forked from enlightenment/efl
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:
parent
44d95d00ea
commit
c7e7c3043e
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@ -10,13 +10,16 @@
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#include <remix/remix.h>
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#include <alsa/asoundlib.h>
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#include <Eina.h>
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#ifdef HAVE_LIBSNDFILE
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#include <sndfile.h>
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#endif
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#define ALSA_PLAYER_BUFFERLEN 2048
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typedef struct _AlsaPlayerData AlsaPlayerData;
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typedef struct _Alsa_Player_Data Alsa_Player_Data;
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typedef short PLAYER_PCM;
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struct _AlsaPlayerData
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struct _Alsa_Player_Data
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{
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RemixPCM databuffer[ALSA_PLAYER_BUFFERLEN];
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snd_pcm_t *alsa_dev;
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@ -25,17 +28,28 @@ struct _AlsaPlayerData
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unsigned int frequency;
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};
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static int _log_dom = -1;
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static int init_count = 0;
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#ifdef WRN
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# undef WRN
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#endif
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#define WRN(...) EINA_LOG_DOM_WARN(_log_dom, __VA_ARGS__)
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//#define MIXDBG 1
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/* Optimisation dependencies: none */
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static RemixBase *alsa_player_optimise(RemixEnv *env, RemixBase *base);
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static snd_pcm_t *
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alsa_open(int channels, unsigned samplerate)
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alsa_open(int channels, unsigned int samplerate, unsigned int *real_samplerate)
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{
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const char *device = "default";
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snd_pcm_t *alsa_dev = NULL;
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snd_pcm_hw_params_t *hw_params;
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snd_pcm_uframes_t alsa_buffer_frames;
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snd_pcm_uframes_t alsa_period_size;
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unsigned int samplerate_ret = 0;
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int err;
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alsa_buffer_frames = ALSA_PLAYER_BUFFERLEN;
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@ -43,67 +57,76 @@ alsa_open(int channels, unsigned samplerate)
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if ((err = snd_pcm_open(&alsa_dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
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{
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printf("cannot open alsa playback stream (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0)
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{
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printf("cannot alloc snd hw params (%s)\n", snd_strerror(err));
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WRN("cannot open alsa playback stream (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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snd_pcm_hw_params_alloca(&hw_params);
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if ((err = snd_pcm_hw_params_any(alsa_dev, hw_params)) < 0)
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{
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printf("cannot initialize snd hw params (%s)\n", snd_strerror(err));
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WRN("cannot initialize snd hw params (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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if ((err = snd_pcm_hw_params_set_access(alsa_dev, hw_params,
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SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
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{
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printf("cannot set access type (%s)\n", snd_strerror(err));
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WRN("cannot set interleaved access (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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if ((err = snd_pcm_hw_params_set_format(alsa_dev, hw_params,
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SND_PCM_FORMAT_FLOAT)) < 0)
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{
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// FIXME: handle if float format not possible
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printf("cannot set sample format (%s)\n", snd_strerror(err));
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WRN("cannot set float sample format (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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#ifdef MIXDBG // testing/debugging by making output samplerate be 48khz
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samplerate_ret = 48000;
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if ((err = snd_pcm_hw_params_set_rate_near(alsa_dev, hw_params,
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&samplerate_ret, 0)) < 0)
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{
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WRN("cannot set sample rate (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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#else
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if ((err = snd_pcm_hw_params_set_rate_near(alsa_dev, hw_params,
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&samplerate, 0)) < 0)
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{
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// FIXME: get actual sample rate and tell remix
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printf("cannot set sample rate (%s)\n", snd_strerror(err));
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WRN("cannot set sample rate (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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#endif
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if ((err = snd_pcm_hw_params_set_channels(alsa_dev, hw_params, channels)) < 0)
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{
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printf("cannot set channel count (%s)\n", snd_strerror(err));
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WRN("cannot set channel count (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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if ((err = snd_pcm_hw_params_set_buffer_size_near(alsa_dev, hw_params,
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&alsa_buffer_frames)) < 0)
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{
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fprintf(stderr, "cannot set buffer size (%s)\n", snd_strerror(err));
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WRN("cannot set buffer size (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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if ((err = snd_pcm_hw_params_set_period_size_near(alsa_dev, hw_params,
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&alsa_period_size, 0)) < 0)
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{
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fprintf(stderr, "cannot set period size (%s)\n", snd_strerror(err));
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WRN("cannot set period size (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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if ((err = snd_pcm_hw_params(alsa_dev, hw_params)) < 0)
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{
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printf("cannot set parameters (%s)\n", snd_strerror(err));
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WRN("cannot set parameters (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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if ((err = snd_pcm_hw_params_get_rate(hw_params, &samplerate_ret, 0)) < 0)
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{
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WRN("cannot get samplerate (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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snd_pcm_hw_params_free(hw_params);
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if ((err = snd_pcm_prepare(alsa_dev)) < 0)
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{
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printf("cannot prepare audio for use (%s)\n", snd_strerror(err));
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WRN("cannot prepare audio for use (%s)\n", snd_strerror(err));
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goto catch_error;
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}
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if (real_samplerate) *real_samplerate = samplerate_ret;
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catch_error:
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if ((err < 0) && (alsa_dev != NULL))
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static RemixBase *
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alsa_player_reset_device(RemixEnv *env, RemixBase *base)
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{
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AlsaPlayerData *player_data = remix_base_get_instance_data(env, base);
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Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
