Index: /trunk/include/VBox/vmm/pdmaudioifs.h =================================================================== --- /trunk/include/VBox/vmm/pdmaudioifs.h (revision 68131) +++ /trunk/include/VBox/vmm/pdmaudioifs.h (revision 68132) @@ -199,7 +199,9 @@ /** - * A single audio sample, representing left and right channels (stereo). - */ -typedef struct PDMAUDIOSAMPLE + * A single audio frame. + * + * Currently only two (2) channels, left and right, are supported. + */ +typedef struct PDMAUDIOFRAME { /** Left channel. */ @@ -207,9 +209,9 @@ /** Right channel. */ int64_t i64RSample; -} PDMAUDIOSAMPLE; -/** Pointer to a single (stereo) audio sample. */ -typedef PDMAUDIOSAMPLE *PPDMAUDIOSAMPLE; -/** Pointer to a const single (stereo) audio sample. */ -typedef PDMAUDIOSAMPLE const *PCPDMAUDIOSAMPLE; +} PDMAUDIOFRAME; +/** Pointer to a single (stereo) audio frame. */ +typedef PDMAUDIOFRAME *PPDMAUDIOFRAME; +/** Pointer to a const single (stereo) audio frame. */ +typedef PDMAUDIOFRAME const *PCPDMAUDIOFRAME; typedef enum PDMAUDIOENDIANNESS @@ -287,5 +289,5 @@ /** Raw (pass through) data, with no data layout processing done. * - * This means that this stream will operate on PDMAUDIOSAMPLE data + * This means that this stream will operate on PDMAUDIOFRAME data * directly. Don't use this if you don't have to. */ PDMAUDIOSTREAMLAYOUT_RAW, @@ -322,5 +324,5 @@ /** Frame size (in bytes) of this channel. */ size_t cbFrame; - /** Offset (in bytes) to first sample in the data block. */ + /** Offset (in bytes) to first frame in the data block. */ size_t cbFirst; /** Currente offset (in bytes) in the data stream. */ @@ -356,5 +358,5 @@ uint32_t uHz; /** Shift count used for faster calculation of various - * values, such as the alignment, bytes to samples and so on. + * values, such as the alignment, bytes to frames and so on. * Depends on number of stream channels and the stream format * being used. @@ -367,5 +369,5 @@ } PDMAUDIOPCMPROPS, *PPDMAUDIOPCMPROPS; -/** Calculates the cShift value of given samples bits and audio channels. +/** Calculates the cShift value of given sample bits and audio channels. * Note: Does only support mono/stereo channels for now. */ #define PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(cBits, cChannels) ((cChannels == 2) + (cBits / 16)) @@ -373,10 +375,10 @@ * Note: Does only support mono/stereo channels for now. */ #define PDMAUDIOPCMPROPS_MAKE_SHIFT(pProps) PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS((pProps)->cChannels == 2) + (pProps)->cBits / 16) -/** Converts (audio) samples to bytes. +/** Converts (audio) frames to bytes. * Needs the cShift value set correctly, using PDMAUDIOPCMPROPS_MAKE_SHIFT. */ -#define PDMAUDIOPCMPROPS_S2B(pProps, samples) ((samples) << (pProps)->cShift) -/** Converts bytes to (audio) samples. +#define PDMAUDIOPCMPROPS_F2B(pProps, frames) ((frames) << (pProps)->cShift) +/** Converts bytes to (audio) frames. * Needs the cShift value set correctly, using PDMAUDIOPCMPROPS_MAKE_SHIFT. */ -#define PDMAUDIOPCMPROPS_B2S(pProps, cb) (cb >> (pProps)->cShift) +#define PDMAUDIOPCMPROPS_B2F(pProps, cb) (cb >> (pProps)->cShift) /** @@ -399,20 +401,20 @@ * * PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED: - * One stream at once. The consecutive audio data is exactly in the format and sample width + * One stream at once. The consecutive audio data is exactly in the format and frame width * like defined in the PCM properties. This is the default. * * PDMAUDIOSTREAMLAYOUT_RAW: * Can be one or many streams at once, depending on the stream's mixing buffer setup. - * The audio data will get handled as PDMAUDIOSAMPLE samples without any modification done. */ + * The audio data will get handled as PDMAUDIOFRAME frames without any modification done. */ PDMAUDIOSTREAMLAYOUT enmLayout; - /** Hint about the optimal sample buffer size (in audio samples). + /** Hint about the optimal frame buffer size (in audio frames). * 0 if no hint is given. */ - uint32_t cSampleBufferHint; + uint32_t cFrameBufferHint; } PDMAUDIOSTREAMCFG, *PPDMAUDIOSTREAMCFG; -/** Converts (audio) samples to bytes. */ -#define PDMAUDIOSTREAMCFG_S2B(pCfg, samples) ((samples) << (pCfg->Props).cShift) -/** Converts bytes to (audio) samples. */ -#define PDMAUDIOSTREAMCFG_B2S(pCfg, cb) (cb >> (pCfg->Props).cShift) +/** Converts (audio) frames to bytes. */ +#define PDMAUDIOSTREAMCFG_F2B(pCfg, frames) ((frames) << (pCfg->Props).cShift) +/** Converts bytes to (audio) frames. */ +#define PDMAUDIOSTREAMCFG_B2F(pCfg, cb) (cb >> (pCfg->Props).cShift) #if defined(RT_LITTLE_ENDIAN) @@ -507,7 +509,7 @@ * stream. */ uint32_t srcOffset; - /** Last processed sample of the input stream. + /** Last processed frame of the input stream. * Needed for interpolation. */ - PDMAUDIOSAMPLE srcSampleLast; + PDMAUDIOFRAME srcFrameLast; } PDMAUDIOSTRMRATE, *PPDMAUDIOSTRMRATE; @@ -532,11 +534,11 @@ /** - * Structure for holding sample conversion parameters for + * Structure for holding frame conversion parameters for * the audioMixBufConvFromXXX / audioMixBufConvToXXX macros. */ typedef struct PDMAUDMIXBUFCONVOPTS { - /** Number of audio samples to convert. */ - uint32_t cSamples; + /** Number of audio frames to convert. */ + uint32_t cFrames; union { @@ -554,5 +556,5 @@ /** - * Note: All internal handling is done in samples, + * Note: All internal handling is done in audio frames, * not in bytes! */ @@ -563,11 +565,11 @@ * Convertion-from function used by the PDM audio buffer mixer. * - * @returns Number of samples returned. - * @param paDst Where to return the converted samples. - * @param pvSrc The source samples bytes. + * @returns Number of audio frames returned. + * @param paDst Where to return the converted frames. + * @param pvSrc The source frame bytes. * @param cbSrc Number of bytes to convert. * @param pOpts Conversion options. */ -typedef DECLCALLBACK(uint32_t) FNPDMAUDIOMIXBUFCONVFROM(PPDMAUDIOSAMPLE paDst, const void *pvSrc, uint32_t cbSrc, +typedef DECLCALLBACK(uint32_t) FNPDMAUDIOMIXBUFCONVFROM(PPDMAUDIOFRAME paDst, const void *pvSrc, uint32_t cbSrc, PCPDMAUDMIXBUFCONVOPTS pOpts); /** Pointer to a convertion-from function used by the PDM audio buffer mixer. */ @@ -578,8 +580,8 @@ * * @param pvDst Output buffer. - * @param paSrc The input samples. + * @param paSrc The input frames. * @param pOpts Conversion options. */ -typedef DECLCALLBACK(void) FNPDMAUDIOMIXBUFCONVTO(void *pvDst, PCPDMAUDIOSAMPLE paSrc, PCPDMAUDMIXBUFCONVOPTS pOpts); +typedef DECLCALLBACK(void) FNPDMAUDIOMIXBUFCONVTO(void *pvDst, PCPDMAUDIOFRAME paSrc, PCPDMAUDMIXBUFCONVOPTS pOpts); /** Pointer to a convertion-to function used by the PDM audio buffer mixer. */ typedef FNPDMAUDIOMIXBUFCONVTO *PFNPDMAUDIOMIXBUFCONVTO; @@ -591,21 +593,21 @@ /** Name of the buffer. */ char *pszName; - /** Sample buffer. */ - PPDMAUDIOSAMPLE pSamples; - /** Size of the sample buffer (in samples). */ - uint32_t cSamples; - /** The current read position (in samples). */ + /** Frame buffer. */ + PPDMAUDIOFRAME pFrames; + /** Size of the frame buffer (in audio frames). */ + uint32_t cFrames; + /** The current read position (in frames). */ uint32_t offRead; - /** The current write position (in samples). */ + /** The current write position (in frames). */ uint32_t offWrite; /** - * Total samples already mixed down to the parent buffer (if any). Always starting at + * Total frames already mixed down to the parent buffer (if any). Always starting at * the parent's offRead position. * - * Note: Count always is specified in parent samples, as the sample count can differ between parent + * Note: Count always is specified in parent frames, as the sample count can differ between parent * and child. */ uint32_t cMixed; - /** How much audio samples are currently being used + /** How much audio frames are currently being used * in this buffer. * Note: This also is known as the distance in ring buffer terms. */ @@ -638,5 +640,5 @@ */ int64_t iFreqRatio; - /** For quickly converting samples <-> bytes and vice versa. */ + /** For quickly converting frames <-> bytes and vice versa. */ uint8_t cShift; } PDMAUDIOMIXBUF; @@ -750,5 +752,5 @@ STAMCOUNTER StatBytesElapsed; STAMCOUNTER StatBytesTotalRead; - STAMCOUNTER StatSamplesCaptured; + STAMCOUNTER StatFramesCaptured; #endif } PDMAUDIOSTREAMIN, *PPDMAUDIOSTREAMIN; @@ -765,5 +767,5 @@ STAMCOUNTER StatBytesElapsed; STAMCOUNTER StatBytesTotalWritten; - STAMCOUNTER StatSamplesPlayed; + STAMCOUNTER StatFramesPlayed; #endif } PDMAUDIOSTREAMOUT, *PPDMAUDIOSTREAMOUT; @@ -1037,22 +1039,22 @@ /** - * Plays (transfers) available audio samples via the host backend. Only works with output streams. + * Plays (transfers) available audio frames via the host backend. Only works with output streams. * * @returns VBox status code. * @param pInterface Pointer to the interface structure containing the called function pointer. * @param pStream Pointer to audio stream. - * @param pcSamplesPlayed Number of samples played. Optional. - */ - DECLR3CALLBACKMEMBER(int, pfnStreamPlay, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcSamplesPlayed)); - - /** - * Captures (transfers) available audio samples from the host backend. Only works with input streams. + * @param pcFramesPlayed Number of frames played. Optional. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamPlay, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcFramesPlayed)); + + /** + * Captures (transfers) available audio frames from the host backend. Only works with input streams. * * @returns VBox status code. * @param pInterface Pointer to the interface structure containing the called function pointer. * @param pStream Pointer to audio stream. - * @param pcSamplesCaptured Number of samples captured. Optional. - */ - DECLR3CALLBACKMEMBER(int, pfnStreamCapture, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcSamplesCaptured)); + * @param pcFramesCaptured Number of frames captured. Optional. + */ + DECLR3CALLBACKMEMBER(int, pfnStreamCapture, (PPDMIAUDIOCONNECTOR pInterface, PPDMAUDIOSTREAM pStream, uint32_t *pcFramesCaptured)); #ifdef VBOX_WITH_AUDIO_DEVICE_CALLBACKS @@ -1183,5 +1185,5 @@ * * @returns For non-raw layout streams: Number of readable bytes. - * for raw layout streams : Number of readable audio samples. + * for raw layout streams : Number of readable audio frames. * @param pInterface Pointer to the interface structure containing the called function pointer. * @param pStream Pointer to audio stream. @@ -1193,5 +1195,5 @@ * * @returns For non-raw layout streams: Number of writable bytes. - * for raw layout streams : Number of writable audio samples. + * for raw layout streams : Number of writable audio frames. * @param pInterface Pointer to the interface structure containing the called function pointer. * @param pStream Pointer to audio stream. @@ -1233,7 +1235,7 @@ * @param pvBuf Pointer to audio data buffer to play. * @param cxBuf For non-raw layout streams: Size (in bytes) of audio data buffer, - * for raw layout streams : Size (in audio samples) of audio data buffer. - * @param pcxWritten For non-raw layout streams: Returns number of bytes written. Optional. - * for raw layout streams : Returns number of samples written. Optional. + * for raw layout streams : Size (in audio frames) of audio data buffer. + * @param pcxWritten For non-raw layout streams: Returns number of bytes written. Optional. + * for raw layout streams : Returns number of frames written. Optional. */ DECLR3CALLBACKMEMBER(int, pfnStreamPlay, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, const void *pvBuf, uint32_t cxBuf, uint32_t *pcxWritten)); @@ -1263,7 +1265,7 @@ * @param pvBuf Buffer where to store read audio data. * @param cxBuf For non-raw layout streams: Size (in bytes) of audio data buffer, - * for raw layout streams : Size (in audio samples) of audio data buffer. - * @param pcxRead For non-raw layout streams: Returns number of bytes read. Optional. - * for raw layout streams : Returns number of samples read. Optional. + * for raw layout streams : Size (in audio frames) of audio data buffer. + * @param pcxRead For non-raw layout streams: Returns number of bytes read. Optional. + * for raw layout streams : Returns number of frames read. Optional. */ DECLR3CALLBACKMEMBER(int, pfnStreamCapture, (PPDMIHOSTAUDIO pInterface, PPDMAUDIOBACKENDSTREAM pStream, void *pvBuf, uint32_t cxBuf, uint32_t *pcxRead)); Index: /trunk/src/VBox/Devices/Audio/AudioMixBuffer.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/AudioMixBuffer.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/AudioMixBuffer.cpp (revision 68132) @@ -1,6 +1,5 @@ /* $Id$ */ /** @file - * VBox audio: Audio mixing buffer for converting reading/writing audio - * samples. + * VBox audio: Audio mixing buffer for converting reading/writing audio data. */ @@ -137,49 +136,49 @@ /** - * Peeks for audio samples without any conversion done. - * This will get the raw sample data out of a mixing buffer. + * Peeks for audio frames without any conversion done. + * This will get the raw frame data out of a mixing buffer. * * @return IPRT status code or VINF_AUDIO_MORE_DATA_AVAILABLE if more data is available to read. * - * @param pMixBuf Mixing buffer to acquire audio samples from. - * @param cSamplesToRead Number of audio samples to read. - * @param paSampleBuf Buffer where to store the returned audio samples. - * @param cSampleBuf Size (in samples) of the buffer to store audio samples into. - * @param pcSamplesRead Returns number of read audio samples. Optional. + * @param pMixBuf Mixing buffer to acquire audio frames from. + * @param cFramesToRead Number of audio frames to read. + * @param paFrameBuf Buffer where to store the returned audio frames. + * @param cFrameBuf Size (in frames) of the buffer to store audio frames into. + * @param pcFramesRead Returns number of read audio frames. Optional. * * @remark This function is not thread safe! */ -int AudioMixBufPeek(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamplesToRead, - PPDMAUDIOSAMPLE paSampleBuf, uint32_t cSampleBuf, uint32_t *pcSamplesRead) -{ - AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER); - AssertPtrReturn(paSampleBuf, VERR_INVALID_POINTER); - AssertReturn(cSampleBuf, VERR_INVALID_PARAMETER); +int AudioMixBufPeek(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFramesToRead, + PPDMAUDIOFRAME paFrameBuf, uint32_t cFrameBuf, uint32_t *pcFramesRead) +{ + AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER); + AssertPtrReturn(paFrameBuf, VERR_INVALID_POINTER); + AssertReturn(cFrameBuf, VERR_INVALID_PARAMETER); /* pcRead is optional. */ int rc; - if (!cSamplesToRead) - { - if (pcSamplesRead) - *pcSamplesRead = 0; + if (!cFramesToRead) + { + if (pcFramesRead) + *pcFramesRead = 0; return VINF_SUCCESS; } uint32_t cRead; - if (pMixBuf->offRead + cSamplesToRead > pMixBuf->cSamples) - { - cRead = pMixBuf->cSamples - pMixBuf->offRead; + if (pMixBuf->offRead + cFramesToRead > pMixBuf->cFrames) + { + cRead = pMixBuf->cFrames - pMixBuf->offRead; rc = VINF_AUDIO_MORE_DATA_AVAILABLE; } else { - cRead = cSamplesToRead; + cRead = cFramesToRead; rc = VINF_SUCCESS; } - if (cRead > cSampleBuf) - { - cRead = cSampleBuf; + if (cRead > cFrameBuf) + { + cRead = cFrameBuf; rc = VINF_AUDIO_MORE_DATA_AVAILABLE; } @@ -187,13 +186,13 @@ if (cRead) { - memcpy(paSampleBuf, &pMixBuf->pSamples[pMixBuf->offRead], sizeof(PDMAUDIOSAMPLE) * cRead); - - pMixBuf->offRead = (pMixBuf->offRead + cRead) % pMixBuf->cSamples; - Assert(pMixBuf->offRead <= pMixBuf->cSamples); + memcpy(paFrameBuf, &pMixBuf->pFrames[pMixBuf->offRead], sizeof(PDMAUDIOFRAME) * cRead); + + pMixBuf->offRead = (pMixBuf->offRead + cRead) % pMixBuf->cFrames; + Assert(pMixBuf->offRead <= pMixBuf->cFrames); pMixBuf->cUsed -= RT_MIN(cRead, pMixBuf->cUsed); } - if (pcSamplesRead) - *pcSamplesRead = cRead; + if (pcFramesRead) + *pcFramesRead = cRead; return rc; @@ -201,50 +200,50 @@ /** - * Returns a mutable pointer to the mixing buffer's audio sample buffer for writing raw - * audio samples. + * Returns a mutable pointer to the mixing buffer's audio frame buffer for writing raw + * audio frames. * * @return IPRT status code. VINF_TRY_AGAIN for getting next pointer at beginning (circular). - * @param pMixBuf Mixing buffer to acquire audio samples from. - * @param cSamples Number of requested audio samples to write. - * @param ppvSamples Returns a mutable pointer to the buffer's audio sample data. - * @param pcSamplesToWrite Number of available audio samples to write. + * @param pMixBuf Mixing buffer to acquire audio frames from. + * @param cFrames Number of requested audio frames to write. + * @param ppvFrames Returns a mutable pointer to the buffer's audio frame data. + * @param pcFramesToWrite Number of available audio frames to write. * * @remark This function is not thread safe! */ -int AudioMixBufPeekMutable(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamples, - PPDMAUDIOSAMPLE *ppvSamples, uint32_t *pcSamplesToWrite) -{ - AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER); - AssertPtrReturn(ppvSamples, VERR_INVALID_POINTER); - AssertPtrReturn(pcSamplesToWrite, VERR_INVALID_POINTER); +int AudioMixBufPeekMutable(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFrames, + PPDMAUDIOFRAME *ppvFrames, uint32_t *pcFramesToWrite) +{ + AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER); + AssertPtrReturn(ppvFrames, VERR_INVALID_POINTER); + AssertPtrReturn(pcFramesToWrite, VERR_INVALID_POINTER); int rc; - if (!cSamples) - { - *pcSamplesToWrite = 0; + if (!cFrames) + { + *pcFramesToWrite = 0; return VINF_SUCCESS; } - uint32_t cSamplesToWrite; - if (pMixBuf->offWrite + cSamples > pMixBuf->cSamples) - { - cSamplesToWrite = pMixBuf->cSamples - pMixBuf->offWrite; + uint32_t cFramesToWrite; + if (pMixBuf->offWrite + cFrames > pMixBuf->cFrames) + { + cFramesToWrite = pMixBuf->cFrames - pMixBuf->offWrite; rc = VINF_TRY_AGAIN; } else { - cSamplesToWrite = cSamples; + cFramesToWrite = cFrames; rc = VINF_SUCCESS; } - *ppvSamples = &pMixBuf->pSamples[pMixBuf->offWrite]; - AssertPtr(ppvSamples); - - pMixBuf->offWrite = (pMixBuf->offWrite + cSamplesToWrite) % pMixBuf->cSamples; - Assert(pMixBuf->offWrite <= pMixBuf->cSamples); - pMixBuf->cUsed += RT_MIN(cSamplesToWrite, pMixBuf->cUsed); - - *pcSamplesToWrite = cSamplesToWrite; + *ppvFrames = &pMixBuf->pFrames[pMixBuf->offWrite]; + AssertPtr(ppvFrames); + + pMixBuf->offWrite = (pMixBuf->offWrite + cFramesToWrite) % pMixBuf->cFrames; + Assert(pMixBuf->offWrite <= pMixBuf->cFrames); + pMixBuf->cUsed += RT_MIN(cFramesToWrite, pMixBuf->cUsed); + + *pcFramesToWrite = cFramesToWrite; return rc; @@ -252,5 +251,5 @@ /** - * Clears the entire sample buffer. + * Clears the entire frame buffer. * * @param pMixBuf Mixing buffer to clear. @@ -261,18 +260,18 @@ AssertPtrReturnVoid(pMixBuf); - if (pMixBuf->cSamples) - RT_BZERO(pMixBuf->pSamples, pMixBuf->cSamples * sizeof(PDMAUDIOSAMPLE)); -} - -/** - * Clears (zeroes) the buffer by a certain amount of (used) samples and + if (pMixBuf->cFrames) + RT_BZERO(pMixBuf->pFrames, pMixBuf->cFrames * sizeof(PDMAUDIOFRAME)); +} + +/** + * Clears (zeroes) the buffer by a certain amount of (used) frames and * keeps track to eventually assigned children buffers. * * @param pMixBuf Mixing buffer to clear. - * @param cSamplesToClear Number of audio samples to clear. - */ -void AudioMixBufFinish(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamplesToClear) -{ - AUDMIXBUF_LOG(("cSamplesToClear=%RU32\n", cSamplesToClear)); + * @param cFramesToClear Number of audio frames to clear. + */ +void AudioMixBufFinish(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFramesToClear) +{ + AUDMIXBUF_LOG(("cFramesToClear=%RU32\n", cFramesToClear)); AUDMIXBUF_LOG(("%s: offRead=%RU32, cUsed=%RU32\n", pMixBuf->pszName, pMixBuf->offRead, pMixBuf->cUsed)); @@ -282,11 +281,11 @@ { AUDMIXBUF_LOG(("\t%s: cMixed=%RU32 -> %RU32\n", - pIter->pszName, pIter->cMixed, pIter->cMixed - cSamplesToClear)); - - pIter->cMixed -= RT_MIN(pIter->cMixed, cSamplesToClear); + pIter->pszName, pIter->cMixed, pIter->cMixed - cFramesToClear)); + + pIter->cMixed -= RT_MIN(pIter->cMixed, cFramesToClear); /* Note: Do not increment pIter->cUsed here, as this gets done when reading from that buffer using AudioMixBufReadXXX. */ } - Assert(cSamplesToClear <= pMixBuf->cSamples); + Assert(cFramesToClear <= pMixBuf->cFrames); uint32_t cClearOff; @@ -294,31 +293,31 @@ /* Clear end of buffer (wrap around). */ - if (cSamplesToClear > pMixBuf->offRead) - { - cClearOff = pMixBuf->cSamples - (cSamplesToClear - pMixBuf->offRead); - cClearLen = pMixBuf->cSamples - cClearOff; + if (cFramesToClear > pMixBuf->offRead) + { + cClearOff = pMixBuf->cFrames - (cFramesToClear - pMixBuf->offRead); + cClearLen = pMixBuf->cFrames - cClearOff; AUDMIXBUF_LOG(("Clearing1: %RU32 - %RU32\n", cClearOff, cClearOff + cClearLen)); - RT_BZERO(pMixBuf->pSamples + cClearOff, cClearLen * sizeof(PDMAUDIOSAMPLE)); - - Assert(cSamplesToClear >= cClearLen); - cSamplesToClear -= cClearLen; + RT_BZERO(pMixBuf->pFrames + cClearOff, cClearLen * sizeof(PDMAUDIOFRAME)); + + Assert(cFramesToClear >= cClearLen); + cFramesToClear -= cClearLen; } /* Clear beginning of buffer. */ - if ( cSamplesToClear + if ( cFramesToClear && pMixBuf->offRead) { - Assert(pMixBuf->offRead >= cSamplesToClear); - - cClearOff = pMixBuf->offRead - cSamplesToClear; - cClearLen = cSamplesToClear; - - Assert(cClearOff + cClearLen <= pMixBuf->cSamples); + Assert(pMixBuf->offRead >= cFramesToClear); + + cClearOff = pMixBuf->offRead - cFramesToClear; + cClearLen = cFramesToClear; + + Assert(cClearOff + cClearLen <= pMixBuf->cFrames); AUDMIXBUF_LOG(("Clearing2: %RU32 - %RU32\n", cClearOff, cClearOff + cClearLen)); - RT_BZERO(pMixBuf->pSamples + cClearOff, cClearLen * sizeof(PDMAUDIOSAMPLE)); + RT_BZERO(pMixBuf->pFrames + cClearOff, cClearLen * sizeof(PDMAUDIOFRAME)); } } @@ -350,19 +349,19 @@ } - if (pMixBuf->pSamples) - { - Assert(pMixBuf->cSamples); - - RTMemFree(pMixBuf->pSamples); - pMixBuf->pSamples = NULL; - } - - pMixBuf->cSamples = 0; -} - -/** - * Returns the size (in audio samples) of free audio buffer space. - * - * @return uint32_t Size (in audio samples) of free audio buffer space. + if (pMixBuf->pFrames) + { + Assert(pMixBuf->cFrames); + + RTMemFree(pMixBuf->pFrames); + pMixBuf->pFrames = NULL; + } + + pMixBuf->cFrames = 0; +} + +/** + * Returns the size (in audio frames) of free audio buffer space. + * + * @return uint32_t Size (in audio frames) of free audio buffer space. * @param pMixBuf Mixing buffer to return free size for. */ @@ -371,25 +370,25 @@ AssertPtrReturn(pMixBuf, 0); - uint32_t cSamples, cSamplesFree; + uint32_t cFrames, cFramesFree; if (pMixBuf->pParent) { /* - * As a linked child buffer we want to know how many samples + * As a linked child buffer we want to know how many frames * already have been consumed by the parent. */ - cSamples = pMixBuf->pParent->cSamples; - - Assert(pMixBuf->cMixed <= cSamples); - cSamplesFree = cSamples - pMixBuf->cMixed; + cFrames = pMixBuf->pParent->cFrames; + + Assert(pMixBuf->cMixed <= cFrames); + cFramesFree = cFrames - pMixBuf->cMixed; } else /* As a parent. */ { - cSamples = pMixBuf->cSamples; - Assert(cSamples >= pMixBuf->cUsed); - cSamplesFree = pMixBuf->cSamples - pMixBuf->cUsed; - } - - AUDMIXBUF_LOG(("%s: %RU32 of %RU32\n", pMixBuf->pszName, cSamplesFree, cSamples)); - return cSamplesFree; + cFrames = pMixBuf->cFrames; + Assert(cFrames >= pMixBuf->cUsed); + cFramesFree = pMixBuf->cFrames - pMixBuf->cUsed; + } + + AUDMIXBUF_LOG(("%s: %RU32 of %RU32\n", pMixBuf->pszName, cFramesFree, cFrames)); + return cFramesFree; } @@ -402,26 +401,26 @@ uint32_t AudioMixBufFreeBytes(PPDMAUDIOMIXBUF pMixBuf) { - return AUDIOMIXBUF_S2B(pMixBuf, AudioMixBufFree(pMixBuf)); -} - -/** - * Allocates the internal audio sample buffer. + return AUDIOMIXBUF_F2B(pMixBuf, AudioMixBufFree(pMixBuf)); +} + +/** + * Allocates the internal audio frame buffer. * * @return IPRT status code. - * @param pMixBuf Mixing buffer to allocate sample buffer for. - * @param cSamples Number of audio samples to allocate. - */ -static int audioMixBufAlloc(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamples) + * @param pMixBuf Mixing buffer to allocate frame buffer for. + * @param cFrames Number of audio frames to allocate. + */ +static int audioMixBufAlloc(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFrames) { AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER); - AssertReturn(cSamples, VERR_INVALID_PARAMETER); - - AUDMIXBUF_LOG(("%s: cSamples=%RU32\n", pMixBuf->pszName, cSamples)); - - size_t cbSamples = cSamples * sizeof(PDMAUDIOSAMPLE); - pMixBuf->pSamples = (PPDMAUDIOSAMPLE)RTMemAllocZ(cbSamples); - if (pMixBuf->pSamples) - { - pMixBuf->cSamples = cSamples; + AssertReturn(cFrames, VERR_INVALID_PARAMETER); + + AUDMIXBUF_LOG(("%s: cFrames=%RU32\n", pMixBuf->pszName, cFrames)); + + size_t cbFrames = cFrames * sizeof(PDMAUDIOFRAME); + pMixBuf->pFrames = (PPDMAUDIOFRAME)RTMemAllocZ(cbFrames); + if (pMixBuf->pFrames) + { + pMixBuf->cFrames = cFrames; return VINF_SUCCESS; } @@ -467,12 +466,12 @@ } \ \ - DECLCALLBACK(uint32_t) audioMixBufConvFrom##_aName##Stereo(PPDMAUDIOSAMPLE paDst, const void *pvSrc, uint32_t cbSrc, \ + DECLCALLBACK(uint32_t) audioMixBufConvFrom##_aName##Stereo(PPDMAUDIOFRAME paDst, const void *pvSrc, uint32_t cbSrc, \ PCPDMAUDMIXBUFCONVOPTS pOpts) \ { \ _aType const *pSrc = (_aType const *)pvSrc; \ - uint32_t cSamples = RT_MIN(pOpts->cSamples, cbSrc / sizeof(_aType)); \ - AUDMIXBUF_MACRO_LOG(("cSamples=%RU32, BpS=%zu, lVol=%RU32, rVol=%RU32\n", \ - pOpts->cSamples, sizeof(_aType), pOpts->From.Volume.uLeft, pOpts->From.Volume.uRight)); \ - for (uint32_t i = 0; i < cSamples; i++) \ + uint32_t cFrames = RT_MIN(pOpts->cFrames, cbSrc / sizeof(_aType)); \ + AUDMIXBUF_MACRO_LOG(("cFrames=%RU32, BpS=%zu, lVol=%RU32, rVol=%RU32\n", \ + pOpts->cFrames, sizeof(_aType), pOpts->From.Volume.uLeft, pOpts->From.Volume.uRight)); \ + for (uint32_t i = 0; i < cFrames; i++) \ { \ paDst->i64LSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc++), pOpts->From.Volume.uLeft ) >> AUDIOMIXBUF_VOL_SHIFT; \ @@ -481,15 +480,15 @@ } \ \ - return cSamples; \ + return cFrames; \ } \ \ - DECLCALLBACK(uint32_t) audioMixBufConvFrom##_aName##Mono(PPDMAUDIOSAMPLE paDst, const void *pvSrc, uint32_t cbSrc, \ + DECLCALLBACK(uint32_t) audioMixBufConvFrom##_aName##Mono(PPDMAUDIOFRAME paDst, const void *pvSrc, uint32_t cbSrc, \ PCPDMAUDMIXBUFCONVOPTS pOpts) \ { \ _aType const *pSrc = (_aType const *)pvSrc; \ - const uint32_t cSamples = RT_MIN(pOpts->cSamples, cbSrc / sizeof(_aType)); \ - AUDMIXBUF_MACRO_LOG(("cSamples=%RU32, BpS=%zu, lVol=%RU32, rVol=%RU32\n", \ - cSamples, sizeof(_aType), pOpts->From.Volume.uLeft, pOpts->From.Volume.uRight)); \ - for (uint32_t i = 0; i < cSamples; i++) \ + const uint32_t cFrames = RT_MIN(pOpts->cFrames, cbSrc / sizeof(_aType)); \ + AUDMIXBUF_MACRO_LOG(("cFrames=%RU32, BpS=%zu, lVol=%RU32, rVol=%RU32\n", \ + cFrames, sizeof(_aType), pOpts->From.Volume.uLeft, pOpts->From.Volume.uRight)); \ + for (uint32_t i = 0; i < cFrames; i++) \ { \ paDst->i64LSample = ASMMult2xS32RetS64((int32_t)audioMixBufClipFrom##_aName(*pSrc), pOpts->From.Volume.uLeft) >> AUDIOMIXBUF_VOL_SHIFT; \ @@ -499,14 +498,14 @@ } \ \ - return cSamples; \ + return cFrames; \ } \ \ - DECLCALLBACK(void) audioMixBufConvTo##_aName##Stereo(void *pvDst, PCPDMAUDIOSAMPLE paSrc, PCPDMAUDMIXBUFCONVOPTS pOpts) \ + DECLCALLBACK(void) audioMixBufConvTo##_aName##Stereo(void *pvDst, PCPDMAUDIOFRAME paSrc, PCPDMAUDMIXBUFCONVOPTS pOpts) \ { \ - PCPDMAUDIOSAMPLE pSrc = paSrc; \ + PCPDMAUDIOFRAME pSrc = paSrc; \ _aType *pDst = (_aType *)pvDst; \ _aType l, r; \ - uint32_t cSamples = pOpts->cSamples; \ - while (cSamples--) \ + uint32_t cFrames = pOpts->cFrames; \ + while (cFrames--) \ { \ AUDMIXBUF_MACRO_LOG(("%p: l=%RI64, r=%RI64\n", pSrc, pSrc->i64LSample, pSrc->i64RSample)); \ @@ -520,10 +519,10 @@ } \ \ - DECLCALLBACK(void) audioMixBufConvTo##_aName##Mono(void *pvDst, PCPDMAUDIOSAMPLE paSrc, PCPDMAUDMIXBUFCONVOPTS pOpts) \ + DECLCALLBACK(void) audioMixBufConvTo##_aName##Mono(void *pvDst, PCPDMAUDIOFRAME paSrc, PCPDMAUDMIXBUFCONVOPTS pOpts) \ { \ - PCPDMAUDIOSAMPLE pSrc = paSrc; \ + PCPDMAUDIOFRAME pSrc = paSrc; \ _aType *pDst = (_aType *)pvDst; \ - uint32_t cSamples = pOpts->cSamples; \ - while (cSamples--) \ + uint32_t cFrames = pOpts->cFrames; \ + while (cFrames--) \ { \ *pDst++ = audioMixBufClipTo##_aName((pSrc->i64LSample + pSrc->i64RSample) / 2); \ @@ -548,10 +547,10 @@ #define AUDMIXBUF_MIXOP(_aName, _aOp) \ - static void audioMixBufOp##_aName(PPDMAUDIOSAMPLE paDst, uint32_t cDstSamples, \ - PPDMAUDIOSAMPLE paSrc, uint32_t cSrcSamples, \ + static void audioMixBufOp##_aName(PPDMAUDIOFRAME paDst, uint32_t cDstFrames, \ + PPDMAUDIOFRAME paSrc, uint32_t cSrcFrames, \ PPDMAUDIOSTRMRATE pRate, \ uint32_t *pcDstWritten, uint32_t *pcSrcRead) \ { \ - AUDMIXBUF_MACRO_LOG(("cSrcSamples=%RU32, cDstSamples=%RU32\n", cSrcSamples, cDstSamples)); \ + AUDMIXBUF_MACRO_LOG(("cSrcFrames=%RU32, cDstFrames=%RU32\n", cSrcFrames, cDstFrames)); \ AUDMIXBUF_MACRO_LOG(("Rate: srcOffset=%RU32, dstOffset=%RU32, dstInc=%RU32\n", \ pRate->srcOffset, \ @@ -560,7 +559,7 @@ if (pRate->dstInc == (UINT64_C(1) + UINT32_MAX)) /* No conversion needed? */ \ { \ - uint32_t cSamples = RT_MIN(cSrcSamples, cDstSamples); \ - AUDMIXBUF_MACRO_LOG(("cSamples=%RU32\n", cSamples)); \ - for (uint32_t i = 0; i < cSamples; i++) \ + uint32_t cFrames = RT_MIN(cSrcFrames, cDstFrames); \ + AUDMIXBUF_MACRO_LOG(("cFrames=%RU32\n", cFrames)); \ + for (uint32_t i = 0; i < cFrames; i++) \ { \ paDst[i].i64LSample _aOp paSrc[i].i64LSample; \ @@ -569,17 +568,17 @@ \ if (pcDstWritten) \ - *pcDstWritten = cSamples; \ + *pcDstWritten = cFrames; \ if (pcSrcRead) \ - *pcSrcRead = cSamples; \ + *pcSrcRead = cFrames; \ return; \ } \ \ - PPDMAUDIOSAMPLE paSrcStart = paSrc; \ - PPDMAUDIOSAMPLE paSrcEnd = paSrc + cSrcSamples; \ - PPDMAUDIOSAMPLE paDstStart = paDst; \ - PPDMAUDIOSAMPLE paDstEnd = paDst + cDstSamples; \ - PDMAUDIOSAMPLE samCur = { 0 }; \ - PDMAUDIOSAMPLE samOut; \ - PDMAUDIOSAMPLE samLast = pRate->srcSampleLast; \ + PPDMAUDIOFRAME paSrcStart = paSrc; \ + PPDMAUDIOFRAME paSrcEnd = paSrc + cSrcFrames; \ + PPDMAUDIOFRAME paDstStart = paDst; \ + PPDMAUDIOFRAME paDstEnd = paDst + cDstFrames; \ + PDMAUDIOFRAME frameCur = { 0 }; \ + PDMAUDIOFRAME frameOut; \ + PDMAUDIOFRAME frameLast = pRate->srcFrameLast; \ \ while (paDst < paDstEnd) \ @@ -593,5 +592,5 @@ { \ Assert(paSrc <= paSrcEnd); \ - samLast = *paSrc++; \ + frameLast = *paSrc++; \ pRate->srcOffset++; \ if (paSrc == paSrcEnd) \ @@ -603,19 +602,19 @@ break; \ \ - samCur = *paSrc; \ + frameCur = *paSrc; \ \ /* Interpolate. */ \ int64_t iDstOffInt = pRate->dstOffset & UINT32_MAX; \ \ - samOut.i64LSample = (samLast.i64LSample * ((int64_t) (INT64_C(1) << 32) - iDstOffInt) + samCur.i64LSample * iDstOffInt) >> 32; \ - samOut.i64RSample = (samLast.i64RSample * ((int64_t) (INT64_C(1) << 32) - iDstOffInt) + samCur.i64RSample * iDstOffInt) >> 32; \ + frameOut.i64LSample = (frameLast.i64LSample * ((int64_t) (INT64_C(1) << 32) - iDstOffInt) + frameCur.i64LSample * iDstOffInt) >> 32; \ + frameOut.i64RSample = (frameLast.i64RSample * ((int64_t) (INT64_C(1) << 32) - iDstOffInt) + frameCur.i64RSample * iDstOffInt) >> 32; \ \ - paDst->i64LSample _aOp samOut.i64LSample; \ - paDst->i64RSample _aOp samOut.i64RSample; \ + paDst->i64LSample _aOp frameOut.i64LSample; \ + paDst->i64RSample _aOp frameOut.i64RSample; \ \ AUDMIXBUF_MACRO_LOG(("\tiDstOffInt=%RI64, l=%RI64, r=%RI64 (cur l=%RI64, r=%RI64)\n", \ iDstOffInt, \ paDst->i64LSample >> 32, paDst->i64RSample >> 32, \ - samCur.i64LSample >> 32, samCur.i64RSample >> 32)); \ + frameCur.i64LSample >> 32, frameCur.i64RSample >> 32)); \ \ paDst++; \ @@ -626,10 +625,10 @@ } \ \ - AUDMIXBUF_MACRO_LOG(("%zu source samples -> %zu dest samples\n", paSrc - paSrcStart, paDst - paDstStart)); \ + AUDMIXBUF_MACRO_LOG(("%zu source frames -> %zu dest frames\n", paSrc - paSrcStart, paDst - paDstStart)); \ \ - pRate->srcSampleLast = samLast; \ + pRate->srcFrameLast = frameLast; \ \ AUDMIXBUF_MACRO_LOG(("pRate->srcSampleLast l=%RI64, r=%RI64\n", \ - pRate->srcSampleLast.i64LSample, pRate->srcSampleLast.i64RSample)); \ + pRate->srcFrameLast.i64LSample, pRate->srcFrameLast.i64RSample)); \ \ if (pcDstWritten) \ @@ -651,16 +650,16 @@ /** Dummy conversion used when the source is muted. */ static DECLCALLBACK(uint32_t) -audioMixBufConvFromSilence(PPDMAUDIOSAMPLE paDst, const void *pvSrc, uint32_t cbSrc, PCPDMAUDMIXBUFCONVOPTS pOpts) +audioMixBufConvFromSilence(PPDMAUDIOFRAME paDst, const void *pvSrc, uint32_t cbSrc, PCPDMAUDMIXBUFCONVOPTS pOpts) { RT_NOREF(cbSrc, pvSrc); /* Internally zero always corresponds to silence. */ - RT_BZERO(paDst, pOpts->cSamples * sizeof(paDst[0])); - return pOpts->cSamples; + RT_BZERO(paDst, pOpts->cFrames * sizeof(paDst[0])); + return pOpts->cFrames; } /** * Looks up the matching conversion (macro) routine for converting - * audio samples from a source format. + * audio frames from a source format. * ** @todo Speed up the lookup by binding it to the actual stream state. @@ -722,5 +721,5 @@ /** * Looks up the matching conversion (macro) routine for converting - * audio samples to a destination format. + * audio frames to a destination format. * ** @todo Speed up the lookup by binding it to the actual stream state. @@ -811,7 +810,7 @@ * @param pszName Name of mixing buffer for easier identification. Optional. * @param pProps PCM audio properties to use for the mixing buffer. - * @param cSamples Maximum number of audio samples the mixing buffer can hold. - */ -int AudioMixBufInit(PPDMAUDIOMIXBUF pMixBuf, const char *pszName, PPDMAUDIOPCMPROPS pProps, uint32_t cSamples) + * @param cFrames Maximum number of audio frames the mixing buffer can hold. + */ +int AudioMixBufInit(PPDMAUDIOMIXBUF pMixBuf, const char *pszName, PPDMAUDIOPCMPROPS pProps, uint32_t cFrames) { AssertPtrReturn(pMixBuf, VERR_INVALID_POINTER); @@ -824,6 +823,6 @@ pMixBuf->cChildren = 0; - pMixBuf->pSamples = NULL; - pMixBuf->cSamples = 0; + pMixBuf->pFrames = NULL; + pMixBuf->cFrames = 0; pMixBuf->offRead = 0; @@ -863,12 +862,12 @@ RT_BOOL(AUDMIXBUF_FMT_SIGNED(pMixBuf->AudioFmt)))); - return audioMixBufAlloc(pMixBuf, cSamples); -} - -/** - * Returns @c true if there are any audio samples available for processing, + return audioMixBufAlloc(pMixBuf, cFrames); +} + +/** + * Returns @c true if there are any audio frames available for processing, * @c false if not. * - * @return bool @c true if there are any audio samples available for processing, @c false if not. + * @return bool @c true if there are any audio frames available for processing, @c false if not. * @param