bool toFixedPoint(class mem_block_container & out, uint32_t bps, uint32_t bpsValid, bool useUpperBits = true, audio_sample scale = 1.0) const; + + //! Convert a buffer of audio_samples to fixed-point PCM format. + //! @param useUpperBits relevant if bps != bpsValid, signals whether upper or lower bits of each sample should be used. + static bool g_toFixedPoint(const audio_sample * in, void * out, size_t count, uint32_t bps, uint32_t bpsValid, bool useUpperBits = true, audio_sample scale = 1.0); + + + //! Helper, calculates peak value of data in the chunk. The optional parameter specifies initial peak value, to simplify calling code. + audio_sample get_peak(audio_sample p_peak) const; + audio_sample get_peak() const; + + //! Helper function; scales entire chunk content by specified value. + void scale(audio_sample p_value); + + //! Helper; copies content of another audio chunk to this chunk. + //! @param bQuicker Avoid memory allocation, permit up to 2x memory used + void copy(const audio_chunk & p_source, bool bQuicker = false) { + set_data(p_source.get_data(),p_source.get_sample_count(),p_source.get_spec(), bQuicker); + } + + const audio_chunk & operator=(const audio_chunk & p_source) { + copy(p_source); + return *this; + } + + struct spec_t { + uint32_t sampleRate = 0; + uint32_t chanCount = 0, chanMask = 0; + + static bool equals( const spec_t & v1, const spec_t & v2 ); + bool operator==(const spec_t & other) const { return equals(*this, other);} + bool operator!=(const spec_t & other) const { return !equals(*this, other);} + bool is_valid() const; + void clear() { sampleRate = 0; chanCount = 0; chanMask = 0; } + +#ifdef _WIN32 + //! Creates WAVE_FORMAT_IEEE_FLOAT WAVEFORMATEX structure + WAVEFORMATEX toWFX() const; + //! Creates WAVE_FORMAT_IEEE_FLOAT WAVEFORMATEXTENSIBLE structure + WAVEFORMATEXTENSIBLE toWFXEX() const; + //! Creates WAVE_FORMAT_PCM WAVEFORMATEX structure + WAVEFORMATEX toWFXWithBPS(uint32_t bps) const; + //! Creates WAVE_FORMAT_PCM WAVEFORMATEXTENSIBLE structure + WAVEFORMATEXTENSIBLE toWFXEXWithBPS(uint32_t bps) const; +#endif + + pfc::string8 toString( const char * delim = " " ) const; + }; + static spec_t makeSpec(uint32_t rate, uint32_t channels); + static spec_t makeSpec(uint32_t rate, uint32_t channels, uint32_t chanMask); + static spec_t emptySpec() { return makeSpec(0, 0, 0); } + + spec_t get_spec() const; + void set_spec(const spec_t &); + + void append(const audio_chunk& other); +protected: + audio_chunk() {} + ~audio_chunk() {} +}; + +//! Implementation of audio_chunk. Takes pfc allocator template as template parameter. +template<typename container_t = pfc::mem_block_aligned_t<audio_sample, 16> > +class audio_chunk_impl_t : public audio_chunk { + typedef audio_chunk_impl_t<container_t> t_self; + container_t m_data; + unsigned m_srate = 0, m_nch = 0, m_setup = 0; + t_size m_samples = 0; + +public: + audio_chunk_impl_t() {} + audio_chunk_impl_t(const audio_sample * src,unsigned samples,unsigned nch,unsigned srate) {set_data(src,samples,nch,srate);} + audio_chunk_impl_t(const audio_chunk & p_source) {copy(p_source);} + + + virtual audio_sample * get_data() {return m_data.get_ptr();} + virtual const audio_sample * get_data() const {return m_data.get_ptr();} + virtual t_size get_data_size() const {return m_data.get_size();} + virtual void set_data_size(t_size new_size) {m_data.set_size(new_size);} + + virtual unsigned get_srate() const {return m_srate;} + virtual void set_srate(unsigned val) {m_srate=val;} + virtual unsigned get_channels() const {return m_nch;} + virtual unsigned get_channel_config() const {return m_setup;} + virtual void set_channels(unsigned val,unsigned setup) {m_nch = val;m_setup = setup;} + void set_channels(unsigned val) {set_channels(val,g_guess_channel_config(val));} + + virtual t_size get_sample_count() const {return m_samples;} + virtual void set_sample_count(t_size val) {m_samples = val;} + + const t_self & operator=(const audio_chunk & p_source) {copy(p_source);return *this;} +}; + +typedef audio_chunk_impl_t<> audio_chunk_impl; +typedef audio_chunk_impl_t<pfc::mem_block_aligned_incremental_t<audio_sample, 16> > audio_chunk_fast_impl; + +//! Implements const methods of audio_chunk only, referring to an external buffer. For temporary use only (does not maintain own storage), e.g.: somefunc( audio_chunk_temp_impl(mybuffer,....) ); +class audio_chunk_memref_impl : public audio_chunk { +public: + audio_chunk_memref_impl(const audio_sample * p_data,t_size p_samples,t_uint32 p_sample_rate,t_uint32 p_channels,t_uint32 p_channel_config) : + m_samples(p_samples), m_sample_rate(p_sample_rate), m_channels(p_channels), m_channel_config(p_channel_config), m_data(p_data) + { +#if PFC_DEBUG + assert_valid(__FUNCTION__); +#endif + } + + audio_sample * get_data() {throw pfc::exception_not_implemented();} + const audio_sample * get_data() const {return m_data;} + t_size get_data_size() const {return m_samples * m_channels;} + void set_data_size(t_size) {throw pfc::exception_not_implemented();} + + unsigned get_srate() const {return m_sample_rate;} + void set_srate(unsigned) {throw pfc::exception_not_implemented();} + unsigned get_channels() const {return m_channels;} + unsigned get_channel_config() const {return m_channel_config;} + void set_channels(unsigned,unsigned) {throw pfc::exception_not_implemented();} + + t_size get_sample_count() const {return m_samples;} + + void set_sample_count(t_size) {throw pfc::exception_not_implemented();} + +private: + t_size m_samples; + t_uint32 m_sample_rate,m_channels,m_channel_config; + const audio_sample * m_data; +}; + + +// Compatibility typedefs. +typedef audio_chunk_fast_impl audio_chunk_impl_temporary; +typedef audio_chunk_impl audio_chunk_i; +typedef audio_chunk_memref_impl audio_chunk_temp_impl; + +class audio_chunk_partial_ref : public audio_chunk_temp_impl { +public: + audio_chunk_partial_ref(const audio_chunk & chunk, t_size base, t_size count) : audio_chunk_temp_impl(chunk.get_data() + base * chunk.get_channels(), count, chunk.get_sample_rate(), chunk.get_channels(), chunk.get_channel_config()) {} +};
