static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec) { GstAlsaSrc *alsa; gint err; alsa = GST_ALSA_SRC (asrc); if (!alsasrc_parse_spec (alsa, spec)) goto spec_parse; CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block); CHECK (set_hwparams (alsa), hw_params_failed); CHECK (set_swparams (alsa), sw_params_failed); CHECK (snd_pcm_prepare (alsa->handle), prepare_failed); alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info); spec->segsize = alsa->period_size * alsa->bpf; spec->segtotal = alsa->buffer_size / alsa->period_size; { snd_output_t *out_buf = NULL; char *msg = NULL; snd_output_buffer_open (&out_buf); snd_pcm_dump_hw_setup (alsa->handle, out_buf); snd_output_buffer_string (out_buf, &msg); GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg); snd_output_close (out_buf); snd_output_buffer_open (&out_buf); snd_pcm_dump_sw_setup (alsa->handle, out_buf); snd_output_buffer_string (out_buf, &msg); GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg); snd_output_close (out_buf); } #ifdef SND_CHMAP_API_VERSION if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9) { snd_pcm_chmap_t *chmap = snd_pcm_get_chmap (alsa->handle); if (chmap && chmap->channels == alsa->channels) { GstAudioChannelPosition pos[8]; if (alsa_chmap_to_channel_positions (chmap, pos)) gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC (alsa)->ringbuffer, pos); } free (chmap); } #endif /* SND_CHMAP_API_VERSION */ return TRUE; /* ERRORS */ spec_parse: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Error parsing spec")); return FALSE; } non_block: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Could not set device to blocking: %s", snd_strerror (err))); return FALSE; } hw_params_failed: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Setting of hwparams failed: %s", snd_strerror (err))); return FALSE; } sw_params_failed: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Setting of swparams failed: %s", snd_strerror (err))); return FALSE; } prepare_failed: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Prepare failed: %s", snd_strerror (err))); return FALSE; } }
static gboolean gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec) { pa_buffer_attr wanted; const pa_buffer_attr *actual; GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc); pa_stream_flags_t flags; pa_operation *o; GstAudioClock *clock; pa_threaded_mainloop_lock (pulsesrc->mainloop); { GstAudioRingBufferSpec s = *spec; const pa_channel_map *m; m = pa_stream_get_channel_map (pulsesrc->stream); gst_pulse_channel_map_to_gst (m, &s); gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC (pulsesrc)->ringbuffer, s.info.position); } /* enable event notifications */ GST_LOG_OBJECT (pulsesrc, "subscribing to context events"); if (!(o = pa_context_subscribe (pulsesrc->context, PA_SUBSCRIPTION_MASK_SOURCE_OUTPUT, NULL, NULL))) { GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("pa_context_subscribe() failed: %s", pa_strerror (pa_context_errno (pulsesrc->context))), (NULL)); goto unlock_and_fail; } pa_operation_unref (o); wanted.maxlength = -1; wanted.tlength = -1; wanted.prebuf = 0; wanted.minreq = -1; wanted.fragsize = spec->segsize; GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength); GST_INFO_OBJECT (pulsesrc, "tlength: %d", wanted.tlength); GST_INFO_OBJECT (pulsesrc, "prebuf: %d", wanted.prebuf); GST_INFO_OBJECT (pulsesrc, "minreq: %d", wanted.minreq); GST_INFO_OBJECT (pulsesrc, "fragsize: %d", wanted.fragsize); flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY | PA_STREAM_START_CORKED; if (pulsesrc->mute_set && pulsesrc->mute) flags |= PA_STREAM_START_MUTED; if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted, flags) < 0) { goto connect_failed; } /* our