static int voice_test() { int rtn = 0; char* name; fd_set rfds, wfds; Audio_t hPcm_Mic; Audio_t hPcm_Spk; snd_pcm_channel_params_t params; bool bQuit = false; /************************ CAPTURE **************************/ // configuring capture (mic) name = "voice"; // get audioman handle rtn = audio_manager_get_handle(AUDIO_TYPE_VIDEO_CHAT, getpid(), (bool)false, &hPcm_Mic.hAudioman); if(rtn < 0) { cli_print("audio_manager_get_handle (mic) failed %s", strerror(-rtn)); return -1; } cli_print("Opening %s - for capture", name); rtn = snd_pcm_open_name(&hPcm_Mic.pDs, name, SND_PCM_OPEN_CAPTURE); if (rtn < 0) { (void)audio_manager_free_handle(hPcm_Mic.hAudioman); cli_print("snd_pcm_open_name (mic) failed %s", snd_strerror(rtn)); return -1; // snd_pcm calls return negative values; make positive } rtn = snd_pcm_set_audioman_handle(hPcm_Mic.pDs, hPcm_Mic.hAudioman); if (rtn < 0) { (void)snd_pcm_close(hPcm_Mic.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); cli_print("snd_pcm_set_audioman_handle (mic) failed %s", snd_strerror(rtn)); (void)snd_pcm_close(hPcm_Mic.pDs); return -1; } // disable mmap (void)snd_pcm_plugin_set_disable(hPcm_Mic.pDs, (unsigned int)PLUGIN_DISABLE_MMAP); // set parameters memset(¶ms, 0, sizeof(params)); params.mode = SND_PCM_MODE_BLOCK; params.channel = SND_PCM_CHANNEL_CAPTURE; params.start_mode = SND_PCM_START_GO; params.stop_mode = SND_PCM_STOP_ROLLOVER; params.buf.block.frag_size = fragsize; params.buf.block.frags_max = 1; params.buf.block.frags_min = 1; params.format.rate = 8000; //samplerate; params.format.interleave = 1; params.format.voices = channels; params.format.format = SND_PCM_SFMT_S16_LE; rtn = snd_pcm_plugin_params(hPcm_Mic.pDs, ¶ms); if (rtn < 0) { cli_print("snd_pcm_plugin_params (mic) failed %s", snd_strerror(rtn)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); return -1; } // get file descriptor for use with the select() call hPcm_Mic.hFD = snd_pcm_file_descriptor(hPcm_Mic.pDs, SND_PCM_CHANNEL_CAPTURE); if (hPcm_Mic.hFD < 0) { cli_print("snd_pcm_file_descriptor (mic) failed %s", snd_strerror(hPcm_Mic.hFD)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); return -1; } // Signal the driver to ready the capture channel rtn = snd_pcm_plugin_prepare(hPcm_Mic.pDs, SND_PCM_CHANNEL_CAPTURE); if (rtn < 0) { cli_print("snd_pcm_plugin_prepare (mic) failed %s", snd_strerror(errno)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); return -1; } fragsize = params.buf.block.frag_size; /************************ PLAYBACK **************************/ name = "voice"; // get and set audioman handle rtn = audio_manager_get_handle(AUDIO_TYPE_VIDEO_CHAT, getpid(), (bool)false, &hPcm_Spk.hAudioman); if (rtn < 0) { cli_print("audioman audio_manager_get_handle (spk) failed %s", strerror(-rtn) ); (void)snd_pcm_close(hPcm_Mic.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); return -1; } #ifdef HANDSET // set audio manager handle type rtn = audio_manager_set_handle_type(hPcm_Spk.hAudioman, AUDIO_TYPE_VIDEO_CHAT, device_type, device_type); if (rtn < 0) { (void)snd_pcm_close(hPcm_Mic.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); (void)audio_manager_free_handle(hPcm_Spk.hAudioman); cli_print("audio_manager_set_handle_type (spk) failed %s", strerror(-rtn)); return -1; } #endif // Create a handle and open a connection to an audio interface specified by name cli_print("Opening %s - for playback", name); rtn = snd_pcm_open_name(&hPcm_Spk.pDs, name, SND_PCM_OPEN_PLAYBACK); if (rtn < 0) { cli_print("snd_pcm_open_name (spk) failed %s", snd_strerror(rtn)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); (void)audio_manager_free_handle(hPcm_Spk.hAudioman); return -1; } rtn = snd_pcm_set_audioman_handle(hPcm_Spk.pDs, hPcm_Spk.hAudioman); if (rtn < 0) { cli_print("snd_pcm_set_audioman_handle (spk) failed %s", snd_strerror(rtn)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)snd_pcm_close(hPcm_Spk.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); (void)audio_manager_free_handle(hPcm_Spk.hAudioman); return -1; } // disable mmap (void)snd_pcm_plugin_set_disable(hPcm_Spk.pDs, (unsigned int)PLUGIN_DISABLE_MMAP); // set parameters memset(¶ms, 0, sizeof(params)); params.mode = SND_PCM_MODE_BLOCK; params.channel = SND_PCM_CHANNEL_PLAYBACK; params.start_mode = SND_PCM_START_GO; params.stop_mode = SND_PCM_STOP_ROLLOVER; params.buf.block.frag_size = fragsize; params.buf.block.frags_max = 1; params.buf.block.frags_min = 1; params.format.rate = samplerate; params.format.interleave = 1; params.format.voices = channels; params.format.format = SND_PCM_SFMT_S16_LE; // Set the configurable parameters for a PCM channel rtn = snd_pcm_plugin_params(hPcm_Spk.pDs, ¶ms); if (rtn < 0) { cli_print("snd_pcm_plugin_params (spk) failed %s", snd_strerror(rtn)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)snd_pcm_close(hPcm_Spk.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); (void)audio_manager_free_handle(hPcm_Spk.hAudioman); return -1; } // get file descriptor for use with the select() call hPcm_Spk.hFD = snd_pcm_file_descriptor(hPcm_Spk.pDs, SND_PCM_CHANNEL_PLAYBACK); if (hPcm_Spk.hFD < 0) { cli_print("snd_pcm_file_descriptor (spk) failed %s", snd_strerror(hPcm_Spk.hFD)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)snd_pcm_close(hPcm_Spk.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); (void)audio_manager_free_handle(hPcm_Spk.hAudioman); return -1; } // Signal the driver to ready the playback channel rtn = snd_pcm_plugin_prepare(hPcm_Spk.pDs, SND_PCM_CHANNEL_PLAYBACK); if (rtn < 0) { cli_print("snd_pcm_plugin_prepare (spk) failed %s", snd_strerror(rtn)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)snd_pcm_close(hPcm_Spk.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); (void)audio_manager_free_handle(hPcm_Spk.hAudioman); return -1; } fragsize = params.buf.block.frag_size; rtn = snd_pcm_capture_go(hPcm_Mic.pDs); if( rtn < 0) { cli_print("snd_pcm_capture_go (mic) failed %s", snd_strerror(rtn)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)snd_pcm_close(hPcm_Spk.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); (void)audio_manager_free_handle(hPcm_Spk.hAudioman); return -1; } rtn = snd_pcm_playback_go(hPcm_Spk.pDs); if (rtn < 0) { cli_print("snd_pcm_playback_go (spk) failed %s", snd_strerror(rtn)); (void)snd_pcm_close(hPcm_Mic.pDs); (void)snd_pcm_close(hPcm_Spk.pDs); (void)audio_manager_free_handle(hPcm_Mic.hAudioman); (void)audio_manager_free_handle(hPcm_Spk.hAudioman); return -1; } /******************* PLAYBACK/RECORD LOOP **************************/ while(!bQuit) { int width; struct timeval timeout; timeout.tv_sec = 0; timeout.tv_usec = 350000; // 350 ms FD_ZERO(&rfds); FD_ZERO(&wfds); FD_SET(hPcm_Mic.hFD, &rfds); FD_SET(hPcm_Spk.hFD, &wfds); width = ((hPcm_Spk.hFD > hPcm_Mic.hFD) ? hPcm_Spk.hFD : hPcm_Mic.hFD) + 1; rtn = select(width, &rfds, &wfds, NULL, &timeout); if (rtn > 0) { if (FD_ISSET(hPcm_Spk.hFD, &wfds)) { bQuit = process_write(hPcm_Spk.pDs); } if (FD_ISSET(hPcm_Mic.hFD, &rfds)) { bQuit = process_read(hPcm_Mic.pDs); } } else if (rtn == 0){ cli_print("select: timed out"); bQuit = true; } else { // (rtn < 0) cli_print("select: %s", strerror(errno)); bQuit = true; } } // Ensure audio processing is stopped if ((rtn = snd_pcm_plugin_playback_drain(hPcm_Spk.pDs)) < 0) { cli_print("snd_pcm_plugin_playback_drain (spk) failed %s", snd_strerror(rtn)); } if ((rtn = snd_pcm_plugin_flush(hPcm_Mic.pDs, SND_PCM_CHANNEL_CAPTURE)) < 0) { cli_print("snd_pcm_plugin_flush (mic) failed %s", snd_strerror(rtn)); } if ((rtn = snd_pcm_close(hPcm_Spk.pDs)) < 0) { cli_print("snd_pcm_close (spk) failed %s", snd_strerror(rtn)); } if ((rtn = snd_pcm_close(hPcm_Mic.pDs)) < 0) { cli_print("snd_pcm_close (mic) failed %s", snd_strerror(rtn)); } if (hPcm_Spk.hAudioman) { (void)audio_manager_free_handle(hPcm_Spk.hAudioman); } if (hPcm_Mic.hAudioman) { (void)audio_manager_free_handle(hPcm_Mic.hAudioman); } return 0; }