pMixBuf Mixing buffer to return value for. */ @@ -926,5 +925,5 @@ AssertMsgReturn(AUDMIXBUF_FMT_SAMPLE_FREQ(pParent->AudioFmt), - ("Parent sample frequency (Hz) not set\n"), VERR_INVALID_PARAMETER); + ("Parent frame frequency (Hz) not set\n"), VERR_INVALID_PARAMETER); AssertMsgReturn(AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->AudioFmt), ("Buffer sample frequency (Hz) not set\n"), VERR_INVALID_PARAMETER); @@ -950,17 +949,17 @@ int rc = VINF_SUCCESS; #if 0 - uint32_t cSamples = (uint32_t)RT_MIN( ((uint64_t)pParent->cSamples << 32) - / pMixBuf->iFreqRatio, _64K /* 64K samples max. */); - if (!cSamples) - cSamples = pParent->cSamples; + uint32_t cFrames = (uint32_t)RT_MIN( ((uint64_t)pParent->cFrames << 32) + / pMixBuf->iFreqRatio, _64K /* 64K frames max. */); + if (!cFrames) + cFrames = pParent->cFrames; int rc = VINF_SUCCESS; - if (cSamples != pMixBuf->cSamples) - { - AUDMIXBUF_LOG(("%s: Reallocating samples %RU32 -> %RU32\n", - pMixBuf->pszName, pMixBuf->cSamples, cSamples)); - - uint32_t cbSamples = cSamples * sizeof(PDMAUDIOSAMPLE); + if (cFrames != pMixBuf->cFrames) + { + AUDMIXBUF_LOG(("%s: Reallocating frames %RU32 -> %RU32\n", + pMixBuf->pszName, pMixBuf->cFrames, cFrames)); + + uint32_t cbSamples = cFrames * sizeof(PDMAUDIOSAMPLE); Assert(cbSamples); pMixBuf->pSamples = (PPDMAUDIOSAMPLE)RTMemRealloc(pMixBuf->pSamples, cbSamples); @@ -970,5 +969,5 @@ if (RT_SUCCESS(rc)) { - pMixBuf->cSamples = cSamples; + pMixBuf->cFrames = cFrames; /* Make sure to zero the reallocated buffer so that it can be @@ -994,11 +993,11 @@ / AUDMIXBUF_FMT_SAMPLE_FREQ(pParent->AudioFmt); - AUDMIXBUF_LOG(("uThisHz=%RU32, uParentHz=%RU32, iFreqRatio=0x%RX64 (%RI64), uRateInc=0x%RX64 (%RU64), cSamples=%RU32 (%RU32 parent)\n", + AUDMIXBUF_LOG(("uThisHz=%RU32, uParentHz=%RU32, iFreqRatio=0x%RX64 (%RI64), uRateInc=0x%RX64 (%RU64), cFrames=%RU32 (%RU32 parent)\n", AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->AudioFmt), AUDMIXBUF_FMT_SAMPLE_FREQ(pParent->AudioFmt), pMixBuf->iFreqRatio, pMixBuf->iFreqRatio, pMixBuf->pRate->dstInc, pMixBuf->pRate->dstInc, - pMixBuf->cSamples, - pParent->cSamples)); + pMixBuf->cFrames, + pParent->cFrames)); AUDMIXBUF_LOG(("%s (%RU32Hz) -> %s (%RU32Hz)\n", pMixBuf->pszName, AUDMIXBUF_FMT_SAMPLE_FREQ(pMixBuf->AudioFmt), @@ -1010,14 +1009,14 @@ /** - * Returns number of available live samples, that is, samples that + * Returns number of available live frames, that is, frames that * have been written into the mixing buffer but not have been processed yet. * - * For a parent buffer, this simply returns the currently used number of samples + * For a parent buffer, this simply returns the currently used number of frames * in the buffer. * - * For a child buffer, this returns the number of samples which have been mixed + * For a child buffer, this returns the number of frames which have been mixed * to the parent and were not processed by the parent yet. * - * @return uint32_t Number of live samples available. + * @return uint32_t Number of live frames available. * @param pMixBuf Mixing buffer to return value for. */ @@ -1027,5 +1026,5 @@ #ifdef RT_STRICT - uint32_t cSamples; + uint32_t cFrames; #endif uint32_t cAvail; @@ -1033,8 +1032,8 @@ { #ifdef RT_STRICT - /* Use the sample count from the parent, as - * pMixBuf->cMixed specifies the sample count - * in parent samples. */ - cSamples = pMixBuf->pParent->cSamples; + /* Use the frame count from the parent, as + * pMixBuf->cMixed specifies the frame count + * in parent frames. */ + cFrames = pMixBuf->pParent->cFrames; #endif cAvail = pMixBuf->cMixed; @@ -1043,15 +1042,15 @@ { #ifdef RT_STRICT - cSamples = pMixBuf->cSamples; + cFrames = pMixBuf->cFrames; #endif cAvail = pMixBuf->cUsed; } - Assert(cAvail <= cSamples); + Assert(cAvail <= cFrames); return cAvail; } /** - * Mixes audio samples from a source mixing buffer to a destination mixing buffer. + * Mixes audio frames from a source mixing buffer to a destination mixing buffer. * * @return IPRT status code. @@ -1061,9 +1060,9 @@ * @param pDst Destination mixing buffer. * @param pSrc Source mixing buffer. - * @param cSrcOff Offset of source audio samples to mix. - * @param cSrcSamples Number of source audio samples to mix. - * @param pcSrcMixed Number of source audio samples successfully mixed. Optional. - */ -static int audioMixBufMixTo(PPDMAUDIOMIXBUF pDst, PPDMAUDIOMIXBUF pSrc, uint32_t cSrcOff, uint32_t cSrcSamples, + * @param cSrcOff Offset of source audio frames to mix. + * @param cSrcFrames Number of source audio frames to mix. + * @param pcSrcMixed Number of source audio frames successfully mixed. Optional. + */ +static int audioMixBufMixTo(PPDMAUDIOMIXBUF pDst, PPDMAUDIOMIXBUF pSrc, uint32_t cSrcOff, uint32_t cSrcFrames, uint32_t *pcSrcMixed) { @@ -1077,8 +1076,8 @@ uint32_t cWrittenTotal = 0; - Assert(pSrc->cMixed <= pDst->cSamples); + Assert(pSrc->cMixed <= pDst->cFrames); Assert(pSrc->cUsed >= pDst->cMixed); - Assert(pDst->cUsed <= pDst->cSamples); + Assert(pDst->cUsed <= pDst->cFrames); uint32_t offSrcRead = cSrcOff; @@ -1087,6 +1086,6 @@ uint32_t cDstMixed = pSrc->cMixed; - uint32_t cSrcAvail = RT_MIN(cSrcSamples, pSrc->cUsed); - uint32_t cDstAvail = pDst->cSamples - pDst->cUsed; /** @todo Use pDst->cMixed later? */ + uint32_t cSrcAvail = RT_MIN(cSrcFrames, pSrc->cUsed); + uint32_t cDstAvail = pDst->cFrames - pDst->cUsed; /** @todo Use pDst->cMixed later? */ AUDMIXBUF_LOG(("%s (%RU32 available) -> %s (%RU32 available)\n", @@ -1112,6 +1111,6 @@ while (cSrcAvail && cDstAvail) { - cSrcToRead = RT_MIN(cSrcAvail, pSrc->cSamples - offSrcRead); - cDstToWrite = RT_MIN(cDstAvail, pDst->cSamples - offDstWrite); + cSrcToRead = RT_MIN(cSrcAvail, pSrc->cFrames - offSrcRead); + cDstToWrite = RT_MIN(cDstAvail, pDst->cFrames - offDstWrite); AUDMIXBUF_LOG(("\tSource: %RU32 @ %RU32 -> reading %RU32\n", cSrcAvail, offSrcRead, cSrcToRead)); @@ -1126,12 +1125,12 @@ cDstWritten = cSrcRead = 0; - Assert(offSrcRead < pSrc->cSamples); - Assert(offSrcRead + cSrcToRead <= pSrc->cSamples); - - Assert(offDstWrite < pDst->cSamples); - Assert(offDstWrite + cDstToWrite <= pDst->cSamples); - - audioMixBufOpAssign(pDst->pSamples + offDstWrite, cDstToWrite, - pSrc->pSamples + offSrcRead, cSrcToRead, + Assert(offSrcRead < pSrc->cFrames); + Assert(offSrcRead + cSrcToRead <= pSrc->cFrames); + + Assert(offDstWrite < pDst->cFrames); + Assert(offDstWrite + cDstToWrite <= pDst->cFrames); + + audioMixBufOpAssign(pDst->pFrames + offDstWrite, cDstToWrite, + pSrc->pFrames + offSrcRead, cSrcToRead, pSrc->pRate, &cDstWritten, &cSrcRead); @@ -1139,6 +1138,6 @@ cWrittenTotal += cDstWritten; - offSrcRead = (offSrcRead + cSrcRead) % pSrc->cSamples; - offDstWrite = (offDstWrite + cDstWritten) % pDst->cSamples; + offSrcRead = (offSrcRead + cSrcRead) % pSrc->cFrames; + offDstWrite = (offDstWrite + cDstWritten) % pDst->cFrames; cDstMixed += cDstWritten; @@ -1159,22 +1158,22 @@ pSrc->cUsed -= RT_MIN(pSrc->cUsed, cReadTotal); - /* Note: Always count in parent samples, as the rate can differ! */ - pSrc->cMixed = RT_MIN(cDstMixed, pDst->cSamples); + /* Note: Always count in parent frames, as the rate can differ! */ + pSrc->cMixed = RT_MIN(cDstMixed, pDst->cFrames); pDst->offWrite = offDstWrite; - Assert(pDst->offWrite <= pDst->cSamples); - Assert((pDst->cUsed + cWrittenTotal) <= pDst->cSamples); + Assert(pDst->offWrite <= pDst->cFrames); + Assert((pDst->cUsed + cWrittenTotal) <= pDst->cFrames); pDst->cUsed += cWrittenTotal; - /* If there are more used samples than fitting in the destination buffer, + /* If there are more used frames than fitting in the destination buffer, * adjust the values accordingly. * * This can happen if this routine has been called too often without * actually processing the destination buffer in between. */ - if (pDst->cUsed > pDst->cSamples) - { - LogFunc(("%s: Warning: Destination buffer used %RU32 / %RU32 samples\n", pDst->pszName, pDst->cUsed, pDst->cSamples)); + if (pDst->cUsed > pDst->cFrames) + { + LogFunc(("%s: Warning: Destination buffer used %RU32 / %RU32 frames\n", pDst->pszName, pDst->cUsed, pDst->cFrames)); pDst->offWrite = 0; - pDst->cUsed = pDst->cSamples; + pDst->cUsed = pDst->cFrames; rc = VERR_BUFFER_OVERFLOW; @@ -1185,5 +1184,5 @@ audioMixBufDbgValidate(pDst); - Assert(pSrc->cMixed <= pDst->cSamples); + Assert(pSrc->cMixed <= pDst->cFrames); #endif @@ -1200,12 +1199,12 @@ Assert(sizeof(auBuf) % 4 == 0); - uint32_t cToRead = RT_MIN(AUDIOMIXBUF_B2S(pDst, sizeof(auBuf)), RT_MIN(cLeft, pDst->cSamples - offRead)); + uint32_t cToRead = RT_MIN(AUDIOMIXBUF_B2F(pDst, sizeof(auBuf)), RT_MIN(cLeft, pDst->cFrames - offRead)); Assert(cToRead <= pDst->cUsed); PDMAUDMIXBUFCONVOPTS convOpts; RT_ZERO(convOpts); - convOpts.cSamples = cToRead; - - pDst->pfnConvTo(auBuf, pDst->pSamples + offRead, &convOpts); + convOpts.cFrames = cToRead; + + pDst->pfnConvTo(auBuf, pDst->pFrames + offRead, &convOpts); RTFILE fh; @@ -1214,9 +1213,9 @@ if (RT_SUCCESS(rc2)) { - RTFileWrite(fh, auBuf, AUDIOMIXBUF_S2B(pDst, cToRead), NULL); + RTFileWrite(fh, auBuf, AUDIOMIXBUF_F2B(pDst, cToRead), NULL); RTFileClose(fh); } - offRead = (offRead + cToRead) % pDst->cSamples; + offRead = (offRead + cToRead) % pDst->cFrames; cLeft -= cToRead; } @@ -1236,13 +1235,13 @@ /** - * Mixes audio samples down to the parent mixing buffer, extended version. + * Mixes audio frames down to the parent mixing buffer, extended version. * * @return IPRT status code. See audioMixBufMixTo() for a more detailed explanation. * @param pMixBuf Source mixing buffer to mix to its parent. - * @param cSrcOffset Offset (in samples) of source mixing buffer. - * @param cSrcSamples Number of source audio samples to mix to its parent. - * @param pcSrcMixed Number of source audio samples successfully mixed. Optional. - */ -int AudioMixBufMixToParentEx(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcOffset, uint32_t cSrcSamples, uint32_t *pcSrcMixed) + * @param cSrcOffset Offset (in frames) of source mixing buffer. + * @param cSrcFrames Number of source audio frames to mix to its parent. + * @param pcSrcMixed Number of source audio frames successfully mixed. Optional. + */ +int AudioMixBufMixToParentEx(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcOffset, uint32_t cSrcFrames, uint32_t *pcSrcMixed) { AssertMsgReturn(VALID_PTR(pMixBuf->pParent), @@ -1250,18 +1249,18 @@ VERR_INVALID_PARAMETER); - return audioMixBufMixTo(pMixBuf->pParent, pMixBuf, cSrcOffset, cSrcSamples, pcSrcMixed); -} - -/** - * Mixes audio samples down to the parent mixing buffer. + return audioMixBufMixTo(pMixBuf->pParent, pMixBuf, cSrcOffset, cSrcFrames, pcSrcMixed); +} + +/** + * Mixes audio frames down to the parent mixing buffer. * * @return IPRT status code. See audioMixBufMixTo() for a more detailed explanation. * @param pMixBuf Source mixing buffer to mix to its parent. - * @param cSrcSamples Number of source audio samples to mix to its parent. - * @param pcSrcMixed Number of source audio samples successfully mixed. Optional. - */ -int AudioMixBufMixToParent(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcSamples, uint32_t *pcSrcMixed) -{ - return audioMixBufMixTo(pMixBuf->pParent, pMixBuf, pMixBuf->offRead, cSrcSamples, pcSrcMixed); + * @param cSrcFrames Number of source audio frames to mix to its parent. + * @param pcSrcMixed Number of source audio frames successfully mixed. Optional. + */ +int AudioMixBufMixToParent(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcFrames, uint32_t *pcSrcMixed) +{ + return audioMixBufMixTo(pMixBuf->pParent, pMixBuf, pMixBuf->offRead, cSrcFrames, pcSrcMixed); } @@ -1281,5 +1280,5 @@ Log(("%s: %*s[%s] %s: offRead=%RU32, offWrite=%RU32, cMixed=%RU32 -> %RU32/%RU32\n", pszFunc, uIdtLvl * 4, "", fIsParent ? "PARENT" : "CHILD", - pMixBuf->pszName, pMixBuf->offRead, pMixBuf->offWrite, pMixBuf->cMixed, pMixBuf->cUsed, pMixBuf->cSamples)); + pMixBuf->pszName, pMixBuf->offRead, pMixBuf->offWrite, pMixBuf->cMixed, pMixBuf->cUsed, pMixBuf->cFrames)); } @@ -1292,12 +1291,12 @@ DECL_FORCE_INLINE(bool) audioMixBufDbgValidate(PPDMAUDIOMIXBUF pMixBuf) { - //const uint32_t offReadEnd = (pMixBuf->offRead + pMixBuf->cUsed) % pMixBuf->cSamples; - //const uint32_t offWriteEnd = (pMixBuf->offWrite + (pMixBuf->cSamples - pMixBuf->cUsed)) % pMixBuf->cSamples; + //const uint32_t offReadEnd = (pMixBuf->offRead + pMixBuf->cUsed) % pMixBuf->cFrames; + //const uint32_t offWriteEnd = (pMixBuf->offWrite + (pMixBuf->cFrames - pMixBuf->cUsed)) % pMixBuf->cFrames; bool fValid = true; - AssertStmt(pMixBuf->offRead <= pMixBuf->cSamples, fValid = false); - AssertStmt(pMixBuf->offWrite <= pMixBuf->cSamples, fValid = false); - AssertStmt(pMixBuf->cUsed <= pMixBuf->cSamples, fValid = false); + AssertStmt(pMixBuf->offRead <= pMixBuf->cFrames, fValid = false); + AssertStmt(pMixBuf->offWrite <= pMixBuf->cFrames, fValid = false); + AssertStmt(pMixBuf->cUsed <= pMixBuf->cFrames, fValid = false); if (pMixBuf->offWrite > pMixBuf->offRead) @@ -1308,5 +1307,5 @@ else if (pMixBuf->offWrite < pMixBuf->offRead) { - if (pMixBuf->offWrite + pMixBuf->cSamples - pMixBuf->offRead != pMixBuf->cUsed) + if (pMixBuf->offWrite + pMixBuf->cFrames - pMixBuf->offRead != pMixBuf->cUsed) fValid = false; } @@ -1409,5 +1408,5 @@ /** - * Returns the total number of samples used. + * Returns the total number of frames used. * * @return uint32_t @@ -1421,9 +1420,9 @@ /** - * Reads audio samples at a specific offset. + * Reads audio frames at a specific offset. * * @return IPRT status code. - * @param pMixBuf Mixing buffer to read audio samples from. - * @param offSamples Offset (in audio samples) to start reading from. + * @param pMixBuf Mixing buffer to read audio frames from. + * @param offFrames Offset (in audio frames) to start reading from. * @param pvBuf Pointer to buffer to write output to. * @param cbBuf Size (in bytes) of buffer to write to. @@ -1431,21 +1430,21 @@ */ int AudioMixBufReadAt(PPDMAUDIOMIXBUF pMixBuf, - uint32_t offSamples, + uint32_t offFrames, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead) { return AudioMixBufReadAtEx(pMixBuf, pMixBuf->AudioFmt, - offSamples, pvBuf, cbBuf, pcbRead); -} - -/** - * Reads audio samples at a specific offset. + offFrames, pvBuf, cbBuf, pcbRead); +} + +/** + * Reads audio frames at a specific offset. * If the audio format of the mixing buffer and the requested audio format do * not match the output will be converted accordingly. * * @return IPRT status code. - * @param pMixBuf Mixing buffer to read audio samples from. + * @param pMixBuf Mixing buffer to read audio frames from. * @param enmFmt Audio format to use for output. - * @param offSamples Offset (in audio samples) to start reading from. + * @param offFrames Offset (in audio frames) to start reading from. * @param pvBuf Pointer to buffer to write output to. * @param cbBuf Size (in bytes) of buffer to write to. @@ -1453,5 +1452,5 @@ */ int AudioMixBufReadAtEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt, - uint32_t offSamples, + uint32_t offFrames, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead) @@ -1461,13 +1460,13 @@ /* pcbRead is optional. */ - uint32_t cDstSamples = pMixBuf->cSamples; + uint32_t cDstFrames = pMixBuf->cFrames; uint32_t cLive = pMixBuf->cUsed; - uint32_t cDead = cDstSamples - cLive; - uint32_t cToProcess = (uint32_t)AUDIOMIXBUF_S2S_RATIO(pMixBuf, cDead); - cToProcess = RT_MIN(cToProcess, AUDIOMIXBUF_B2S(pMixBuf, cbBuf)); - - AUDMIXBUF_LOG(("%s: offSamples=%RU32, cLive=%RU32, cDead=%RU32, cToProcess=%RU32\n", - pMixBuf->pszName, offSamples, cLive, cDead, cToProcess)); + uint32_t cDead = cDstFrames - cLive; + uint32_t cToProcess = (uint32_t)AUDIOMIXBUF_F2F_RATIO(pMixBuf, cDead); + cToProcess = RT_MIN(cToProcess, AUDIOMIXBUF_B2F(pMixBuf, cbBuf)); + + AUDMIXBUF_LOG(("%s: offFrames=%RU32, cLive=%RU32, cDead=%RU32, cToProcess=%RU32\n", + pMixBuf->pszName, offFrames, cLive, cDead, cToProcess)); int rc; @@ -1486,7 +1485,7 @@ /* Note: No volume handling/conversion done in the conversion-to macros (yet). */ - convOpts.cSamples = cToProcess; - - pfnConvTo(pvBuf, pMixBuf->pSamples + offSamples, &convOpts); + convOpts.cFrames = cToProcess; + + pfnConvTo(pvBuf, pMixBuf->pFrames + offFrames, &convOpts); #ifdef DEBUG @@ -1507,19 +1506,19 @@ { if (pcbRead) - *pcbRead = AUDIOMIXBUF_S2B(pMixBuf, cToProcess); - } - - AUDMIXBUF_LOG(("cbRead=%RU32, rc=%Rrc\n", AUDIOMIXBUF_S2B(pMixBuf, cToProcess), rc)); + *pcbRead = AUDIOMIXBUF_F2B(pMixBuf, cToProcess); + } + + AUDMIXBUF_LOG(("cbRead=%RU32, rc=%Rrc\n", AUDIOMIXBUF_F2B(pMixBuf, cToProcess), rc)); return rc; } /** - * Reads audio samples. The audio format of the mixing buffer will be used. + * Reads audio frames. The audio format of the mixing buffer will be used. * * @return IPRT status code. - * @param pMixBuf Mixing buffer to read audio samples from. + * @param pMixBuf Mixing buffer to read audio frames from. * @param pvBuf Pointer to buffer to write output to. * @param cbBuf Size (in bytes) of buffer to write to. - * @param pcRead Number of audio samples read. Optional. + * @param pcRead Number of audio frames read. Optional. */ int AudioMixBufReadCirc(PPDMAUDIOMIXBUF pMixBuf, void *pvBuf, uint32_t cbBuf, uint32_t *pcRead) @@ -1529,14 +1528,14 @@ /** - * Reads audio samples in a specific audio format. + * Reads audio frames in a specific audio format. * If the audio format of the mixing buffer and the requested audio format