/**************************************************************************** * audio/pcm_decode.c * * Licensed to the Apache Software Foundation (ASF) under one or more * contributor license agreements. See the NOTICE file distributed with * this work for additional information regarding copyright ownership. The * ASF licenses this file to you under the Apache License, Version 2.0 (the * "License"); you may not use this file except in compliance with the * License. You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, WITHOUT * WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. See the * License for the specific language governing permissions and limitations * under the License. * ****************************************************************************/ /**************************************************************************** * Included Files ****************************************************************************/ #include #include #include #include #include #include #include #include #include #include #include #include #include #if defined(CONFIG_AUDIO) && defined(CONFIG_AUDIO_FORMAT_PCM) /**************************************************************************** * Pre-processor Definitions ****************************************************************************/ /* Configuration ************************************************************/ #define CONFIG_PCM_DEBUG 1 /* For now */ /* Often defined and re-defined macros */ #ifndef MIN # define MIN(a,b) (((a) < (b)) ? (a) : (b)) #endif #ifndef MAX # define MAX(a,b) (((a) > (b)) ? (a) : (b)) #endif /**************************************************************************** * Private Types ****************************************************************************/ /* This structure describes the internal state of the PCM decoder */ struct pcm_decode_s { /* This is is our our appearance to the outside world. This *MUST* be the * first element of the structure so that we can freely cast between types * struct audio_lowerhalf and struct pcm_decode_s. */ struct audio_lowerhalf_s export; /* These are our operations that intervene between the player application * and the lower level driver. Unlike the ops in the struct * audio_lowerhalf_s, these are writeable because we need to customize a * few of the methods based upon what is supported by the lower level * driver. */ struct audio_ops_s ops; /* This is the contained, low-level DAC-type device and will receive the * decoded PCM audio data. */ FAR struct audio_lowerhalf_s *lower; /* Session returned from the lower level driver */ #ifdef CONFIG_AUDIO_MULTI_SESSION FAR void *session; #endif /* These are values extracted from WAV file header */ uint32_t samprate; /* 8000, 44100, ... */ uint32_t byterate; /* samprate * nchannels * bpsamp / 8 */ uint8_t align; /* nchannels * bpsamp / 8 */ uint8_t bpsamp; /* Bits per sample: 8 bits = 8, 16 bits = 16 */ uint8_t nchannels; /* Mono=1, Stereo=2 */ bool streaming; /* Streaming PCM data chunk */ #ifndef CONFIG_AUDIO_EXCLUDE_FFORWARD /* Fast forward support */ uint8_t subsample; /* Fast forward rate: See AUDIO_SUBSAMPLE_* defns */ uint8_t skip; /* Number of sample bytes to be skipped */ uint8_t npartial; /* Size of the partially copied sample */ #endif }; /**************************************************************************** * Private Function Prototypes ****************************************************************************/ /* Helper functions *********************************************************/ #ifdef CONFIG_PCM_DEBUG static void pcm_dump(FAR const struct wav_header_s *wav); #else # define pcm_dump(w) #endif #ifdef CONFIG_ENDIAN_BIG static uint16_t pcm_leuint16(uint16_t value); static uint16_t pcm_leuint32(uint32_t value); #else # define pcm_leuint16(v) (v) # define pcm_leuint32(v) (v) #endif #ifndef CONFIG_AUDIO_FORMAT_RAW static inline bool pcm_validwav(FAR const struct wav_header_s *wav); static bool pcm_parsewav(FAR struct pcm_decode_s *priv, uint8_t *data); #endif #ifndef CONFIG_AUDIO_EXCLUDE_FFORWARD static void pcm_subsample_configure(FAR struct pcm_decode_s *priv, uint8_t subsample); static void pcm_subsample(FAR struct pcm_decode_s *priv, FAR struct ap_buffer_s *apb); #endif /* struct audio_lowerhalf_s methods *****************************************/ static int pcm_getcaps(FAR struct audio_lowerhalf_s *dev, int type, FAR struct audio_caps_s *caps); #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_configure(FAR struct audio_lowerhalf_s *dev, FAR void *session, FAR const struct audio_caps_s *caps); #else static int pcm_configure(FAR struct audio_lowerhalf_s *dev, FAR const struct audio_caps_s *caps); #endif static int pcm_shutdown(FAR struct audio_lowerhalf_s *dev); #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_start(FAR struct audio_lowerhalf_s *dev, FAR void *session); #else static int pcm_start(FAR struct audio_lowerhalf_s *dev); #endif #ifndef CONFIG_AUDIO_EXCLUDE_STOP #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_stop(FAR struct audio_lowerhalf_s *dev, FAR void *session); #else static int pcm_stop(FAR struct audio_lowerhalf_s *dev); #endif #endif #ifndef CONFIG_AUDIO_EXCLUDE_PAUSE_RESUME #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_pause(FAR struct audio_lowerhalf_s *dev, FAR void *session); #else static int pcm_pause(FAR struct audio_lowerhalf_s *dev); #endif #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_resume(FAR struct audio_lowerhalf_s *dev, FAR void *session); #else static int pcm_resume(FAR struct audio_lowerhalf_s *dev); #endif #endif static int pcm_allocbuffer(FAR struct audio_lowerhalf_s *dev, FAR struct audio_buf_desc_s *apb); static int pcm_freebuffer(FAR struct audio_lowerhalf_s *dev, FAR struct audio_buf_desc_s *apb); static int pcm_enqueuebuffer(FAR struct audio_lowerhalf_s *dev, FAR struct ap_buffer_s *apb); static int pcm_cancelbuffer(FAR struct audio_lowerhalf_s *dev, FAR struct ap_buffer_s *apb); static int pcm_ioctl(FAR struct audio_lowerhalf_s *dev, int cmd, unsigned long arg); #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_reserve(FAR struct audio_lowerhalf_s *dev, FAR void **session); #else static int pcm_reserve(FAR struct audio_lowerhalf_s *dev); #endif #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_release(FAR struct audio_lowerhalf_s *dev, FAR void *session); #else static int pcm_release(FAR struct audio_lowerhalf_s *dev); #endif /* Audio callback */ #ifdef CONFIG_AUDIO_MULTI_SESSION static void pcm_callback(FAR void *arg, uint16_t reason, FAR struct ap_buffer_s *apb, uint16_t status, FAR void *session); #else static void pcm_callback(FAR void *arg, uint16_t reason, FAR struct ap_buffer_s *apb, uint16_t status); #endif /**************************************************************************** * Private Data ****************************************************************************/ /**************************************************************************** * Public Data ****************************************************************************/ /**************************************************************************** * Private Functions ****************************************************************************/ /**************************************************************************** * Name: pcm_dump * * Description: * Dump a WAV file header. * ****************************************************************************/ #ifdef CONFIG_PCM_DEBUG static void pcm_dump(FAR const struct wav_header_s *wav) { _info("Wave file header\n"); _info(" Header Chunk:\n"); _info(" Chunk ID: 0x%08" PRIx32 "\n", wav->hdr.chunkid); _info(" Chunk Size: %" PRIu32 "\n", wav->hdr.chunklen); _info(" Format: 0x%08" PRIx32 "\n", wav->hdr.format); _info(" Format Chunk:\n"); _info(" Chunk ID: 0x%08" PRIx32 "\n", wav->fmt.chunkid); _info(" Chunk Size: %" PRIu32 "\n", wav->fmt.chunklen); _info(" Audio Format: 0x%04x\n", wav->fmt.format); _info(" Num. Channels: %d\n", wav->fmt.nchannels); _info(" Sample Rate: %" PRIu32 "\n", wav->fmt.samprate); _info(" Byte Rate: %" PRIu32 "\n", wav->fmt.byterate); _info(" Block Align: %d\n", wav->fmt.align); _info(" Bits Per Sample: %d\n", wav->fmt.bpsamp); _info(" Data Chunk:\n"); _info(" Chunk ID: 0x%08" PRIx32 "\n", wav->data.chunkid); _info(" Chunk Size: %" PRIu32 "\n", wav->data.chunklen); } #endif /**************************************************************************** * Name: pcm_leuint16 * * Description: * Get a 16-bit value stored in little endian order for a big-endian * machine. * ****************************************************************************/ #ifdef CONFIG_ENDIAN_BIG static uint16_t pcm_leuint16(uint16_t value) { return (((value & 0x00ff) << 8) | ((value >> 8) & 0x00ff)); } #endif /**************************************************************************** * Name: pcm_leuint16 * * Description: * Get a 16-bit value stored in little endian order for a big-endian * machine. * ****************************************************************************/ #ifdef CONFIG_ENDIAN_BIG static uint16_t pcm_leuint32(uint32_t value) { return (((value & 0x000000ff) << 24) | ((value & 0x0000ff00) << 8) | ((value & 0x00ff0000) >> 8) | ((value & 0xff000000) >> 24)); } #endif /**************************************************************************** * Name: pcm_validwav * * Description: * Return true if this is a valid WAV file header * ****************************************************************************/ #ifndef CONFIG_AUDIO_FORMAT_RAW static inline bool pcm_validwav(FAR const struct wav_header_s *wav) { return (wav->hdr.chunkid == WAV_HDR_CHUNKID && wav->hdr.format == WAV_HDR_FORMAT && wav->fmt.chunkid == WAV_FMT_CHUNKID && wav->fmt.chunklen == WAV_FMT_CHUNKLEN && wav->fmt.format == WAV_FMT_FORMAT && wav->fmt.nchannels < 256 && wav->fmt.align < 256 && wav->fmt.bpsamp < 256 && wav->data.chunkid == WAV_DATA_CHUNKID); } /**************************************************************************** * Name: pcm_parsewav * * Description: * Parse and verify the WAV file header. A WAV file is a particular * packaging of an audio file; on PCM encoded WAV files are accepted by * this driver. * ****************************************************************************/ static bool pcm_parsewav(FAR struct pcm_decode_s *priv, uint8_t *data) { FAR const struct wav_header_s *wav = (FAR const struct wav_header_s *)data; struct wav_header_s localwav; bool ret; /* Transfer the purported WAV file header into our stack storage, * correcting for endian issues as needed. */ localwav.hdr.chunkid = pcm_leuint32(wav->hdr.chunkid); localwav.hdr.chunklen = pcm_leuint32(wav->hdr.chunklen); localwav.hdr.format = pcm_leuint32(wav->hdr.format); localwav.fmt.chunkid = pcm_leuint32(wav->fmt.chunkid); localwav.fmt.chunklen = pcm_leuint32(wav->fmt.chunklen); localwav.fmt.format = pcm_leuint16(wav->fmt.format); localwav.fmt.nchannels = pcm_leuint16(wav->fmt.nchannels); localwav.fmt.samprate = pcm_leuint32(wav->fmt.samprate); localwav.fmt.byterate = pcm_leuint32(wav->fmt.byterate); localwav.fmt.align = pcm_leuint16(wav->fmt.align); localwav.fmt.bpsamp = pcm_leuint16(wav->fmt.bpsamp); localwav.data.chunkid = pcm_leuint32(wav->data.chunkid); localwav.data.chunklen = pcm_leuint32(wav->data.chunklen); /* Dump the converted wave header information */ pcm_dump(&localwav); /* Check if the file is a valid PCM WAV header */ ret = pcm_validwav(&localwav); if (ret) { /* Yes... pick off the relevant format values and save then in the * device structure. */ priv->samprate = localwav.fmt.samprate; /* 8000, 44100, ... */ priv->byterate = localwav.fmt.byterate; /* samprate * nchannels * bpsamp / 8 */ priv->align = localwav.fmt.align; /* nchannels * bpsamp / 8 */ priv->bpsamp = localwav.fmt.bpsamp; /* Bits per sample: 8 bits = 8, 16 bits = 16 */ priv->nchannels = localwav.fmt.nchannels; /* Mono=1, Stereo=2 */ #ifndef CONFIG_AUDIO_EXCLUDE_FFORWARD /* We are going to subsample, there then are some restrictions on the * number of channels and sample sizes that we can handle. */ if (priv->bpsamp != 8 && priv->bpsamp != 16) { auderr("ERROR: %d bits per sample are not supported in this " "mode\n", priv->bpsamp); return -EINVAL; } if (priv->nchannels != 1 && priv->nchannels != 2) { auderr("ERROR: %d channels are not supported in this mode\n", priv->nchannels); return -EINVAL; } DEBUGASSERT(priv->align == priv->nchannels * priv->bpsamp / 8); #endif } /* And return true if the file is a valid WAV header file */ return ret; } #endif /**************************************************************************** * Name: pcm_subsample_configure * * Description: * Configure to perform sub-sampling (or not) on the following audio * buffers. * ****************************************************************************/ #ifndef CONFIG_AUDIO_EXCLUDE_FFORWARD static void pcm_subsample_configure(FAR struct pcm_decode_s *priv, uint8_t subsample) { audinfo("subsample: %d\n", subsample); /* Three possibilities: * * 1. We were playing normally and we have been requested to begin fast * forwarding. */ if (priv->subsample == AUDIO_SUBSAMPLE_NONE) { /* Ignore request to stop fast forwarding if we are already playing * normally. */ if (subsample != AUDIO_SUBSAMPLE_NONE) { audinfo("Start sub-sampling\n"); /* Save the current subsample setting. Subsampling will begin on * then next audio buffer that we receive. */ priv->npartial = 0; priv->skip = 0; priv->subsample = subsample; } } /* 2. We were already fast forwarding and we have been asked to return to * normal play. */ else if (subsample == AUDIO_SUBSAMPLE_NONE) { audinfo("Stop sub-sampling\n"); /* Indicate that we are in normal play mode. This will take effect * when the next audio buffer is received. */ priv->npartial = 0; priv->skip = 0; priv->subsample = AUDIO_SUBSAMPLE_NONE; } /* 3. Were already fast forwarding and we have been asked to change the * sub-sampling rate. */ else if (priv->subsample != subsample) { /* Just save the new subsample setting. It will take effect on the * next audio buffer that we receive. */ priv->subsample = subsample; } } #endif /**************************************************************************** * Name: pcm_subsample * * Description: * Given a newly