cst_audiodev *audio_open_alsa(int sps, int channels, cst_audiofmt fmt) { snd_pcm_channel_info_t pinfo; snd_pcm_channel_params_t params; snd_pcm_channel_setup_t setup; snd_pcm_t *pcm; cst_audiodev *ad; int err; #ifdef __QNXNTO__ if (snd_pcm_open_preferred(&pcm,&alsa_card,&alsa_device,SND_PCM_OPEN_PLAYBACK) < 0) { cst_errmsg("alsa_audio: failed to open audio device\n"); cst_error(); } if (snd_pcm_plugin_set_disable(pcm,PLUGIN_DISABLE_MMAP) < 0) { cst_errmsg("alsa_audio: failed to disable mmap\n"); snd_pcm_close(pcm); cst_error(); } #else if (snd_pcm_open(&pcm,alsa_card,alsa_device,SND_PCM_OPEN_PLAYBACK) < 0) { cst_errmsg("alsa_audio: failed to open audio device\n"); cst_error(); } #endif memset(&pinfo, 0, sizeof(pinfo)); memset(¶ms, 0, sizeof(params)); memset(&setup, 0, sizeof(setup)); pinfo.channel = SND_PCM_CHANNEL_PLAYBACK; snd_pcm_plugin_info(pcm,&pinfo); params.mode = SND_PCM_MODE_BLOCK; params.channel = SND_PCM_CHANNEL_PLAYBACK; params.start_mode = SND_PCM_START_DATA; params.stop_mode = SND_PCM_STOP_STOP; params.buf.block.frag_size = pinfo.max_fragment_size; params.buf.block.frags_max = 1; params.buf.block.frags_min = 1; params.format.interleave = 1; params.format.rate = sps; params.format.voices = channels; switch (fmt) { case CST_AUDIO_LINEAR16: if (CST_LITTLE_ENDIAN) params.format.format = SND_PCM_SFMT_S16_LE; else params.format.format = SND_PCM_SFMT_S16_BE; break; case CST_AUDIO_LINEAR8: params.format.format = SND_PCM_SFMT_U8; break; case CST_AUDIO_MULAW: params.format.format = SND_PCM_SFMT_MU_LAW; break; } if((err = snd_pcm_plugin_params(pcm,¶ms)) < 0) { cst_errmsg("alsa_audio params setting failed: %s\n",snd_strerror(err)); snd_pcm_close(pcm); cst_error(); } if((err = snd_pcm_plugin_setup(pcm,SND_PCM_CHANNEL_PLAYBACK)) > 0) { cst_errmsg("alsa_audio sound prepare setting failed: %s\n",snd_strerror(err)); snd_pcm_close(pcm); cst_error(); } if((err = snd_pcm_plugin_prepare(pcm,SND_PCM_CHANNEL_PLAYBACK)) > 0) { cst_errmsg("alsa_audio sound prepare setting failed: %s\n",snd_strerror(err)); snd_pcm_close(pcm); cst_error(); } pinfo.channel = SND_PCM_CHANNEL_PLAYBACK; snd_pcm_plugin_info(pcm,&pinfo); ad = cst_alloc(cst_audiodev, 1); ad->platform_data = pcm; ad->sps = ad->real_sps = sps; ad->channels = ad->real_channels = channels; ad->fmt = ad->real_fmt = fmt; return ad; }
int main (int argc, char **argv) { int card = -1; int dev = 0; snd_pcm_t *pcm_handle; FILE *file; wave_hdr wav_header; int samples; int sample_rate; int sample_channels; int sample_bits; char *sample_buffer; int fragsize = -1; int verbose = 0; int rtn; int final_return_code = -1; snd_pcm_channel_info_t pi; snd_mixer_t *mixer_handle; snd_mixer_group_t group; snd_pcm_channel_params_t pp; snd_pcm_channel_setup_t setup; int bsize, bytes_read, total_written = 0; fd_set rfds, wfds; uint32_t voice_mask[] = { 0, 0, 0, 0 }; snd_pcm_voice_conversion_t voice_conversion; int voice_override = 0; int num_frags = -1; char input_file[PATH_MAX]; char cwd[PATH_MAX]; screen_context_t screen_cxt; screen_window_t screen_win; screen_buffer_t screen_buf; int screen_fill_attribs[] = { SCREEN_BLIT_COLOR, COLOR_PURPLE, SCREEN_BLIT_END }; int screen_dirty[4] = { 0, 0, 1024, 600 }; //start with sane default values int idle_mode = SCREEN_IDLE_MODE_KEEP_AWAKE; int usage = SCREEN_USAGE_NATIVE; if (screen_create_context(&screen_cxt, 0) != 0) { return err("failed to create context"); } if (screen_create_window(&screen_win, screen_cxt) != 0) { err("failed to create window"); goto fail1; } if (screen_set_window_property_iv(screen_win, SCREEN_PROPERTY_USAGE, &usage) != 0) { err("failed to set native usage mode"); goto fail2; } if (screen_create_window_buffers(screen_win, 1) != 0) { err("failed to set native usage mode"); goto fail2; } if(screen_get_window_property_pv(screen_win, SCREEN_PROPERTY_RENDER_BUFFERS, (void **)&screen_buf) != 0) { err("failed to get screen buffer"); goto fail2; } if (screen_fill(screen_cxt, screen_buf, screen_fill_attribs) != 0) { err("failed to fill the screen"); goto fail3; } if (screen_get_window_property_iv(screen_win, SCREEN_PROPERTY_BUFFER_SIZE, screen_dirty+2) != 0) { err("failed to get window size"); goto fail3; } if (screen_set_window_property_iv(screen_win, SCREEN_PROPERTY_IDLE_MODE, &idle_mode) != 0) { err("failed to set idle mode"); goto fail3; } if (screen_post_window(screen_win, screen_buf, 1, screen_dirty, 0) != 0) { err("failed to post the window"); goto fail3; } if ((rtn = snd_pcm_open_preferred (&pcm_handle, &card, &dev, SND_PCM_OPEN_PLAYBACK)) < 0) { err ("device open"); goto fail3; } getcwd(cwd, PATH_MAX); rtn = snprintf(input_file, PATH_MAX, "%s%s", cwd, WAV_RELATIVE_PATH); if (rtn > PATH_MAX - 1) { err ("File name and path too long"); goto fail4; } if ((file = fopen (input_file, "r")) == 0) { err ("File open failed"); goto fail4; } if (check_hdr (file) == -1) { err ("check_hdr failed"); goto fail5; } samples = find_tag (file, "fmt "); fread (&wav_header, sizeof (wav_header), 1, file); fseek (file, (samples - sizeof (wave_hdr)), SEEK_CUR); sample_rate = ENDIAN_LE32 (wav_header.samples_per_sec); sample_channels = ENDIAN_LE16 (wav_header.channels); sample_bits = ENDIAN_LE16 (wav_header.bits_per_sample); printf ("SampleRate = %d, channels = %d, SampleBits = %d\n", sample_rate, sample_channels, sample_bits); /* disabling mmap is not actually required in this example but it is included to * demonstrate how it is used when it is required. */ if ((rtn = snd_pcm_plugin_set_disable (pcm_handle, PLUGIN_DISABLE_MMAP)) < 0) { fprintf (stderr, "snd_pcm_plugin_set_disable failed: %s\n", snd_strerror (rtn)); goto fail5; } memset (&pi, 0, sizeof (pi)); pi.channel = SND_PCM_CHANNEL_PLAYBACK; if ((rtn = snd_pcm_plugin_info (pcm_handle, &pi)) < 0) { fprintf (stderr, "snd_pcm_plugin_info failed: %s\n", snd_strerror (rtn)); goto fail5; } memset (&pp, 0, sizeof (pp)); pp.mode = SND_PCM_MODE_BLOCK; pp.channel = SND_PCM_CHANNEL_PLAYBACK; pp.start_mode = SND_PCM_START_FULL; pp.stop_mode = SND_PCM_STOP_STOP; pp.buf.block.frag_size = pi.max_fragment_size; if (fragsize != -1) { pp.buf.block.frag_size = fragsize; } pp.buf.block.frags_max = num_frags; pp.buf.block.frags_min = 1; pp.format.interleave = 1; pp.format.rate = sample_rate; pp.format.voices = sample_channels; if (ENDIAN_LE16 (wav_header.format_tag) == 6) pp.format.format = SND_PCM_SFMT_A_LAW; else if (ENDIAN_LE16 (wav_header.format_tag) == 7) pp.format.format = SND_PCM_SFMT_MU_LAW; else if (sample_bits == 8) pp.format.format = SND_PCM_SFMT_U8; else if (sample_bits == 24) pp.format.format = SND_PCM_SFMT_S24; else pp.format.format = SND_PCM_SFMT_S16_LE; strcpy (pp.sw_mixer_subchn_name, "Wave playback channel"); if ((rtn = snd_pcm_plugin_params (pcm_handle, &pp)) < 0) { fprintf (stderr, "snd_pcm_plugin_params failed: %s\n", snd_strerror (rtn)); goto fail5; } if ((rtn = snd_pcm_plugin_prepare (pcm_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0) { fprintf (stderr, "snd_pcm_plugin_prepare failed: %s\n", snd_strerror (rtn)); goto fail5; } if (voice_override) { snd_pcm_plugin_get_voice_conversion (pcm_handle, SND_PCM_CHANNEL_PLAYBACK, &voice_conversion); voice_conversion.matrix[0] = voice_mask[0]; voice_conversion.matrix[1] = voice_mask[1]; voice_conversion.matrix[2] = voice_mask[2]; voice_conversion.matrix[3] = voice_mask[3]; snd_pcm_plugin_set_voice_conversion (pcm_handle, SND_PCM_CHANNEL_PLAYBACK, &voice_conversion); } memset (&setup, 0, sizeof (setup)); memset (&group, 0, sizeof (group)); setup.channel = SND_PCM_CHANNEL_PLAYBACK; setup.mixer_gid = &group.gid; if ((rtn = snd_pcm_plugin_setup (pcm_handle, &setup)) < 0) { fprintf (stderr, "snd_pcm_plugin_setup failed: %s\n", snd_strerror (rtn)); goto fail5; } printf ("Format %s \n", snd_pcm_get_format_name (setup.format.format)); printf ("Frag Size %d \n", setup.buf.block.frag_size); printf ("Total Frags %d \n", setup.buf.block.frags); printf ("Rate %d \n", setup.format.rate); printf ("Voices %d \n", setup.format.voices); bsize = setup.buf.block.frag_size; if (group.gid.name[0] == 0) { fprintf (stderr, "Mixer Pcm Group [%s] Not Set \n", group.gid.name); goto fail5; } printf ("Mixer Pcm Group [%s]\n", group.gid.name); if ((rtn = snd_mixer_open (&mixer_handle, card, setup.mixer_device)) < 0) { fprintf (stderr, "snd_mixer_open failed: %s\n", snd_strerror (rtn)); goto fail5; } samples = find_tag (file, "data"); sample_buffer = malloc (bsize); FD_ZERO (&rfds); FD_ZERO (&wfds); bytes_read = 1; while (total_written < samples && bytes_read > 0) { if (tcgetpgrp (0) == getpid ()) FD_SET (STDIN_FILENO, &rfds); FD_SET (snd_mixer_file_descriptor (mixer_handle), &rfds); FD_SET (snd_pcm_file_descriptor (pcm_handle, SND_PCM_CHANNEL_PLAYBACK), &wfds); rtn = max (snd_mixer_file_descriptor (mixer_handle), snd_pcm_file_descriptor (pcm_handle, SND_PCM_CHANNEL_PLAYBACK)); if (select (rtn + 1, &rfds, &wfds, NULL, NULL) == -1) { err ("select"); goto fail6; } if (FD_ISSET (snd_pcm_file_descriptor (pcm_handle, SND_PCM_CHANNEL_PLAYBACK), &wfds)) { snd_pcm_channel_status_t status; int written = 0; if ((bytes_read = fread (sample_buffer, 1, min (samples - total_written, bsize), file)) <= 0) continue; written = snd_pcm_plugin_write (pcm_handle, sample_buffer, bytes_read); if (verbose) printf ("bytes written = %d \n", written); if (written < bytes_read) { memset (&status, 0, sizeof (status)); status.channel = SND_PCM_CHANNEL_PLAYBACK; if (snd_pcm_plugin_status (pcm_handle, &status) < 0) { fprintf (stderr, "underrun: playback channel status error\n"); goto fail6; } if (status.status == SND_PCM_STATUS_READY || status.status == SND_PCM_STATUS_UNDERRUN) { if (snd_pcm_plugin_prepare (pcm_handle, SND_PCM_CHANNEL_PLAYBACK) < 0) { fprintf (stderr, "underrun: playback channel prepare error\n"); goto fail6; } } if (written < 0) written = 0; written += snd_pcm_plugin_write (pcm_handle, sample_buffer + written, bytes_read - written); } total_written += written; } } bytes_read = snd_pcm_plugin_flush (pcm_handle, SND_PCM_CHANNEL_PLAYBACK); final_return_code = 0; fail6: rtn = snd_mixer_close (mixer_handle); fail5: fclose (file); fail4: rtn = snd_pcm_close (pcm_handle); fail3: screen_destroy_buffer(screen_buf); fail2: screen_destroy_window(screen_win); fail1: screen_destroy_context(screen_cxt); return final_return_code; }
static pj_status_t bb10_open_playback (struct bb10_stream *stream, const pjmedia_aud_param *param) { int ret = 0; snd_pcm_channel_info_t pi; snd_pcm_channel_setup_t setup; snd_mixer_group_t group; snd_pcm_channel_params_t pp; unsigned int rate; unsigned long tmp_buf_size; if (param->play_id < 0 || param->play_id >= stream->af->dev_cnt) { return PJMEDIA_EAUD_INVDEV; } PJ_ASSERT_RETURN(param->bits_per_sample == 16, PJMEDIA_EAUD_SAMPFORMAT); /* Use the bb10 audio manager API to open as opposed to QNX core audio * Echo cancellation built in */ if ((ret = audio_manager_snd_pcm_open_name( AUDIO_TYPE_VIDEO_CHAT, &stream->pb_pcm, &stream->pb_audio_manager_handle, (char*)"voice", SND_PCM_OPEN_PLAYBACK)) < 0) { TRACE_((THIS_FILE, "audio_manager_snd_pcm_open_name ret = %d", ret)); return PJMEDIA_EAUD_SYSERR; } /* Required call from January 2013 gold OS release */ snd_pcm_plugin_set_disable(stream->pb_pcm, PLUGIN_DISABLE_MMAP); /* Required call from January 2013 gold OS release */ snd_pcm_plugin_set_enable(stream->pb_pcm, PLUGIN_ROUTING); memset (&pi, 0, sizeof (pi)); pi.channel = SND_PCM_CHANNEL_PLAYBACK; if ((ret = snd_pcm_plugin_info (stream->pb_pcm, &pi)) < 0) { TRACE_((THIS_FILE, "snd_pcm_plugin_info ret = %d", ret)); return PJMEDIA_EAUD_SYSERR; } memset (&pp, 0, sizeof (pp)); /* Request VoIP compatible capabilities */ pp.mode = SND_PCM_MODE_BLOCK; pp.channel = SND_PCM_CHANNEL_PLAYBACK; pp.start_mode = SND_PCM_START_FULL; pp.stop_mode = SND_PCM_STOP_ROLLOVER; pp.buf.block.frag_size = param->samples_per_frame * param->bits_per_sample / 8; /* RIM recommends maximum of 5 */ pp.buf.block.frags_max = 5; pp.buf.block.frags_min = 1; pp.format.interleave = 1; pp.format.rate = param->clock_rate; pp.format.voices = param->channel_count; pp.format.format = get_alsa_pcm_fmt(param); /* Make the calls as per the wave sample */ if ((ret = snd_pcm_plugin_params (stream->pb_pcm, &pp)) < 0) { TRACE_((THIS_FILE, "snd_pcm_plugin_params ret = %d", ret)); return PJMEDIA_EAUD_SYSERR; } memset (&setup, 0, sizeof (setup)); memset (&group, 0, sizeof (group)); setup.channel = SND_PCM_CHANNEL_PLAYBACK; setup.mixer_gid = &group.gid; if ((ret = snd_pcm_plugin_setup (stream->pb_pcm, &setup)) < 0) { TRACE_((THIS_FILE, "snd_pcm_plugin_setup ret = %d", ret)); return PJMEDIA_EAUD_SYSERR; } if (group.gid.name[0] == 0) { return PJMEDIA_EAUD_SYSERR; } rate = param->clock_rate; /* Set the sound device buffer size and latency */ if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY) { tmp_buf_size = rate * param->output_latency_ms / 1000; } else { tmp_buf_size = rate * PJMEDIA_SND_DEFAULT_PLAY_LATENCY / 1000; } /* Set period size to samples_per_frame frames. */ stream->pb_frames = param->samples_per_frame / param->channel_count; stream->param.output_latency_ms = tmp_buf_size * 1000 / rate; /* Set our buffer */ stream->pb_buf_size = stream->pb_frames * param->channel_count * param->bits_per_sample / 8; stream->pb_buf = (char *) pj_pool_alloc(stream->pool, stream->pb_buf_size); TRACE_((THIS_FILE, "bb10_open_playback: pb_frames = %d clock = %d", stream->pb_frames, param->clock_rate)); return PJ_SUCCESS; }
static pj_status_t bb10_open_capture (struct bb10_stream *stream, const pjmedia_aud_param *param) { int ret = 0; unsigned int rate; unsigned long tmp_buf_size; int frame_size; snd_pcm_channel_info_t pi; snd_mixer_group_t group; snd_pcm_channel_params_t pp; snd_pcm_channel_setup_t setup; if (param->rec_id < 0 || param->rec_id >= stream->af->dev_cnt) return PJMEDIA_EAUD_INVDEV; PJ_ASSERT_RETURN(param->bits_per_sample == 16, PJMEDIA_EAUD_SAMPFORMAT); if ((ret=audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT, &stream->ca_pcm, &stream->ca_audio_manager_handle, (char*)"voice", SND_PCM_OPEN_CAPTURE)) < 0) { TRACE_((THIS_FILE, "audio_manager_snd_pcm_open_name ret = %d", ret)); return PJMEDIA_EAUD_SYSERR; } /* Required call from January 2013 gold OS release */ snd_pcm_plugin_set_disable (stream->ca_pcm, PLUGIN_DISABLE_MMAP); /* Required call from January 2013 gold OS release */ snd_pcm_plugin_set_enable(stream->ca_pcm, PLUGIN_ROUTING); /* sample reads the capabilities of the capture */ memset (&pi, 0, sizeof (pi)); pi.channel = SND_PCM_CHANNEL_CAPTURE; if ((ret = snd_pcm_plugin_info (stream->ca_pcm, &pi)) < 0) { TRACE_((THIS_FILE, "snd_pcm_plugin_info ret = %d", ret)); return PJMEDIA_EAUD_SYSERR; } /* Request the VoIP parameters * These parameters are different to waverec sample */ memset (&pp, 0, sizeof (pp)); /* Blocking read */ pp.mode = SND_PCM_MODE_BLOCK; pp.channel = SND_PCM_CHANNEL_CAPTURE; pp.start_mode = SND_PCM_START_FULL; /* Auto-recover from errors */ pp.stop_mode = SND_PCM_STOP_ROLLOVER; pp.buf.block.frag_size = param->samples_per_frame * param->bits_per_sample / 8; /* From January 2013 gold OS release. RIM recommend these for capture */ pp.buf.block.frags_max = 3; pp.buf.block.frags_min = 1; pp.format.interleave = 1; pp.format.rate = param->clock_rate; pp.format.voices = param->channel_count; pp.format.format = get_alsa_pcm_fmt(param); /* make the request */ if ((ret = snd_pcm_plugin_params (stream->ca_pcm, &pp)) < 0) { TRACE_((THIS_FILE, "snd_pcm_plugin_params ret = %d", ret)); return PJMEDIA_EAUD_SYSERR; } /* Again based on the sample */ memset (&setup, 0, sizeof (setup)); memset (&group, 0, sizeof (group)); setup.channel = SND_PCM_CHANNEL_CAPTURE; setup.mixer_gid = &group.gid; if ((ret = snd_pcm_plugin_setup (stream->ca_pcm, &setup)) < 0) { TRACE_((THIS_FILE, "snd_pcm_plugin_setup ret = %d", ret)); return PJMEDIA_EAUD_SYSERR; } frame_size = setup.buf.block.frag_size; if (group.gid.name[0] == 0) { } else { } frame_size = setup.buf.block.frag_size; PJ_UNUSED_ARG(frame_size); /* Warning about unused var */ /* Set clock rate */ rate = param->clock_rate; stream->ca_frames = (unsigned long) param->samples_per_frame / param->channel_count; /* Set the sound device buffer size and latency */ if (param->flags & PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY) { tmp_buf_size = rate * param->input_latency_ms / 1000; } else { tmp_buf_size = rate * PJMEDIA_SND_DEFAULT_REC_LATENCY / 1000; } stream->param.input_latency_ms = tmp_buf_size * 1000 / rate; /* Set our buffer */ stream->ca_buf_size = stream->ca_frames * param->channel_count * param->bits_per_sample / 8; stream->ca_buf = (char *)pj_pool_alloc (stream->pool, stream->ca_buf_size); TRACE_((THIS_FILE, "bb10_open_capture: ca_frames = %d clock = %d", stream->ca_frames, param->clock_rate)); return PJ_SUCCESS; }
static void capturesetup() { snd_pcm_channel_setup_t setup; int ret; snd_pcm_channel_info_t pi; snd_mixer_group_t group; snd_pcm_channel_params_t pp; int card = setup.mixer_card; if ((ret = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT, &g_pcm_handle_c, &g_audio_manager_handle_c, (char*) "voice", SND_PCM_OPEN_CAPTURE)) < 0) { return; } snd_pcm_plugin_set_disable(g_pcm_handle_c, PLUGIN_DISABLE_MMAP); snd_pcm_plugin_set_enable(g_pcm_handle_c, PLUGIN_ROUTING); // sample reads the capabilities of the capture memset(&pi, 0, sizeof(pi)); pi.channel = SND_PCM_CHANNEL_CAPTURE; if ((ret = snd_pcm_plugin_info(g_pcm_handle_c, &pi)) < 0) { fprintf(stderr, "snd_pcm_plugin_info failed: %s\n", snd_strerror (ret)); return; } fprintf(stderr,"CAPTURE Minimum Rate = %d\n",pi.min_rate); // Request the VoIP parameters // These parameters are different to waverec sample memset(&pp, 0, sizeof(pp)); fprintf(stderr,"CAPTURE Minimum fragment size = %d\n",pi.min_fragment_size); // Blocking read pp.mode = SND_PCM_MODE_BLOCK; pp.channel = SND_PCM_CHANNEL_CAPTURE; //pp.start_mode = SND_PCM_START_DATA; // Auto-recover from errors pp.stop_mode = SND_PCM_STOP_ROLLOVER; pp.start_mode = SND_PCM_START_DATA; pp.buf.block.frag_size = PREFERRED_FRAME_SIZE; pp.buf.block.frags_max = 3; pp.buf.block.frags_min = 1; pp.format.interleave = 1; pp.format.rate = VOIP_SAMPLE_RATE; pp.format.voices = 1; pp.format.format = SND_PCM_SFMT_S16_LE; // make the request if ((ret = snd_pcm_plugin_params(g_pcm_handle_c, &pp)) < 0) { fprintf(stderr, "ca snd_pcm_plugin_params failed: %s\n", snd_strerror (ret)); return; } // Again based on the sample memset(&setup, 0, sizeof(setup)); memset(&group, 0, sizeof(group)); setup.channel = SND_PCM_CHANNEL_CAPTURE; setup.mixer_gid = &group.gid; if ((ret = snd_pcm_plugin_setup(g_pcm_handle_c, &setup)) < 0) { fprintf(stderr, "snd_pcm_plugin_setup failed: %s\n", snd_strerror (ret)); return; } // On the simulator at least, our requested capabilities are accepted. fprintf(stderr,"CAPTURE Format %s card = %d\n", snd_pcm_get_format_name (setup.format.format),card); fprintf(stderr,"CAPTURE Rate %d \n", setup.format.rate); g_frame_size_c = setup.buf.block.frag_size; if (group.gid.name[0] == 0) { printf("Mixer Pcm Group [%s] Not Set \n", group.gid.name); printf("***>>>> Input Gain Controls Disabled <<<<*** \n"); } else { printf("Mixer Pcm Group [%s]\n", group.gid.name); } g_frame_size_c = setup.buf.block.frag_size; fprintf(stderr, "CAPTURE frame_size = %d\n", g_frame_size_c); // Sample calls prepare() if ((ret = snd_pcm_plugin_prepare(g_pcm_handle_c, SND_PCM_CHANNEL_CAPTURE)) < 0) { fprintf(stderr, "snd_pcm_plugin_prepare failed: %s\n", snd_strerror (ret)); } }
static void playsetup() { snd_pcm_channel_setup_t setup; //Enabling Echo canceller int ret; snd_pcm_channel_info_t pi; snd_mixer_group_t group; snd_pcm_channel_params_t pp; //audio routing enabled with AEC if ((ret = audio_manager_snd_pcm_open_name(AUDIO_TYPE_VIDEO_CHAT, &g_pcm_handle_p, &g_audio_manager_handle_p, (char*) "voice", SND_PCM_OPEN_PLAYBACK)) < 0) { return; } snd_pcm_plugin_set_disable(g_pcm_handle_p, PLUGIN_DISABLE_MMAP); snd_pcm_plugin_set_enable(g_pcm_handle_p, PLUGIN_ROUTING); memset(&pi, 0, sizeof(pi)); pi.channel = SND_PCM_CHANNEL_PLAYBACK; if ((ret = snd_pcm_plugin_info(g_pcm_handle_p, &pi)) < 0) { fprintf(stderr, "snd_pcm_plugin_info failed: %s\n", snd_strerror (ret)); return; } fprintf(stderr,"PLAY Minimum Rate = %d\n",pi.min_rate); // Interestingly on the simulator this returns 4096 but in reality always 170 is the result fprintf(stderr,"PLAY Minimum fragment size = %d\n",pi.min_fragment_size); memset(&pp, 0, sizeof(pp)); // Request VoIP compatible capabilities // On simulator frag_size is always negotiated to 170 pp.mode = SND_PCM_MODE_BLOCK; pp.channel = SND_PCM_CHANNEL_PLAYBACK; //pp.start_mode = SND_PCM_START_DATA; pp.stop_mode = SND_PCM_STOP_ROLLOVER; pp.start_mode = SND_PCM_START_FULL; pp.buf.block.frag_size = PREFERRED_FRAME_SIZE; // Increasing this internal buffer count delays write failure in the loop pp.buf.block.frags_max = 5; pp.buf.block.frags_min = 1; pp.format.interleave = 1; pp.format.rate = VOIP_SAMPLE_RATE; pp.format.voices = 1; pp.format.format = SND_PCM_SFMT_S16_LE; // Make the calls as per the wave sample if ((ret = snd_pcm_plugin_params(g_pcm_handle_p, &pp)) < 0) { fprintf(stderr, "pb snd_pcm_plugin_params failed: %s\n", snd_strerror (ret)); return; } memset(&setup, 0, sizeof(setup)); memset(&group, 0, sizeof(group)); setup.channel = SND_PCM_CHANNEL_PLAYBACK; setup.mixer_gid = &group.gid; if ((ret = snd_pcm_plugin_setup(g_pcm_handle_p, &setup)) < 0) { fprintf(stderr, "snd_pcm_plugin_setup failed: %s\n", snd_strerror (ret)); return; } fprintf(stderr,"PLAY frame_size %d \n", setup.buf.block.frag_size); fprintf(stderr,"PLAY Rate %d \n", setup.format.rate); g_frame_size_p = setup.buf.block.frag_size; if (group.gid.name[0] == 0) { fprintf(stderr,"FATAL Mixer Pcm Group [%s] Not Set \n", group.gid.name); exit (-1); } printf("Mixer Pcm Group [%s]\n", group.gid.name); if ((ret = snd_pcm_plugin_prepare(g_pcm_handle_p, SND_PCM_CHANNEL_PLAYBACK)) < 0) { fprintf(stderr, "snd_pcm_plugin_prepare failed: %s\n", snd_strerror (ret)); } }
static ALCenum qsa_open_capture(ALCdevice* device, const ALCchar* deviceName) { qsa_data *data; int card, dev; int format=-1; int status; data=(qsa_data*)calloc(1, sizeof(qsa_data)); if (data==NULL) { return ALC_OUT_OF_MEMORY; } if(!deviceName) deviceName = qsaDevice; if(strcmp(deviceName, qsaDevice) == 0) status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_CAPTURE); else { const DevMap *iter; if(VECTOR_SIZE(CaptureNameMap) == 0) deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap); #define MATCH_DEVNAME(iter) ((iter)->name && strcmp(deviceName, (iter)->name)==0) VECTOR_FIND_IF(iter, const DevMap, CaptureNameMap, MATCH_DEVNAME); #undef MATCH_DEVNAME if(iter == VECTOR_ITER_END(CaptureNameMap)) { free(data); return ALC_INVALID_DEVICE; } status = snd_pcm_open(&data->pcmHandle, iter->card, iter->dev, SND_PCM_OPEN_CAPTURE); } if(status < 0) { free(data); return ALC_INVALID_DEVICE; } data->audio_fd = snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE); if(data->audio_fd < 0) { snd_pcm_close(data->pcmHandle); free(data); return ALC_INVALID_DEVICE; } al_string_copy_cstr(&device->DeviceName, deviceName); device->ExtraData = data; switch (device->FmtType) { case DevFmtByte: format=SND_PCM_SFMT_S8; break; case DevFmtUByte: format=SND_PCM_SFMT_U8; break; case DevFmtShort: format=SND_PCM_SFMT_S16_LE; break; case DevFmtUShort: format=SND_PCM_SFMT_U16_LE; break; case