[alsa-devel] [PATCH 6/8] add PCM interface
o-takashi at sakamocchi.jp
o-takashi at sakamocchi.jp
Sat Jun 1 17:55:56 CEST 2013
From: Takashi Sakamoto <o-takashi at sakamocchi.jp>
This file includes PCM callbacks.
With PCM rules, the number of channels and sampling rate are automatically
arranged according to device property.
Currently this module deal with digital and analog interface in the same PCM
device. So there is no way for ALSA application to distinguish these interfaces.
Signed-off-by: Takashi Sakamoto <o-takashi at sakamocchi.jp>
---
sound/firewire/fireworks/fireworks_pcm.c | 530 ++++++++++++++++++++++++++++++
1 file changed, 530 insertions(+)
create mode 100644 sound/firewire/fireworks/fireworks_pcm.c
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
new file mode 100644
index 0000000..e97bfd8
--- /dev/null
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -0,0 +1,530 @@
+/*
+ * fireworks_pcm.c - driver for Firewire devices from Echo Digital Audio
+ *
+ * Copyright (c) 2009-2010 Clemens Ladisch
+ * Copyright (c) 2013 Takashi Sakamoto
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this driver; if not, see <http://www.gnu.org/licenses/>.
+ */
+#include "./fireworks.h"
+
+/*
+ * NOTE:
+ * Fireworks changes its PCM channels according to its sampling rate.
+ * There are three modes. Here "capture" or "playback" is appplied to XXX.
+ * 0: 32.0- 48.0 kHz then snd_efw_hwinfo.nb_1394_XXX_channels applied
+ * 1: 88.2- 96.0 kHz then snd_efw_hwinfo.nb_1394_XXX_channels_2x applied
+ * 2: 176.4-192.0 kHz then snd_efw_hwinfo.nb_1394_XXX_channels_4x applied
+ *
+ * Then the number of PCM channels for analog input and output are always fixed
+ * but the number of PCM channels for digital input and output are differed.
+ *
+ * Additionally, according to AudioFire Owner's Manual Version 2.2,
+ * the number of PCM channels for digital input has more restriction
+ * depending on which digital interface is selected.
+ * - S/PDIF coaxial and optical : use input 1-2
+ * - ADAT optical with 32.0-48.0 kHz : use input 1-8
+ * - ADAT optical with 88.2-96.0 kHz : use input 1-4 (S/MUX format)
+ * If these restriction is applied, the number of channels in stream is decided
+ * according to above modes.
+ *
+ * Currently this module doesn't have rules for the latter.
+ */
+static unsigned int freq_table[] = {
+ /* multiplier mode 0 */
+ [0] = 32000,
+ [1] = 44100,
+ [2] = 48000,
+ /* multiplier mode 1 */
+ [3] = 88200,
+ [4] = 96000,
+ /* multiplier mode 2 */
+ [5] = 176400,
+ [6] = 192000,
+};
+
+static inline int
+get_multiplier_mode_with_index(int index)
+{
+ return ((int)index - 1) / 2;
+}
+
+int snd_efw_get_multiplier_mode(int sampling_rate)
+{
+ int i;
+ for (i = 0; i < sizeof(freq_table); i += 1)
+ if (freq_table[i] == sampling_rate)
+ return get_multiplier_mode_with_index(i);
+
+ return -1;
+}
+
+static int
+hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule,
+ struct snd_efw *efw, unsigned int *channels)
+{
+ struct snd_interval *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ const struct snd_interval *c =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1
+ };
+ unsigned int rate_bit;
+ int mode, i;
+
+ for (i = 0; i < ARRAY_SIZE(freq_table); i += 1) {
+ /* skip unsupported sampling rate */
+ rate_bit = snd_pcm_rate_to_rate_bit(freq_table[i]);
+ if (!(efw->supported_sampling_rate & rate_bit))
+ continue;
+
+ mode = get_multiplier_mode_with_index(i);
+ if (!snd_interval_test(c, channels[mode]))
+ continue;
+
+ t.min = min(t.min, freq_table[i]);
+ t.max = max(t.max, freq_table[i]);
+
+ }
+
+ return snd_interval_refine(r, &t);
+}
+
+static int
+hw_rule_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule,
+ struct snd_efw *efw, unsigned int *channels)
+{
+ struct snd_interval *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ const struct snd_interval *r =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1
+ };
+
+ unsigned int rate_bit;
+ int mode, i;
+
+ for (i = 0; i < ARRAY_SIZE(freq_table); i += 1) {
+ /* skip unsupported sampling rate */
+ rate_bit = snd_pcm_rate_to_rate_bit(freq_table[i]);
+ if (!(efw->supported_sampling_rate & rate_bit))
+ continue;
+
+ mode = get_multiplier_mode_with_index(i);
+ if (!snd_interval_test(r, freq_table[i]))
+ continue;
+
+ t.min = min(t.min, channels[mode]);
+ t.max = max(t.max, channels[mode]);
+
+ }
+
+ return snd_interval_refine(c, &t);
+}
+
+static inline int
+hw_rule_capture_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_efw *efw = rule->private;
+ return hw_rule_rate(params, rule, efw,
+ efw->pcm_capture_channels);
+}
+
+static inline int
