* [PATCH v2 - AAF PCM plugin 1/7] aaf: Introduce plugin skeleton
2018-10-25 1:11 [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Andre Guedes
@ 2018-10-25 1:11 ` Andre Guedes
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 2/7] aaf: Load configuration parameters Andre Guedes
` (6 subsequent siblings)
7 siblings, 0 replies; 12+ messages in thread
From: Andre Guedes @ 2018-10-25 1:11 UTC (permalink / raw)
To: alsa-devel; +Cc: tiwai, pierre-louis.bossart, liam.r.girdwood
The patch introduces the skeleton code from the AAF plugin as well as
the buildsystem bits in order to get the plugin built. Following the
approach from other plugins, the AAF plugin is only built if its
dependency (libavtp) is detected by configure.
Follow-up patches implement support for both playback and capture modes.
Signed-off-by: Andre Guedes <andre.guedes@intel.com>
---
Makefile.am | 3 ++
aaf/Makefile.am | 9 ++++++
aaf/pcm_aaf.c | 85 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++
configure.ac | 10 +++++++
doc/aaf.txt | 18 ++++++++++++
5 files changed, 125 insertions(+)
create mode 100644 aaf/Makefile.am
create mode 100644 aaf/pcm_aaf.c
create mode 100644 doc/aaf.txt
diff --git a/Makefile.am b/Makefile.am
index 27f61a4..af0e9c4 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -35,6 +35,9 @@ endif
if HAVE_SPEEXDSP
SUBDIRS += speex
endif
+if HAVE_AAF
+SUBDIRS += aaf
+endif
EXTRA_DIST = gitcompile version COPYING.GPL m4/attributes.m4
AUTOMAKE_OPTIONS = foreign
diff --git a/aaf/Makefile.am b/aaf/Makefile.am
new file mode 100644
index 0000000..492b883
--- /dev/null
+++ b/aaf/Makefile.am
@@ -0,0 +1,9 @@
+asound_module_pcm_aaf_LTLIBRARIES = libasound_module_pcm_aaf.la
+
+asound_module_pcm_aafdir = @ALSA_PLUGIN_DIR@
+
+AM_CFLAGS = @ALSA_CFLAGS@ @AVTP_CFLAGS@
+AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED)
+
+libasound_module_pcm_aaf_la_SOURCES = pcm_aaf.c
+libasound_module_pcm_aaf_la_LIBADD = @ALSA_LIBS@ @AVTP_LIBS@
diff --git a/aaf/pcm_aaf.c b/aaf/pcm_aaf.c
new file mode 100644
index 0000000..31d8759
--- /dev/null
+++ b/aaf/pcm_aaf.c
@@ -0,0 +1,85 @@
+/*
+ * AVTP Audio Format (AAF) PCM Plugin
+ *
+ * Copyright (c) 2018, Intel Corporation
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as
+ * published by the Free Software Foundation; either version 2.1 of
+ * the License, or (at your option) any later version.
+ *
+ * This program 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <alsa/asoundlib.h>
+#include <alsa/pcm_external.h>
+
+typedef struct {
+ snd_pcm_ioplug_t io;
+} snd_pcm_aaf_t;
+
+static int aaf_close(snd_pcm_ioplug_t *io)
+{
+ free(io->private_data);
+ return 0;
+}
+
+static snd_pcm_sframes_t aaf_pointer(snd_pcm_ioplug_t *io)
+{
+ return 0;
+}
+
+static int aaf_start(snd_pcm_ioplug_t *io)
+{
+ return 0;
+}
+
+static int aaf_stop(snd_pcm_ioplug_t *io)
+{
+ return 0;
+}
+
+static const snd_pcm_ioplug_callback_t aaf_callback = {
+ .close = aaf_close,
+ .pointer = aaf_pointer,
+ .start = aaf_start,
+ .stop = aaf_stop,
+};
+
+SND_PCM_PLUGIN_DEFINE_FUNC(aaf)
+{
+ snd_pcm_aaf_t *aaf;
+ int res;
+
+ aaf = calloc(1, sizeof(*aaf));
+ if (!aaf) {
+ SNDERR("Failed to allocate memory");
+ return -ENOMEM;
+ }
+
+ aaf->io.version = SND_PCM_IOPLUG_VERSION;
+ aaf->io.name = "AVTP Audio Format (AAF) Plugin";
+ aaf->io.callback = &aaf_callback;
+ aaf->io.private_data = aaf;
+ res = snd_pcm_ioplug_create(&aaf->io, name, stream, mode);
+ if (res < 0) {
+ SNDERR("Failed to create ioplug instance");
+ goto err;
+ }
+
+ *pcmp = aaf->io.pcm;
+ return 0;
+
+err:
+ free(aaf);
+ return res;