DevFmtInt: format=SND_PCM_SFMT_S32_LE; break; case DevFmtUInt: format=SND_PCM_SFMT_U32_LE; break; case DevFmtFloat: format=SND_PCM_SFMT_FLOAT_LE; break; } /* we actually don't want to block on reads */ snd_pcm_nonblock_mode(data->pcmHandle, 1); /* Disable mmap to control data transfer to the audio device */ snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP); /* configure a sound channel */ memset(&data->cparams, 0, sizeof(data->cparams)); data->cparams.mode=SND_PCM_MODE_BLOCK; data->cparams.channel=SND_PCM_CHANNEL_CAPTURE; data->cparams.start_mode=SND_PCM_START_GO; data->cparams.stop_mode=SND_PCM_STOP_STOP; data->cparams.buf.block.frag_size=device->UpdateSize* ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType); data->cparams.buf.block.frags_max=device->NumUpdates; data->cparams.buf.block.frags_min=device->NumUpdates; data->cparams.format.interleave=1; data->cparams.format.rate=device->Frequency; data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans); data->cparams.format.format=format; if(snd_pcm_plugin_params(data->pcmHandle, &data->cparams) < 0) { snd_pcm_close(data->pcmHandle); free(data); device->ExtraData=NULL; return ALC_INVALID_VALUE; } return ALC_NO_ERROR; }
static int QSA_OpenDevice(_THIS, const char *devname, int iscapture) { int status = 0; int format = 0; SDL_AudioFormat test_format = 0; int found = 0; snd_pcm_channel_setup_t csetup; snd_pcm_channel_params_t cparams; /* Initialize all variables that we clean on shutdown */ this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof (struct SDL_PrivateAudioData))); if (this->hidden == NULL) { return SDL_OutOfMemory(); } SDL_memset(this->hidden, 0, sizeof(struct SDL_PrivateAudioData)); /* Initialize channel transfer parameters to default */ QSA_InitAudioParams(&cparams); /* Initialize channel direction: capture or playback */ this->hidden->iscapture = iscapture; /* Find deviceid and cardid by device name for playback */ if ((!this->hidden->iscapture) && (devname != NULL)) { uint32_t device; int32_t status; /* Search in the playback devices */ device = 0; do { status = SDL_strcmp(qsa_playback_device[device].name, devname); if (status == 0) { /* Found requested device */ this->hidden->deviceno = qsa_playback_device[device].deviceno; this->hidden->cardno = qsa_playback_device[device].cardno; break; } device++; if (device >= qsa_playback_devices) { QSA_CloseDevice(this); return SDL_SetError("No such playback device"); } } while (1); } /* Find deviceid and cardid by device name for capture */ if ((this->hidden->iscapture) && (devname != NULL)) { /* Search in the capture devices */ uint32_t device; int32_t status; /* Searching in the playback devices */ device = 0; do { status = SDL_strcmp(qsa_capture_device[device].name, devname); if (status == 0) { /* Found requested device */ this->hidden->deviceno = qsa_capture_device[device].deviceno; this->hidden->cardno = qsa_capture_device[device].cardno; break; } device++; if (device >= qsa_capture_devices) { QSA_CloseDevice(this); return SDL_SetError("No such capture device"); } } while (1); } /* Check if SDL requested default audio device */ if (devname == NULL) { /* Open system default audio device */ if (!this->hidden->iscapture) { status = snd_pcm_open_preferred(&this->hidden->audio_handle, &this->hidden->cardno, &this->hidden->deviceno, SND_PCM_OPEN_PLAYBACK); } else { status = snd_pcm_open_preferred(&this->hidden->audio_handle, &this->hidden->cardno, &this->hidden->deviceno, SND_PCM_OPEN_CAPTURE); } } else { /* Open requested audio device */ if (!this->hidden->iscapture) { status = snd_pcm_open(&this->hidden->audio_handle, this->hidden->cardno, this->hidden->deviceno, SND_PCM_OPEN_PLAYBACK); } else { status = snd_pcm_open(&this->hidden->audio_handle, this->hidden->cardno, this->hidden->deviceno, SND_PCM_OPEN_CAPTURE); } } /* Check if requested device is opened */ if (status < 0) { this->hidden->audio_handle = NULL; QSA_CloseDevice(this); return QSA_SetError("snd_pcm_open", status); } if (!QSA_CheckBuggyCards(this, QSA_MMAP_WORKAROUND)) { /* Disable QSA MMAP plugin for buggy audio drivers */ status = snd_pcm_plugin_set_disable(this->hidden->audio_handle, PLUGIN_DISABLE_MMAP); if (status < 0) { QSA_CloseDevice(this); return QSA_SetError("snd_pcm_plugin_set_disable", status); } } /* Try for a closest match on audio format */ format = 0; /* can't use format as SND_PCM_SFMT_U8 = 0 in qsa */ found = 0; for (test_format = SDL_FirstAudioFormat(this->spec.format); !found;) { /* if match found set format to equivalent QSA format */ switch (test_format) { case AUDIO_U8: { format = SND_PCM_SFMT_U8; found = 1; } break; case AUDIO_S8: { format = SND_PCM_SFMT_S8; found = 1; } break; case AUDIO_S16LSB: { format = SND_PCM_SFMT_S16_LE; found = 1; } break; case AUDIO_S16MSB: { format = SND_PCM_SFMT_S16_BE; found = 1; } break; case AUDIO_U16LSB: { format = SND_PCM_SFMT_U16_LE; found = 1; } break; case AUDIO_U16MSB: { format = SND_PCM_SFMT_U16_BE; found = 1; } break; case AUDIO_S32LSB: { format = SND_PCM_SFMT_S32_LE; found = 1; } break; case AUDIO_S32MSB: { format = SND_PCM_SFMT_S32_BE; found = 1; } break; case AUDIO_F32LSB: { format = SND_PCM_SFMT_FLOAT_LE; found = 1; } break; case AUDIO_F32MSB: { format = SND_PCM_SFMT_FLOAT_BE; found = 1; } break; default: { break; } } if (!found) { test_format = SDL_NextAudioFormat(); } } /* assumes test_format not 0 on success */ if (test_format == 0) { QSA_CloseDevice(this); return SDL_SetError("QSA: Couldn't find any hardware audio formats"); } this->spec.format = test_format; /* Set the audio format */ cparams.format.format = format; /* Set mono/stereo/4ch/6ch/8ch audio */ cparams.format.voices = this->spec.channels; /* Set rate */ cparams.format.rate = this->spec.freq; /* Setup the transfer parameters according to cparams */ status = snd_pcm_plugin_params(this->hidden->audio_handle, &cparams); if (status < 0) { QSA_CloseDevice(this); return QSA_SetError("snd_pcm_channel_params", status); } /* Make sure channel is setup right one last time */ SDL_memset(&csetup, 0, sizeof(csetup)); if (!this->hidden->iscapture) { csetup.channel = SND_PCM_CHANNEL_PLAYBACK; } else { csetup.channel = SND_PCM_CHANNEL_CAPTURE; } /* Setup an audio channel */ if (snd_pcm_plugin_setup(this->hidden->audio_handle, &csetup) < 0) { QSA_CloseDevice(this); return SDL_SetError("QSA: Unable to setup channel"); } /* Calculate the final parameters for this audio specification */ SDL_CalculateAudioSpec(&this->spec); this->hidden->pcm_len = this->spec.size; if (this->hidden->pcm_len == 0) { this->hidden->pcm_len = csetup.buf.block.frag_size * this->spec.channels * (snd_pcm_format_width(format) / 8); } /* * Allocate memory to the audio buffer and initialize with silence * (Note that buffer size must be a multiple of fragment size, so find * closest multiple) */ this->hidden->pcm_buf = (Uint8 *) SDL_AllocAudioMem(this->hidden->pcm_len); if (this->hidden->pcm_buf == NULL) { QSA_CloseDevice(this); return SDL_OutOfMemory(); } SDL_memset(this->hidden->pcm_buf, this->spec.silence, this->hidden->pcm_len); /* get the file descriptor */ if (!this->hidden->iscapture) { this->hidden->audio_fd = snd_pcm_file_descriptor(this->hidden->audio_handle, SND_PCM_CHANNEL_PLAYBACK); } else { this->hidden->audio_fd = snd_pcm_file_descriptor(this->hidden->audio_handle, SND_PCM_CHANNEL_CAPTURE); } if (this->hidden->audio_fd < 0) { QSA_CloseDevice(this); return QSA_SetError("snd_pcm_file_descriptor", status); } /* Prepare an audio channel */ if (!this->hidden->iscapture) { /* Prepare audio playback */ status = snd_pcm_plugin_prepare(this->hidden->audio_handle, SND_PCM_CHANNEL_PLAYBACK); } else { /* Prepare audio capture */ status = snd_pcm_plugin_prepare(this->hidden->audio_handle, SND_PCM_CHANNEL_CAPTURE); } if (status < 0) { QSA_CloseDevice(this); return QSA_SetError("snd_pcm_plugin_prepare", status); } /* We're really ready to rock and roll. :-) */ return 0; }
