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see how much morte code i had to add just to use eina_log gustavo? i 

had to add eina_init() and shutdown and init refcounting and log
domain creating and add a macro.. so i can finally use it? eina being
included here actually was never needed... but since include was there
i could use it...

just saying - using eina_log is not convenient. it doesn't save time.
it's benefits are dubious (knowing what envv ar to set to what values
to get log output is black magic as u need to know specific log level
values and log domain names which u only find out by digging thru
code). my point -> if u want eina_log used more.. it needs to be AS
EASY as the printf. :)



SVN revision: 64937
This commit is contained in:
Carsten Haitzler 2011-11-08 07:14:49 +00:00
parent 44d95d00ea
commit c7e7c3043e
1 changed files with 102 additions and 33 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

135
legacy/edje/src/modules/alsa_snd_player/alsa_snd_player.c
View File

@ -10,13 +10,16 @@
#include <remix/remix.h>
#include <alsa/asoundlib.h>
#include <Eina.h>
#ifdef HAVE_LIBSNDFILE
#include <sndfile.h>
#endif
#define ALSA_PLAYER_BUFFERLEN 2048
typedef struct _AlsaPlayerData AlsaPlayerData;
typedef struct _Alsa_Player_Data Alsa_Player_Data;
typedef short PLAYER_PCM;
struct _AlsaPlayerData
struct _Alsa_Player_Data
{
RemixPCM databuffer[ALSA_PLAYER_BUFFERLEN];
snd_pcm_t *alsa_dev;
@ -25,17 +28,28 @@ struct _AlsaPlayerData
unsigned int frequency;
};
static int _log_dom = -1;
static int init_count = 0;
#ifdef WRN
# undef WRN
#endif
#define WRN(...) EINA_LOG_DOM_WARN(_log_dom, __VA_ARGS__)
//#define MIXDBG 1
/* Optimisation dependencies: none */
static RemixBase *alsa_player_optimise(RemixEnv *env, RemixBase *base);
static snd_pcm_t *
alsa_open(int channels, unsigned samplerate)
alsa_open(int channels, unsigned int samplerate, unsigned int *real_samplerate)
{
const char *device = "default";
snd_pcm_t *alsa_dev = NULL;
snd_pcm_hw_params_t *hw_params;
snd_pcm_uframes_t alsa_buffer_frames;
snd_pcm_uframes_t alsa_period_size;
unsigned int samplerate_ret = 0;
int err;
alsa_buffer_frames = ALSA_PLAYER_BUFFERLEN;
@ -43,67 +57,76 @@ alsa_open(int channels, unsigned samplerate)
if ((err = snd_pcm_open(&alsa_dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
printf("cannot open alsa playback stream (%s)\n", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0)
{
printf("cannot alloc snd hw params (%s)\n", snd_strerror(err));
WRN("cannot open alsa playback stream (%s)\n", snd_strerror(err));
goto catch_error;
}
snd_pcm_hw_params_alloca(&hw_params);
if ((err = snd_pcm_hw_params_any(alsa_dev, hw_params)) < 0)
{
printf("cannot initialize snd hw params (%s)\n", snd_strerror(err));
WRN("cannot initialize snd hw params (%s)\n", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_set_access(alsa_dev, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
printf("cannot set access type (%s)\n", snd_strerror(err));
WRN("cannot set interleaved access (%s)\n", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_set_format(alsa_dev, hw_params,
SND_PCM_FORMAT_FLOAT)) < 0)
{
// FIXME: handle if float format not possible
printf("cannot set sample format (%s)\n", snd_strerror(err));