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unsigned int real_samplerate = 0;
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if (player_data->alsa_dev)
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{
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snd_pcm_close(player_data->alsa_dev);
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}
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player_data->alsa_dev = alsa_open(player_data->channels,
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player_data->frequency);
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player_data->frequency,
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&real_samplerate);
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if (!player_data->alsa_dev)
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{
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remix_set_error(env, REMIX_ERROR_SYSTEM);
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return RemixNone;
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}
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// printf("%i != %i\n", real_samplerate, player_data->frequency);
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if (real_samplerate != player_data->frequency)
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{
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player_data->frequency = real_samplerate;
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remix_set_samplerate(env, player_data->frequency);
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}
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return base;
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}
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alsa_player_init(RemixEnv *env, RemixBase *base, CDSet *parameters __UNUSED__)
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{
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CDSet *channels;
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AlsaPlayerData *player_data = calloc(1, sizeof(AlsaPlayerData));
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Alsa_Player_Data *player_data = calloc(1, sizeof(Alsa_Player_Data));
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if (!player_data)
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{
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return RemixNone;
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}
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init_count++;
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if (init_count == 1)
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{
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eina_init();
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_log_dom = eina_log_domain_register("remix-alsa", EINA_COLOR_CYAN);
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}
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remix_base_set_instance_data(env, base, player_data);
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channels = remix_get_channels(env);
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static int
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alsa_player_destroy(RemixEnv *env, RemixBase *base)
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{
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AlsaPlayerData *player_data = remix_base_get_instance_data(env, base);
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Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
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if (player_data->alsa_dev)
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{
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snd_pcm_close(player_data->alsa_dev);
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}
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free(player_data);
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init_count--;
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if (init_count == 0)
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{
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eina_log_domain_unregister(_log_dom);
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_log_dom = -1;
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eina_shutdown();
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}
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return 0;
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}
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static int
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alsa_player_ready(RemixEnv *env, RemixBase *base)
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{
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AlsaPlayerData *player_data = remix_base_get_instance_data(env, base);
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Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
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RemixCount nr_channels;
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CDSet *channels;
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int samplerate;
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}
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static RemixCount
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alsa_player_playbuffer(RemixEnv *env __UNUSED__, AlsaPlayerData *player, RemixPCM *data, RemixCount count)
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alsa_player_playbuffer(RemixEnv *env __UNUSED__, Alsa_Player_Data *player, RemixPCM *data, RemixCount count)
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{
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#ifdef MIXDBG
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{
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static int total = 0;
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static SNDFILE *sfile = NULL;
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static SF_INFO sfinfo;
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if (total == 0)
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{
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sfinfo.frames = 0;
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sfinfo.samplerate = player->frequency;
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sfinfo.channels = 2;
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sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16 | SF_ENDIAN_LITTLE;
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sfinfo.sections = 0;
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sfinfo.seekable = 0;
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sfile = sf_open("out.wav", SFM_WRITE, &sfinfo);
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}
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if (sfile)
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{
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sf_writef_float(sfile, data, count);
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total += count;
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}
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}
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#endif
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return snd_pcm_writei(player->alsa_dev, data, count);
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}
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static RemixCount
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alsa_player_chunk(RemixEnv *env, RemixChunk *chunk, RemixCount offset, RemixCount count, int channelname __UNUSED__, void *data)
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{
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AlsaPlayerData *player = data;
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Alsa_Player_Data *player = data;
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RemixCount remaining = count, written = 0, n, playcount;
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RemixPCM *d;
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static RemixCount
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alsa_player_process(RemixEnv *env, RemixBase *base, RemixCount count, RemixStream *input, RemixStream *output __UNUSED__)
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{
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AlsaPlayerData *player_data = remix_base_get_instance_data(env, base);
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Alsa_Player_Data *player_data = remix_base_get_instance_data(env, base);
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RemixCount nr_channels = remix_stream_nr_channels(env, input);
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RemixCount remaining = count, processed = 0, n, nn;
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}
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else if ((nr_channels == 2) && (player_data->stereo == 1))
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{ /*STEREO*/
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while (processed < count)
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while (remaining > 0)
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{
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n = MIN(remaining, ALSA_PLAYER_BUFFERLEN);
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n = MIN(remaining, ALSA_PLAYER_BUFFERLEN / 2);
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n = remix_stream_interleave_2(env, input,
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REMIX_CHANNEL_LEFT,
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REMIX_CHANNEL_RIGHT,
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}
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return processed;
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}
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printf("[alsa_player_process] unsupported stream/output channel\n");
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printf("combination %ld / %d\n", nr_channels, player_data->stereo ? 2 : 1);
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WRN("[alsa_player_process] unsupported stream/output channel"
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"combination %ld / %d\n", nr_channels, player_data->stereo ? 2 : 1);
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return -1;
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}
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