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nuttx/audio/pcm_decode.c
Alin Jerpelea 231b8518b7 NuttX: Ken Pettit: update licenses to Apache 
Ken Pettit has submitted the ICLA and we can migrate the licenses
 to Apache.

Sebastien Lorquet has submitted the ICLA and we can migrate the licenses
 to Apache.

Gregory Nutt has submitted the SGA and we can migrate the licenses
 to Apache.

Signed-off-by: Alin Jerpelea <alin.jerpelea@sony.com>
2021-04-10 06:42:19 -05:00

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/****************************************************************************
* 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 <nuttx/config.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)
/****************************************************************************
* 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 */
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