/***************************************************************************** * Open : creates a handle and opens an alsa device ***************************************************************************** * This function opens an alsa device, through the alsa API *****************************************************************************/ int E_(OpenAudio)( vlc_object_t *p_this ) { aout_instance_t *p_aout = (aout_instance_t *)p_this; int i_ret; int i_bytes_per_sample; int i_nb_channels; snd_pcm_channel_info_t pi; snd_pcm_channel_params_t pp; aout_instance_t *p_aout = (aout_instance_t *)p_this; /* allocate structure */ p_aout->output.p_sys = malloc( sizeof( aout_sys_t ) ); if( p_aout->output.p_sys == NULL ) { msg_Err( p_aout, "out of memory" ); return -1; } /* open audio device */ if( ( i_ret = snd_pcm_open_preferred( &p_aout->output.p_sys->p_pcm_handle, &p_aout->output.p_sys->i_card, &p_aout->output.p_sys->i_device, SND_PCM_OPEN_PLAYBACK ) ) < 0 ) { msg_Err( p_aout, "unable to open audio device (%s)", snd_strerror( i_ret ) ); free( p_aout->output.p_sys ); return -1; } /* disable mmap */ if( ( i_ret = snd_pcm_plugin_set_disable( p_aout->output.p_sys->p_pcm_handle, PLUGIN_DISABLE_MMAP ) ) < 0 ) { msg_Err( p_aout, "unable to disable mmap (%s)", snd_strerror(i_ret) ); E_(CloseAudio)( p_this ); free( p_aout->output.p_sys ); return -1; } p_aout->output.p_sys->p_silent_buffer = malloc( DEFAULT_FRAME_SIZE * 4 ); p_aout->output.pf_play = Play; aout_VolumeSoftInit( p_aout ); memset( &pi, 0, sizeof(pi) ); memset( &pp, 0, sizeof(pp) ); pi.channel = SND_PCM_CHANNEL_PLAYBACK; if( ( i_ret = snd_pcm_plugin_info( p_aout->output.p_sys->p_pcm_handle, &pi ) ) < 0 ) { msg_Err( p_aout, "unable to get plugin info (%s)", snd_strerror( i_ret ) ); E_(CloseAudio)( p_this ); free( p_aout->output.p_sys ); return -1; } pp.mode = SND_PCM_MODE_BLOCK; pp.channel = SND_PCM_CHANNEL_PLAYBACK; pp.start_mode = SND_PCM_START_FULL; pp.stop_mode = SND_PCM_STOP_STOP; pp.buf.block.frags_max = 3; pp.buf.block.frags_min = 1; pp.format.interleave = 1; pp.format.rate = p_aout->output.output.i_rate; i_nb_channels = aout_FormatNbChannels( &p_aout->output.output ); if ( i_nb_channels > 2 ) { /* I don't know if QNX supports more than two channels. */ i_nb_channels = 2; p_aout->output.output.i_channels = AOUT_CHAN_STEREO; } pp.format.voices = i_nb_channels; p_aout->output.output.i_format = AOUT_FMT_S16_NE; p_aout->output.i_nb_samples = DEFAULT_FRAME_SIZE; pp.format.format = SND_PCM_SFMT_S16; i_bytes_per_sample = 2; pp.buf.block.frag_size = p_aout->output.i_nb_samples * p_aout->output.output.i_channels * i_bytes_per_sample; /* set parameters */ if( ( i_ret = snd_pcm_plugin_params( p_aout->output.p_sys->p_pcm_handle, &pp ) ) < 0 ) { msg_Err( p_aout, "unable to set parameters (%s)", snd_strerror(i_ret) ); E_(CloseAudio)( p_this ); free( p_aout->output.p_sys ); return -1; } /* prepare channel */ if( ( i_ret = snd_pcm_plugin_prepare( p_aout->output.p_sys->p_pcm_handle, SND_PCM_CHANNEL_PLAYBACK ) ) < 0 ) { msg_Err( p_aout, "unable to prepare channel (%s)", snd_strerror( i_ret ) ); E_(CloseAudio)( p_this ); free( p_aout->output.p_sys ); return -1; } /* Create audio thread and wait for its readiness. */ if( vlc_thread_create( p_aout, "aout", QNXaoutThread, VLC_THREAD_PRIORITY_OUTPUT, VLC_FALSE ) ) { msg_Err( p_aout, "cannot create QNX audio thread (%s)", strerror(errno) ); E_(CloseAudio)( p_this ); free( p_aout->output.p_sys ); return -1; } return( 0 ); }
static ALCboolean qsa_reset_playback(ALCdevice* device) { qsa_data* data=(qsa_data*)device->ExtraData; int32_t format=-1; switch(device->FmtType) { case DevFmtByte: format=SND_PCM_SFMT_S8; break; case DevFmtUByte: format=SND_PCM_SFMT_U8; break; case DevFmtShort: format=SND_PCM_SFMT_S16_LE; break; case DevFmtUShort: format=SND_PCM_SFMT_U16_LE; break; case DevFmtInt: format=SND_PCM_SFMT_S32_LE; break; case DevFmtUInt: format=SND_PCM_SFMT_U32_LE; break; case DevFmtFloat: format=SND_PCM_SFMT_FLOAT_LE; break; } /* we actually don't want to block on writes */ snd_pcm_nonblock_mode(data->pcmHandle, 1); /* Disable mmap to control data transfer to the audio device */ snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP); snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_BUFFER_PARTIAL_BLOCKS); // configure a sound channel memset(&data->cparams, 0, sizeof(data->cparams)); data->cparams.channel=SND_PCM_CHANNEL_PLAYBACK; data->cparams.mode=SND_PCM_MODE_BLOCK; data->cparams.start_mode=SND_PCM_START_FULL; data->cparams.stop_mode=SND_PCM_STOP_STOP; data->cparams.buf.block.frag_size=device->UpdateSize* ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType); data->cparams.buf.block.frags_max=device->NumUpdates; data->cparams.buf.block.frags_min=device->NumUpdates; data->cparams.format.interleave=1; data->cparams.format.rate=device->Frequency; data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans); data->cparams.format.format=format; if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0) { int original_rate=data->cparams.format.rate; int original_voices=data->cparams.format.voices; int original_format=data->cparams.format.format; int it; int jt; for (it=0; it<1; it++) { /* Check for second pass */ if (it==1) { original_rate=ratelist[0].rate; original_voices=channellist[0].channels; original_format=formatlist[0].format; } do { /* At first downgrade sample format */ jt=0; do { if (formatlist[jt].format==data->cparams.format.format) { data->cparams.format.format=formatlist[jt+1].format; break; } if (formatlist[jt].format==0) { data->cparams.format.format=0; break; } jt++; } while(1); if (data->cparams.format.format==0) { data->cparams.format.format=original_format; /* At secod downgrade sample rate */ jt=0; do { if (ratelist[jt].rate==data->cparams.format.rate) { data->cparams.format.rate=ratelist[jt+1].rate; break; } if (ratelist[jt].rate==0) { data->cparams.format.rate=0; break; } jt++; } while(1); if (data->cparams.format.rate==0) { data->cparams.format.rate=original_rate; data->cparams.format.format=original_format; /* At third downgrade channels number */ jt=0; do { if(channellist[jt].channels==data->cparams.format.voices) { data->cparams.format.voices=channellist[jt+1].channels; break; } if (channellist[jt].channels==0) { data->cparams.format.voices=0; break; } jt++; } while(1); } if (data->cparams.format.voices==0) { break; } } data->cparams.buf.block.frag_size=device->UpdateSize* data->cparams.format.voices* snd_pcm_format_width(data->cparams.format.format)/8; data->cparams.buf.block.frags_max=device->NumUpdates; data->cparams.buf.block.frags_min=device->NumUpdates; if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0) { continue; } else { break; } } while(1); if (data->cparams.format.voices!=0) { break; } } if (data->cparams.format.voices==0) { return ALC_FALSE; } } if ((snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK))<0) { return ALC_FALSE; } memset(&data->csetup, 0, sizeof(data->csetup)); data->csetup.channel=SND_PCM_CHANNEL_PLAYBACK; if (snd_pcm_plugin_setup(data->pcmHandle, &data->csetup)<0) { return ALC_FALSE; } /* now fill back to the our AL device */ device->Frequency=data->cparams.format.rate; switch (data->cparams.format.voices) { case 1: device->FmtChans=DevFmtMono; break; case 2: device->FmtChans=DevFmtStereo; break; case 4: device->FmtChans=DevFmtQuad; break; case 6: device->FmtChans=DevFmtX51; break; case 7: device->FmtChans=DevFmtX61; break; case 8: device->FmtChans=DevFmtX71; break; default: device->FmtChans=DevFmtMono; break; } switch (data->cparams.format.format) { case SND_PCM_SFMT_S8: device->FmtType=DevFmtByte; break; case SND_PCM_SFMT_U8: device->FmtType=DevFmtUByte; break; case SND_PCM_SFMT_S16_LE: device->FmtType=DevFmtShort; break; case SND_PCM_SFMT_U16_LE: device->FmtType=DevFmtUShort; break; case SND_PCM_SFMT_S32_LE: device->FmtType=DevFmtInt; break; case SND_PCM_SFMT_U32_LE: device->FmtType=DevFmtUInt; break; case SND_PCM_SFMT_FLOAT_LE: device->FmtType=DevFmtFloat; break; default: device->FmtType=DevFmtShort; break; } SetDefaultChannelOrder(device); device->UpdateSize=data->csetup.buf.block.frag_size/ (ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType)); device->NumUpdates=data->csetup.buf.block.frags; data->size=data->csetup.buf.block.frag_size; data->buffer=malloc(data->size); if (!data->buffer) { return ALC_FALSE; } return ALC_TRUE; }