+}
+
+SND_PCM_PLUGIN_SYMBOL(aaf);
diff --git a/configure.ac b/configure.ac
index ecc4c7a..db36f03 100644
--- a/configure.ac
+++ b/configure.ac
@@ -176,6 +176,14 @@ fi
test "x$prefix" = xNONE && prefix=$ac_default_prefix
test "x$exec_prefix" = xNONE && exec_prefix=$prefix
+AC_ARG_ENABLE([aaf],
+ AS_HELP_STRING([--disable-aaf], [Disable building of AAF plugin]))
+
+if test "x$enable_aaf" != "xno"; then
+ PKG_CHECK_MODULES(AVTP, avtp >= 0.1, [HAVE_AAF=yes], [HAVE_AAF=no])
+fi
+AM_CONDITIONAL(HAVE_AAF, test x$HAVE_AAF = xyes)
+
dnl ALSA plugin directory
AC_ARG_WITH(plugindir,
AS_HELP_STRING([--with-plugindir=dir],
@@ -251,6 +259,7 @@ AC_OUTPUT([
usb_stream/Makefile
speex/Makefile
arcam-av/Makefile
+ aaf/Makefile
])
dnl Show the build conditions
@@ -289,3 +298,4 @@ if test "$HAVE_SPEEX" = "yes"; then
echo " speexdsp_CFLAGS: $speexdsp_CFLAGS"
echo " speexdsp_LIBS: $speexdsp_LIBS"
fi
+echo "AAF plugin: $HAVE_AAF"
diff --git a/doc/aaf.txt b/doc/aaf.txt
new file mode 100644
index 0000000..b260a26
--- /dev/null
+++ b/doc/aaf.txt
@@ -0,0 +1,18 @@
+AVTP Audio Format (AAF) Plugin
+==============================
+
+Overview
+--------
+
+The AAF plugin is a PCM plugin that uses Audio Video Transport Protocol (AVTP)
+to transmit/receive audio samples through a Time-Sensitive Network (TSN)
+capable network. The plugin enables media applications to easily implement AVTP
+Talker and Listener functionalities.
+
+Plugin Dependencies
+-------------------
+
+The AAF plugin uses libavtp to handle AVTP packetization. Libavtp source code
+can be found in https://github.com/AVnu/libavtp as well as instructions to
+build and install it. If libavtp isn't detected by configure, the plugin isn't
+built.
--
2.14.4
^ permalink raw reply related [flat|nested] 12+ messages in thread* [PATCH v2 - AAF PCM plugin 2/7] aaf: Load configuration parameters
2018-10-25 1:11 [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Andre Guedes
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 1/7] aaf: Introduce plugin skeleton Andre Guedes
@ 2018-10-25 1:11 ` Andre Guedes
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 3/7] aaf: Implement Playback mode support Andre Guedes
` (5 subsequent siblings)
7 siblings, 0 replies; 12+ messages in thread
From: Andre Guedes @ 2018-10-25 1:11 UTC (permalink / raw)
To: alsa-devel; +Cc: tiwai, pierre-louis.bossart, liam.r.girdwood
This patch implements the infrastructure to load the plugin
configuration from ALSA configuration file. The configuration
is loaded in open() callback.
All configuration parameters are described in details in doc/aaf.txt
file.
Signed-off-by: Andre Guedes <andre.guedes@intel.com>
---
aaf/pcm_aaf.c | 131 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
configure.ac | 1 +
doc/aaf.txt | 33 ++++++++++++++-
3 files changed, 164 insertions(+), 1 deletion(-)
diff --git a/aaf/pcm_aaf.c b/aaf/pcm_aaf.c
index 31d8759..c9d4b70 100644
--- a/aaf/pcm_aaf.c
+++ b/aaf/pcm_aaf.c
@@ -20,11 +20,138 @@
#include <alsa/asoundlib.h>
#include <alsa/pcm_external.h>
+#include <linux/if_ether.h>
+#include <net/if.h>
+#include <string.h>
+#include <stdint.h>
+
+#define NSEC_PER_USEC 1000
typedef struct {
snd_pcm_ioplug_t io;
+
+ char ifname[IFNAMSIZ];
+ unsigned char addr[ETH_ALEN];
+ int prio;
+ uint64_t streamid;
+ int mtt;
+ int t_uncertainty;
+ snd_pcm_uframes_t frames_per_pdu;
} snd_pcm_aaf_t;
+static int aaf_load_config(snd_pcm_aaf_t *aaf, snd_config_t *conf)
+{
+ snd_config_iterator_t cur, next;
+
+ snd_config_for_each(cur, next, conf) {
+ snd_config_t *entry = snd_config_iterator_entry(cur);