+hw_rule_playback_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_efw *efw = rule->private;
+ return hw_rule_rate(params, rule, efw,
+ efw->pcm_playback_channels);
+}
+
+static inline int
+hw_rule_capture_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_efw *efw = rule->private;
+ return hw_rule_channels(params, rule, efw,
+ efw->pcm_capture_channels);
+}
+
+static inline int
+hw_rule_playback_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_efw *efw = rule->private;
+ return hw_rule_channels(params, rule, efw,
+ efw->pcm_playback_channels);
+}
+
+static int
+pcm_init_hw_params(struct snd_efw *efw,
+ struct snd_pcm_substream *substream)
+{
+ unsigned int *pcm_channels;
+ unsigned int rate_bit;
+ int mode, i;
+ int err;
+
+ struct snd_pcm_hardware hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_FIFO_IN_FRAMES |
+ /* for Open Sound System compatibility */
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .rates = efw->supported_sampling_rate,
+ /* set up later */
+ .rate_min = UINT_MAX,
+ .rate_max = 0,
+ /* set up later */
+ .channels_min = UINT_MAX,
+ .channels_max = 0,
+ .buffer_bytes_max = 1024 * 1024 * 1024,
+ .period_bytes_min = 256,
+ .period_bytes_max = 1024 * 1024 * 1024 / 2,
+ .periods_min = 2,
+ .periods_max = 32,
+ .fifo_size = 0,
+ };
+
+ substream->runtime->hw = hardware;
+ substream->runtime->delay = substream->runtime->hw.fifo_size;
+
+ /* add rule between channels and sampling rate */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE;
+ snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_capture_channels, efw,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_capture_rate, efw,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ pcm_channels = efw->pcm_capture_channels;
+ } else {
+ substream->runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_playback_channels, efw,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_playback_rate, efw,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ pcm_channels = efw->pcm_playback_channels;
+ }
+
+ /* preparing min/max sampling rate */
+ snd_pcm_limit_hw_rates(substream->runtime);
+
+ /* preparing the number of channels */
+ for (i = 0; i < ARRAY_SIZE(freq_table); i += 1) {
+ /* skip unsupported sampling rate */
+ rate_bit = snd_pcm_rate_to_rate_bit(freq_table[i]);
+ if (!(efw->supported_sampling_rate & rate_bit))
+ continue;
+
+ mode = get_multiplier_mode_with_index(i);
+ if (pcm_channels[mode] == 0)
+ continue;
+ substream->runtime->hw.channels_min =
+ min(substream->runtime->hw.channels_min,
+ pcm_channels[mode]);
+ substream->runtime->hw.channels_max =
+ max(substream->runtime->hw.channels_max,
+ pcm_channels[mode]);
+ }
+
+ /* AM824 in IEC 61883-6 can deliver 24bit data */
+ err = snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24);
+ if (err < 0)
+ goto end;
+
+ /* format of PCM samples is 16bit or 24bit inner 32bit */
+ err = snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (err < 0)
+ goto end;
+ err = snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (err < 0)
+ goto end;
+
+ /* time for period constraint */
+ err = snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+ 500, UINT_MAX);
+ if (err < 0)
+ goto end;
+
+ err = 0;
+
+end:
+ return err;
+}
+
+static int
+pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_efw *efw = substream->private_data;
+ int sampling_rate;
+ int err;
+
+ err = pcm_init_hw_params(efw, substream);
+ if (err < 0)
+ goto end;
+
+ /*
+ * The same sampling rate must be used for transmit and receive stream
+ * as long as the streams include PCM samples
+ */
+ if ((amdtp_stream_running(&efw->receive_stream) &&
+ amdtp_stream_pcm_running(&efw->receive_stream)) ||
+ (amdtp_stream_running(&efw->transmit_stream) &&
+ amdtp_stream_pcm_running(&efw->transmit_stream))) {
+ err = snd_efw_command_get_sampling_rate(efw, &sampling_rate);
+ if (err < 0)
+ goto end;
+ substream->runtime->hw.rate_min = sampling_rate;
+ substream->runtime->hw.rate_max = sampling_rate;
+ }
+
+ snd_pcm_set_sync(substream);
+
+end:
+ return err;
+}
+
+static int
+pcm_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+static int
+pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_efw *efw = substream->private_data;
+ struct amdtp_stream *stream;
+ struct amdtp_stream *opposite;
+ unsigned int *pcm_channels;
+ int mode;
+ int err;
+
+ /* keep PCM ring buffer */
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ goto end;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ stream = &efw->receive_stream;
+ opposite = &efw->transmit_stream;
+ pcm_channels = efw->pcm_playback_channels;
+ } else {
+ stream = &efw->transmit_stream;
+ opposite = &efw->receive_stream;