/* * General setup of the libasound audio mixer and pcm components. * Some of the settings are based on the format of the wav data. */ int setup_snd(const char * name) { int fragsize = -1; int num_frags = -1; int rtn; char *dev_name; if (NULL == name) { dev_name = "pcmPreferred"; } else { dev_name = (char *) name; } if ((rtn = snd_pcm_open_name(&pcm_handle, dev_name, SND_PCM_OPEN_PLAYBACK)) < 0) { snprintf(msg, MSG_SIZE, "snd_pcm_open_name failed: %s\n", snd_strerror(rtn)); show_dialog_message(msg); return FAILURE; } if ((rtn = snd_pcm_info(pcm_handle, &info)) < 0) { snprintf(msg, MSG_SIZE, "snd_pcm_info failed: %s\n", snd_strerror(rtn)); goto setup_failure; } card = info.card; /* disabling mmap is not actually required in this example but it is included to * demonstrate how it is used when it is required. */ if ((rtn = snd_pcm_plugin_set_disable(pcm_handle, PLUGIN_DISABLE_MMAP)) < 0) { snprintf(msg, MSG_SIZE, "snd_pcm_plugin_set_disable failed: %s\n", snd_strerror(rtn)); goto setup_failure; } memset(&pi, 0, sizeof(pi)); pi.channel = SND_PCM_CHANNEL_PLAYBACK; if ((rtn = snd_pcm_plugin_info(pcm_handle, &pi)) < 0) { snprintf(msg, MSG_SIZE, "snd_pcm_plugin_info failed: %s\n", snd_strerror(rtn)); goto setup_failure; } memset(&pp, 0, sizeof(pp)); pp.mode = SND_PCM_MODE_BLOCK; pp.channel = SND_PCM_CHANNEL_PLAYBACK; pp.start_mode = SND_PCM_START_FULL; pp.stop_mode = SND_PCM_STOP_STOP; pp.buf.block.frag_size = pi.max_fragment_size; if (fragsize != -1) { pp.buf.block.frag_size = fragsize; } pp.buf.block.frags_max = num_frags; pp.buf.block.frags_min = 1; pp.format.interleave = 1; pp.format.rate = sample_rate; pp.format.voices = sample_channels; if (ENDIAN_LE16(wav_header.format_tag) == 6) pp.format.format = SND_PCM_SFMT_A_LAW; else if (ENDIAN_LE16(wav_header.format_tag) == 7) pp.format.format = SND_PCM_SFMT_MU_LAW; else if (sample_bits == 8) pp.format.format = SND_PCM_SFMT_U8; else if (sample_bits == 24) pp.format.format = SND_PCM_SFMT_S24; else pp.format.format = SND_PCM_SFMT_S16_LE; strcpy(pp.sw_mixer_subchn_name, "Wave playback channel"); if ((rtn = snd_pcm_plugin_params(pcm_handle, &pp)) < 0) { snprintf(msg, MSG_SIZE, "snd_pcm_plugin_params failed: %s\n", snd_strerror(rtn)); goto setup_failure; } if ((rtn = snd_pcm_plugin_prepare(pcm_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0) { snprintf(msg, MSG_SIZE, "snd_pcm_plugin_prepare failed: %s\n", snd_strerror(rtn)); goto setup_failure; } memset(&setup, 0, sizeof(setup)); memset(&group, 0, sizeof(group)); setup.channel = SND_PCM_CHANNEL_PLAYBACK; setup.mixer_gid = &group.gid; if ((rtn = snd_pcm_plugin_setup(pcm_handle, &setup)) < 0) { snprintf(msg, MSG_SIZE, "snd_pcm_plugin_setup failed: %s\n", snd_strerror(rtn)); goto setup_failure; } if (group.gid.name[0] == 0) { snprintf(msg, MSG_SIZE, "Mixer Pcm Group [%s] Not Set \n", group.gid.name); goto setup_failure; } if ((rtn = snd_mixer_open(&mixer_handle, card, setup.mixer_device)) < 0) { snprintf(msg, MSG_SIZE, "snd_mixer_open failed: %s\n", snd_strerror(rtn)); goto setup_failure; } char tmp[MSG_SIZE]; snprintf(msg, MSG_SIZE, "Format %s \n", snd_pcm_get_format_name(setup.format.format)); snprintf(tmp, MSG_SIZE, "Frag Size %d \n", setup.buf.block.frag_size); strlcat(msg, tmp, MSG_SIZE); snprintf(tmp, MSG_SIZE, "Total Frags %d \n", setup.buf.block.frags); strlcat(msg, tmp, MSG_SIZE); snprintf(tmp, MSG_SIZE, "Rate %d \n", setup.format.rate); strlcat(msg, tmp, MSG_SIZE); snprintf(tmp, MSG_SIZE, "Voices %d \n", setup.format.voices); strlcat(msg, tmp, MSG_SIZE); snprintf(tmp, MSG_SIZE, "Mixer Pcm Group [%s]\n", group.gid.name); strlcat(msg, tmp, MSG_SIZE); show_dialog_message(msg); return SUCCESS; setup_failure: show_dialog_message(msg); snd_pcm_close(pcm_handle); return FAILURE; }
bool QnxAudioInput::open() { if (!m_format.isValid() || m_format.sampleRate() <= 0) { if (!m_format.isValid()) qWarning("QnxAudioInput: open error, invalid format."); else qWarning("QnxAudioInput: open error, invalid sample rate (%d).", m_format.sampleRate()); return false; } int errorCode = 0; int card = 0; int device = 0; if ((errorCode = snd_pcm_open_preferred(&m_pcmHandle, &card, &device, SND_PCM_OPEN_CAPTURE)) < 0) { qWarning("QnxAudioInput: open error, couldn't open card (0x%x)", -errorCode); return false; } // Necessary so that bytesFree() which uses the "free" member of the status struct works snd_pcm_plugin_set_disable(m_pcmHandle, PLUGIN_DISABLE_MMAP); snd_pcm_channel_info_t info; memset(&info, 0, sizeof(info)); info.channel = SND_PCM_CHANNEL_CAPTURE; if ((errorCode = snd_pcm_plugin_info(m_pcmHandle, &info)) < 0) { qWarning("QnxAudioInput: open error, couldn't get channel info (0x%x)", -errorCode); close(); return false; } snd_pcm_channel_params_t params = QnxAudioUtils::formatToChannelParams(m_format, QAudio::AudioInput, info.max_fragment_size); if ((errorCode = snd_pcm_plugin_params(m_pcmHandle, ¶ms)) < 0) { qWarning("QnxAudioInput: open error, couldn't set channel params (0x%x)", -errorCode); close(); return false; } if ((errorCode = snd_pcm_plugin_prepare(m_pcmHandle, SND_PCM_CHANNEL_CAPTURE)) < 0) { qWarning("QnxAudioInput: open error, couldn't prepare channel (0x%x)", -errorCode); close(); return false; } snd_pcm_channel_setup_t setup; memset(&setup, 0, sizeof(setup)); setup.channel = SND_PCM_CHANNEL_CAPTURE; if ((errorCode = snd_pcm_plugin_setup(m_pcmHandle, &setup)) < 0) { qWarning("QnxAudioInput: open error, couldn't get channel setup (0x%x)", -errorCode); close(); return false; } m_periodSize = qMin(2048, setup.buf.block.frag_size); m_clockStamp.restart(); m_timeStamp.restart(); m_elapsedTimeOffset = 0; m_totalTimeValue = 0; m_bytesRead = 0; m_pcmNotifier = new QSocketNotifier(snd_pcm_file_descriptor(m_pcmHandle, SND_PCM_CHANNEL_CAPTURE), QSocketNotifier::Read, this); connect(m_pcmNotifier, SIGNAL(activated(int)), SLOT(userFeed())); return true; }
static int NTO_OpenAudio(_THIS, SDL_AudioSpec* spec) { int rval; int format; Uint16 test_format; int found; audio_handle = NULL; this->enabled = 0; if (pcm_buf != NULL) { SDL_FreeAudioMem(pcm_buf); pcm_buf = NULL; } /* initialize channel transfer parameters to default */ NTO_InitAudioParams(&cparams); /* Open the audio device */ rval = snd_pcm_open_preferred(&audio_handle, &cardno, &deviceno, OPEN_FLAGS); if (rval < 0) { SDL_SetError("NTO_OpenAudio(): snd_pcm_open failed: %s\n", snd_strerror(rval)); return (-1); } if (!NTO_CheckBuggyCards(this, QSA_MMAP_WORKAROUND)) { /* enable count status parameter */ if ((rval = snd_pcm_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP)) < 0) { SDL_SetError("snd_pcm_plugin_set_disable failed: %s\n", snd_strerror(rval)); return (-1); } } /* Try for a closest match on audio format */ format = 0; /* can't use format as SND_PCM_SFMT_U8 = 0 in nto */ found = 0; for (test_format=SDL_FirstAudioFormat(spec->format); !found ;) { /* if match found set format to equivalent ALSA format */ switch (test_format) { case AUDIO_U8: format = SND_PCM_SFMT_U8; found = 1; break; case AUDIO_S8: format = SND_PCM_SFMT_S8; found = 1; break; case AUDIO_S16LSB: format = SND_PCM_SFMT_S16_LE; found = 1; break; case AUDIO_S16MSB: format = SND_PCM_SFMT_S16_BE; found = 1; break; case AUDIO_U16LSB: format = SND_PCM_SFMT_U16_LE; found = 1; break; case AUDIO_U16MSB: format = SND_PCM_SFMT_U16_BE; found = 1; break; default: break; } if (!found) { test_format = SDL_NextAudioFormat(); } } /* assumes test_format not 0 on success */ if (test_format == 0) { SDL_SetError("NTO_OpenAudio(): Couldn't find any hardware audio formats"); return (-1); } spec->format = test_format; /* Set the audio format */ cparams.format.format = format; /* Set mono or stereo audio (currently only two channels supported) */ cparams.format.voices = spec->channels; /* Set rate */ cparams.format.rate = spec->freq; /* Setup the transfer parameters according to cparams */ rval = snd_pcm_plugin_params(audio_handle, &cparams); if (rval < 0) { SDL_SetError("NTO_OpenAudio(): snd_pcm_channel_params failed: %s\n", snd_strerror(rval)); return (-1); } /* Make sure channel is setup right one last time */ SDL_memset(&csetup, 0x00, sizeof(csetup)); csetup.channel = SND_PCM_CHANNEL_PLAYBACK; if (snd_pcm_plugin_setup(audio_handle, &csetup) < 0) { SDL_SetError("NTO_OpenAudio(): Unable to setup playback channel\n"); return -1; } /* Calculate the final parameters for this audio specification */ SDL_CalculateAudioSpec(spec); pcm_len = spec->size; if (pcm_len==0) { pcm_len = csetup.buf.block.frag_size * spec->channels * (snd_pcm_format_width(format)/8); } /* Allocate memory to the audio buffer and initialize with silence (Note that buffer size must be a multiple of fragment size, so find closest multiple) */ pcm_buf = (Uint8*)SDL_AllocAudioMem(pcm_len); if (pcm_buf == NULL) { SDL_SetError("NTO_OpenAudio(): pcm buffer allocation failed\n"); return (-1); } SDL_memset(pcm_buf, spec->silence, pcm_len); /* get the file descriptor */ if ((audio_fd = snd_pcm_file_descriptor(audio_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0) { SDL_SetError("NTO_OpenAudio(): snd_pcm_file_descriptor failed with error code: %s\n", snd_strerror(rval)); return (-1); } /* Trigger audio playback */ rval = snd_pcm_plugin_prepare(audio_handle, SND_PCM_CHANNEL_PLAYBACK); if (rval < 0) { SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror(rval)); return (-1); } this->enabled = 1; /* Get the parent process id (we're the parent of the audio thread) */ parent = getpid(); /* We're really ready to rock and roll. :-) */ return (0); }
static int open_qsa(audio_output_t* ao) { int status; int cardno; int deviceno; int it; snd_pcm_t* audio_handle; qsa_internal_t* userptr; ao->userptr=NULL; status=snd_pcm_open_preferred(&audio_handle, &cardno, &deviceno, SND_PCM_OPEN_PLAYBACK); if (status<0) { return FALSE; } status=snd_pcm_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP); if (status<0) { return FALSE; } userptr=calloc(1, sizeof(qsa_internal_t)); if (userptr==NULL) { return FALSE; } ao->userptr=userptr; userptr->audio_handle=audio_handle; userptr->cardno=cardno; userptr->deviceno=deviceno; memset(&userptr->cpars, 0, sizeof(userptr->cpars)); userptr->cpars.channel=SND_PCM_CHANNEL_PLAYBACK; userptr->cpars.mode=SND_PCM_MODE_BLOCK; userptr->cpars.start_mode=SND_PCM_START_DATA; userptr->cpars.stop_mode=SND_PCM_STOP_STOP; userptr->cpars.format.format=0; it=0; do { if ((format_map[it].qsa_format==0) && (format_map[it].mp_format==0)) { break; } if (ao->format==format_map[it].mp_format) { userptr->cpars.format.format=format_map[it].qsa_format; break; } it++; } while(1); userptr->cpars.format.interleave=1; userptr->cpars.format.rate=ao->rate; userptr->cpars.format.voices=ao->channels; userptr->cpars.buf.block.frag_size=4096; userptr->cpars.buf.block.frags_min=8; userptr->cpars.buf.block.frags_max=16; if ((ao->channels!=-1) && (ao->rate!=-1)) { status=snd_pcm_plugin_params(userptr->audio_handle, &userptr->cpars); if (status<0) { return FALSE; } status=snd_pcm_plugin_prepare(userptr->audio_handle, SND_PCM_CHANNEL_PLAYBACK); if (status<0) { return FALSE; } } return TRUE; }
// Initialize audio and report back a few parameters of the channel int setupAudio( ) { int error; snd_pcm_channel_info_t pi; snd_pcm_channel_params_t pp; snd_pcm_channel_setup_t ps; //if ((error = snd_pcm_open_name (&playback_handle, "pcmPreferred", SND_PCM_OPEN_PLAYBACK)) < 0) { if ((error = snd_pcm_open_name (&playback_handle, "tones", SND_PCM_OPEN_PLAYBACK)) < 0) { slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to open preferred audio driver\n"); return -error; } sample_handle = snd_pcm_file_descriptor( playback_handle, SND_PCM_CHANNEL_PLAYBACK ); if ((error = snd_pcm_plugin_set_disable (playback_handle, PLUGIN_DISABLE_MMAP)) < 0) { slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to disable mmap\n"); snd_pcm_close(playback_handle); return -error; } if ((error = snd_pcm_plugin_set_enable (playback_handle, PLUGIN_ROUTING)) < 0) { slogf ( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "snd_pcm_plugin_set_enable failed: %s\n", snd_strerror (error)); snd_pcm_close(playback_handle); return -error; } // Find what sample rate and format to use memset( &pi, 0, sizeof(pi) ); pi.channel = SND_PCM_CHANNEL_PLAYBACK; if ((error = snd_pcm_channel_info (playback_handle, &pi)) < 0) { slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to get plugin info\n"); snd_pcm_close(playback_handle); return -error; } // Initialize the playback channel memset(&pp, 0, sizeof(pp) ); pp.mode = SND_PCM_MODE_BLOCK; pp.channel = SND_PCM_CHANNEL_PLAYBACK; pp.start_mode = SND_PCM_START_FULL; pp.stop_mode = SND_PCM_STOP_STOP; pp.buf.block.frag_size = pi.max_fragment_size; pp.buf.block.frags_max = 3; pp.buf.block.frags_min = 1; pp.format.interleave = 1; pp.format.format = SND_PCM_SFMT_S16_LE; pp.format.rate = 48000; pp.format.voices = 2; memset( &ps, 0, sizeof( ps ) ); memset( &group, 0, sizeof( group ) ); ps.channel = SND_PCM_CHANNEL_PLAYBACK; ps.mixer_gid = &group.gid; //strcpy(pp.sw_mixer_subchn_name, "Wave playback channel"); strcpy(pp.sw_mixer_subchn_name, "voice_ringtone"); if ((error = snd_pcm_plugin_params( playback_handle, &pp)) < 0) { slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to set plugin params\n"); snd_pcm_close(playback_handle); return -error; } if ((error = snd_pcm_plugin_prepare (playback_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0) { slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to set prepare plugin\n"); snd_pcm_close(playback_handle); return -error; } if ((error = snd_pcm_plugin_setup (playback_handle, &ps)) < 0) { slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to setup plugin\n"); snd_pcm_close(playback_handle); return -error; } card = ps.mixer_card; device = ps.mixer_device; if( audioman_handle ) { if ((error = snd_pcm_set_audioman_handle (playback_handle, audioman_handle)) < 0) { slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to set audioman handle \n"); snd_pcm_close(playback_handle); return -error; } } sample_frequency = ps.format.rate; frag_samples = ps.buf.block.frag_size / 4; frame_size = ps.buf.block.frag_size; format = pp.format.format; /* // Only support 16-bit samples for now frag_buffer = malloc( ps.buf.block.frag_size * 2 ); if( frag_buffer == NULL ) { slogf( _SLOG_SETCODE(_SLOGC_AUDIO, 0), _SLOG_CRITICAL, "Unable to alloc frag buffer\n"); snd_pcm_close(playback_handle); return ENOMEM; } */ fprintf (stderr,"Playback Format %s card = %d\n", snd_pcm_get_format_name (ps.format.format),card); fprintf (stderr,"Playback preferred frame_size %d \n", pi.max_fragment_size); fprintf (stderr,"Playback frame_size %d \n", ps.buf.block.frag_size); fprintf (stderr,"Playback frame_samples %d \n", frag_samples); fprintf (stderr,"Playback Rate %d \n", ps.format.rate); if (group.gid.name[0] == 0) { fprintf (stderr,"Playback Mixer Pcm Group [%s] Not Set \n", group.gid.name); return -1; } fprintf (stderr, "Playback Mixer Pcm Group [%s]\n", group.gid.name); return EOK; }