received audio buffer, perform sub-sampling in-place in * the audio buffer. Since the sub-sampled data will always be smaller * than the original buffer, no additional buffering should be necessary. * ****************************************************************************/ #ifndef CONFIG_AUDIO_EXCLUDE_FFORWARD static void pcm_subsample(FAR struct pcm_decode_s *priv, FAR struct ap_buffer_s *apb) { FAR const uint8_t *src; FAR uint8_t *dest; unsigned int destsize; unsigned int srcsize; unsigned int skipsize; unsigned int copysize; unsigned int i; /* Are we sub-sampling? */ if (priv->subsample == AUDIO_SUBSAMPLE_NONE) { /* No.. do nothing to the buffer */ return; } /* Yes.. we will need to subsample the newly received buffer in-place by * copying from the upper end of the buffer to the lower end. */ src = &apb->samp[apb->curbyte]; dest = apb->samp; srcsize = apb->nbytes - apb->curbyte; destsize = apb->nmaxbytes; /* This is the number of bytes that we need to skip between samples */ skipsize = priv->align * (priv->subsample - 1); /* Let's deal with any partial samples from the last buffer */ if (priv->npartial > 0) { /* Let's get an impossible corner case out of the way. What if we * received a tiny audio buffer. So small, that it (plus any previous * sample) is smaller than one sample. */ if (priv->npartial + srcsize < priv->align) { /* Update the partial sample size and return the unmodified * buffer. */ priv->npartial += srcsize; return; } /* We do at least have enough to complete the sample. If this data * does not resides at the correct position at the from of the audio * buffer, then we will need to copy it. */ copysize = priv->align - priv->npartial; if (apb->curbyte > 0) { /* We have to copy down */ for (i = 0; i < copysize; i++) { *dest++ = *src++; } } else { /* If the data is already position at the beginning of the audio * buffer, then just increment the buffer pointers around the * data. */ src += copysize; dest += copysize; } /* Update the number of bytes in the working buffer and reset the * skip value */ priv->npartial = 0; srcsize -= copysize; destsize -= copysize; priv->skip = skipsize; } /* Now loop until either the entire audio buffer has been sub-sampling. * This copy in place works because we know that the sub-sampled data * will always be smaller than the original data. */ while (srcsize > 0) { /* Do we need to skip ahead in the buffer? */ if (priv->skip > 0) { /* How much can we skip in this buffer? Depends on the smaller * of (1) the number of bytes that we need to skip and (2) the * number of bytes available in the newly received audio buffer. */ copysize = MIN(priv->skip, srcsize); priv->skip -= copysize; src += copysize; srcsize -= copysize; } /* Copy the sample from the audio buffer into the working buffer. */ else { /* Do we have space for the whole sample? */ if (srcsize < priv->align) { /* No.. this is a partial copy */ copysize = srcsize; priv->npartial = srcsize; } else { /* Copy the whole sample and re-arm the skip size */ copysize = priv->align; priv->skip = skipsize; } /* Now copy the sample from the end of audio buffer * to the beginning. */ for (i = 0; i < copysize; i++) { *dest++ = *src++; } /* Updates bytes available in the source buffer and bytes * remaining in the destination buffer. */ srcsize -= copysize; destsize -= copysize; } } /* Update the size and offset data in the audio buffer */ apb->curbyte = 0; apb->nbytes = apb->nmaxbytes - destsize; } #endif /**************************************************************************** * Name: pcm_getcaps * * Description: * This method is called to retrieve the lower-half device capabilities. * It will be called with device type AUDIO_TYPE_QUERY to request the * overall capabilities, such as to determine the types of devices * supported audio formats supported, etc. * Then it may be called once or more with reported supported device types * to determine the specific capabilities of that device type * (such as MP3 encoder, WMA encoder, PCM output, etc.). * ****************************************************************************/ static int pcm_getcaps(FAR struct audio_lowerhalf_s *dev, int type, FAR struct audio_caps_s *caps) { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; int ret; DEBUGASSERT(priv); /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->getcaps); /* Get the capabilities of the lower-level driver */ ret = lower->ops->getcaps(lower, type, caps); if (ret < 0) { auderr("ERROR: Lower getcaps() failed: %d\n", ret); return ret; } /* Modify the capabilities reported by the lower driver: * PCM is the only supported format that we will report, * regardless of what the lower driver reported. */ if (caps->ac_subtype == AUDIO_TYPE_QUERY) { caps->ac_format.hw = (1 << (AUDIO_FMT_PCM - 1)); } return caps->ac_len; } /**************************************************************************** * Name: pcm_configure * * Description: * This method is called to bind the lower-level driver to the upper-level * driver and to configure the driver for a specific mode of * operation defined by the parameters selected in supplied device caps * structure. The lower-level device should perform any initialization * needed to prepare for operations in the specified mode. It should not, * however, process any audio data until the start method is called. * ****************************************************************************/ #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_configure(FAR struct audio_lowerhalf_s *dev, FAR void *session, FAR const struct audio_caps_s *caps) #else static int pcm_configure(FAR struct audio_lowerhalf_s *dev, FAR const struct audio_caps_s *caps) #endif { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); #ifndef CONFIG_AUDIO_EXCLUDE_FFORWARD /* Pick off commands to perform sub-sampling. Those are done by this * decoder. All of configuration settings are handled by the lower level * audio driver. */ if (caps->ac_type == AUDIO_TYPE_PROCESSING && caps->ac_format.hw == AUDIO_PU_SUBSAMPLE_FORWARD) { /* Configure sub-sampling and return to avoid forwarding the * configuration to the lower level * driver. */ pcm_subsample_configure(priv, caps->ac_controls.b[0]); return OK; } #endif /* Defer all other operations to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->configure); audinfo("Defer to lower configure\n"); #ifdef CONFIG_AUDIO_MULTI_SESSION return lower->ops->configure(lower, session, caps); #else return lower->ops->configure(lower, caps); #endif } /**************************************************************************** * Name: pcm_shutdown * * Description: * This method is called when the driver is closed. The lower half driver * should stop processing audio data, including terminating any active * output generation. It should also disable the audio hardware and put * it into the lowest possible power usage state. * * Any enqueued Audio Pipeline Buffers that have not been * processed / dequeued should be dequeued by this function. * ****************************************************************************/ static int pcm_shutdown(FAR struct audio_lowerhalf_s *dev) { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* We are no longer streaming audio */ priv->streaming = false; /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->start); audinfo("Defer to lower shutdown\n"); return lower->ops->shutdown(lower); } /**************************************************************************** * Name: pcm_start * * Description: * Start audio streaming in the configured mode. * For input and synthesis devices, this means it should begin sending * streaming audio data. For output or processing type device, it means * it should begin processing of any enqueued Audio Pipeline Buffers. * ****************************************************************************/ #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_start(FAR struct audio_lowerhalf_s *dev, FAR void *session) #else static int pcm_start(FAR struct audio_lowerhalf_s *dev) #endif { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->start); audinfo("Defer to lower start\n"); #ifdef CONFIG_AUDIO_MULTI_SESSION return lower->ops->start(lower, session); #else return lower->ops->start(lower); #endif } /**************************************************************************** * Name: pcm_stop * * Description: * Stop audio streaming and/or processing of enqueued Audio Pipeline * Buffers * ****************************************************************************/ #ifndef CONFIG_AUDIO_EXCLUDE_STOP #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_stop(FAR struct audio_lowerhalf_s *dev, FAR void *session) #else static int pcm_stop(FAR struct audio_lowerhalf_s *dev) #endif { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* We are no longer streaming */ priv->streaming = false; /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->stop); audinfo("Defer to lower stop\n"); #ifdef CONFIG_AUDIO_MULTI_SESSION return lower->ops->stop(lower, session); #else return lower->ops->stop(lower); #endif } #endif /* CONFIG_AUDIO_EXCLUDE_STOP */ /**************************************************************************** * Name: pcm_pause * * Description: * Pause the audio stream. * Should keep current playback context active in case a resume is issued. * Could be called and then followed by a stop. * ****************************************************************************/ #ifndef CONFIG_AUDIO_EXCLUDE_PAUSE_RESUME #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_pause(FAR struct audio_lowerhalf_s *dev, FAR void *session) #else static int pcm_pause(FAR struct audio_lowerhalf_s *dev) #endif { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->pause); audinfo("Defer to lower pause\n"); #ifdef CONFIG_AUDIO_MULTI_SESSION return lower->ops->pause(lower, session); #else return lower->ops->pause(lower); #endif } /**************************************************************************** * Name: pcm_resume * * Description: * Resumes audio streaming after a pause. * ****************************************************************************/ #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_resume(FAR struct audio_lowerhalf_s *dev, FAR void *session) #else static int pcm_resume(FAR struct audio_lowerhalf_s *dev) #endif { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->resume); audinfo("Defer to lower resume\n"); #ifdef CONFIG_AUDIO_MULTI_SESSION return lower->ops->resume(lower, session); #else return lower->ops->resume(lower); #endif } #endif /* CONFIG_AUDIO_EXCLUDE_PAUSE_RESUME */ /**************************************************************************** * Name: pcm_allocbuffer * * Description: * Allocate an audio pipeline buffer. This routine provides the * lower-half driver with the opportunity to perform special buffer * allocation if needed, such as allocating from a specific memory * region (DMA-able, etc.). If not supplied, then the top-half * driver will perform a standard kumm_malloc using normal user-space * memory region. * ****************************************************************************/ static int pcm_allocbuffer(FAR struct audio_lowerhalf_s *dev, FAR struct audio_buf_desc_s *apb) { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->allocbuffer); audinfo("Defer to lower allocbuffer\n"); return lower->ops->allocbuffer(lower, apb); } /**************************************************************************** * Name: pcm_freebuffer * * Description: * Free an audio pipeline buffer. If the lower-level driver provides an * allocbuffer routine, it should also provide the freebuffer routine to * perform the free operation. * ****************************************************************************/ static int pcm_freebuffer(FAR struct audio_lowerhalf_s *dev, FAR struct audio_buf_desc_s *apb) { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->freebuffer); audinfo("Defer to lower freebuffer, apb=%p\n", apb); return lower->ops->freebuffer(lower, apb); } /**************************************************************************** * Name: pcm_enqueuebuffer * * Description: * Enqueue a buffer for processing. This is a non-blocking enqueue * operation. If the lower-half driver's buffer queue is full, then it * should return an error code of -ENOMEM, and the upper-half driver can * decide to either block the calling thread or deal with it in a non- * blocking manner. * * For each call to enqueuebuffer, the lower-half driver must call * audio_dequeuebuffer when it is finished processing the bufferr, passing * the previously enqueued apb and a dequeue status so that the upper-half * driver can decide if a waiting thread needs to be release, if the * dequeued buffer should be passed to the next block in the Audio * Pipeline, etc. * ****************************************************************************/ static int pcm_enqueuebuffer(FAR struct audio_lowerhalf_s *dev, FAR struct ap_buffer_s *apb) { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; apb_samp_t bytesleft; int ret; DEBUGASSERT(priv); audinfo("Received buffer %p, streaming=%d\n", apb, priv->streaming); lower = priv->lower; DEBUGASSERT(lower && lower->ops->enqueuebuffer && lower->ops->configure); /* Are we streaming yet? */ if (priv->streaming) { /* Yes, we are streaming */ /* Check for the last audio buffer in the stream */ if ((apb->flags & AUDIO_APB_FINAL) != 0) { /* Yes.. then we are no longer streaming */ priv->streaming = false; } #ifndef CONFIG_AUDIO_EXCLUDE_FFORWARD audinfo("Received: apb=%p curbyte=%d nbytes=%d flags=%04x\n", apb, apb->curbyte, apb->nbytes, apb->flags); /* Perform any necessary sub-sampling operations */ pcm_subsample(priv, apb); #endif /* Then give the audio buffer to the lower driver */ audinfo("Pass to lower enqueuebuffer: apb=%p curbyte=%d nbytes=%d\n", apb, apb->curbyte, apb->nbytes); return lower->ops->enqueuebuffer(lower, apb); } /* No.. then this must be the first buffer that we have seen (since we * will error out out if the first buffer is smaller than the WAV file * header. There is no attempt to reconstruct the full header from * fragments in multiple, tiny audio buffers). */ bytesleft = apb->nbytes - apb->curbyte; audinfo("curbyte=%d nbytes=%d