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Merge audio capture work back into the mainline.

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This commit is contained in:
Ryan C. Gordon
2016-08-28 13:36:13 -04:00
parent 06dcdc7d48 b6daf1f60a
commit 2da1ec8354
73 changed files with 2448 additions and 1144 deletions
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160
src/audio/alsa/SDL_alsa_audio.c
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@@ -32,7 +32,6 @@
#include "SDL_assert.h"
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_alsa_audio.h"
@@ -43,8 +42,10 @@
static int (*ALSA_snd_pcm_open)
(snd_pcm_t **, const char *, snd_pcm_stream_t, int);
static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm);
static snd_pcm_sframes_t(*ALSA_snd_pcm_writei)
static snd_pcm_sframes_t (*ALSA_snd_pcm_writei)
(snd_pcm_t *, const void *, snd_pcm_uframes_t);
static snd_pcm_sframes_t (*ALSA_snd_pcm_readi)
(snd_pcm_t *, void *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_recover) (snd_pcm_t *, int, int);
static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *);
static int (*ALSA_snd_pcm_drain) (snd_pcm_t *);
@@ -86,6 +87,7 @@ static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int);
static int (*ALSA_snd_pcm_wait)(snd_pcm_t *, int);
static int (*ALSA_snd_pcm_sw_params_set_avail_min)
(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_reset)(snd_pcm_t *);
static int (*ALSA_snd_device_name_hint) (int, const char *, void ***);
static char* (*ALSA_snd_device_name_get_hint) (const void *, const char *);
static int (*ALSA_snd_device_name_free_hint) (void **);
@@ -122,6 +124,7 @@ load_alsa_syms(void)
SDL_ALSA_SYM(snd_pcm_open);
SDL_ALSA_SYM(snd_pcm_close);
SDL_ALSA_SYM(snd_pcm_writei);
SDL_ALSA_SYM(snd_pcm_readi);
SDL_ALSA_SYM(snd_pcm_recover);
SDL_ALSA_SYM(snd_pcm_prepare);
SDL_ALSA_SYM(snd_pcm_drain);
@@ -148,6 +151,7 @@ load_alsa_syms(void)
SDL_ALSA_SYM(snd_pcm_nonblock);
SDL_ALSA_SYM(snd_pcm_wait);
SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min);
SDL_ALSA_SYM(snd_pcm_reset);
SDL_ALSA_SYM(snd_device_name_hint);
SDL_ALSA_SYM(snd_device_name_get_hint);
SDL_ALSA_SYM(snd_device_name_free_hint);
@@ -242,37 +246,37 @@ ALSA_WaitDevice(_THIS)
* "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
*/
#define SWIZ6(T) \
T *ptr = (T *) this->hidden->mixbuf; \
#define SWIZ6(T, buf, numframes) \
T *ptr = (T *) buf; \
Uint32 i; \
for (i = 0; i < this->spec.samples; i++, ptr += 6) { \
for (i = 0; i < numframes; i++, ptr += 6) { \
T tmp; \
tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
}
static SDL_INLINE void
swizzle_alsa_channels_6_64bit(_THIS)
swizzle_alsa_channels_6_64bit(void *buffer, Uint32 bufferlen)
{
SWIZ6(Uint64);
SWIZ6(Uint64, buffer, bufferlen);
}
static SDL_INLINE void
swizzle_alsa_channels_6_32bit(_THIS)
swizzle_alsa_channels_6_32bit(void *buffer, Uint32 bufferlen)
{
SWIZ6(Uint32);
SWIZ6(Uint32, buffer, bufferlen);
}
static SDL_INLINE void
swizzle_alsa_channels_6_16bit(_THIS)
swizzle_alsa_channels_6_16bit(void *buffer, Uint32 bufferlen)
{
SWIZ6(Uint16);
SWIZ6(Uint16, buffer, bufferlen);
}
static SDL_INLINE void
swizzle_alsa_channels_6_8bit(_THIS)
swizzle_alsa_channels_6_8bit(void *buffer, Uint32 bufferlen)
{
SWIZ6(Uint8);
SWIZ6(Uint8, buffer, bufferlen);
}
#undef SWIZ6
@@ -283,18 +287,16 @@ swizzle_alsa_channels_6_8bit(_THIS)
* channels from Windows/Mac order to the format alsalib will want.