clock will now start from 0 again */ clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SRC (pulsesrc)->clock); gst_audio_clock_reset (clock, 0); pulsesrc->corked = TRUE; for (;;) { pa_stream_state_t state; state = pa_stream_get_state (pulsesrc->stream); if (!PA_STREAM_IS_GOOD (state)) goto stream_is_bad; if (state == PA_STREAM_READY) break; /* Wait until the stream is ready */ pa_threaded_mainloop_wait (pulsesrc->mainloop); } pulsesrc->stream_connected = TRUE; /* store the source output index so it can be accessed via a property */ pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream); g_object_notify (G_OBJECT (pulsesrc), "source-output-index"); if (pulsesrc->volume_set) { gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume); pulsesrc->volume_set = FALSE; } /* get the actual buffering properties now */ actual = pa_stream_get_buffer_attr (pulsesrc->stream); GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength); GST_INFO_OBJECT (pulsesrc, "tlength: %d (wanted: %d)", actual->tlength, wanted.tlength); GST_INFO_OBJECT (pulsesrc, "prebuf: %d", actual->prebuf); GST_INFO_OBJECT (pulsesrc, "minreq: %d (wanted %d)", actual->minreq, wanted.minreq); GST_INFO_OBJECT (pulsesrc, "fragsize: %d (wanted %d)", actual->fragsize, wanted.fragsize); if (actual->fragsize >= wanted.fragsize) { spec->segsize = actual->fragsize; } else { spec->segsize = actual->fragsize * (wanted.fragsize / actual->fragsize); } spec->segtotal = actual->maxlength / spec->segsize; if (!pulsesrc->paused) { GST_DEBUG_OBJECT (pulsesrc, "uncorking because we are playing"); gst_pulsesrc_set_corked (pulsesrc, FALSE, FALSE); } pa_threaded_mainloop_unlock (pulsesrc->mainloop); return TRUE; /* ERRORS */ connect_failed: { GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect stream: %s", pa_strerror (pa_context_errno (pulsesrc->context))), (NULL)); goto unlock_and_fail; } stream_is_bad: { GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect stream: %s", pa_strerror (pa_context_errno (pulsesrc->context))), (NULL)); goto unlock_and_fail; } unlock_and_fail: { gst_pulsesrc_destroy_stream (pulsesrc); pa_threaded_mainloop_unlock (pulsesrc->mainloop); return FALSE; } }
static gboolean alsasrc_parse_spec (GstAlsaSrc * alsa, GstAudioRingBufferSpec * spec) { switch (spec->type) { case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW: switch (GST_AUDIO_INFO_FORMAT (&spec->info)) { case GST_AUDIO_FORMAT_U8: alsa->format = SND_PCM_FORMAT_U8; break; case GST_AUDIO_FORMAT_S8: alsa->format = SND_PCM_FORMAT_S8; break; case GST_AUDIO_FORMAT_S16LE: alsa->format = SND_PCM_FORMAT_S16_LE; break; case GST_AUDIO_FORMAT_S16BE: alsa->format = SND_PCM_FORMAT_S16_BE; break; case GST_AUDIO_FORMAT_U16LE: alsa->format = SND_PCM_FORMAT_U16_LE; break; case GST_AUDIO_FORMAT_U16BE: alsa->format = SND_PCM_FORMAT_U16_BE; break; case GST_AUDIO_FORMAT_S24_32LE: alsa->format = SND_PCM_FORMAT_S24_LE; break; case GST_AUDIO_FORMAT_S24_32BE: alsa->format = SND_PCM_FORMAT_S24_BE; break; case GST_AUDIO_FORMAT_U24_32LE: alsa->format = SND_PCM_FORMAT_U24_LE; break; case GST_AUDIO_FORMAT_U24_32BE: alsa->format = SND_PCM_FORMAT_U24_BE; break; case GST_AUDIO_FORMAT_S32LE: alsa->format = SND_PCM_FORMAT_S32_LE; break; case GST_AUDIO_FORMAT_S32BE: alsa->format = SND_PCM_FORMAT_S32_BE; break; case GST_AUDIO_FORMAT_U32LE: alsa->format = SND_PCM_FORMAT_U32_LE; break; case GST_AUDIO_FORMAT_U32BE: alsa->format = SND_PCM_FORMAT_U32_BE; break; case GST_AUDIO_FORMAT_S24LE: alsa->format = SND_PCM_FORMAT_S24_3LE; break; case GST_AUDIO_FORMAT_S24BE: alsa->format = SND_PCM_FORMAT_S24_3BE; break; case GST_AUDIO_FORMAT_U24LE: alsa->format = SND_PCM_FORMAT_U24_3LE; break; case GST_AUDIO_FORMAT_U24BE: alsa->format = SND_PCM_FORMAT_U24_3BE; break; case