nmaxbytes=%d bytesleft=%d\n", apb->curbyte, apb->nbytes, apb->nmaxbytes, bytesleft); /* Parse and verify the candidate PCM WAV file header */ #ifndef CONFIG_AUDIO_FORMAT_RAW if (bytesleft >= sizeof(struct wav_header_s) && pcm_parsewav(priv, &apb->samp[apb->curbyte])) { struct audio_caps_s caps; /* Configure the lower level for the number of channels, bitrate, * and sample bitwidth. */ DEBUGASSERT(priv->samprate < 65535); caps.ac_len = sizeof(struct audio_caps_s); caps.ac_type = AUDIO_TYPE_OUTPUT; caps.ac_channels = priv->nchannels; caps.ac_controls.hw[0] = (uint16_t)priv->samprate; caps.ac_controls.b[2] = priv->bpsamp; #ifdef CONFIG_AUDIO_MULTI_SESSION ret = lower->ops->configure(lower, priv->session, &caps); #else ret = lower->ops->configure(lower, &caps); #endif if (ret < 0) { auderr("ERROR: Failed to set PCM configuration: %d\n", ret); return ret; } /* Bump up the data offset */ apb->curbyte += sizeof(struct wav_header_s); #endif #ifndef CONFIG_AUDIO_EXCLUDE_FFORWARD audinfo("Begin streaming: apb=%p curbyte=%d nbytes=%d\n", apb, apb->curbyte, apb->nbytes); /* Perform any necessary sub-sampling operations */ pcm_subsample(priv, apb); #endif /* Then give the audio buffer to the lower driver */ audinfo( "Pass to lower enqueuebuffer: apb=%p curbyte=%d nbytes=%d\n", apb, apb->curbyte, apb->nbytes); ret = lower->ops->enqueuebuffer(lower, apb); if (ret == OK) { /* Now we are streaming. Unless for some reason there is only * one audio buffer in the audio stream. In that case, this * will be marked as the final buffer */ priv->streaming = ((apb->flags & AUDIO_APB_FINAL) == 0); return OK; } /* The normal protocol for streaming errors is as follows: * * (1) Fail the enqueueing by returned a negated error value. The * upper level then knows that this buffer was not queue. * (2) Return all queued buffers to the caller using the * AUDIO_CALLBACK_DEQUEUE callback * (3) Terminate playing using the AUDIO_CALLBACK_COMPLETE * callback. * * In this case we fail on the very first buffer and we need only * do (1) and (3). */ #ifdef CONFIG_AUDIO_MULTI_SESSION priv->export.upper(priv->export.priv, AUDIO_CALLBACK_COMPLETE, NULL, OK, NULL); #else priv->export.upper(priv->export.priv, AUDIO_CALLBACK_COMPLETE, NULL, OK); #endif } #ifndef CONFIG_AUDIO_FORMAT_RAW /* This is not a WAV file! */ auderr("ERROR: Invalid PCM WAV file\n"); return -EINVAL; #endif } /**************************************************************************** * Name: pcm_cancelbuffer * * Description: * Cancel a previously enqueued buffer. * ****************************************************************************/ static int pcm_cancelbuffer(FAR struct audio_lowerhalf_s *dev, FAR struct ap_buffer_s *apb) { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->cancelbuffer); audinfo("Defer to lower cancelbuffer, apb=%p\n", apb); return lower->ops->cancelbuffer(lower, apb); } /**************************************************************************** * Name: pcm_ioctl * * Description: * Lower-half logic may support platform-specific ioctl commands. * ****************************************************************************/ static int pcm_ioctl(FAR struct audio_lowerhalf_s *dev, int cmd, unsigned long arg) { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* Defer the operation to the lower device driver */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->ioctl); audinfo("Defer to lower ioctl, cmd=%d arg=%ld\n", cmd, arg); return lower->ops->ioctl(lower, cmd, arg); } /**************************************************************************** * Name: pcm_reserve * * Description: * Reserve a session (may only be one per device or may be multiple) for * use by a client. Client software can open audio devices and issue * AUDIOIOC_GETCAPS calls freely, but other operations require a * reservation. A session reservation will assign a context that must * be passed with * ****************************************************************************/ #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_reserve(FAR struct audio_lowerhalf_s *dev, FAR void **session) #else static int pcm_reserve(FAR struct audio_lowerhalf_s *dev) #endif { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; int ret; #ifdef CONFIG_AUDIO_MULTI_SESSION DEBUGASSERT(priv && session); #else DEBUGASSERT(priv); #endif /* It is not necessary to reserve the upper half. What we really need to * do is to reserved the lower device driver for exclusive use by the PCM * decoder. That effectively reserves the upper PCM decoder along with * the lower driver (which is then not available for use by other * decoders). * * We do, however, need to remember the session returned by the lower * level. */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->reserve); audinfo("Defer