*/
static SDL_INLINE void
swizzle_alsa_channels(_THIS)
swizzle_alsa_channels(_THIS, void *buffer, Uint32 bufferlen)
{
if (this->spec.channels == 6) {
const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
if (fmtsize == 16)
swizzle_alsa_channels_6_16bit(this);
else if (fmtsize == 8)
swizzle_alsa_channels_6_8bit(this);
else if (fmtsize == 32)
swizzle_alsa_channels_6_32bit(this);
else if (fmtsize == 64)
swizzle_alsa_channels_6_64bit(this);
switch (SDL_AUDIO_BITSIZE(this->spec.format)) {
case 8: swizzle_alsa_channels_6_8bit(buffer, bufferlen); break;
case 16: swizzle_alsa_channels_6_16bit(buffer, bufferlen); break;
case 32: swizzle_alsa_channels_6_32bit(buffer, bufferlen); break;
case 64: swizzle_alsa_channels_6_64bit(buffer, bufferlen); break;
default: SDL_assert(!"unhandled bitsize"); break;
}
}
/* !!! FIXME: update this for 7.1 if needed, later. */
@@ -304,19 +306,18 @@ swizzle_alsa_channels(_THIS)
static void
ALSA_PlayDevice(_THIS)
{
int status;
const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf;
const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) *
const int frame_size = (((int) SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
this->spec.channels;
snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples);
swizzle_alsa_channels(this);
swizzle_alsa_channels(this, this->hidden->mixbuf, frames_left);
while ( frames_left > 0 && this->enabled ) {
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
/* !!! FIXME: This works, but needs more testing before going live */
/* ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1); */
status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
int status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
if (status == -EAGAIN) {
@@ -346,20 +347,66 @@ ALSA_GetDeviceBuf(_THIS)
return (this->hidden->mixbuf);
}
static int
ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
Uint8 *sample_buf = (Uint8 *) buffer;
const int frame_size = (((int) SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
this->spec.channels;
const int total_frames = buflen / frame_size;
snd_pcm_uframes_t frames_left = total_frames;
SDL_assert((buflen % frame_size) == 0);
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
/* !!! FIXME: This works, but needs more testing before going live */
/* ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1); */
int status = ALSA_snd_pcm_readi(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
/*printf("ALSA: capture error %d\n", status);*/
if (status == -EAGAIN) {
/* Apparently snd_pcm_recover() doesn't handle this case -
does it assume snd_pcm_wait() above? */
SDL_Delay(1);
continue;
}
status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA read failed (unrecoverable): %s\n",
ALSA_snd_strerror(status));
return -1;
}
continue;
}
/*printf("ALSA: captured %d bytes\n", status * frame_size);*/
sample_buf += status * frame_size;
frames_left -= status;
}
swizzle_alsa_channels(this, buffer, total_frames - frames_left);
return (total_frames - frames_left) * frame_size;
}
static void
ALSA_FlushCapture(_THIS)
{
ALSA_snd_pcm_reset(this->hidden->pcm_handle);
}
static void
ALSA_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
if (this->hidden->pcm_handle) {
ALSA_snd_pcm_drain(this->hidden->pcm_handle);
ALSA_snd_pcm_close(this->hidden->pcm_handle);
this->hidden->pcm_handle = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
if (this->hidden->pcm_handle) {
ALSA_snd_pcm_drain(this->hidden->pcm_handle);
ALSA_snd_pcm_close(this->hidden->pcm_handle);
}
SDL_free(this->hidden->mixbuf);
SDL_free(this->hidden);
}
static int
@@ -492,16 +539,16 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
SDL_zerop(this->hidden);
/* Open the audio device */
/* Name of device should depend on # channels in spec */
status = ALSA_snd_pcm_open(&pcm_handle,
get_audio_device(handle, this->spec.channels),
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
get_audio_device(handle, this->spec.channels),
iscapture ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't open audio device: %s",
ALSA_snd_strerror(status));
}
@@ -512,7 +559,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
snd_pcm_hw_params_alloca(&hwparams);
status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't get hardware config: %s",
ALSA_snd_strerror(status));
}
@@ -521,7 +567,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set interleaved access: %s",
ALSA_snd_strerror(status));
}
@@ -575,7 +620,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
}
}
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
@@ -587,7 +631,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (status < 0) {
status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set audio channels");
}
this->spec.channels = channels;
@@ -598,7 +641,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
&rate, NULL);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set audio frequency: %s",
ALSA_snd_strerror(status));
}
@@ -610,7 +652,6 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Failed to set desired buffer size, do the best you can... */
status = ALSA_set_period_size(this, hwparams, 1);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status));
}
}
@@ -618,26 +659,22 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
snd_pcm_sw_params_alloca(&swparams);
status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't get software config: %s",
ALSA_snd_strerror(status));
}
status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, this->spec.samples);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("Couldn't set minimum available samples: %s",
ALSA_snd_strerror(status));
}
status =
ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("ALSA: Couldn't set start threshold: %s",
ALSA_snd_strerror(status));
}
status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);
if (status < 0) {
ALSA_CloseDevice(this);
return SDL_SetError("Couldn't set software audio parameters: %s",
ALSA_snd_strerror(status));
}
@@ -646,13 +683,14 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ALSA_CloseDevice(this);
return SDL_OutOfMemory();
if (!iscapture) {
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
/* Switch to blocking mode for playback */
ALSA_snd_pcm_nonblock(pcm_handle, 0);
@@ -866,6 +904,10 @@ ALSA_Init(SDL_AudioDriverImpl * impl)
impl->PlayDevice = ALSA_PlayDevice;
impl->CloseDevice = ALSA_CloseDevice;
impl->Deinitialize = ALSA_Deinitialize;
impl->CaptureFromDevice = ALSA_CaptureFromDevice;
impl->FlushCapture = ALSA_FlushCapture;
impl->HasCaptureSupport = SDL_TRUE;
return 1; /* this audio target is available. */
}