WRN("cannot set float sample format (%s)\n", snd_strerror(err));
goto catch_error;
}
#ifdef MIXDBG // testing/debugging by making output samplerate be 48khz
samplerate_ret = 48000;
if ((err = snd_pcm_hw_params_set_rate_near(alsa_dev, hw_params,
&samplerate_ret, 0)) < 0)
{
WRN("cannot set sample rate (%s)\n", snd_strerror(err));
goto catch_error;
}
#else
if ((err = snd_pcm_hw_params_set_rate_near(alsa_dev, hw_params,
&samplerate, 0)) < 0)
{
// FIXME: get actual sample rate and tell remix
printf("cannot set sample rate (%s)\n", snd_strerror(err));
WRN("cannot set sample rate (%s)\n", snd_strerror(err));
goto catch_error;
}
#endif
if ((err = snd_pcm_hw_params_set_channels(alsa_dev, hw_params, channels)) < 0)
{
printf("cannot set channel count (%s)\n", snd_strerror(err));
WRN("cannot set channel count (%s)\n", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_set_buffer_size_near(alsa_dev, hw_params,
&alsa_buffer_frames)) < 0)
{
fprintf(stderr, "cannot set buffer size (%s)\n", snd_strerror(err));
WRN("cannot set buffer size (%s)\n", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_set_period_size_near(alsa_dev, hw_params,
&alsa_period_size, 0)) < 0)
{
fprintf(stderr, "cannot set period size (%s)\n", snd_strerror(err));
WRN("cannot set period size (%s)\n", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params(alsa_dev, hw_params)) < 0)
{
printf("cannot set parameters (%s)\n", snd_strerror(err));
WRN("cannot set parameters (%s)\n", snd_strerror(err));
goto catch_error;
}
if ((err = snd_pcm_hw_params_get_rate(hw_params, &samplerate_ret, 0)) < 0)
{
WRN("cannot get samplerate (%s)\n", snd_strerror(err));
goto catch_error;
}
snd_pcm_hw_params_free(hw_params);
if ((err = snd_pcm_prepare(alsa_dev)) < 0)
{
printf("cannot prepare audio for use (%s)\n", snd_strerror(err));
WRN("cannot prepare audio for use (%s)\n", snd_strerror(err));
goto catch_error;
}
if (real_samplerate) *real_samplerate = samplerate_ret;
catch_error:
if ((err < 0) && (alsa_dev != NULL))
@ -117,7 +140,8 @@ catch_error:
static RemixBase *
alsa_player_reset_device(RemixEnv *env, RemixBase *base)
{
AlsaPlayerData *player_data = remix_base_get_instance_data(env, base);
Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
unsigned int real_samplerate = 0;
if (player_data->alsa_dev)
{
@ -125,12 +149,19 @@ alsa_player_reset_device(RemixEnv *env, RemixBase *base)
snd_pcm_close(player_data->alsa_dev);
}
player_data->alsa_dev = alsa_open(player_data->channels,
player_data->frequency);
player_data->frequency,
&real_samplerate);
if (!player_data->alsa_dev)
{
remix_set_error(env, REMIX_ERROR_SYSTEM);
return RemixNone;
}
// printf("%i != %i\n", real_samplerate, player_data->frequency);
if (real_samplerate != player_data->frequency)
{
player_data->frequency = real_samplerate;
remix_set_samplerate(env, player_data->frequency);
}
return base;
}
@ -138,13 +169,20 @@ static RemixBase *
alsa_player_init(RemixEnv *env, RemixBase *base, CDSet *parameters __UNUSED__)
{
CDSet *channels;
AlsaPlayerData *player_data = calloc(1, sizeof(AlsaPlayerData));
Alsa_Player_Data *player_data = calloc(1, sizeof(Alsa_Player_Data));
if (!player_data)
{
remix_set_error(env, REMIX_ERROR_SYSTEM);
return RemixNone;
}
init_count++;
if (init_count == 1)
{
eina_init();
_log_dom = eina_log_domain_register("remix-alsa", EINA_COLOR_CYAN);