GST_AUDIO_FORMAT_S20LE: alsa->format = SND_PCM_FORMAT_S20_3LE; break; case GST_AUDIO_FORMAT_S20BE: alsa->format = SND_PCM_FORMAT_S20_3BE; break; case GST_AUDIO_FORMAT_U20LE: alsa->format = SND_PCM_FORMAT_U20_3LE; break; case GST_AUDIO_FORMAT_U20BE: alsa->format = SND_PCM_FORMAT_U20_3BE; break; case GST_AUDIO_FORMAT_S18LE: alsa->format = SND_PCM_FORMAT_S18_3LE; break; case GST_AUDIO_FORMAT_S18BE: alsa->format = SND_PCM_FORMAT_S18_3BE; break; case GST_AUDIO_FORMAT_U18LE: alsa->format = SND_PCM_FORMAT_U18_3LE; break; case GST_AUDIO_FORMAT_U18BE: alsa->format = SND_PCM_FORMAT_U18_3BE; break; case GST_AUDIO_FORMAT_F32LE: alsa->format = SND_PCM_FORMAT_FLOAT_LE; break; case GST_AUDIO_FORMAT_F32BE: alsa->format = SND_PCM_FORMAT_FLOAT_BE; break; case GST_AUDIO_FORMAT_F64LE: alsa->format = SND_PCM_FORMAT_FLOAT64_LE; break; case GST_AUDIO_FORMAT_F64BE: alsa->format = SND_PCM_FORMAT_FLOAT64_BE; break; default: goto error; } break; case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW: alsa->format = SND_PCM_FORMAT_A_LAW; break; case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW: alsa->format = SND_PCM_FORMAT_MU_LAW; break; default: goto error; } alsa->rate = GST_AUDIO_INFO_RATE (&spec->info); alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info); alsa->buffer_time = spec->buffer_time; alsa->period_time = spec->latency_time; alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED; if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9) gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC (alsa)->ringbuffer, alsa_position[alsa->channels - 1]); return TRUE; /* ERRORS */ error: { return FALSE; } }
static gboolean gst_alsasink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec) { GstAlsaSink *alsa; gint err; alsa = GST_ALSA_SINK (asink); if (alsa->iec958) { snd_pcm_close (alsa->handle); alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa), alsa->device); if (G_UNLIKELY (!alsa->handle)) { goto no_iec958; } } if (!alsasink_parse_spec (alsa, spec)) goto spec_parse; CHECK (set_hwparams (alsa), hw_params_failed); CHECK (set_swparams (alsa), sw_params_failed); alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info); spec->segsize = alsa->period_size * alsa->bpf; spec->segtotal = alsa->buffer_size / alsa->period_size; { snd_output_t *out_buf = NULL; char *msg = NULL; snd_output_buffer_open (&out_buf); snd_pcm_dump_hw_setup (alsa->handle, out_buf); snd_output_buffer_string (out_buf, &msg); GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg); snd_output_close (out_buf); snd_output_buffer_open (&out_buf); snd_pcm_dump_sw_setup (alsa->handle, out_buf); snd_output_buffer_string (out_buf, &msg); GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg); snd_output_close (out_buf); } #ifdef SND_CHMAP_API_VERSION if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9) { snd_pcm_chmap_t *chmap = snd_pcm_get_chmap (alsa->handle); if (chmap && chmap->channels == alsa->channels) { GstAudioChannelPosition pos[8]; if (alsa_chmap_to_channel_positions (chmap, pos)) gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK (alsa)->ringbuffer, pos); } free (chmap); } #endif /* SND_CHMAP_API_VERSION */ return TRUE; /* ERRORS */ no_iec958: { GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL), ("Could not open IEC958 (SPDIF) device for playback")); return FALSE; } spec_parse: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Error parsing spec")); return FALSE; } hw_params_failed: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Setting of hwparams failed: %s", snd_strerror (err))); return FALSE; } sw_params_failed: { GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), ("Setting of swparams failed: %s", snd_strerror (err))); return FALSE; } }