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nuttx/audio/pcm_decode.c
Alin Jerpelea 95ef3b77c2 audio: migrate to SPDX identifier 
Most tools used for compliance and SBOM generation use SPDX identifiers
This change brings us a step closer to an easy SBOM generation.

Signed-off-by: Alin Jerpelea <alin.jerpelea@sony.com>
2024-09-10 11:33:26 +08:00

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/****************************************************************************
* audio/pcm_decode.c
*
* SPDX-License-Identifier: Apache-2.0
*
* 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 <nuttx/config.h>
#include <sys/param.h>
#include <sys/types.h>
#include <inttypes.h>
#include <stdint.h>
#include <stdbool.h>
#include <stdlib.h>
#include <string.h>
#include <assert.h>
#include <errno.h>
#include <debug.h>
#include <nuttx/kmalloc.h>
#include <nuttx/audio/audio.h>
#include <nuttx/audio/pcm.h>
#if defined(CONFIG_AUDIO) && defined(CONFIG_AUDIO_FORMAT_PCM)
/****************************************************************************
* 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_DEBUG_AUDIO_INFO
static void pcm_dump(FAR const struct wav_header_s *wav);
#else
# define pcm_dump(w)
#endif
#ifndef CONFIG_AUDIO_FORMAT_RAW
static inline bool pcm_validwav(FAR const struct wav_header_s *wav);
static ssize_t pcm_parsewav(FAR struct pcm_decode_s *priv, uint8_t *data,
apb_samp_t len);
#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_DEBUG_AUDIO_INFO
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. Unaligned address is
* acceptable.
*
****************************************************************************/
#ifndef CONFIG_AUDIO_FORMAT_RAW
static uint16_t pcm_leuint16(FAR const uint16_t *ptr)
{
FAR const uint8_t *p = (FAR const uint8_t *)ptr;
return ((p[0] << 0) |
(p[1] << 8));
}
/****************************************************************************
* Name: pcm_leuint32
*
* Description:
* Get a 32-bit value stored in little endian order. Unaligned address is
* acceptable.
*
****************************************************************************/
static uint32_t pcm_leuint32(FAR const uint32_t *ptr)
{
FAR const uint8_t *p = (FAR const uint8_t *)ptr;
return ((p[0] << 0) |
(p[1] << 8) |
(p[2] << 16) |
(p[3] << 24));
}
/****************************************************************************
* Name: pcm_validwav
*
* Description:
* Return true if this is a valid WAV file header
*
****************************************************************************/
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);
}
/****************************************************************************
* 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 ssize_t pcm_parsewav(FAR struct pcm_decode_s *priv, uint8_t *data,
apb_samp_t len)
{
FAR const struct wav_header_s *wav = (FAR const struct wav_header_s *)data;
FAR const struct wav_datachunk_s *dchunk;
struct wav_header_s localwav;
size_t ret = sizeof(struct wav_header_s);
if (len < sizeof(struct wav_header_s))
{
return -EINVAL;
}
/* 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);
/* Find the data chunk */
dchunk = &wav->data;
for (; ; )
{
/* NOTE: The data chunk is possible to be not word-aligned if extra
* chunks exist before it.
*/
localwav.data.chunkid = pcm_leuint32(&dchunk->chunkid);
localwav.data.chunklen = pcm_leuint32(&dchunk->chunklen);
if (localwav.data.chunkid == WAV_DATA_CHUNKID)
{
break;
}
/* Not data chunk. Skip it. */
ret += localwav.data.chunklen + 8;
if (ret >= len)
{
/* Data chunk not found */
return -EINVAL;
}
dchunk = (FAR const struct wav_datachunk_s *)
((uintptr_t)dchunk + localwav.data.chunklen + 8);
}
/* Dump the converted wave header information */
pcm_dump(&localwav);
/* Check if the file is a valid PCM WAV header */
if (!pcm_validwav(&localwav))
{
return -EINVAL;
}
else
{
/* 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
ssize_t headersize = pcm_parsewav(priv, &apb->samp[apb->curbyte],
bytesleft);
if (headersize > 0)
{
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 += headersize;
#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");
#endif
return -EINVAL;
}
/****************************************************************************
* 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 = 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 */
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