+ pcm_channels = efw->pcm_capture_channels;
+ }
+
+ /* stop stream if it's just for MIDI streams */
+ if (amdtp_stream_running(stream) &&
+ !amdtp_stream_pcm_running(stream) &&
+ amdtp_stream_midi_running(stream))
+ snd_efw_stream_stop(efw, stream);
+ if (amdtp_stream_running(opposite) &&
+ !amdtp_stream_pcm_running(opposite) &&
+ amdtp_stream_midi_running(opposite))
+ snd_efw_stream_stop(efw, opposite);
+
+ /* set sampling rate if both streams are not running */
+ if (!amdtp_stream_running(stream) ||
+ !amdtp_stream_running(opposite)) {
+ err = snd_efw_command_set_sampling_rate(efw,
+ params_rate(hw_params));
+ if (err < 0)
+ return err;
+ snd_ctl_notify(efw->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ efw->control_id_sampling_rate);
+ }
+
+ /* set AMDTP parameters for transmit stream */
+ amdtp_stream_set_rate(stream, params_rate(hw_params));
+ amdtp_stream_set_pcm(stream, params_channels(hw_params));
+ amdtp_stream_set_pcm_format(stream, params_format(hw_params));
+
+ /* need to start if opposite stream has MIDI stream */
+ if (!opposite->pcm && amdtp_stream_midi_running(opposite)) {
+ amdtp_stream_set_rate(opposite, params_rate(hw_params));
+ mode = snd_efw_get_multiplier_mode(params_rate(hw_params));
+ amdtp_stream_set_pcm(opposite, pcm_channels[mode]);
+ err = snd_efw_stream_start(efw, opposite);
+ }
+
+end:
+ return err;
+}
+
+static int
+pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_efw *efw = substream->private_data;
+ struct amdtp_stream *stream;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ stream = &efw->receive_stream;
+ else
+ stream = &efw->transmit_stream;
+
+ /* don't stop stream if MIDI stream is still running */
+ if (!amdtp_stream_midi_running(stream))
+ snd_efw_stream_stop(efw, stream);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int
+pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_efw *efw = substream->private_data;
+ struct amdtp_stream *stream;
+ int err = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ stream = &efw->receive_stream;
+ else
+ stream = &efw->transmit_stream;
+
+ /* start stream */
+ err = snd_efw_stream_start(efw, stream);
+ if (err < 0)
+ goto end;
+
+ /* initialize buffer pointer */
+ amdtp_stream_pcm_prepare(stream);
+
+end:
+ return err;
+}
+
+static int
+pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_efw *efw = substream->private_data;
+ struct snd_pcm_substream *pcm;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ pcm = substream;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pcm = NULL;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ amdtp_stream_pcm_trigger(&efw->receive_stream, pcm);
+ else
+ amdtp_stream_pcm_trigger(&efw->transmit_stream, pcm);
+
+ return 0;
+}
+
+static snd_pcm_uframes_t
+pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_efw *efw = substream->private_data;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return amdtp_stream_pcm_pointer(&efw->receive_stream);
+ else
+ return amdtp_stream_pcm_pointer(&efw->transmit_stream);
+}
+
+static struct snd_pcm_ops pcm_playback_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_hw_params,
+ .hw_free = pcm_hw_free,
+ .prepare = pcm_prepare,
+ .trigger = pcm_trigger,
+ .pointer = pcm_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
+};
+
+static struct snd_pcm_ops pcm_capture_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_hw_params,
+ .hw_free = pcm_hw_free,
+ .prepare = pcm_prepare,
+ .trigger = pcm_trigger,
+ .pointer = pcm_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+};
+
+int snd_efw_create_pcm_devices(struct snd_efw *efw)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(efw->card, efw->card->driver, 0, 1, 1, &pcm);
+ if (err < 0)
+ goto end;
+
+ pcm->private_data = efw;
+ snprintf(pcm->name, sizeof(pcm->name), "%s PCM", efw->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+
+ /* for host transmit and target input */
+ err = snd_efw_stream_init(efw, &efw->transmit_stream);
+ if (err < 0)
+ goto end;
+
+ /* for target output and host receive */
+ err = snd_efw_stream_init(efw, &efw->receive_stream);
+ if (err < 0) {
+ snd_efw_stream_destroy(efw, &efw->transmit_stream);
+ goto end;
+ }
+
+end:
+ return err;
+}
+
+void snd_efw_destroy_pcm_devices(struct snd_efw *efw)
+{
+ amdtp_stream_pcm_abort(&efw->receive_stream);
+ amdtp_stream_stop(&efw->receive_stream);
+ snd_efw_stream_destroy(efw, &efw->receive_stream);
+
+ amdtp_stream_pcm_abort(&efw->transmit_stream);
+ amdtp_stream_stop(&efw->transmit_stream);
+ snd_efw_stream_destroy(efw, &efw->transmit_stream);
+
+ return;
+}
--
1.7.10.4
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