to lower reserve\n"); #ifdef CONFIG_AUDIO_MULTI_SESSION ret = lower->ops->reserve(lower, &priv->session); /* Return a copy of the session to the caller */ *session = priv->session; #else ret = lower->ops->reserve(lower); #endif return ret; } /**************************************************************************** * Name: pcm_release * * Description: * Release a session. * ****************************************************************************/ #ifdef CONFIG_AUDIO_MULTI_SESSION static int pcm_release(FAR struct audio_lowerhalf_s *dev, FAR void *session) #else static int pcm_release(FAR struct audio_lowerhalf_s *dev) #endif { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)dev; FAR struct audio_lowerhalf_s *lower; DEBUGASSERT(priv); /* Release the lower driver. It is then available for use by other * decoders (and we cannot use the lower driver either unless we re- * reserve it). */ lower = priv->lower; DEBUGASSERT(lower && lower->ops->release); audinfo("Defer to lower release\n"); #ifdef CONFIG_AUDIO_MULTI_SESSION return lower->ops->release(lower, session); #else return lower->ops->release(lower); #endif } /**************************************************************************** * Name: pcm_callback * * Description: * Lower-to-upper level callback for buffer dequeueing. * * Input Parameters: * priv - The value of the 'priv' field from out audio_lowerhalf_s. * * Returned Value: * None * ****************************************************************************/ #ifdef CONFIG_AUDIO_MULTI_SESSION static void pcm_callback(FAR void *arg, uint16_t reason, FAR struct ap_buffer_s *apb, uint16_t status, FAR void *session) #else static void pcm_callback(FAR void *arg, uint16_t reason, FAR struct ap_buffer_s *apb, uint16_t status) #endif { FAR struct pcm_decode_s *priv = (FAR struct pcm_decode_s *)arg; DEBUGASSERT(priv && priv->export.upper); /* The buffer belongs to an upper level. Just forward the event to * the next level up. */ #ifdef CONFIG_AUDIO_MULTI_SESSION priv->export.upper(priv->export.priv, reason, apb, status, session); #else priv->export.upper(priv->export.priv, reason, apb, status); #endif } /**************************************************************************** * Public Functions ****************************************************************************/ /**************************************************************************** * Name: pcm_decode_initialize * * Description: * Initialize the PCM device. The PCM device accepts and contains a * low-level audio DAC-type device. It then returns a new audio lower * half interface at adds a PCM decoding from end to the low-level * audio device * * Input Parameters: * dev - A reference to the low-level audio DAC-type device to contain. * * Returned Value: * On success, a new audio device instance is returned that wraps the * low-level device and provides a PCM decoding front end. NULL is * returned on failure. * ****************************************************************************/ FAR struct audio_lowerhalf_s * pcm_decode_initialize(FAR struct audio_lowerhalf_s *dev) { FAR struct pcm_decode_s *priv; FAR struct audio_ops_s *ops; /* Allocate an instance of our private data structure */ priv = (FAR struct pcm_decode_s *)kmm_zalloc(sizeof(struct pcm_decode_s)); if (!priv) { auderr("ERROR: Failed to allocate driver structure\n"); return NULL; } /* Initialize our private data structure. Since kmm_zalloc() was used for * the allocation, we need to initialize only non-zero, non-NULL, non- * false fields. */ /* Setup our operations */ ops = &priv->ops; ops->getcaps = pcm_getcaps; ops->configure = pcm_configure; ops->shutdown = pcm_shutdown; ops->start = pcm_start; #ifndef CONFIG_AUDIO_EXCLUDE_STOP ops->stop = pcm_stop; #endif #ifndef CONFIG_AUDIO_EXCLUDE_PAUSE_RESUME ops->pause = pcm_pause; ops->resume = pcm_resume; #endif if (dev->ops->allocbuffer) { DEBUGASSERT(dev->ops->freebuffer); ops->allocbuffer = pcm_allocbuffer; ops->freebuffer = pcm_freebuffer; } ops->enqueuebuffer = pcm_enqueuebuffer; ops->cancelbuffer = pcm_cancelbuffer; ops->ioctl = pcm_ioctl; ops->reserve = pcm_reserve; ops->release = pcm_release; /* Set up our struct audio_lower_half that we will register with the * system. The registration process will fill in the priv->export.upper * and priv fields with the correct callback information. */ priv->export.ops = &priv->ops; /* Save the struct audio_lower_half of the low-level audio device. Set * out callback information for the lower-level audio device. Our * callback will simply forward to the upper callback. */ priv->lower = dev; dev->upper = pcm_callback; dev->priv = priv; return &priv->export; } #endif /* CONFIG_AUDIO && CONFIG_AUDIO_FORMAT_PCM */