do * not match the output will be converted accordingly. * * @return IPRT status code. - * @param pMixBuf Mixing buffer to read audio samples from. + * @param pMixBuf Mixing buffer to read audio frames from. * @param enmFmt Audio format to use for output. * @param pvBuf Pointer to buffer to write output to. * @param cbBuf Size (in bytes) of buffer to write to. - * @param pcRead Number of audio samples read. Optional. + * @param pcRead Number of audio frames read. Optional. */ int AudioMixBufReadCircEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt, void *pvBuf, uint32_t cbBuf, uint32_t *pcRead) @@ -1547,11 +1546,11 @@ /* pcRead is optional. */ - /* Make sure that we at least have space for a full audio sample. */ - AssertReturn(AUDIOMIXBUF_B2S(pMixBuf, cbBuf), VERR_INVALID_PARAMETER); - - uint32_t cToRead = RT_MIN(pMixBuf->cUsed, AUDIOMIXBUF_B2S(pMixBuf, cbBuf)); - - AUDMIXBUF_LOG(("%s: cbBuf=%RU32 (%RU32 samples), cToRead=%RU32, fmtSrc=0x%x, fmtDst=0x%x\n", - pMixBuf->pszName, cbBuf, AUDIOMIXBUF_B2S(pMixBuf, cbBuf), cToRead, pMixBuf->AudioFmt, enmFmt)); + /* Make sure that we at least have space for a full audio frame. */ + AssertReturn(AUDIOMIXBUF_B2F(pMixBuf, cbBuf), VERR_INVALID_PARAMETER); + + uint32_t cToRead = RT_MIN(pMixBuf->cUsed, AUDIOMIXBUF_B2F(pMixBuf, cbBuf)); + + AUDMIXBUF_LOG(("%s: cbBuf=%RU32 (%RU32 frames), cToRead=%RU32, fmtSrc=0x%x, fmtDst=0x%x\n", + pMixBuf->pszName, cbBuf, AUDIOMIXBUF_B2F(pMixBuf, cbBuf), cToRead, pMixBuf->AudioFmt, enmFmt)); if (!cToRead) @@ -1577,14 +1576,14 @@ } - cToRead = RT_MIN(cToRead, pMixBuf->cSamples - pMixBuf->offRead); + cToRead = RT_MIN(cToRead, pMixBuf->cFrames - pMixBuf->offRead); if (cToRead) { PDMAUDMIXBUFCONVOPTS convOpts; RT_ZERO(convOpts); - convOpts.cSamples = cToRead; + convOpts.cFrames = cToRead; AUDMIXBUF_LOG(("cToRead=%RU32\n", cToRead)); - pfnConvTo(pvBuf, pMixBuf->pSamples + pMixBuf->offRead, &convOpts); + pfnConvTo(pvBuf, pMixBuf->pFrames + pMixBuf->offRead, &convOpts); #ifdef AUDIOMIXBUF_DEBUG_DUMP_PCM_DATA @@ -1594,9 +1593,9 @@ if (RT_SUCCESS(rc2)) { - RTFileWrite(fh, pvBuf, AUDIOMIXBUF_S2B(pMixBuf, cToRead), NULL); + RTFileWrite(fh, pvBuf, AUDIOMIXBUF_F2B(pMixBuf, cToRead), NULL); RTFileClose(fh); } #endif - pMixBuf->offRead = (pMixBuf->offRead + cToRead) % pMixBuf->cSamples; + pMixBuf->offRead = (pMixBuf->offRead + cToRead) % pMixBuf->cFrames; Assert(pMixBuf->cUsed >= cToRead); pMixBuf->cUsed -= cToRead; @@ -1610,5 +1609,5 @@ #endif - AUDMIXBUF_LOG(("cRead=%RU32 (%RU32 bytes)\n", cToRead, AUDIOMIXBUF_S2B(pMixBuf, cToRead))); + AUDMIXBUF_LOG(("cRead=%RU32 (%RU32 bytes)\n", cToRead, AUDIOMIXBUF_F2B(pMixBuf, cToRead))); return VINF_SUCCESS; } @@ -1664,7 +1663,7 @@ /** - * Returns the maximum amount of audio samples this buffer can hold. - * - * @return uint32_t Size (in audio samples) the mixing buffer can hold. + * Returns the maximum amount of audio frames this buffer can hold. + * + * @return uint32_t Size (in audio frames) the mixing buffer can hold. * @param pMixBuf Mixing buffer to retrieve maximum for. */ @@ -1672,5 +1671,5 @@ { AssertPtrReturn(pMixBuf, 0); - return pMixBuf->cSamples; + return pMixBuf->cFrames; } @@ -1684,5 +1683,5 @@ { AssertPtrReturn(pMixBuf, 0); - return AUDIOMIXBUF_S2B(pMixBuf, pMixBuf->cSamples); + return AUDIOMIXBUF_F2B(pMixBuf, pMixBuf->cFrames); } @@ -1749,24 +1748,24 @@ /** - * Writes audio samples at a specific offset. + * Writes audio frames at a specific offset. * The sample format being written must match the format of the mixing buffer. * * @return IPRT status code. * @param pMixBuf Pointer to mixing buffer to write to. - * @param offSamples Offset (in samples) starting to write at. + * @param offFrames Offset (in frames) starting to write at. * @param pvBuf Pointer to audio buffer to be written. * @param cbBuf Size (in bytes) of audio buffer. - * @param pcWritten Returns number of audio samples written. Optional. - */ -int AudioMixBufWriteAt(PPDMAUDIOMIXBUF pMixBuf, uint32_t offSamples, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten) -{ - return AudioMixBufWriteAtEx(pMixBuf, pMixBuf->AudioFmt, offSamples, pvBuf, cbBuf, pcWritten); -} - -/** - * Writes audio samples at a specific offset. + * @param pcWritten Returns number of audio frames written. Optional. + */ +int AudioMixBufWriteAt(PPDMAUDIOMIXBUF pMixBuf, uint32_t offFrames, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten) +{ + return AudioMixBufWriteAtEx(pMixBuf, pMixBuf->AudioFmt, offFrames, pvBuf, cbBuf, pcWritten); +} + +/** + * Writes audio frames at a specific offset. * * Note that this operation also modifies the current read and write position - * to \a offSamples + written samples on success. + * to \a offFrames + written frames on success. * * The audio sample format to be written can be different from the audio format @@ -1776,11 +1775,11 @@ * @param pMixBuf Pointer to mixing buffer to write to. * @param enmFmt Audio format supplied in the buffer. - * @param offSamples Offset (in samples) starting to write at. + * @param offFrames Offset (in frames) starting to write at. * @param pvBuf Pointer to audio buffer to be written. * @param cbBuf Size (in bytes) of audio buffer. - * @param pcWritten Returns number of audio samples written. Optional. + * @param pcWritten Returns number of audio frames written. Optional. */ int AudioMixBufWriteAtEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt, - uint32_t offSamples, const void *pvBuf, uint32_t cbBuf, + uint32_t offFrames, const void *pvBuf, uint32_t cbBuf, uint32_t *pcWritten) { @@ -1790,5 +1789,5 @@ /* pcbWritten is optional. */ - if (offSamples >= pMixBuf->cSamples) + if (offFrames >= pMixBuf->cFrames) { if (pcWritten) @@ -1800,5 +1799,5 @@ * Adjust cToWrite so we don't overflow our buffers. */ - uint32_t cToWrite = RT_MIN(AUDIOMIXBUF_B2S(pMixBuf, cbBuf), pMixBuf->cSamples - offSamples); + uint32_t cToWrite = RT_MIN(AUDIOMIXBUF_B2F(pMixBuf, cbBuf), pMixBuf->cFrames - offFrames); #ifdef AUDIOMIXBUF_DEBUG_DUMP_PCM_DATA @@ -1811,5 +1810,5 @@ if (RT_SUCCESS(rc2)) { - RTFileWrite(hFile, pvBuf, AUDIOMIXBUF_S2B(pMixBuf, cToWrite), NULL); + RTFileWrite(hFile, pvBuf, AUDIOMIXBUF_F2B(pMixBuf, cToWrite), NULL); RTFileClose(hFile); } @@ -1838,10 +1837,10 @@ PDMAUDMIXBUFCONVOPTS convOpts; - convOpts.cSamples = cToWrite; + convOpts.cFrames = cToWrite; convOpts.From.Volume.fMuted = pMixBuf->Volume.fMuted; convOpts.From.Volume.uLeft = pMixBuf->Volume.uLeft; convOpts.From.Volume.uRight = pMixBuf->Volume.uRight; - cWritten = pfnConvFrom(pMixBuf->pSamples + offSamples, pvBuf, AUDIOMIXBUF_S2B(pMixBuf, cToWrite), &convOpts); + cWritten = pfnConvFrom(pMixBuf->pFrames + offFrames, pvBuf, AUDIOMIXBUF_F2B(pMixBuf, cToWrite), &convOpts); } else @@ -1855,13 +1854,13 @@ } - AUDMIXBUF_LOG(("%s: offSamples=%RU32, cbBuf=%RU32, cToWrite=%RU32 (%zu bytes), cWritten=%RU32 (%zu bytes), rc=%Rrc\n", - pMixBuf->pszName, offSamples, cbBuf, - cToWrite, AUDIOMIXBUF_S2B(pMixBuf, cToWrite), - cWritten, AUDIOMIXBUF_S2B(pMixBuf, cWritten), rc)); + AUDMIXBUF_LOG(("%s: offFrames=%RU32, cbBuf=%RU32, cToWrite=%RU32 (%zu bytes), cWritten=%RU32 (%zu bytes), rc=%Rrc\n", + pMixBuf->pszName, offFrames, cbBuf, + cToWrite, AUDIOMIXBUF_F2B(pMixBuf, cToWrite), + cWritten, AUDIOMIXBUF_F2B(pMixBuf, cWritten), rc)); if (RT_SUCCESS(rc)) { - pMixBuf->offRead = offSamples % pMixBuf->cSamples; - pMixBuf->offWrite = (offSamples + cWritten) % pMixBuf->cSamples; + pMixBuf->offRead = offFrames % pMixBuf->cFrames; + pMixBuf->offWrite = (offFrames + cWritten) % pMixBuf->cFrames; pMixBuf->cUsed = cWritten; pMixBuf->cMixed = 0; @@ -1880,14 +1879,14 @@ /** - * Writes audio samples. + * Writes audio frames. * * The sample format being written must match the format of the mixing buffer. * - * @return IPRT status code, or VERR_BUFFER_OVERFLOW if samples which not have + * @return IPRT status code, or VERR_BUFFER_OVERFLOW if frames which not have * been processed yet have been overwritten (due to cyclic buffer). * @param pMixBuf Pointer to mixing buffer to write to. * @param pvBuf Pointer to audio buffer to be written. * @param cbBuf Size (in bytes) of audio buffer. - * @param pcWritten Returns number of audio samples written. Optional. + * @param pcWritten Returns number of audio frames written. Optional. */ int AudioMixBufWriteCirc(PPDMAUDIOMIXBUF pMixBuf, @@ -1899,5 +1898,5 @@ /** - * Writes audio samples of a specific format. + * Writes audio frames of a specific format. * This function might write less data at once than requested. * @@ -1907,5 +1906,5 @@ * @param pvBuf Pointer to audio buffer to be written. * @param cbBuf Size (in bytes) of audio buffer. - * @param pcWritten Returns number of audio samples written. Optional. + * @param pcWritten Returns number of audio frames written. Optional. */ int AudioMixBufWriteCircEx(PPDMAUDIOMIXBUF pMixBuf, PDMAUDIOMIXBUFFMT enmFmt, @@ -1923,9 +1922,9 @@ } - /* Make sure that we at least write a full audio sample. */ - AssertReturn(AUDIOMIXBUF_B2S(pMixBuf, cbBuf), VERR_INVALID_PARAMETER); - - Assert(pMixBuf->cSamples); - AssertPtr(pMixBuf->pSamples); + /* Make sure that we at least write a full audio frame. */ + AssertReturn(AUDIOMIXBUF_B2F(pMixBuf, cbBuf), VERR_INVALID_PARAMETER); + + Assert(pMixBuf->cFrames); + AssertPtr(pMixBuf->pFrames); PFNPDMAUDIOMIXBUFCONVFROM pfnConvFrom = NULL; @@ -1950,11 +1949,11 @@ uint32_t cWritten = 0; - uint32_t cFree = pMixBuf->cSamples - pMixBuf->cUsed; + uint32_t cFree = pMixBuf->cFrames - pMixBuf->cUsed; if (cFree) { - if ((pMixBuf->cSamples - pMixBuf->offWrite) == 0) + if ((pMixBuf->cFrames - pMixBuf->offWrite) == 0) pMixBuf->offWrite = 0; - uint32_t cToWrite = RT_MIN(AUDIOMIXBUF_B2S(pMixBuf, cbBuf), RT_MIN(pMixBuf->cSamples - pMixBuf->offWrite, cFree)); + uint32_t cToWrite = RT_MIN(AUDIOMIXBUF_B2F(pMixBuf, cbBuf), RT_MIN(pMixBuf->cFrames - pMixBuf->offWrite, cFree)); Assert(cToWrite); @@ -1966,8 +1965,8 @@ convOpts.From.Volume.uRight = pMixBuf->Volume.uRight; - convOpts.cSamples = cToWrite; - - cWritten = pfnConvFrom(pMixBuf->pSamples + pMixBuf->offWrite, - pvBuf, AUDIOMIXBUF_S2B(pMixBuf, cToWrite), &convOpts); + convOpts.cFrames = cToWrite; + + cWritten = pfnConvFrom(pMixBuf->pFrames + pMixBuf->offWrite, + pvBuf, AUDIOMIXBUF_F2B(pMixBuf, cToWrite), &convOpts); Assert(cWritten == cToWrite); @@ -1976,12 +1975,12 @@ RTFileOpen(&fh, AUDIOMIXBUF_DEBUG_DUMP_PCM_DATA_PATH "mixbuf_writecirc_ex.pcm", RTFILE_O_OPEN_CREATE | RTFILE_O_APPEND | RTFILE_O_WRITE | RTFILE_O_DENY_NONE); - RTFileWrite(fh, pvBuf, AUDIOMIXBUF_S2B(pMixBuf, cToWrite), NULL); + RTFileWrite(fh, pvBuf, AUDIOMIXBUF_F2B(pMixBuf, cToWrite), NULL); RTFileClose(fh); #endif pMixBuf->cUsed += cWritten; - Assert(pMixBuf->cUsed <= pMixBuf->cSamples); - - pMixBuf->offWrite = (pMixBuf->offWrite + cWritten) % pMixBuf->cSamples; - Assert(pMixBuf->offWrite <= pMixBuf->cSamples); + Assert(pMixBuf->cUsed <= pMixBuf->cFrames); + + pMixBuf->offWrite = (pMixBuf->offWrite + cWritten) % pMixBuf->cFrames; + Assert(pMixBuf->offWrite <= pMixBuf->cFrames); } else @@ -1996,6 +1995,6 @@ *pcWritten = cWritten; - AUDMIXBUF_LOG(("%s: enmFmt=0x%x, cbBuf=%RU32 (%RU32 samples), cWritten=%RU32, rc=%Rrc\n", - pMixBuf->pszName, enmFmt, cbBuf, AUDIOMIXBUF_B2S(pMixBuf, cbBuf), cWritten, rc)); + AUDMIXBUF_LOG(("%s: enmFmt=0x%x, cbBuf=%RU32 (%RU32 frames), cWritten=%RU32, rc=%Rrc\n", + pMixBuf->pszName, enmFmt, cbBuf, AUDIOMIXBUF_B2F(pMixBuf, cbBuf), cWritten, rc)); return rc; } Index: /trunk/src/VBox/Devices/Audio/AudioMixBuffer.h =================================================================== --- /trunk/src/VBox/Devices/Audio/AudioMixBuffer.h (revision 68131) +++ /trunk/src/VBox/Devices/Audio/AudioMixBuffer.h (revision 68132) @@ -37,14 +37,14 @@ #define AUDMIXBUF_FMT_BYTES_PER_SAMPLE(a) ((AUDMIXBUF_AUDIO_FMT_BITS_PER_SAMPLE(a) + 7) / 8) -/** Converts samples to bytes. */ -#define AUDIOMIXBUF_S2B(pBuf, samples) ((samples) << (pBuf)->cShift) -/** Converts samples to bytes, respecting the conversion ratio to +/** Converts frames to bytes. */ +#define AUDIOMIXBUF_F2B(pBuf, frames) ((frames) << (pBuf)->cShift) +/** Converts frames to bytes, respecting the conversion ratio to * a linked buffer. */ -#define AUDIOMIXBUF_S2B_RATIO(pBuf, samples) ((((int64_t) samples << 32) / (pBuf)->iFreqRatio) << (pBuf)->cShift) -/** Converts bytes to samples, *not* taking the conversion ratio +#define AUDIOMIXBUF_F2B_RATIO(pBuf, frames) ((((int64_t) frames << 32) / (pBuf)->iFreqRatio) << (pBuf)->cShift) +/** Converts bytes to frames, *not* taking the conversion ratio * into account. */ -#define AUDIOMIXBUF_B2S(pBuf, cb) (cb >> (pBuf)->cShift) -/** Converts number of samples according to the buffer's ratio. */ -#define AUDIOMIXBUF_S2S_RATIO(pBuf, samples) (((int64_t) samples << 32) / (pBuf)->iFreqRatio) +#define AUDIOMIXBUF_B2F(pBuf, cb) (cb >> (pBuf)->cShift) +/** Converts number of frames according to the buffer's ratio. */ +#define AUDIOMIXBUF_F2F_RATIO(pBuf, frames) (((int64_t) frames << 32) / (pBuf)->iFreqRatio) @@ -52,15 +52,15 @@ void AudioMixBufClear(PPDMAUDIOMIXBUF pMixBuf); void AudioMixBufDestroy(PPDMAUDIOMIXBUF pMixBuf); -void AudioMixBufFinish(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamplesToClear); +void AudioMixBufFinish(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFramesToClear); uint32_t AudioMixBufFree(PPDMAUDIOMIXBUF pMixBuf); uint32_t AudioMixBufFreeBytes(PPDMAUDIOMIXBUF pMixBuf); -int AudioMixBufInit(PPDMAUDIOMIXBUF pMixBuf, const char *pszName, PPDMAUDIOPCMPROPS pProps, uint32_t cSamples); +int AudioMixBufInit(PPDMAUDIOMIXBUF pMixBuf, const char *pszName, PPDMAUDIOPCMPROPS pProps, uint32_t cFrames); bool AudioMixBufIsEmpty(PPDMAUDIOMIXBUF pMixBuf); int AudioMixBufLinkTo(PPDMAUDIOMIXBUF pMixBuf, PPDMAUDIOMIXBUF pParent); uint32_t AudioMixBufLive(PPDMAUDIOMIXBUF pMixBuf); -int AudioMixBufMixToParent(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcSamples, uint32_t *pcSrcMixed); -int AudioMixBufMixToParentEx(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcOffset, uint32_t cSrcSamples, uint32_t *pcSrcMixed); -int AudioMixBufPeek(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamplesToRead, PPDMAUDIOSAMPLE paSampleBuf, uint32_t cSampleBuf, uint32_t *pcSamplesRead); -int AudioMixBufPeekMutable(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSamplesToRead, PPDMAUDIOSAMPLE *ppvSamples, uint32_t *pcSamplesRead); +int AudioMixBufMixToParent(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcFrames, uint32_t *pcSrcMixed); +int AudioMixBufMixToParentEx(PPDMAUDIOMIXBUF pMixBuf, uint32_t cSrcOffset, uint32_t cSrcFrames, uint32_t *pcSrcMixed); +int AudioMixBufPeek(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFramesToRead, PPDMAUDIOFRAME paSampleBuf, uint32_t cSampleBuf, uint32_t *pcFramesRead); +int AudioMixBufPeekMutable(PPDMAUDIOMIXBUF pMixBuf, uint32_t cFramesToRead, PPDMAUDIOFRAME *ppvSamples, uint32_t *pcFramesRead); uint32_t AudioMixBufUsed(PPDMAUDIOMIXBUF pMixBuf); int AudioMixBufReadAt(PPDMAUDIOMIXBUF pMixBuf, uint32_t offSamples, void *pvBuf, uint32_t cbBuf, uint32_t *pcbRead); Index: /trunk/src/VBox/Devices/Audio/AudioMixer.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/AudioMixer.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/AudioMixer.cpp (revision 68132) @@ -1440,5 +1440,5 @@ AssertPtr(pConn); - uint32_t csProc = 0; + uint32_t cfProc = 0; int rc2 = pConn->pfnStreamIterate(pConn, pStream); @@ -1447,5 +1447,5 @@ if (pSink->enmDir == AUDMIXSINKDIR_INPUT) { - rc = pConn->pfnStreamCapture(pConn, pStream, &csProc); + rc = pConn->pfnStreamCapture(pConn, pStream, &cfProc); if (RT_FAILURE(rc2)) { @@ -1456,10 +1456,10 @@ } - if (csProc) + if (cfProc) pSink->fStatus |= AUDMIXSINK_STS_DIRTY; } else if (pSink->enmDir == AUDMIXSINKDIR_OUTPUT) { - rc2 = pConn->pfnStreamPlay(pConn, pStream, &csProc); + rc2 = pConn->pfnStreamPlay(pConn, pStream, &cfProc); if (RT_FAILURE(rc2)) { @@ -1495,5 +1495,5 @@ } - Log3Func(("\t%s: cPlayed/cCaptured=%RU32, rc2=%Rrc\n", pStream->szName, csProc, rc2)); + Log3Func(("\t%s: cPlayed/cCaptured=%RU32, rc2=%Rrc\n", pStream->szName, cfProc, rc2)); } Index: /trunk/src/VBox/Devices/Audio/DevHDA.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DevHDA.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DevHDA.cpp (revision 68132) @@ -2690,7 +2690,7 @@ RTListForEach(&pThis->lstDrv, pDrv, HDADRIVER, Node) { - uint32_t cSamplesPlayed; - int rc2 = pDrv->pConnector->pfnPlay(pDrv->pConnector, &cSamplesPlayed); - LogFlowFunc(("LUN#%RU8: cSamplesPlayed=%RU32, rc=%Rrc\n", pDrv->uLUN, cSamplesPlayed, rc2)); + uint32_t cFramesPlayed; + int rc2 = pDrv->pConnector->pfnPlay(pDrv->pConnector, &cFramesPlayed); + LogFlowFunc(("LUN#%RU8: cFramesPlayed=%RU32, rc=%Rrc\n", pDrv->uLUN, cFramesPlayed, rc2)); } } Index: /trunk/src/VBox/Devices/Audio/DrvAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvAudio.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvAudio.cpp (revision 68132) @@ -645,8 +645,8 @@ #endif - /* No sample buffer size hint given by the backend? Default to some sane value. */ - if (!CfgHostAcq.cSampleBufferHint) - { - CfgHostAcq.cSampleBufferHint = _1K; /** @todo Make this configurable? */ + /* No frame buffer size hint given by the backend? Default to some sane value. */ + if (!CfgHostAcq.cFrameBufferHint) + { + CfgHostAcq.cFrameBufferHint = _1K; /** @todo Make this configurable? */ } @@ -658,10 +658,10 @@ /* Set set host buffer size multiplicator. */ - const unsigned cSampleBufferHostFactor = 2; /** @todo Make this configurable. */ - - LogFunc(("[%s] cSamples=%RU32 (x %u)\n", pHstStream->szName, CfgHostAcq.cSampleBufferHint, cSampleBufferHostFactor)); + const unsigned cFrameBufferHostFactor = 2; /** @todo Make this configurable. */ + + LogFunc(("[%s] cFrames=%RU32 (x %u)\n", pHstStream->szName, CfgHostAcq.cFrameBufferHint, cFrameBufferHostFactor)); int rc2 = AudioMixBufInit(&pHstStream->MixBuf, pHstStream->szName, &CfgHostAcq.Props, - CfgHostAcq.cSampleBufferHint * cSampleBufferHostFactor); + CfgHostAcq.cFrameBufferHint * cFrameBufferHostFactor); AssertRC(rc2); @@ -684,10 +684,10 @@ /* Set set guest buffer size multiplicator. */ - const unsigned cSampleBufferGuestFactor = 10; /** @todo Make this configurable. */ - - LogFunc(("[%s] cSamples=%RU32 (x %u)\n", pGstStream->szName, CfgHostAcq.cSampleBufferHint, cSampleBufferGuestFactor)); + const unsigned cFrameBufferGuestFactor = 10; /** @todo Make this configurable. */ + + LogFunc(("[%s] cFrames=%RU32 (x %u)\n", pGstStream->szName, CfgHostAcq.cFrameBufferHint, cFrameBufferGuestFactor)); rc2 = AudioMixBufInit(&pGstStream->MixBuf, pGstStream->szName, &pCfgGuest->Props, - CfgHostAcq.cSampleBufferHint * cSampleBufferGuestFactor); + CfgHostAcq.cFrameBufferHint * cFrameBufferGuestFactor); AssertRC(rc2); @@ -834,5 +834,5 @@ STAM_COUNTER_RESET(&pHstStream->In.StatBytesElapsed); STAM_COUNTER_RESET(&pHstStream->In.StatBytesTotalRead); - STAM_COUNTER_RESET(&pHstStream->In.StatSamplesCaptured); + STAM_COUNTER_RESET(&pHstStream->In.StatFramesCaptured); if (pGstStream) @@ -842,5 +842,5 @@ STAM_COUNTER_RESET(&pGstStream->In.StatBytesElapsed); STAM_COUNTER_RESET(&pGstStream->In.StatBytesTotalRead); - STAM_COUNTER_RESET(&pGstStream->In.StatSamplesCaptured); + STAM_COUNTER_RESET(&pGstStream->In.StatFramesCaptured); } } @@ -849,5 +849,5 @@ STAM_COUNTER_RESET(&pHstStream->Out.StatBytesElapsed); STAM_COUNTER_RESET(&pHstStream->Out.StatBytesTotalWritten); - STAM_COUNTER_RESET(&pHstStream->Out.StatSamplesPlayed); + STAM_COUNTER_RESET(&pHstStream->Out.StatFramesPlayed); if (pGstStream) @@ -857,5 +857,5 @@ STAM_COUNTER_RESET(&pGstStream->Out.StatBytesElapsed); STAM_COUNTER_RESET(&pGstStream->Out.StatBytesTotalWritten); - STAM_COUNTER_RESET(&pGstStream->Out.StatSamplesPlayed); + STAM_COUNTER_RESET(&pGstStream->Out.StatFramesPlayed); } } @@ -953,11 +953,11 @@ /* We use the guest side mixing buffer as an intermediate buffer to do some * (first) processing (if needed), so always write the incoming data at offset 0. */ - uint32_t csWritten = 0; - rc = AudioMixBufWriteAt(&pGstStream->MixBuf, 0 /* offSamples */, pvBuf, cbBuf, &csWritten); + uint32_t cfWritten = 0; + rc = AudioMixBufWriteAt(&pGstStream->MixBuf, 0 /* offFrames */, pvBuf, cbBuf, &cfWritten); if ( RT_FAILURE(rc) - || !csWritten) - { - AssertMsgFailed(("[%s] Write failed: cbBuf=%RU32, csWritten=%RU32, rc=%Rrc\n", - pGstStream->szName, cbBuf, csWritten, rc)); + || !cfWritten) + { + AssertMsgFailed(("[%s] Write failed: cbBuf=%RU32, cfWritten=%RU32, rc=%Rrc\n", + pGstStream->szName, cbBuf, cfWritten, rc)); break; } @@ -968,18 +968,18 @@ #ifdef VBOX_WITH_STATISTICS - STAM_COUNTER_ADD(&pThis->Stats.TotalSamplesWritten, csWritten); -#endif - uint32_t csMixed = 0; - if (csWritten) - { - int rc2 = AudioMixBufMixToParentEx(&pGstStream->MixBuf, 0 /* Offset */, csWritten /* Samples */, &csMixed); + STAM_COUNTER_ADD(&pThis->Stats.TotalFramesWritten, cfWritten); +#endif + uint32_t cfMixed = 0; + if (cfWritten) + { + int rc2 = AudioMixBufMixToParentEx(&pGstStream->MixBuf, 0 /* Offset */, cfWritten /* Frames */, &cfMixed); if ( RT_FAILURE(rc2) - || csMixed < csWritten) + || cfMixed < cfWritten) { - AssertMsgFailed(("[%s] Mixing failed: cbBuf=%RU32, csWritten=%RU32, csMixed=%RU32, rc=%Rrc\n", - pGstStream->szName, cbBuf, csWritten, csMixed, rc2)); - - LogRel2(("Audio: Lost audio samples (%RU32) due to full host stream '%s', expect stuttering audio output\n", - csWritten - csMixed, pHstStream->szName)); + AssertMsgFailed(("[%s] Mixing failed: cbBuf=%RU32, cfWritten=%RU32, cfMixed=%RU32, rc=%Rrc\n", + pGstStream->szName, cbBuf, cfWritten, cfMixed, rc2)); + + LogRel2(("Audio: Lost audio frames (%RU32) due to full host stream '%s', expect stuttering audio output\n", + cfWritten - cfMixed, pHstStream->szName)); /* Keep going. */ @@ -989,10 +989,10 @@ rc = rc2; - cbWritten = AUDIOMIXBUF_S2B(&pGstStream->MixBuf, csWritten); + cbWritten = AUDIOMIXBUF_F2B(&pGstStream->MixBuf, cfWritten); #ifdef VBOX_WITH_STATISTICS - STAM_COUNTER_ADD(&pThis->Stats.TotalSamplesMixedOut, csMixed); - Assert(csWritten >= csMixed); - STAM_COUNTER_ADD(&pThis->Stats.TotalSamplesLostOut, csWritten - csMixed); + STAM_COUNTER_ADD(&pThis->Stats.TotalFramesMixedOut, cfMixed); + Assert(cfWritten >= cfMixed); + STAM_COUNTER_ADD(&pThis->Stats.TotalFramesLostOut, cfWritten - cfMixed); STAM_COUNTER_ADD(&pThis->Stats.TotalBytesWritten, cbWritten); STAM_COUNTER_ADD(&pGstStream->Out.StatBytesTotalWritten, cbWritten); @@ -1002,5 +1002,5 @@ Log3Func(("[%s] cbBuf=%RU32, cUsed=%RU32, cLive=%RU32, cWritten=%RU32, cMixed=%RU32, rc=%Rrc\n", pGstStream->szName, - cbBuf, AudioMixBufUsed(&pGstStream->MixBuf), AudioMixBufLive(&pGstStream->MixBuf), csWritten, csMixed, rc)); + cbBuf, AudioMixBufUsed(&pGstStream->MixBuf), AudioMixBufLive(&pGstStream->MixBuf), cfWritten, cfMixed, rc)); } while (0); @@ -1143,5 +1143,5 @@ do { - uint32_t csMixed = 0; + uint32_t cfMixed = 0; rc = pThis->pHostDrvAudio->pfnStreamIterate(pThis->pHostDrvAudio, pHstStream->pvBackend); @@ -1151,11 +1151,11 @@ if (pHstStream->enmDir == PDMAUDIODIR_IN) { - /* Has the host captured any samples which were not mixed to the guest side yet? */ - uint32_t csCaptured = AudioMixBufUsed(&pHstStream->MixBuf); - if (csCaptured) + /* Has the host captured any frames which were not mixed to the guest side yet? */ + uint32_t cfCaptured = AudioMixBufUsed(&pHstStream->MixBuf); + if (cfCaptured) { - /* When capturing samples, the guest is the parent while the host is the child. - * So try mixing not yet mixed host-side samples to the guest-side buffer. */ - rc = AudioMixBufMixToParent(&pHstStream->MixBuf, csCaptured, &csMixed); + /* When capturing frames, the guest is the parent while the host is the child. + * So try mixing not yet mixed host-side frames to the guest-side buffer. */ + rc = AudioMixBufMixToParent(&pHstStream->MixBuf, cfCaptured, &cfMixed); if (RT_FAILURE(rc)) { @@ -1170,9 +1170,9 @@ #ifdef VBOX_WITH_STATISTICS - STAM_COUNTER_ADD(&pThis->Stats.TotalSamplesMixedIn, csMixed); - Assert(csCaptured >= csMixed); - STAM_COUNTER_ADD(&pThis->Stats.TotalSamplesLostIn, csCaptured - csMixed); -#endif - Log3Func(("[%s] %RU32/%RU32 input samples mixed, rc=%Rrc\n", pHstStream->szName, csMixed, csCaptured, rc)); + STAM_COUNTER_ADD(&pThis->Stats.TotalFramesMixedIn, cfMixed); + Assert(cfCaptured >= cfMixed); + STAM_COUNTER_ADD(&pThis->Stats.TotalFramesLostIn, cfCaptured - cfMixed); +#endif + Log3Func(("[%s] %RU32/%RU32 input frames mixed, rc=%Rrc\n", pHstStream->szName, cfMixed, cfCaptured, rc)); } else @@ -1257,13 +1257,13 @@ * @param pThis Pointer to driver instance. * @param pHstStream Host stream to play. - * @param csToPlay Number of audio samples to play. - * @param pcsPlayed Returns number of audio samples played. Optional. + * @param cfToPlay Number of audio frames to play. + * @param pcfPlayed Returns number of audio frames played. Optional. */ static int drvAudioStreamPlayNonInterleaved(PDRVAUDIO pThis, - PPDMAUDIOSTREAM pHstStream, uint32_t csToPlay, uint32_t *pcsPlayed) + PPDMAUDIOSTREAM pHstStream, uint32_t cfToPlay, uint32_t *pcfPlayed) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pHstStream, VERR_INVALID_POINTER); - /* pcsPlayed is optional. */ + /* pcfPlayed is optional. */ /* Sanity. */ @@ -1272,8 +1272,8 @@ Assert(pHstStream->Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED); - if (!csToPlay) - { - if (pcsPlayed) - *pcsPlayed = 0; + if (!cfToPlay) + { + if (pcfPlayed) + *pcfPlayed = 0; return VINF_SUCCESS; } @@ -1281,5 +1281,5 @@ int rc = VINF_SUCCESS; - uint32_t csPlayedTotal = 0; + uint32_t cfPlayedTotal = 0; AssertPtr(pThis->pHostDrvAudio->pfnStreamGetWritable); @@ -1287,19 +1287,19 @@ if (cbWritable) { - if (csToPlay > AUDIOMIXBUF_B2S(&pHstStream->MixBuf, cbWritable)) /* More samples available than we can write? Limit. */ - csToPlay = AUDIOMIXBUF_B2S(&pHstStream->MixBuf, cbWritable); - - if (csToPlay) + if (cfToPlay > AUDIOMIXBUF_B2F(&pHstStream->MixBuf, cbWritable)) /* More frames available than we can write? Limit. */ + cfToPlay = AUDIOMIXBUF_B2F(&pHstStream->MixBuf, cbWritable); + + if (cfToPlay) { uint8_t auBuf[256]; /** @todo Get rid of this here. */ - uint32_t cbLeft = AUDIOMIXBUF_S2B(&pHstStream->MixBuf, csToPlay); + uint32_t cbLeft = AUDIOMIXBUF_F2B(&pHstStream->MixBuf, cfToPlay); uint32_t cbChunk = sizeof(auBuf); while (cbLeft) { - uint32_t csRead = 0; - rc = AudioMixBufReadCirc(&pHstStream->MixBuf, auBuf, RT_MIN(cbChunk, cbLeft), &csRead); - if ( !csRead + uint32_t cfRead = 0; + rc = AudioMixBufReadCirc(&pHstStream->MixBuf, auBuf, RT_MIN(cbChunk, cbLeft), &cfRead); + if ( !cfRead || RT_FAILURE(rc)) { @@ -1307,5 +1307,5 @@ } - uint32_t cbRead = AUDIOMIXBUF_S2B(&pHstStream->MixBuf, csRead); + uint32_t cbRead = AUDIOMIXBUF_F2B(&pHstStream->MixBuf, cfRead); Assert(cbRead <= cbChunk); @@ -1326,8 +1326,8 @@ #if 0 /** @todo Also handle mono channels. Needs fixing */ AssertMsg(cbPlayed % 2 == 0, - ("Backend for stream '%s' returned uneven played bytes count (csRead=%RU32, cbPlayed=%RU32)\n", - pHstStream->szName, csRead, cbPlayed));*/ -#endif - csPlayedTotal += AUDIOMIXBUF_B2S(&pHstStream->MixBuf, cbPlayed); + ("Backend for stream '%s' returned uneven played bytes count (cfRead=%RU32, cbPlayed=%RU32)\n", + pHstStream->szName, cfRead, cbPlayed));*/ +#endif + cfPlayedTotal += AUDIOMIXBUF_B2F(&pHstStream->MixBuf, cbPlayed); Assert(cbLeft >= cbPlayed); cbLeft -= cbPlayed; @@ -1336,10 +1336,10 @@ } - Log3Func(("[%s] Played %RU32/%RU32 samples, rc=%Rrc\n", pHstStream->szName, csPlayedTotal, csToPlay, rc)); + Log3Func(("[%s] Played %RU32/%RU32 frames, rc=%Rrc\n", pHstStream->szName, cfPlayedTotal, cfToPlay, rc)); if (RT_SUCCESS(rc)) { - if (pcsPlayed) - *pcsPlayed = csPlayedTotal; + if (pcfPlayed) + *pcfPlayed = cfPlayedTotal; } @@ -1353,13 +1353,13 @@ * @param pThis Pointer to driver instance. * @param pHstStream Host stream to play. - * @param csToPlay Number of audio samples to play. - * @param pcsPlayed Returns number of audio samples played. Optional. + * @param cfToPlay Number of audio frames to play. + * @param pcfPlayed Returns number of audio frames played. Optional. */ static int drvAudioStreamPlayRaw(PDRVAUDIO pThis, - PPDMAUDIOSTREAM pHstStream, uint32_t csToPlay, uint32_t *pcsPlayed) + PPDMAUDIOSTREAM pHstStream, uint32_t cfToPlay, uint32_t *pcfPlayed) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pHstStream, VERR_INVALID_POINTER); - /* pcsPlayed is optional. */ + /* pcfPlayed is optional. */ /* Sanity. */ @@ -1368,8 +1368,8 @@ Assert(pHstStream->Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_RAW); - if (!csToPlay) - { - if (pcsPlayed) - *pcsPlayed = 0; + if (!cfToPlay) + { + if (pcfPlayed) + *pcfPlayed = 0; return VINF_SUCCESS; } @@ -1377,44 +1377,44 @@ int rc = VINF_SUCCESS; - uint32_t csPlayedTotal = 0; + uint32_t cfPlayedTotal = 0; AssertPtr(pThis->pHostDrvAudio->pfnStreamGetWritable); - uint32_t csWritable = pThis->pHostDrvAudio->pfnStreamGetWritable(pThis->pHostDrvAudio, pHstStream->pvBackend); - if (csWritable) - { - if (csToPlay > csWritable) /* More samples available than we can write? Limit. */ - csToPlay = csWritable; - - PDMAUDIOSAMPLE aSampleBuf[256]; /** @todo Get rid of this here. */ - - uint32_t csLeft = csToPlay; - while (csLeft) - { - uint32_t csRead = 0; - rc = AudioMixBufPeek(&pHstStream->MixBuf, csLeft, aSampleBuf, - RT_MIN(csLeft, RT_ELEMENTS(aSampleBuf)), &csRead); + uint32_t cfWritable = pThis->pHostDrvAudio->pfnStreamGetWritable(pThis->pHostDrvAudio, pHstStream->pvBackend); + if (cfWritable) + { + if (cfToPlay > cfWritable) /* More frames available than we can write? Limit. */ + cfToPlay = cfWritable; + + PDMAUDIOFRAME aFrameBuf[256]; /** @todo Get rid of this here. */ + + uint32_t cfLeft = cfToPlay; + while (cfLeft) + { + uint32_t cfRead = 0; + rc = AudioMixBufPeek(&pHstStream->MixBuf, cfLeft, aFrameBuf, + RT_MIN(cfLeft, RT_ELEMENTS(aFrameBuf)), &cfRead); if (RT_SUCCESS(rc)) { - if (csRead) + if (cfRead) { - uint32_t csPlayed; + uint32_t cfPlayed; /* Note: As the stream layout is RPDMAUDIOSTREAMLAYOUT_RAW, operate on audio frames * rather on bytes. */ - Assert(csRead <= RT_ELEMENTS(aSampleBuf)); + Assert(cfRead <= RT_ELEMENTS(aFrameBuf)); rc = pThis->pHostDrvAudio->pfnStreamPlay(pThis->pHostDrvAudio, pHstStream->pvBackend, - aSampleBuf, csRead, &csPlayed); + aFrameBuf, cfRead, &cfPlayed); if ( RT_FAILURE(rc) - || !csPlayed) + || !cfPlayed) { break; } - csPlayedTotal += csPlayed; - Assert(csPlayedTotal <= csToPlay); - - Assert(csLeft >= csRead); - csLeft -= csRead; + cfPlayedTotal += cfPlayed; + Assert(cfPlayedTotal <= cfToPlay); + + Assert(cfLeft >= cfRead); + cfLeft -= cfRead; } else @@ -1431,10 +1431,10 @@ } - Log3Func(("[%s] Played %RU32/%RU32 samples, rc=%Rrc\n", pHstStream->szName, csPlayedTotal, csToPlay, rc)); + Log3Func(("[%s] Played %RU32/%RU32 frames, rc=%Rrc\n", pHstStream->szName, cfPlayedTotal, cfToPlay, rc)); if (RT_SUCCESS(rc)) { - if (pcsPlayed) - *pcsPlayed = csPlayedTotal; + if (pcfPlayed) + *pcfPlayed = cfPlayedTotal; } @@ -1447,9 +1447,9 @@ */ static DECLCALLBACK(int) drvAudioStreamPlay(PPDMIAUDIOCONNECTOR pInterface, - PPDMAUDIOSTREAM pStream, uint32_t *pcSamplesPlayed) + PPDMAUDIOSTREAM pStream, uint32_t *pcFramesPlayed) { AssertPtrReturn(pInterface, VERR_INVALID_POINTER); AssertPtrReturn(pStream, VERR_INVALID_POINTER); - /* pcSamplesPlayed is optional. */ + /* pcFramesPlayed is optional. */ PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); @@ -1463,5 +1463,5 @@ pStream->szName, pStream->enmDir)); - uint32_t csPlayedTotal = 0; + uint32_t cfPlayedTotal = 0; do @@ -1501,5 +1501,5 @@ break; - uint32_t csToPlay = AudioMixBufLive(&pHstStream->MixBuf); + uint32_t cfToPlay = AudioMixBufLive(&pHstStream->MixBuf); if (pThis->pHostDrvAudio->pfnStreamPlayBegin) @@ -1508,9 +1508,9 @@ if (RT_LIKELY(pHstStream->Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED)) { - rc = drvAudioStreamPlayNonInterleaved(pThis, pHstStream, csToPlay, &csPlayedTotal); + rc = drvAudioStreamPlayNonInterleaved(pThis, pHstStream, cfToPlay, &cfPlayedTotal); } else if (pHstStream->Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_RAW) { - rc = drvAudioStreamPlayRaw(pThis, pHstStream, csToPlay, &csPlayedTotal); + rc = drvAudioStreamPlayRaw(pThis, pHstStream, cfToPlay, &cfPlayedTotal); } else @@ -1520,26 +1520,26 @@ pThis->pHostDrvAudio->pfnStreamPlayEnd(pThis->pHostDrvAudio, pHstStream->pvBackend); - uint32_t csLive = 0; + uint32_t cfLive = 0; if (RT_SUCCESS(rc)) { - AudioMixBufFinish(&pHstStream->MixBuf, csPlayedTotal); + AudioMixBufFinish(&pHstStream->MixBuf, cfPlayedTotal); #ifdef VBOX_WITH_STATISTICS - STAM_COUNTER_ADD(&pThis->Stats.TotalSamplesOut, csPlayedTotal); + STAM_COUNTER_ADD (&pThis->Stats.TotalFramesOut, cfPlayedTotal); STAM_PROFILE_ADV_STOP(&pThis->Stats.DelayOut, out); - STAM_COUNTER_ADD(&pHstStream->Out.StatSamplesPlayed, csPlayedTotal); -#endif - csLive = AudioMixBufLive(&pHstStream->MixBuf); + STAM_COUNTER_ADD (&pHstStream->Out.StatFramesPlayed, cfPlayedTotal); +#endif + cfLive = AudioMixBufLive(&pHstStream->MixBuf); } #ifdef LOG_ENABLED pszBackendSts = dbgAudioStreamStatusToStr(stsBackend); - Log3Func(("[%s] End: stsBackend=%s, csLive=%RU32, csPlayedTotal=%RU32, rc=%Rrc\n", - pHstStream->szName, pszBackendSts, csLive, csPlayedTotal, rc)); + Log3Func(("[%s] End: stsBackend=%s, cfLive=%RU32, cfPlayedTotal=%RU32, rc=%Rrc\n", + pHstStream->szName, pszBackendSts, cfLive, cfPlayedTotal, rc)); RTStrFree(pszBackendSts); #endif /* LOG_ENABLED */ - if (!csLive) + if (!cfLive) { /* Has the host stream marked as disabled but there still were guest streams relying @@ -1566,6 +1566,6 @@ if (RT_SUCCESS(rc)) { - if (pcSamplesPlayed) - *pcSamplesPlayed = csPlayedTotal; + if (pcFramesPlayed) + *pcFramesPlayed = cfPlayedTotal; } @@ -1582,11 +1582,11 @@ * @param pThis Driver instance. * @param pHstStream Host stream to capture from. - * @param pcsCaptured Number of (host) audio samples captured. Optional. - */ -static int drvAudioStreamCaptureNonInterleaved(PDRVAUDIO pThis, PPDMAUDIOSTREAM pHstStream, uint32_t *pcsCaptured) + * @param pcfCaptured Number of (host) audio frames captured. Optional. + */ +static int drvAudioStreamCaptureNonInterleaved(PDRVAUDIO pThis, PPDMAUDIOSTREAM pHstStream, uint32_t *pcfCaptured) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pHstStream, VERR_INVALID_POINTER); - /* pcsCaptured is optional. */ + /* pcfCaptured is optional. */ /* Sanity. */ @@ -1597,5 +1597,5 @@ int rc = VINF_SUCCESS; - uint32_t csCapturedTotal = 0; + uint32_t cfCapturedTotal = 0; AssertPtr(pThis->pHostDrvAudio->pfnStreamGetReadable); @@ -1610,5 +1610,5 @@ break; - uint32_t cbFree = AUDIOMIXBUF_S2B(&pHstStream->MixBuf, AudioMixBufFree(&pHstStream->MixBuf)); + uint32_t cbFree = AUDIOMIXBUF_F2B(&pHstStream->MixBuf, AudioMixBufFree(&pHstStream->MixBuf)); if (!cbFree) break; @@ -1640,8 +1640,8 @@ cbCaptured = (uint32_t)cbBuf; - uint32_t csCaptured = 0; - rc = AudioMixBufWriteCirc(&pHstStream->MixBuf, auBuf, cbCaptured, &csCaptured); + uint32_t cfCaptured = 0; + rc = AudioMixBufWriteCirc(&pHstStream->MixBuf, auBuf, cbCaptured, &cfCaptured); if (RT_SUCCESS(rc)) - csCapturedTotal += csCaptured; + cfCapturedTotal += cfCaptured; } else /* Nothing captured -- bail out. */ @@ -1652,8 +1652,8 @@ } - Log2Func(("[%s] %RU32 samples captured, rc=%Rrc\n", pHstStream->szName, csCapturedTotal, rc)); - - if (pcsCaptured) - *pcsCaptured = csCapturedTotal; + Log2Func(("[%s] %RU32 frames captured, rc=%Rrc\n", pHstStream->szName, cfCapturedTotal, rc)); + + if (pcfCaptured) + *pcfCaptured = cfCapturedTotal; return rc; @@ -1662,5 +1662,5 @@ /** * Captures raw input from a host stream. - * Raw input means that the backend directly operates on PDMAUDIOSAMPLE structs without + * Raw input means that the backend directly operates on PDMAUDIOFRAME structs without * no data layout processing done in between. * @@ -1670,11 +1670,11 @@ * @param pThis Driver instance. * @param pHstStream Host stream to capture from. - * @param pcsCaptured Number of (host) audio samples captured. Optional. - */ -static int drvAudioStreamCaptureRaw(PDRVAUDIO pThis, PPDMAUDIOSTREAM pHstStream, uint32_t *pcsCaptured) + * @param pcfCaptured Number of (host) audio frames captured. Optional. + */ +static int drvAudioStreamCaptureRaw(PDRVAUDIO pThis, PPDMAUDIOSTREAM pHstStream, uint32_t *pcfCaptured) { AssertPtrReturn(pThis, VERR_INVALID_POINTER); AssertPtrReturn(pHstStream, VERR_INVALID_POINTER); - /* pcsCaptured is optional. */ + /* pcfCaptured is optional. */ /* Sanity. */ @@ -1685,5 +1685,5 @@ int rc = VINF_SUCCESS; - uint32_t csCapturedTotal = 0; + uint32_t cfCapturedTotal = 0; AssertPtr(pThis->pHostDrvAudio->pfnStreamGetReadable); @@ -1702,17 +1702,17 @@ cbReadable = cbFree; - PPDMAUDIOSAMPLE paSamples; - uint32_t csWritable; - rc = AudioMixBufPeekMutable(&pHstStream->MixBuf, AUDIOMIXBUF_B2S(&pHstStream->MixBuf, cbReadable), - &paSamples, &csWritable); + PPDMAUDIOFRAME paFrames; + uint32_t cfWritable; + rc = AudioMixBufPeekMutable(&pHstStream->MixBuf, AUDIOMIXBUF_B2F(&pHstStream->MixBuf, cbReadable), + &paFrames, &cfWritable); if ( RT_FAILURE(rc) - || !csWritable) - { - break; - } - - uint32_t csCaptured; + || !cfWritable) + { + break; + } + + uint32_t cfCaptured; rc = pThis->pHostDrvAudio->pfnStreamCapture(pThis->pHostDrvAudio, pHstStream->pvBackend, - paSamples, csWritable, &csCaptured); + paFrames, cfWritable, &cfCaptured); if (RT_FAILURE(rc)) { @@ -1720,11 +1720,11 @@ AssertRC(rc2); } - else if (csCaptured) - { - Assert(csCaptured <= csWritable); - if (csCaptured > csWritable) /* Paranoia. */ - csCaptured = csWritable; - - csCapturedTotal += csCaptured; + else if (cfCaptured) + { + Assert(cfCaptured <= cfWritable); + if (cfCaptured > cfWritable) /* Paranoia. */ + cfCaptured = cfWritable; + + cfCapturedTotal += cfCaptured; } else /* Nothing captured -- bail out. */ @@ -1735,8 +1735,8 @@ } - Log2Func(("[%s] %RU32 samples captured, rc=%Rrc\n", pHstStream->szName, csCapturedTotal, rc)); - - if (pcsCaptured) - *pcsCaptured = csCapturedTotal; + Log2Func(("[%s] %RU32 frames captured, rc=%Rrc\n", pHstStream->szName, cfCapturedTotal, rc)); + + if (pcfCaptured) + *pcfCaptured = cfCapturedTotal; return rc; @@ -1747,5 +1747,5 @@ */ static DECLCALLBACK(int) drvAudioStreamCapture(PPDMIAUDIOCONNECTOR pInterface, - PPDMAUDIOSTREAM pStream, uint32_t *pcSamplesCaptured) + PPDMAUDIOSTREAM pStream, uint32_t *pcFramesCaptured) { PDRVAUDIO pThis = PDMIAUDIOCONNECTOR_2_DRVAUDIO(pInterface); @@ -1759,5 +1759,5 @@ pStream->szName, pStream->enmDir)); - uint32_t csCaptured = 0; + uint32_t cfCaptured = 0; do @@ -1800,9 +1800,9 @@ if (RT_LIKELY(pHstStream->Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_NON_INTERLEAVED)) { - rc = drvAudioStreamCaptureNonInterleaved(pThis, pHstStream, &csCaptured); + rc = drvAudioStreamCaptureNonInterleaved(pThis, pHstStream, &cfCaptured); } else if (pHstStream->Cfg.enmLayout == PDMAUDIOSTREAMLAYOUT_RAW) { - rc = drvAudioStreamCaptureRaw(pThis, pHstStream, &csCaptured); + rc = drvAudioStreamCaptureRaw(pThis, pHstStream, &cfCaptured); } else @@ -1814,6 +1814,6 @@ #ifdef LOG_ENABLED pszBackendSts = dbgAudioStreamStatusToStr(stsBackend); - Log3Func(("[%s] End: stsBackend=%s, csCaptured=%RU32, rc=%Rrc\n", - pHstStream->szName, pszBackendSts, csCaptured, rc)); + Log3Func(("[%s] End: stsBackend=%s, cfCaptured=%RU32, rc=%Rrc\n", + pHstStream->szName, pszBackendSts, cfCaptured, rc)); RTStrFree(pszBackendSts); #endif /* LOG_ENABLED */ @@ -1821,9 +1821,9 @@ if (RT_SUCCESS(rc)) { - Log3Func(("[%s] %RU32 samples captured, rc=%Rrc\n", pHstStream->szName, csCaptured, rc)); + Log3Func(("[%s] %RU32 frames captured, rc=%Rrc\n", pHstStream->szName, cfCaptured, rc)); #ifdef VBOX_WITH_STATISTICS - STAM_COUNTER_ADD(&pThis->Stats.TotalSamplesIn, csCaptured); - STAM_COUNTER_ADD(&pHstStream->In.StatSamplesCaptured, csCaptured); + STAM_COUNTER_ADD(&pThis->Stats.TotalFramesIn, cfCaptured); + STAM_COUNTER_ADD(&pHstStream->In.StatFramesCaptured, cfCaptured); #endif } @@ -1838,6 +1838,6 @@ } while (0); - if (pcSamplesCaptured) - *pcSamplesCaptured = csCaptured; + if (pcFramesCaptured) + *pcFramesCaptured = cfCaptured; int rc2 = RTCritSectLeave(&pThis->CritSect); @@ -2365,15 +2365,15 @@ uint32_t cReadTotal = 0; - uint32_t cToRead = RT_MIN(AUDIOMIXBUF_B2S(&pGstStream->MixBuf, cbBuf), AudioMixBufUsed(&pGstStream->MixBuf)); + uint32_t cToRead = RT_MIN(AUDIOMIXBUF_B2F(&pGstStream->MixBuf, cbBuf), AudioMixBufUsed(&pGstStream->MixBuf)); while (cToRead) { uint32_t cRead; - rc = AudioMixBufReadCirc(&pGstStream->MixBuf, (uint8_t *)pvBuf + AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cReadTotal), - AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cToRead), &cRead); + rc = AudioMixBufReadCirc(&pGstStream->MixBuf, (uint8_t *)pvBuf + AUDIOMIXBUF_F2B(&pGstStream->MixBuf, cReadTotal), + AUDIOMIXBUF_F2B(&pGstStream->MixBuf, cToRead), &cRead); if (RT_FAILURE(rc)) break; #if defined (VBOX_WITH_STATISTICS) || defined (VBOX_AUDIO_DEBUG_DUMP_PCM_DATA) - const uint32_t cbRead = AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cRead); + const uint32_t cbRead = AUDIOMIXBUF_F2B(&pGstStream->MixBuf, cRead); #endif @@ -2392,5 +2392,5 @@ #ifdef VBOX_AUDIO_DEBUG_DUMP_PCM_DATA drvAudioDbgPCMDump(pThis, VBOX_AUDIO_DEBUG_DUMP_PCM_DATA_PATH, "StreamRead.pcm", - pvBuf, AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cReadTotal)); + pvBuf, AUDIOMIXBUF_F2B(&pGstStream->MixBuf, cReadTotal)); #endif AudioMixBufFinish(&pGstStream->MixBuf, cReadTotal); @@ -2398,5 +2398,5 @@ pGstStream->In.tsLastReadMS = RTTimeMilliTS(); - cbReadTotal = AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cReadTotal); + cbReadTotal = AUDIOMIXBUF_F2B(&pGstStream->MixBuf, cReadTotal); } @@ -2548,7 +2548,7 @@ szStatName, STAMUNIT_BYTES, "Total bytes read."); - RTStrPrintf(szStatName, sizeof(szStatName), "Host/%s/SamplesCaptured", pHstStrm->szName); - PDMDrvHlpSTAMRegCounterEx(pThis->pDrvIns, &pHstStrm->In.StatSamplesCaptured, - szStatName, STAMUNIT_COUNT, "Total samples captured."); + RTStrPrintf(szStatName, sizeof(szStatName), "Host/%s/FramesCaptured", pHstStrm->szName); + PDMDrvHlpSTAMRegCounterEx(pThis->pDrvIns, &pHstStrm->In.StatFramesCaptured, + szStatName, STAMUNIT_COUNT, "Total frames captured."); } else if (pCfgGuest->enmDir == PDMAUDIODIR_OUT) @@ -2562,7 +2562,7 @@ szStatName, STAMUNIT_BYTES, "Total bytes written."); - RTStrPrintf(szStatName, sizeof(szStatName), "Host/%s/SamplesPlayed", pHstStrm->szName); - PDMDrvHlpSTAMRegCounterEx(pThis->pDrvIns, &pHstStrm->Out.StatSamplesPlayed, - szStatName, STAMUNIT_COUNT, "Total samples played."); + RTStrPrintf(szStatName, sizeof(szStatName), "Host/%s/FramesPlayed", pHstStrm->szName); + PDMDrvHlpSTAMRegCounterEx(pThis->pDrvIns, &pHstStrm->Out.StatFramesPlayed, + szStatName, STAMUNIT_COUNT, "Total frames played."); } else @@ -2723,12 +2723,12 @@ Log3Func(("[%s] cbReadable=%RU32 (%zu bytes)\n", pHstStream->szName, cReadable, - AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cReadable))); - - uint32_t cbReadable = AUDIOMIXBUF_S2B(&pGstStream->MixBuf, cReadable); + AUDIOMIXBUF_F2B(&pGstStream->MixBuf, cReadable))); + + uint32_t cbReadable = AUDIOMIXBUF_F2B(&pGstStream->MixBuf, cReadable); rc2 = RTCritSectLeave(&pThis->CritSect); AssertRC(rc2); - /* Return bytes instead of audio samples. */ + /* Return bytes instead of audio frames. */ return cbReadable; } @@ -2862,9 +2862,9 @@ if (pHstStream->enmDir == PDMAUDIODIR_IN) { - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pHstStream->In.StatSamplesCaptured); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pHstStream->In.StatFramesCaptured); } else if (pHstStream->enmDir == PDMAUDIODIR_OUT) { - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pHstStream->Out.StatSamplesPlayed); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pHstStream->Out.StatFramesPlayed); } else @@ -2891,5 +2891,5 @@ PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pGstStream->In.StatBytesElapsed); PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pGstStream->In.StatBytesTotalRead); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pGstStream->In.StatSamplesCaptured); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pGstStream->In.StatFramesCaptured); } else if (pGstStream->enmDir == PDMAUDIODIR_OUT) @@ -2897,5 +2897,5 @@ PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pGstStream->Out.StatBytesElapsed); PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pGstStream->Out.StatBytesTotalWritten); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pGstStream->Out.StatSamplesPlayed); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pGstStream->Out.StatFramesPlayed); } else @@ -3217,20 +3217,20 @@ PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalStreamsCreated, "TotalStreamsCreated", STAMUNIT_COUNT, "Total created audio streams."); - PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalSamplesRead, "TotalSamplesRead", - STAMUNIT_COUNT, "Total samples read by device emulation."); - PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalSamplesWritten, "TotalSamplesWritten", - STAMUNIT_COUNT, "Total samples written by device emulation "); - PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalSamplesMixedIn, "TotalSamplesMixedIn", - STAMUNIT_COUNT, "Total input samples mixed."); - PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalSamplesMixedOut, "TotalSamplesMixedOut", - STAMUNIT_COUNT, "Total output samples mixed."); - PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalSamplesLostIn, "TotalSamplesLostIn", - STAMUNIT_COUNT, "Total input samples lost."); - PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalSamplesLostOut, "TotalSamplesLostOut", - STAMUNIT_COUNT, "Total output samples lost."); - PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalSamplesOut, "TotalSamplesPlayed", - STAMUNIT_COUNT, "Total samples played by backend."); - PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalSamplesIn, "TotalSamplesCaptured", - STAMUNIT_COUNT, "Total samples captured by backend."); + PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalFramesRead, "TotalFramesRead", + STAMUNIT_COUNT, "Total frames read by device emulation."); + PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalFramesWritten, "TotalFramesWritten", + STAMUNIT_COUNT, "Total frames written by device emulation "); + PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalFramesMixedIn, "TotalFramesMixedIn", + STAMUNIT_COUNT, "Total input frames mixed."); + PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalFramesMixedOut, "TotalFramesMixedOut", + STAMUNIT_COUNT, "Total output frames mixed."); + PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalFramesLostIn, "TotalFramesLostIn", + STAMUNIT_COUNT, "Total input frames lost."); + PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalFramesLostOut, "TotalFramesLostOut", + STAMUNIT_COUNT, "Total output frames lost."); + PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalFramesOut, "TotalFramesPlayed", + STAMUNIT_COUNT, "Total frames played by backend."); + PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalFramesIn, "TotalFramesCaptured", + STAMUNIT_COUNT, "Total frames captured by backend."); PDMDrvHlpSTAMRegCounterEx(pDrvIns, &pThis->Stats.TotalBytesRead, "TotalBytesRead", STAMUNIT_BYTES, "Total bytes read."); @@ -3334,12 +3334,12 @@ PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalStreamsActive); PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalStreamsCreated); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalSamplesRead); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalSamplesWritten); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalSamplesMixedIn); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalSamplesMixedOut); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalSamplesLostIn); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalSamplesLostOut); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalSamplesOut); - PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalSamplesIn); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalFramesRead); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalFramesWritten); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalFramesMixedIn); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalFramesMixedOut); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalFramesLostIn); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalFramesLostOut); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalFramesOut); + PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalFramesIn); PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalBytesRead); PDMDrvHlpSTAMDeregister(pThis->pDrvIns, &pThis->Stats.TotalBytesWritten); Index: /trunk/src/VBox/Devices/Audio/DrvAudio.h =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvAudio.h (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvAudio.h (revision 68132) @@ -94,12 +94,12 @@ STAMCOUNTER TotalStreamsActive; STAMCOUNTER TotalStreamsCreated; - STAMCOUNTER TotalSamplesRead; - STAMCOUNTER TotalSamplesWritten; - STAMCOUNTER TotalSamplesMixedIn; - STAMCOUNTER TotalSamplesMixedOut; - STAMCOUNTER TotalSamplesLostIn; - STAMCOUNTER TotalSamplesLostOut; - STAMCOUNTER TotalSamplesOut; - STAMCOUNTER TotalSamplesIn; + STAMCOUNTER TotalFramesRead; + STAMCOUNTER TotalFramesWritten; + STAMCOUNTER TotalFramesMixedIn; + STAMCOUNTER TotalFramesMixedOut; + STAMCOUNTER TotalFramesLostIn; + STAMCOUNTER TotalFramesLostOut; + STAMCOUNTER TotalFramesOut; + STAMCOUNTER TotalFramesIn; STAMCOUNTER TotalBytesRead; STAMCOUNTER TotalBytesWritten; @@ -165,5 +165,5 @@ uint8_t DrvAudioHlpAudFmtToBits(PDMAUDIOFMT enmFmt); const char *DrvAudioHlpAudFmtToStr(PDMAUDIOFMT enmFmt); -void DrvAudioHlpClearBuf(const PPDMAUDIOPCMPROPS pPCMInfo, void *pvBuf, size_t cbBuf, uint32_t cSamples); +void DrvAudioHlpClearBuf(const PPDMAUDIOPCMPROPS pPCMInfo, void *pvBuf, size_t cbBuf, uint32_t cFrames); uint32_t DrvAudioHlpCalcBitrate(uint8_t cBits, uint32_t uHz, uint8_t cChannels); uint32_t DrvAudioHlpCalcBitrate(const PPDMAUDIOPCMPROPS pProps); Index: /trunk/src/VBox/Devices/Audio/DrvHostALSAAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostALSAAudio.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvHostALSAAudio.cpp (revision 68132) @@ -1035,5 +1035,5 @@ { case SND_PCM_STATE_PREPARED: - cAvail = PDMAUDIOSTREAMCFG_B2S(pCfg, cxBuf); + cAvail = PDMAUDIOSTREAMCFG_B2F(pCfg, cxBuf); break; @@ -1065,5 +1065,5 @@ * the mixer buffer. */ - size_t cbToRead = RT_MIN((size_t)PDMAUDIOSTREAMCFG_S2B(pCfg, cAvail), cxBuf); + size_t cbToRead = RT_MIN((size_t)PDMAUDIOSTREAMCFG_F2B(pCfg, cAvail), cxBuf); LogFlowFunc(("cbToRead=%zu, cAvail=%RI32\n", cbToRead, cAvail)); @@ -1077,6 +1077,6 @@ && RT_SUCCESS(rc)) { - cToRead = RT_MIN(PDMAUDIOSTREAMCFG_B2S(pCfg, cbToRead), - PDMAUDIOSTREAMCFG_B2S(pCfg, pStreamALSA->cbBuf)); + cToRead = RT_MIN(PDMAUDIOSTREAMCFG_B2F(pCfg, cbToRead), + PDMAUDIOSTREAMCFG_B2F(pCfg, pStreamALSA->cbBuf)); AssertBreakStmt(cToRead, rc = VERR_NO_DATA); cRead = snd_pcm_readi(pStreamALSA->phPCM, pStreamALSA->pvBuf, cToRead); @@ -1128,5 +1128,5 @@ * capture device for example). */ - uint32_t cbRead = PDMAUDIOSTREAMCFG_S2B(pCfg, cRead); + uint32_t cbRead = PDMAUDIOSTREAMCFG_F2B(pCfg, cRead); memcpy(pvBuf, pStreamALSA->pvBuf, cbRead); @@ -1181,5 +1181,5 @@ break; - size_t cbToWrite = RT_MIN((unsigned)PDMAUDIOSTREAMCFG_S2B(pCfg, csAvail), pStreamALSA->cbBuf); + size_t cbToWrite = RT_MIN((unsigned)PDMAUDIOSTREAMCFG_F2B(pCfg, csAvail), pStreamALSA->cbBuf); if (!cbToWrite) break; @@ -1198,5 +1198,5 @@ { csWritten = snd_pcm_writei(pStreamALSA->phPCM, pStreamALSA->pvBuf, - PDMAUDIOSTREAMCFG_B2S(pCfg, cbToWrite)); + PDMAUDIOSTREAMCFG_B2F(pCfg, cbToWrite)); if (csWritten <= 0) { @@ -1251,5 +1251,5 @@ break; - cbWrittenTotal = PDMAUDIOSTREAMCFG_S2B(pCfg, csWritten); + cbWrittenTotal = PDMAUDIOSTREAMCFG_F2B(pCfg, csWritten); } while (0); @@ -1320,9 +1320,9 @@ break; - pCfgAcq->cSampleBufferHint = obt.samples * 4; + pCfgAcq->cFrameBufferHint = obt.samples * 4; AssertBreakStmt(obt.samples, rc = VERR_INVALID_PARAMETER); - size_t cbBuf = obt.samples * PDMAUDIOSTREAMCFG_S2B(pCfgAcq, 1); + size_t cbBuf = obt.samples * PDMAUDIOSTREAMCFG_F2B(pCfgAcq, 1); AssertBreakStmt(cbBuf, rc = VERR_INVALID_PARAMETER); @@ -1375,9 +1375,9 @@ break; - pCfgAcq->cSampleBufferHint = obt.samples; + pCfgAcq->cFrameBufferHint = obt.samples; AssertBreakStmt(obt.samples, rc = VERR_INVALID_PARAMETER); - size_t cbBuf = obt.samples * PDMAUDIOSTREAMCFG_S2B(pCfgAcq, 1); + size_t cbBuf = obt.samples * PDMAUDIOSTREAMCFG_F2B(pCfgAcq, 1); AssertBreakStmt(cbBuf, rc = VERR_INVALID_PARAMETER); @@ -1644,5 +1644,5 @@ int rc = alsaStreamGetAvail(pStreamALSA->phPCM, &cFramesAvail); if (RT_SUCCESS(rc)) - cbAvail = PDMAUDIOSTREAMCFG_S2B(pStreamALSA->pCfg, cFramesAvail); + cbAvail = PDMAUDIOSTREAMCFG_F2B(pStreamALSA->pCfg, cFramesAvail); return cbAvail; @@ -1666,5 +1666,5 @@ && (uint32_t)cFramesAvail >= pStreamALSA->Out.cSamplesMin) { - cbAvail = PDMAUDIOSTREAMCFG_S2B(pStreamALSA->pCfg, cFramesAvail); + cbAvail = PDMAUDIOSTREAMCFG_F2B(pStreamALSA->pCfg, cFramesAvail); } Index: /trunk/src/VBox/Devices/Audio/DrvHostCoreAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostCoreAudio.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvHostCoreAudio.cpp (revision 68132) @@ -1591,5 +1591,5 @@ } - rc = RTCircBufCreate(&pCAStream->pCircBuf, PDMAUDIOSTREAMCFG_S2B(pCfgReq, 4096)); /** @todo Make this configurable. */ + rc = RTCircBufCreate(&pCAStream->pCircBuf, PDMAUDIOSTREAMCFG_F2B(pCfgReq, 4096)); /** @todo Make this configurable. */ if (RT_FAILURE(rc)) return rc; @@ -2308,5 +2308,5 @@ if (RT_SUCCESS(rc)) { - pCfgAcq->cSampleBufferHint = _4K; /** @todo Make this configurable. */ + pCfgAcq->cFrameBufferHint = _4K; /** @todo Make this configurable. */ } if (RT_SUCCESS(rc)) Index: /trunk/src/VBox/Devices/Audio/DrvHostDSound.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostDSound.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvHostDSound.cpp (revision 68132) @@ -733,5 +733,5 @@ #endif /* VBOX_WITH_AUDIO_DEVICE_CALLBACKS */ - pCfgAcq->cSampleBufferHint = PDMAUDIOSTREAMCFG_B2S(pCfgAcq, pThis->cfg.cbBufferOut); + pCfgAcq->cFrameBufferHint = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, pThis->cfg.cbBufferOut); } while (0); @@ -758,6 +758,6 @@ if (SUCCEEDED(hr)) { - DWORD len1 = PDMAUDIOPCMPROPS_B2S(pProps, cb1); - DWORD len2 = PDMAUDIOPCMPROPS_B2S(pProps, cb2); + DWORD len1 = PDMAUDIOPCMPROPS_B2F(pProps, cb1); + DWORD len2 = PDMAUDIOPCMPROPS_B2F(pProps, cb2); if (pv1 && len1) @@ -1170,5 +1170,5 @@ pStreamDS->In.offCaptureBufRead, pStreamDS->In.cbCaptureBuf)); - pCfgAcq->cSampleBufferHint = PDMAUDIOSTREAMCFG_B2S(pCfgAcq, pThis->cfg.cbBufferIn); + pCfgAcq->cFrameBufferHint = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, pThis->cfg.cbBufferIn); } while (0); @@ -1582,5 +1582,5 @@ * i.e. always leave a free space for 1 audio sample. */ - const DWORD cbSample = PDMAUDIOPCMPROPS_S2B(pProps, 1); + const DWORD cbSample = PDMAUDIOPCMPROPS_F2B(pProps, 1); if (cbFree <= cbSample) break; Index: /trunk/src/VBox/Devices/Audio/DrvHostDebugAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostDebugAudio.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvHostDebugAudio.cpp (revision 68132) @@ -125,5 +125,5 @@ if (pCfgAcq) - pCfgAcq->cSampleBufferHint = _1K; + pCfgAcq->cFrameBufferHint = _1K; return VINF_SUCCESS; @@ -138,5 +138,5 @@ int rc = VINF_SUCCESS; - pStreamDbg->Out.cbPlayBuffer = _1K * PDMAUDIOSTREAMCFG_S2B(pCfgReq, 1); /** @todo Make this configurable? */ + pStreamDbg->Out.cbPlayBuffer = _1K * PDMAUDIOSTREAMCFG_F2B(pCfgReq, 1); /** @todo Make this configurable? */ pStreamDbg->Out.auPlayBuffer = (uint8_t *)RTMemAlloc(pStreamDbg->Out.cbPlayBuffer); if (!pStreamDbg->Out.auPlayBuffer) @@ -170,5 +170,5 @@ { if (pCfgAcq) - pCfgAcq->cSampleBufferHint = PDMAUDIOSTREAMCFG_B2S(pCfgAcq, pStreamDbg->Out.cbPlayBuffer); + pCfgAcq->cFrameBufferHint = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, pStreamDbg->Out.cbPlayBuffer); } Index: /trunk/src/VBox/Devices/Audio/DrvHostNullAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostNullAudio.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvHostNullAudio.cpp (revision 68132) @@ -176,5 +176,5 @@ if (pCfgAcq) - pCfgAcq->cSampleBufferHint = _1K; + pCfgAcq->cFrameBufferHint = _1K; return VINF_SUCCESS; @@ -187,5 +187,5 @@ if (pCfgAcq) - pCfgAcq->cSampleBufferHint = _1K; /** @todo Make this configurable. */ + pCfgAcq->cFrameBufferHint = _1K; /** @todo Make this configurable. */ return VINF_SUCCESS; Index: /trunk/src/VBox/Devices/Audio/DrvHostOSSAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostOSSAudio.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvHostOSSAudio.cpp (revision 68132) @@ -360,5 +360,5 @@ { DrvAudioHlpClearBuf(&pStreamOSS->pCfg->Props, pStreamOSS->pvBuf, pStreamOSS->cbBuf, - PDMAUDIOPCMPROPS_B2S(&pStreamOSS->pCfg->Props, pStreamOSS->cbBuf)); + PDMAUDIOPCMPROPS_B2F(&pStreamOSS->pCfg->Props, pStreamOSS->cbBuf)); int mask = PCM_ENABLE_OUTPUT; @@ -647,5 +647,5 @@ } - uint32_t cSamples = PDMAUDIOSTREAMCFG_B2S(pCfgAcq, ossAcq.cFragments * ossAcq.cbFragmentSize); + uint32_t cSamples = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, ossAcq.cFragments * ossAcq.cbFragmentSize); if (!cSamples) rc = VERR_INVALID_PARAMETER; @@ -653,5 +653,5 @@ if (RT_SUCCESS(rc)) { - size_t cbBuf = PDMAUDIOSTREAMCFG_S2B(pCfgAcq, cSamples); + size_t cbBuf = PDMAUDIOSTREAMCFG_F2B(pCfgAcq, cSamples); void *pvBuf = RTMemAlloc(cbBuf); if (!pvBuf) @@ -665,5 +665,5 @@ pStreamOSS->cbBuf = cbBuf; - pCfgAcq->cSampleBufferHint = cSamples; + pCfgAcq->cFrameBufferHint = cSamples; } } @@ -700,5 +700,5 @@ memcpy(&pCfgAcq->Props, &obtStream.Props, sizeof(PDMAUDIOPCMPROPS)); - cSamples = PDMAUDIOSTREAMCFG_B2S(pCfgAcq, obtStream.cFragments * obtStream.cbFragmentSize); + cSamples = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, obtStream.cFragments * obtStream.cbFragmentSize); if (obtStream.cFragments * obtStream.cbFragmentSize & pStreamOSS->uAlign) @@ -713,5 +713,5 @@ pStreamOSS->Out.fMMIO = false; - size_t cbBuf = PDMAUDIOSTREAMCFG_S2B(pCfgAcq, cSamples); + size_t cbBuf = PDMAUDIOSTREAMCFG_F2B(pCfgAcq, cSamples); Assert(cbBuf); @@ -781,5 +781,5 @@ } #endif - pCfgAcq->cSampleBufferHint = cSamples; + pCfgAcq->cFrameBufferHint = cSamples; } Index: /trunk/src/VBox/Devices/Audio/DrvHostPulseAudio.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostPulseAudio.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvHostPulseAudio.cpp (revision 68132) @@ -776,5 +776,5 @@ if (cbBuf) { - pCfgAcq->cSampleBufferHint = PDMAUDIOSTREAMCFG_B2S(pCfgAcq, cbBuf); + pCfgAcq->cFrameBufferHint = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, cbBuf); pStreamPA->pDrv = pThis; @@ -815,5 +815,5 @@ pCfgAcq->Props.uHz = pStreamPA->SampleSpec.rate; pCfgAcq->Props.cChannels = pStreamPA->SampleSpec.channels; - pCfgAcq->cSampleBufferHint = PDMAUDIOSTREAMCFG_B2S(pCfgAcq, + pCfgAcq->cFrameBufferHint = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, RT_MIN(pStreamPA->BufAttr.fragsize * 10, pStreamPA->BufAttr.maxlength)); Index: /trunk/src/VBox/Devices/Audio/DrvHostValidationKit.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/DrvHostValidationKit.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/DrvHostValidationKit.cpp (revision 68132) @@ -132,5 +132,5 @@ if (pCfgAcq) - pCfgAcq->cSampleBufferHint = _1K; + pCfgAcq->cFrameBufferHint = _1K; return VINF_SUCCESS; @@ -148,5 +148,5 @@ pStreamDbg->uSamplesSinceStarted = 0; pStreamDbg->Out.tsLastPlayed = 0; - pStreamDbg->Out.cbPlayBuffer = 16 * _1K * PDMAUDIOSTREAMCFG_S2B(pCfgReq, 1); /** @todo Make this configurable? */ + pStreamDbg->Out.cbPlayBuffer = 16 * _1K * PDMAUDIOSTREAMCFG_F2B(pCfgReq, 1); /** @todo Make this configurable? */ pStreamDbg->Out.pu8PlayBuffer = (uint8_t *)RTMemAlloc(pStreamDbg->Out.cbPlayBuffer); if (!pStreamDbg->Out.pu8PlayBuffer) @@ -200,5 +200,5 @@ { if (pCfgAcq) - pCfgAcq->cSampleBufferHint = PDMAUDIOSTREAMCFG_B2S(pCfgAcq, pStreamDbg->Out.cbPlayBuffer); + pCfgAcq->cFrameBufferHint = PDMAUDIOSTREAMCFG_B2F(pCfgAcq, pStreamDbg->Out.cbPlayBuffer); } Index: /trunk/src/VBox/Devices/Audio/testcase/tstAudioMixBuffer.cpp =================================================================== --- /trunk/src/VBox/Devices/Audio/testcase/tstAudioMixBuffer.cpp (revision 68131) +++ /trunk/src/VBox/Devices/Audio/testcase/tstAudioMixBuffer.cpp (revision 68132) @@ -62,41 +62,40 @@ RTTESTI_CHECK_RC_OK(AudioMixBufInit(&mb, "Single", &config, cBufSize)); RTTESTI_CHECK(AudioMixBufSize(&mb) == cBufSize); - RTTESTI_CHECK(AUDIOMIXBUF_B2S(&mb, AudioMixBufSizeBytes(&mb)) == cBufSize); - RTTESTI_CHECK(AUDIOMIXBUF_S2B(&mb, AudioMixBufSize(&mb)) == AudioMixBufSizeBytes(&mb)); + RTTESTI_CHECK(AUDIOMIXBUF_B2F(&mb, AudioMixBufSizeBytes(&mb)) == cBufSize); + RTTESTI_CHECK(AUDIOMIXBUF_F2B(&mb, AudioMixBufSize(&mb)) == AudioMixBufSizeBytes(&mb)); RTTESTI_CHECK(AudioMixBufFree(&mb) == cBufSize); - RTTESTI_CHECK(AUDIOMIXBUF_S2B(&mb, AudioMixBufFree(&mb)) == AudioMixBufFreeBytes(&mb)); + RTTESTI_CHECK(AUDIOMIXBUF_F2B(&mb, AudioMixBufFree(&mb)) == AudioMixBufFreeBytes(&mb)); /* * Absolute writes. */ - uint32_t cSamplesRead = 0, cSamplesWritten = 0, cSamplesWrittenAbs = 0; - int8_t samples8 [2] = { 0x12, 0x34 }; - int16_t samples16[2] = { 0xAA, 0xBB }; - int32_t samples32[2] = { 0xCC, 0xDD }; - /* int64_t samples64[2] = { 0xEE, 0xFF }; - unused */ - - RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 0 /* Offset */, &samples8, sizeof(samples8), &cSamplesWritten)); - RTTESTI_CHECK(cSamplesWritten == 0 /* Samples */); + uint32_t cFramesRead = 0, cFramesWritten = 0, cFramesWrittenAbs = 0; + int8_t aFrames8 [2] = { 0x12, 0x34 }; + int16_t aFrames16[2] = { 0xAA, 0xBB }; + int32_t aFrames32[2] = { 0xCC, 0xDD }; + + RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 0 /* Offset */, &aFrames8, sizeof(aFrames8), &cFramesWritten)); + RTTESTI_CHECK(cFramesWritten == 0 /* Frames */); RTTESTI_CHECK(AudioMixBufUsed(&mb) == 0); - RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 0 /* Offset */, &samples16, sizeof(samples16), &cSamplesWritten)); - RTTESTI_CHECK(cSamplesWritten == 1 /* Samples */); + RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 0 /* Offset */, &aFrames16, sizeof(aFrames16), &cFramesWritten)); + RTTESTI_CHECK(cFramesWritten == 1 /* Frames */); RTTESTI_CHECK(AudioMixBufUsed(&mb) == 1); - RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 2 /* Offset */, &samples32, sizeof(samples32), &cSamplesWritten)); - RTTESTI_CHECK(cSamplesWritten == 2 /* Samples */); + RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 2 /* Offset */, &aFrames32, sizeof(aFrames32), &cFramesWritten)); + RTTESTI_CHECK(cFramesWritten == 2 /* Frames */); RTTESTI_CHECK(AudioMixBufUsed(&mb) == 2); /* Beyond buffer. */ - RTTESTI_CHECK_RC(AudioMixBufWriteAt(&mb, AudioMixBufSize(&mb) + 1, &samples16, sizeof(samples16), - &cSamplesWritten), VERR_BUFFER_OVERFLOW); - - /* Offset wrap-around: When writing as much (or more) samples the mixing buffer can hold. */ + RTTESTI_CHECK_RC(AudioMixBufWriteAt(&mb, AudioMixBufSize(&mb) + 1, &aFrames16, sizeof(aFrames16), + &cFramesWritten), VERR_BUFFER_OVERFLOW); + + /* Offset wrap-around: When writing as much (or more) frames the mixing buffer can hold. */ uint32_t cbSamples = cBufSize * sizeof(int16_t) * 2 /* Channels */; RTTESTI_CHECK(cbSamples); uint16_t *paSamples = (uint16_t *)RTMemAlloc(cbSamples); RTTESTI_CHECK(paSamples); - RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 0 /* Offset */, paSamples, cbSamples, &cSamplesWritten)); - RTTESTI_CHECK(cSamplesWritten == cBufSize /* Samples */); + RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 0 /* Offset */, paSamples, cbSamples, &cFramesWritten)); + RTTESTI_CHECK(cFramesWritten == cBufSize /* Frames */); RTTESTI_CHECK(AudioMixBufUsed(&mb) == cBufSize); RTTESTI_CHECK(AudioMixBufReadPos(&mb) == 0); @@ -110,46 +109,46 @@ AudioMixBufReset(&mb); - RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 2 /* Offset */, &samples32, sizeof(samples32), &cSamplesWritten)); - RTTESTI_CHECK(cSamplesWritten == 2 /* Samples */); + RTTESTI_CHECK_RC_OK(AudioMixBufWriteAt(&mb, 2 /* Offset */, &aFrames32, sizeof(aFrames32), &cFramesWritten)); + RTTESTI_CHECK(cFramesWritten == 2 /* Frames */); RTTESTI_CHECK(AudioMixBufUsed(&mb) == 2); - cSamplesWrittenAbs = AudioMixBufUsed(&mb); - - uint32_t cToWrite = AudioMixBufSize(&mb) - cSamplesWrittenAbs - 1; /* -1 as padding plus -2 samples for above. */ + cFramesWrittenAbs = AudioMixBufUsed(&mb); + + uint32_t cToWrite = AudioMixBufSize(&mb) - cFramesWrittenAbs - 1; /* -1 as padding plus -2 frames for above. */ for (uint32_t i = 0; i < cToWrite; i++) { - RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&mb, &samples16, sizeof(samples16), &cSamplesWritten)); - RTTESTI_CHECK(cSamplesWritten == 1); + RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&mb, &aFrames16, sizeof(aFrames16), &cFramesWritten)); + RTTESTI_CHECK(cFramesWritten == 1); } RTTESTI_CHECK(!AudioMixBufIsEmpty(&mb)); RTTESTI_CHECK(AudioMixBufFree(&mb) == 1); - RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_S2B(&mb, 1U)); - RTTESTI_CHECK(AudioMixBufUsed(&mb) == cToWrite + cSamplesWrittenAbs /* + last absolute write */); - - RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&mb, &samples16, sizeof(samples16), &cSamplesWritten)); - RTTESTI_CHECK(cSamplesWritten == 1); + RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_F2B(&mb, 1U)); + RTTESTI_CHECK(AudioMixBufUsed(&mb) == cToWrite + cFramesWrittenAbs /* + last absolute write */); + + RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&mb, &aFrames16, sizeof(aFrames16), &cFramesWritten)); + RTTESTI_CHECK(cFramesWritten == 1); RTTESTI_CHECK(AudioMixBufFree(&mb) == 0); - RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_S2B(&mb, 0U)); + RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_F2B(&mb, 0U)); RTTESTI_CHECK(AudioMixBufUsed(&mb) == cBufSize); /* Circular reads. */ - uint32_t cToRead = AudioMixBufSize(&mb) - cSamplesWrittenAbs - 1; + uint32_t cToRead = AudioMixBufSize(&mb) - cFramesWrittenAbs - 1; for (uint32_t i = 0; i < cToRead; i++) { - RTTESTI_CHECK_RC_OK(AudioMixBufReadCirc(&mb, &samples16, sizeof(samples16), &cSamplesRead)); - RTTESTI_CHECK(cSamplesRead == 1); - AudioMixBufFinish(&mb, cSamplesRead); + RTTESTI_CHECK_RC_OK(AudioMixBufReadCirc(&mb, &aFrames16, sizeof(aFrames16), &cFramesRead)); + RTTESTI_CHECK(cFramesRead == 1); + AudioMixBufFinish(&mb, cFramesRead); } RTTESTI_CHECK(!AudioMixBufIsEmpty(&mb)); - RTTESTI_CHECK(AudioMixBufFree(&mb) == AudioMixBufSize(&mb) - cSamplesWrittenAbs - 1); - RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_S2B(&mb, cBufSize - cSamplesWrittenAbs - 1)); + RTTESTI_CHECK(AudioMixBufFree(&mb) == AudioMixBufSize(&mb) - cFramesWrittenAbs - 1); + RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_F2B(&mb, cBufSize - cFramesWrittenAbs - 1)); RTTESTI_CHECK(AudioMixBufUsed(&mb) == cBufSize - cToRead); - RTTESTI_CHECK_RC_OK(AudioMixBufReadCirc(&mb, &samples16, sizeof(samples16), &cSamplesRead)); - RTTESTI_CHECK(cSamplesRead == 1); - AudioMixBufFinish(&mb, cSamplesRead); - RTTESTI_CHECK(AudioMixBufFree(&mb) == cBufSize - cSamplesWrittenAbs); - RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_S2B(&mb, cBufSize - cSamplesWrittenAbs)); - RTTESTI_CHECK(AudioMixBufUsed(&mb) == cSamplesWrittenAbs); + RTTESTI_CHECK_RC_OK(AudioMixBufReadCirc(&mb, &aFrames16, sizeof(aFrames16), &cFramesRead)); + RTTESTI_CHECK(cFramesRead == 1); + AudioMixBufFinish(&mb, cFramesRead); + RTTESTI_CHECK(AudioMixBufFree(&mb) == cBufSize - cFramesWrittenAbs); + RTTESTI_CHECK(AudioMixBufFreeBytes(&mb) == AUDIOMIXBUF_F2B(&mb, cBufSize - cFramesWrittenAbs)); + RTTESTI_CHECK(AudioMixBufUsed(&mb) == cFramesWrittenAbs); AudioMixBufDestroy(&mb); @@ -191,6 +190,6 @@ RTTESTI_CHECK(DrvAudioHlpPCMPropsAreValid(&cfg_c1)); - uint32_t cSamples = 16; - uint32_t cChildBufSize = RTRandU32Ex(cSamples /* Min */, 64 /* Max */); + uint32_t cFrames = 16; + uint32_t cChildBufSize = RTRandU32Ex(cFrames /* Min */, 64 /* Max */); PDMAUDIOMIXBUF child1; @@ -220,15 +219,15 @@ uint32_t cbBuf = _1K; char pvBuf[_1K]; - int16_t samples[32] = { 0xAA, 0xBB }; - uint32_t cSamplesRead, cSamplesWritten, cSamplesMixed; - - uint32_t cSamplesChild1 = cSamples; - uint32_t cSamplesChild2 = cSamples; + int16_t aFrames16[32] = { 0xAA, 0xBB }; + uint32_t cFramesRead, cFramesWritten, cFramesMixed; + + uint32_t cFramesChild1 = cFrames; + uint32_t cFramesChild2 = cFrames; uint32_t t = RTRandU32() % 32; RTTestPrintf(hTest, RTTESTLVL_DEBUG, - "cParentBufSize=%RU32, cChildBufSize=%RU32, %RU32 samples -> %RU32 iterations total\n", - cParentBufSize, cChildBufSize, cSamples, t); + "cParentBufSize=%RU32, cChildBufSize=%RU32, %RU32 frames -> %RU32 iterations total\n", + cParentBufSize, cChildBufSize, cFrames, t); /* @@ -248,12 +247,12 @@ { /* Child 1. */ - RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufWriteAt(&child1, 0, &samples, sizeof(samples), &cSamplesWritten)); - RTTESTI_CHECK_MSG_BREAK(cSamplesWritten == cSamplesChild1, ("Child1: Expected %RU32 written samples, got %RU32\n", cSamplesChild1, cSamplesWritten)); - RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufMixToParent(&child1, cSamplesWritten, &cSamplesMixed)); - - cChildrenSamplesMixedTotal += cSamplesMixed; - - RTTESTI_CHECK_MSG_BREAK(cSamplesWritten == cSamplesMixed, ("Child1: Expected %RU32 mixed samples, got %RU32\n", cSamplesWritten, cSamplesMixed)); - RTTESTI_CHECK_MSG_BREAK(AudioMixBufUsed(&child1) == 0, ("Child1: Expected %RU32 used samples, got %RU32\n", 0, AudioMixBufUsed(&child1))); + RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufWriteAt(&child1, 0, &aFrames16, sizeof(aFrames16), &cFramesWritten)); + RTTESTI_CHECK_MSG_BREAK(cFramesWritten == cFramesChild1, ("Child1: Expected %RU32 written frames, got %RU32\n", cFramesChild1, cFramesWritten)); + RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufMixToParent(&child1, cFramesWritten, &cFramesMixed)); + + cChildrenSamplesMixedTotal += cFramesMixed; + + RTTESTI_CHECK_MSG_BREAK(cFramesWritten == cFramesMixed, ("Child1: Expected %RU32 mixed frames, got %RU32\n", cFramesWritten, cFramesMixed)); + RTTESTI_CHECK_MSG_BREAK(AudioMixBufUsed(&child1) == 0, ("Child1: Expected %RU32 used frames, got %RU32\n", 0, AudioMixBufUsed(&child1))); } @@ -263,16 +262,16 @@ { /* Child 2. */ - RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufWriteAt(&child2, 0, &samples, sizeof(samples), &cSamplesWritten)); - RTTESTI_CHECK_MSG_BREAK(cSamplesWritten == cSamplesChild2, ("Child2: Expected %RU32 written samples, got %RU32\n", cSamplesChild2, cSamplesWritten)); - RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufMixToParent(&child2, cSamplesWritten, &cSamplesMixed)); - - cChildrenSamplesMixedTotal += cSamplesMixed; - - RTTESTI_CHECK_MSG_BREAK(cSamplesWritten == cSamplesMixed, ("Child2: Expected %RU32 mixed samples, got %RU32\n", cSamplesWritten, cSamplesMixed)); - RTTESTI_CHECK_MSG_BREAK(AudioMixBufUsed(&child2) == 0, ("Child2: Expected %RU32 used samples, got %RU32\n", 0, AudioMixBufUsed(&child2))); + RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufWriteAt(&child2, 0, &aFrames16, sizeof(aFrames16), &cFramesWritten)); + RTTESTI_CHECK_MSG_BREAK(cFramesWritten == cFramesChild2, ("Child2: Expected %RU32 written frames, got %RU32\n", cFramesChild2, cFramesWritten)); + RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufMixToParent(&child2, cFramesWritten, &cFramesMixed)); + + cChildrenSamplesMixedTotal += cFramesMixed; + + RTTESTI_CHECK_MSG_BREAK(cFramesWritten == cFramesMixed, ("Child2: Expected %RU32 mixed frames, got %RU32\n", cFramesWritten, cFramesMixed)); + RTTESTI_CHECK_MSG_BREAK(AudioMixBufUsed(&child2) == 0, ("Child2: Expected %RU32 used frames, got %RU32\n", 0, AudioMixBufUsed(&child2))); } /* - * Read out all samples from the parent buffer and also mark the just-read samples as finished + * Read out all frames from the parent buffer and also mark the just-read frames as finished * so that both connected children buffers can keep track of their stuff. */ @@ -280,12 +279,12 @@ while (cParentSamples) { - RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, pvBuf, cbBuf, &cSamplesRead)); - if (!cSamplesRead) + RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, pvBuf, cbBuf, &cFramesRead)); + if (!cFramesRead) break; - AudioMixBufFinish(&parent, cSamplesRead); - - RTTESTI_CHECK(cParentSamples >= cSamplesRead); - cParentSamples -= cSamplesRead; + AudioMixBufFinish(&parent, cFramesRead); + + RTTESTI_CHECK(cParentSamples >= cFramesRead); + cParentSamples -= cFramesRead; } @@ -331,6 +330,6 @@ * take shortcuts and performs conversion. Because conversion to double * the sample rate effectively inserts one additional sample between every - * two source samples, N source samples will be converted to N * 2 - 1 - * samples. However, the last source sample will be saved for later + * two source frames, N source frames will be converted to N * 2 - 1 + * frames. However, the last source sample will be saved for later * interpolation and not immediately output. */ @@ -353,7 +352,7 @@ RTTESTI_CHECK_RC_OK(AudioMixBufLinkTo(&child, &parent)); - /* 8-bit unsigned samples. Often used with SB16 device. */ - uint8_t samples[16] = { 0xAA, 0xBB, 0, 1, 43, 125, 126, 127, - 128, 129, 130, 131, 132, UINT8_MAX - 1, UINT8_MAX, 0 }; + /* 8-bit unsigned frames. Often used with SB16 device. */ + uint8_t aFrames8U[16] = { 0xAA, 0xBB, 0, 1, 43, 125, 126, 127, + 128, 129, 130, 131, 132, UINT8_MAX - 1, UINT8_MAX, 0 }; /* @@ -362,17 +361,17 @@ uint32_t cbBuf = 256; char achBuf[256]; - uint32_t cSamplesRead, cSamplesWritten, cSamplesMixed; - - uint32_t cSamplesChild = 16; - uint32_t cSamplesParent = cSamplesChild * 2 - 2; - uint32_t cSamplesTotalRead = 0; + uint32_t cFramesRead, cFramesWritten, cFramesMixed; + + uint32_t cFramesChild = 16; + uint32_t cFramesParent = cFramesChild * 2 - 2; + uint32_t cFramesTotalRead = 0; /**** 8-bit unsigned samples ****/ RTTestPrintf(hTest, RTTESTLVL_DEBUG, "Conversion test %uHz %uch 8-bit\n", cfg_c.uHz, cfg_c.cChannels); - RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &samples, sizeof(samples), &cSamplesWritten)); - RTTESTI_CHECK_MSG(cSamplesWritten == cSamplesChild, ("Child: Expected %RU32 written samples, got %RU32\n", cSamplesChild, cSamplesWritten)); - RTTESTI_CHECK_RC_OK(AudioMixBufMixToParent(&child, cSamplesWritten, &cSamplesMixed)); - uint32_t cSamples = AudioMixBufUsed(&parent); - RTTESTI_CHECK_MSG(AudioMixBufLive(&child) == cSamples, ("Child: Expected %RU32 mixed samples, got %RU32\n", AudioMixBufLive(&child), cSamples)); + RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &aFrames8U, sizeof(aFrames8U), &cFramesWritten)); + RTTESTI_CHECK_MSG(cFramesWritten == cFramesChild, ("Child: Expected %RU32 written frames, got %RU32\n", cFramesChild, cFramesWritten)); + RTTESTI_CHECK_RC_OK(AudioMixBufMixToParent(&child, cFramesWritten, &cFramesMixed)); + uint32_t cFrames = AudioMixBufUsed(&parent); + RTTESTI_CHECK_MSG(AudioMixBufLive(&child) == cFrames, ("Child: Expected %RU32 mixed frames, got %RU32\n", AudioMixBufLive(&child), cFrames)); RTTESTI_CHECK(AudioMixBufUsed(&parent) == AudioMixBufLive(&child)); @@ -380,19 +379,19 @@ for (;;) { - RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, achBuf, cbBuf, &cSamplesRead)); - if (!cSamplesRead) + RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, achBuf, cbBuf, &cFramesRead)); + if (!cFramesRead) break; - cSamplesTotalRead += cSamplesRead; - AudioMixBufFinish(&parent, cSamplesRead); - } - - RTTESTI_CHECK_MSG(cSamplesTotalRead == cSamplesParent, ("Parent: Expected %RU32 mixed samples, got %RU32\n", cSamplesParent, cSamplesTotalRead)); - - /* Check that the samples came out unharmed. Every other sample is interpolated and we ignore it. */ + cFramesTotalRead += cFramesRead; + AudioMixBufFinish(&parent, cFramesRead); + } + + RTTESTI_CHECK_MSG(cFramesTotalRead == cFramesParent, ("Parent: Expected %RU32 mixed frames, got %RU32\n", cFramesParent, cFramesTotalRead)); + + /* Check that the frames came out unharmed. Every other sample is interpolated and we ignore it. */ /* NB: This also checks that the default volume setting is 0dB attenuation. */ - uint8_t *pSrc8 = &samples[0]; + uint8_t *pSrc8 = &aFrames8U[0]; uint8_t *pDst8 = (uint8_t *)achBuf; - for (i = 0; i < cSamplesChild - 1; ++i) + for (i = 0; i < cFramesChild - 1; ++i) { RTTESTI_CHECK_MSG(*pSrc8 == *pDst8, ("index %u: Dst=%d, Src=%d\n", i, *pDst8, *pSrc8)); @@ -452,6 +451,6 @@ /* 16-bit signed. More or less exclusively used as output, and usually as input, too. */ - int16_t samples[16] = { 0xAA, 0xBB, INT16_MIN, INT16_MIN + 1, INT16_MIN / 2, -3, -2, -1, - 0, 1, 2, 3, INT16_MAX / 2, INT16_MAX - 1, INT16_MAX, 0 }; + int16_t aFrames16S[16] = { 0xAA, 0xBB, INT16_MIN, INT16_MIN + 1, INT16_MIN / 2, -3, -2, -1, + 0, 1, 2, 3, INT16_MAX / 2, INT16_MAX - 1, INT16_MAX, 0 }; /* @@ -460,17 +459,17 @@ uint32_t cbBuf = 256; char achBuf[256]; - uint32_t cSamplesRead, cSamplesWritten, cSamplesMixed; - - uint32_t cSamplesChild = 16; - uint32_t cSamplesParent = cSamplesChild * 2 - 2; - uint32_t cSamplesTotalRead = 0; + uint32_t cFramesRead, cFramesWritten, cFramesMixed; + + uint32_t cFramesChild = 16; + uint32_t cFramesParent = cFramesChild * 2 - 2; + uint32_t cFramesTotalRead = 0; /**** 16-bit signed samples ****/ RTTestPrintf(hTest, RTTESTLVL_DEBUG, "Conversion test %uHz %uch 16-bit\n", cfg_c.uHz, cfg_c.cChannels); - RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &samples, sizeof(samples), &cSamplesWritten)); - RTTESTI_CHECK_MSG(cSamplesWritten == cSamplesChild, ("Child: Expected %RU32 written samples, got %RU32\n", cSamplesChild, cSamplesWritten)); - RTTESTI_CHECK_RC_OK(AudioMixBufMixToParent(&child, cSamplesWritten, &cSamplesMixed)); - uint32_t cSamples = AudioMixBufUsed(&parent); - RTTESTI_CHECK_MSG(AudioMixBufLive(&child) == cSamples, ("Child: Expected %RU32 mixed samples, got %RU32\n", AudioMixBufLive(&child), cSamples)); + RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &aFrames16S, sizeof(aFrames16S), &cFramesWritten)); + RTTESTI_CHECK_MSG(cFramesWritten == cFramesChild, ("Child: Expected %RU32 written frames, got %RU32\n", cFramesChild, cFramesWritten)); + RTTESTI_CHECK_RC_OK(AudioMixBufMixToParent(&child, cFramesWritten, &cFramesMixed)); + uint32_t cFrames = AudioMixBufUsed(&parent); + RTTESTI_CHECK_MSG(AudioMixBufLive(&child) == cFrames, ("Child: Expected %RU32 mixed frames, got %RU32\n", AudioMixBufLive(&child), cFrames)); RTTESTI_CHECK(AudioMixBufUsed(&parent) == AudioMixBufLive(&child)); @@ -478,18 +477,18 @@ for (;;) { - RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, achBuf, cbBuf, &cSamplesRead)); - if (!cSamplesRead) + RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, achBuf, cbBuf, &cFramesRead)); + if (!cFramesRead) break; - cSamplesTotalRead += cSamplesRead; - AudioMixBufFinish(&parent, cSamplesRead); - } - RTTESTI_CHECK_MSG(cSamplesTotalRead == cSamplesParent, ("Parent: Expected %RU32 mixed samples, got %RU32\n", cSamplesParent, cSamplesTotalRead)); - - /* Check that the samples came out unharmed. Every other sample is interpolated and we ignore it. */ + cFramesTotalRead += cFramesRead; + AudioMixBufFinish(&parent, cFramesRead); + } + RTTESTI_CHECK_MSG(cFramesTotalRead == cFramesParent, ("Parent: Expected %RU32 mixed frames, got %RU32\n", cFramesParent, cFramesTotalRead)); + + /* Check that the frames came out unharmed. Every other sample is interpolated and we ignore it. */ /* NB: This also checks that the default volume setting is 0dB attenuation. */ - int16_t *pSrc16 = &samples[0]; + int16_t *pSrc16 = &aFrames16S[0]; int16_t *pDst16 = (int16_t *)achBuf; - for (i = 0; i < cSamplesChild - 1; ++i) + for (i = 0; i < cFramesChild - 1; ++i) { RTTESTI_CHECK_MSG(*pSrc16 == *pDst16, ("index %u: Dst=%d, Src=%d\n", i, *pDst16, *pSrc16)); @@ -538,6 +537,6 @@ /* A few 16-bit signed samples. */ - int16_t samples[16] = { INT16_MIN, INT16_MIN + 1, -128, -64, -4, -1, 0, 1, - 2, 255, 256, INT16_MAX / 2, INT16_MAX - 2, INT16_MAX - 1, INT16_MAX, 0 }; + int16_t aFrames16S[16] = { INT16_MIN, INT16_MIN + 1, -128, -64, -4, -1, 0, 1, + 2, 255, 256, INT16_MAX / 2, INT16_MAX - 2, INT16_MAX - 1, INT16_MAX, 0 }; /* @@ -546,9 +545,9 @@ uint32_t cbBuf = 256; char achBuf[256]; - uint32_t cSamplesRead, cSamplesWritten, cSamplesMixed; - - uint32_t cSamplesChild = 8; - uint32_t cSamplesParent = cSamplesChild; - uint32_t cSamplesTotalRead; + uint32_t cFramesRead, cFramesWritten, cFramesMixed; + + uint32_t cFramesChild = 8; + uint32_t cFramesParent = cFramesChild; + uint32_t cFramesTotalRead; int16_t *pSrc16; int16_t *pDst16; @@ -561,24 +560,24 @@ AudioMixBufSetVolume(&child, &vol); - RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &samples, sizeof(samples), &cSamplesWritten)); - RTTESTI_CHECK_MSG(cSamplesWritten == cSamplesChild, ("Child: Expected %RU32 written samples, got %RU32\n", cSamplesChild, cSamplesWritten)); - RTTESTI_CHECK_RC_OK(AudioMixBufMixToParent(&child, cSamplesWritten, &cSamplesMixed)); - - cSamplesTotalRead = 0; + RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &aFrames16S, sizeof(aFrames16S), &cFramesWritten)); + RTTESTI_CHECK_MSG(cFramesWritten == cFramesChild, ("Child: Expected %RU32 written frames, got %RU32\n", cFramesChild, cFramesWritten)); + RTTESTI_CHECK_RC_OK(AudioMixBufMixToParent(&child, cFramesWritten, &cFramesMixed)); + + cFramesTotalRead = 0; for (;;) { - RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, achBuf, cbBuf, &cSamplesRead)); - if (!cSamplesRead) + RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, achBuf, cbBuf, &cFramesRead)); + if (!cFramesRead) break; - cSamplesTotalRead += cSamplesRead; - AudioMixBufFinish(&parent, cSamplesRead); - } - RTTESTI_CHECK_MSG(cSamplesTotalRead == cSamplesParent, ("Parent: Expected %RU32 mixed samples, got %RU32\n", cSamplesParent, cSamplesTotalRead)); - - /* Check that at 0dB the samples came out unharmed. */ - pSrc16 = &samples[0]; + cFramesTotalRead += cFramesRead; + AudioMixBufFinish(&parent, cFramesRead); + } + RTTESTI_CHECK_MSG(cFramesTotalRead == cFramesParent, ("Parent: Expected %RU32 mixed frames, got %RU32\n", cFramesParent, cFramesTotalRead)); + + /* Check that at 0dB the frames came out unharmed. */ + pSrc16 = &aFrames16S[0]; pDst16 = (int16_t *)achBuf; - for (i = 0; i < cSamplesParent * 2 /* stereo */; ++i) + for (i = 0; i < cFramesParent * 2 /* stereo */; ++i) { RTTESTI_CHECK_MSG(*pSrc16 == *pDst16, ("index %u: Dst=%d, Src=%d\n", i, *pDst16, *pSrc16)); @@ -592,24 +591,24 @@ AudioMixBufSetVolume(&child, &vol); - RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &samples, sizeof(samples), &cSamplesWritten)); - RTTESTI_CHECK_MSG(cSamplesWritten == cSamplesChild, ("Child: Expected %RU32 written samples, got %RU32\n", cSamplesChild, cSamplesWritten)); - RTTESTI_CHECK_RC_OK(AudioMixBufMixToParent(&child, cSamplesWritten, &cSamplesMixed)); - - cSamplesTotalRead = 0; + RTTESTI_CHECK_RC_OK(AudioMixBufWriteCirc(&child, &aFrames16S, sizeof(aFrames16S), &cFramesWritten)); + RTTESTI_CHECK_MSG(cFramesWritten == cFramesChild, ("Child: Expected %RU32 written frames, got %RU32\n", cFramesChild, cFramesWritten)); + RTTESTI_CHECK_RC_OK(AudioMixBufMixToParent(&child, cFramesWritten, &cFramesMixed)); + + cFramesTotalRead = 0; for (;;) { - RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, achBuf, cbBuf, &cSamplesRead)); - if (!cSamplesRead) + RTTESTI_CHECK_RC_OK_BREAK(AudioMixBufReadCirc(&parent, achBuf, cbBuf, &cFramesRead)); + if (!cFramesRead) break; - cSamplesTotalRead += cSamplesRead; - AudioMixBufFinish(&parent, cSamplesRead); - } - RTTESTI_CHECK_MSG(cSamplesTotalRead == cSamplesParent, ("Parent: Expected %RU32 mixed samples, got %RU32\n", cSamplesParent, cSamplesTotalRead)); + cFramesTotalRead += cFramesRead; + AudioMixBufFinish(&parent, cFramesRead); + } + RTTESTI_CHECK_MSG(cFramesTotalRead == cFramesParent, ("Parent: Expected %RU32 mixed frames, got %RU32\n", cFramesParent, cFramesTotalRead)); /* Check that at -6dB the sample values are halved. */ - pSrc16 = &samples[0]; + pSrc16 = &aFrames16S[0]; pDst16 = (int16_t *)achBuf; - for (i = 0; i < cSamplesParent * 2 /* stereo */; ++i) + for (i = 0; i < cFramesParent * 2 /* stereo */; ++i) { /* Watch out! For negative values, x >> 1 is not the same as x / 2. */ Index: /trunk/src/VBox/Main/src-client/DrvAudioVRDE.cpp =================================================================== --- /trunk/src/VBox/Main/src-client/DrvAudioVRDE.cpp (revision 68131) +++ /trunk/src/VBox/Main/src-client/DrvAudioVRDE.cpp (revision 68132) @@ -86,5 +86,5 @@ /* Sanity. */ -AssertCompileSize(PDMAUDIOSAMPLE, sizeof(int64_t) * 2 /* st_sample_t using by VRDP server */); +AssertCompileSize(PDMAUDIOFRAME, sizeof(int64_t) * 2 /* st_sample_t using by VRDP server */); static int vrdeCreateStreamIn(PVRDESTREAM pStreamVRDE, PPDMAUDIOSTREAMCFG pCfgReq, PPDMAUDIOSTREAMCFG pCfgAcq) @@ -105,5 +105,5 @@ */ pCfgAcq->enmLayout = PDMAUDIOSTREAMLAYOUT_RAW; - pCfgAcq->cSampleBufferHint = pStreamVRDE->In.csMax; + pCfgAcq->cFrameBufferHint = pStreamVRDE->In.csMax; } } @@ -127,5 +127,5 @@ */ pCfgAcq->enmLayout = PDMAUDIOSTREAMLAYOUT_RAW; - pCfgAcq->cSampleBufferHint = _4K; /** @todo Make this configurable. */ + pCfgAcq->cFrameBufferHint = _4K; /** @todo Make this configurable. */ } @@ -292,5 +292,5 @@ int rc = VINF_SUCCESS; - PPDMAUDIOSAMPLE paSampleBuf = (PPDMAUDIOSAMPLE)pvBuf; + PPDMAUDIOFRAME paSampleBuf = (PPDMAUDIOFRAME)pvBuf; AssertPtr(paSampleBuf); @@ -501,5 +501,5 @@ /* Return samples instead of bytes here * (since we specified PDMAUDIOSTREAMLAYOUT_RAW as the audio data layout). */ - return (uint32_t)PDMAUDIOSTREAMCFG_B2S(pStreamVRDE->pCfg, RTCircBufUsed(pStreamVRDE->In.pCircBuf)); + return (uint32_t)PDMAUDIOSTREAMCFG_B2F(pStreamVRDE->pCfg, RTCircBufUsed(pStreamVRDE->In.pCircBuf)); } Index: /trunk/src/VBox/Main/src-client/DrvAudioVideoRec.cpp =================================================================== --- /trunk/src/VBox/Main/src-client/DrvAudioVideoRec.cpp (revision 68131) +++ /trunk/src/VBox/Main/src-client/DrvAudioVideoRec.cpp (revision 68132) @@ -375,5 +375,5 @@ if (pCfgAcq) - pCfgAcq->cSampleBufferHint = 0; + pCfgAcq->cFrameBufferHint = 0; LogRel2(("VideoRec: Support for surround audio not implemented yet\n")); @@ -409,5 +409,5 @@ pCfgAcq->Props.uHz = pSink->Codec.Parms.uHz; pCfgAcq->Props.cShift = PDMAUDIOPCMPROPS_MAKE_SHIFT_PARMS(pCfgAcq->Props.cBits, pCfgAcq->Props.cChannels); - pCfgAcq->cSampleBufferHint = _4K; /** @todo Make this configurable. */ + pCfgAcq->cFrameBufferHint = _4K; /** @todo Make this configurable. */ } } @@ -586,5 +586,5 @@ const uint32_t csFrame = pSink->Codec.Opus.csFrame; - const uint32_t cbFrame = PDMAUDIOSTREAMCFG_S2B(pStreamAV->pCfg, csFrame); + const uint32_t cbFrame = PDMAUDIOSTREAMCFG_F2B(pStreamAV->pCfg, csFrame); while (RTCircBufUsed(pCircBuf) >= cbFrame)