Files
foobar2000-sdk/foobar2000/SDK/audio_chunk.h
2021-12-14 00:28:25 -07:00

384 lines
19 KiB
C++

#pragma once
#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:
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,
channel_config_mono = channel_front_center,
channel_config_stereo = channel_front_left | channel_front_right,
channel_config_4point0 = channel_front_left | channel_front_right | channel_back_left | channel_back_right,
channel_config_5point0 = channel_front_left | channel_front_right | channel_front_center | channel_back_left | channel_back_right,
channel_config_5point1 = channel_front_left | channel_front_right | channel_front_center | channel_lfe | channel_back_left | channel_back_right,
channel_config_5point1_side = channel_front_left | channel_front_right | channel_front_center | channel_lfe | channel_side_left | channel_side_right,
channel_config_7point1 = channel_config_5point1 | channel_side_left | channel_side_right,
channels_back_left_right = channel_back_left | channel_back_right,
channels_side_left_right = channel_side_left | channel_side_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 uint32_t g_channel_config_to_wfx(unsigned p_config);
//! Helper function; translates WAVEFORMATEXTENSIBLE channel map to audio_chunk channel map.
static unsigned g_channel_config_from_wfx(uint32_t 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 unsigned g_count_channels(unsigned 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);
//! 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.
void allocate(size_t size) { set_data_size( size ); }
//! 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 {}
#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();}
#ifdef _MSC_VER
#pragma deprecated( get_data_length )
#endif
//! 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,t_size samples,unsigned nch,unsigned srate,unsigned channel_config);
//! Helper, sets chunk data to contents of specified buffer, with specified number of channels / sample rate, using default channel map for specified channel count.
inline void set_data(const audio_sample * src,t_size samples,unsigned nch,unsigned srate) {set_data(src,samples,nch,srate,g_guess_channel_config(nch));}
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
inline void set_data_32(const float * src,t_size samples,unsigned nch,unsigned srate) {return set_data(src,samples,nch,srate);}
//! 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 Number of silent samples to append.s
void pad_with_silence(t_size samples);
//! Inserts silence at the beginning of the audio chunk.
//! @param 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 samples 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, float 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, float 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, float 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.
void copy(const audio_chunk & p_source) {
set_data(p_source.get_data(),p_source.get_sample_count(),p_source.get_channels(),p_source.get_srate(),p_source.get_channel_config());
}
const audio_chunk & operator=(const audio_chunk & p_source) {
copy(p_source);
return *this;
}
struct spec_t {
uint32_t sampleRate;
uint32_t chanCount, chanMask;
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()) {}
};