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E SVN: raster trunk/edje/src/modules/alsa_snd_player
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From: Enlightenment S. <no-...@en...> - 2011-11-08 07:14:55
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Log: 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. :) Author: raster Date: 2011-11-07 23:14:49 -0800 (Mon, 07 Nov 2011) New Revision: 64937 Trac: http://trac.enlightenment.org/e/changeset/64937 Modified: trunk/edje/src/modules/alsa_snd_player/alsa_snd_player.c Modified: trunk/edje/src/modules/alsa_snd_player/alsa_snd_player.c =================================================================== --- trunk/edje/src/modules/alsa_snd_player/alsa_snd_player.c 2011-11-08 07:10:33 UTC (rev 64936) +++ trunk/edje/src/modules/alsa_snd_player/alsa_snd_player.c 2011-11-08 07:14:49 UTC (rev 64937) @@ -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 @@ 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 @@ 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)); + WRN("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)); - 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; } - snd_pcm_hw_params_free(hw_params); + 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; + } 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 @@ 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 @@ 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 @@ 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 @@ 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 @@ 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 @@ } 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 @@ 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 @@ } 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 @@ } 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; } |