}
remix_base_set_instance_data(env, base, player_data);
channels = remix_get_channels(env);
@ -170,7 +208,7 @@ alsa_player_clone(RemixEnv *env, RemixBase *base __UNUSED__)
static int
alsa_player_destroy(RemixEnv *env, RemixBase *base)
{
AlsaPlayerData *player_data = remix_base_get_instance_data(env, base);
Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
if (player_data->alsa_dev)
{
@ -178,13 +216,20 @@ alsa_player_destroy(RemixEnv *env, RemixBase *base)
snd_pcm_close(player_data->alsa_dev);
}
free(player_data);
init_count--;
if (init_count == 0)
{
eina_log_domain_unregister(_log_dom);
_log_dom = -1;
eina_shutdown();
}
return 0;
}
static int
alsa_player_ready(RemixEnv *env, RemixBase *base)
{
AlsaPlayerData *player_data = remix_base_get_instance_data(env, base);
Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
RemixCount nr_channels;
CDSet *channels;
int samplerate;
@ -205,15 +250,39 @@ alsa_player_prepare(RemixEnv *env, RemixBase *base)
}
static RemixCount
alsa_player_playbuffer(RemixEnv *env __UNUSED__, AlsaPlayerData *player, RemixPCM *data, RemixCount count)
alsa_player_playbuffer(RemixEnv *env __UNUSED__, Alsa_Player_Data *player, RemixPCM *data, RemixCount count)
{
#ifdef MIXDBG
{
static int total = 0;
static SNDFILE *sfile = NULL;
static SF_INFO sfinfo;
if (total == 0)
{
sfinfo.frames = 0;
sfinfo.samplerate = player->frequency;
sfinfo.channels = 2;
sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16 | SF_ENDIAN_LITTLE;
sfinfo.sections = 0;
sfinfo.seekable = 0;
sfile = sf_open("out.wav", SFM_WRITE, &sfinfo);
}
if (sfile)
{
sf_writef_float(sfile, data, count);
total += count;
}
}
#endif
return snd_pcm_writei(player->alsa_dev, data, count);
}
static RemixCount
alsa_player_chunk(RemixEnv *env, RemixChunk *chunk, RemixCount offset, RemixCount count, int channelname __UNUSED__, void *data)
{
AlsaPlayerData *player = data;
Alsa_Player_Data *player = data;
RemixCount remaining = count, written = 0, n, playcount;
RemixPCM *d;
@ -237,7 +306,7 @@ alsa_player_chunk(RemixEnv *env, RemixChunk *chunk, RemixCount offset, RemixCoun
static RemixCount
alsa_player_process(RemixEnv *env, RemixBase *base, RemixCount count, RemixStream *input, RemixStream *output __UNUSED__)
{
AlsaPlayerData *player_data = remix_base_get_instance_data(env, base);
Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
RemixCount nr_channels = remix_stream_nr_channels(env, input);
RemixCount remaining = count, processed = 0, n, nn;
@ -248,9 +317,9 @@ alsa_player_process(RemixEnv *env, RemixBase *base, RemixCount count, RemixStrea
}
else if ((nr_channels == 2) && (player_data->stereo == 1))
{ /*STEREO*/
while (processed < count)
while (remaining > 0)
{
n = MIN(remaining, ALSA_PLAYER_BUFFERLEN);
n = MIN(remaining, ALSA_PLAYER_BUFFERLEN / 2);
n = remix_stream_interleave_2(env, input,
REMIX_CHANNEL_LEFT,
REMIX_CHANNEL_RIGHT,
@ -262,8 +331,8 @@ alsa_player_process(RemixEnv *env, RemixBase *base, RemixCount count, RemixStrea
}
return processed;
}
printf("[alsa_player_process] unsupported stream/output channel\n");
printf("combination %ld / %d\n", nr_channels, player_data->stereo ? 2 : 1);
WRN("[alsa_player_process] unsupported stream/output channel"
"combination %ld / %d\n", nr_channels, player_data->stereo ? 2 : 1);
return -1;
}