+ const char *id;
+
+ if (snd_config_get_id(entry, &id) < 0)
+ goto err;
+
+ if (strcmp(id, "comment") == 0 ||
+ strcmp(id, "type") == 0 ||
+ strcmp(id, "hint") == 0)
+ continue;
+
+ if (strcmp(id, "ifname") == 0) {
+ const char *ifname;
+
+ if (snd_config_get_string(entry, &ifname) < 0)
+ goto err;
+
+ snprintf(aaf->ifname, sizeof(aaf->ifname), "%s",
+ ifname);
+ } else if (strcmp(id, "addr") == 0) {
+ const char *addr;
+ int n;
+
+ if (snd_config_get_string(entry, &addr) < 0)
+ goto err;
+
+ n = sscanf(addr, "%hhx:%hhx:%hhx:%hhx:%hhx:%hhx",
+ &aaf->addr[0], &aaf->addr[1],
+ &aaf->addr[2], &aaf->addr[3],
+ &aaf->addr[4], &aaf->addr[5]);
+ if (n != 6)
+ goto err;
+ } else if (strcmp(id, "prio") == 0) {
+ long prio;
+
+ if (snd_config_get_integer(entry, &prio) < 0)
+ goto err;
+
+ if (prio < 0)
+ goto err;
+
+ aaf->prio = prio;
+ } else if (strcmp(id, "streamid") == 0) {
+ const char *streamid;
+ unsigned char addr[6];
+ unsigned short unique_id;
+ int n;
+
+ if (snd_config_get_string(entry, &streamid) < 0)
+ goto err;
+
+ n = sscanf(streamid,
+ "%hhx:%hhx:%hhx:%hhx:%hhx:%hhx:%hx",
+ &addr[0], &addr[1], &addr[2], &addr[3],
+ &addr[4], &addr[5], &unique_id);
+ if (n != 7)
+ goto err;
+
+ aaf->streamid = (uint64_t) addr[0] << 56 |
+ (uint64_t) addr[1] << 48 |
+ (uint64_t) addr[2] << 40 |
+ (uint64_t) addr[3] << 32 |
+ (uint64_t) addr[4] << 24 |
+ (uint64_t) addr[5] << 16 |
+ unique_id;
+ } else if (strcmp(id, "mtt") == 0) {
+ long mtt;
+
+ if (snd_config_get_integer(entry, &mtt) < 0)
+ goto err;
+
+ if (mtt < 0)
+ goto err;
+
+ aaf->mtt = mtt * NSEC_PER_USEC;
+ } else if (strcmp(id, "time_uncertainty") == 0) {
+ long t_uncertainty;
+
+ if (snd_config_get_integer(entry, &t_uncertainty) < 0)
+ goto err;
+
+ if (t_uncertainty < 0)
+ goto err;
+
+ aaf->t_uncertainty = t_uncertainty * NSEC_PER_USEC;
+ } else if (strcmp(id, "frames_per_pdu") == 0) {
+ long frames_per_pdu;
+
+ if (snd_config_get_integer(entry, &frames_per_pdu) < 0)
+ goto err;
+
+ if (frames_per_pdu < 0)
+ goto err;
+
+ aaf->frames_per_pdu = frames_per_pdu;
+ } else {
+ SNDERR("Invalid configuration: %s", id);
+ goto err;
+ }
+ }
+
+ return 0;
+
+err:
+ SNDERR("Error loading device configuration");
+ return -EINVAL;
+}
+
static int aaf_close(snd_pcm_ioplug_t *io)
{
free(io->private_data);
@@ -64,6 +191,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(aaf)
return -ENOMEM;
}
+ res = aaf_load_config(aaf, conf);
+ if (res < 0)
+ goto err;
+
aaf->io.version = SND_PCM_IOPLUG_VERSION;
aaf->io.name = "AVTP Audio Format (AAF) Plugin";
aaf->io.callback = &aaf_callback;
diff --git a/configure.ac b/configure.ac
index db36f03..eadaed6 100644
--- a/configure.ac
+++ b/configure.ac
@@ -181,6 +181,7 @@ AC_ARG_ENABLE([aaf],
if test "x$enable_aaf" != "xno"; then
PKG_CHECK_MODULES(AVTP, avtp >= 0.1, [HAVE_AAF=yes], [HAVE_AAF=no])
+ AC_CHECK_HEADERS([linux/if_ether.h], [], [HAVE_AAF=no])
fi
AM_CONDITIONAL(HAVE_AAF, test x$HAVE_AAF = xyes)
diff --git a/doc/aaf.txt b/doc/aaf.txt
index b260a26..afc794b 100644
--- a/doc/aaf.txt
+++ b/doc/aaf.txt
@@ -14,5 +14,36 @@ Plugin Dependencies
The AAF plugin uses libavtp to handle AVTP packetization. Libavtp source code
can be found in https://github.com/AVnu/libavtp as well as instructions to
-build and install it. If libavtp isn't detected by configure, the plugin isn't
+build and install it.
+
+The plugin also depends on some kernel API headers such as linux/if_ether.h so
+make sure you have them installed in your system.
+
+If libavtp or the kernel headers aren't detected by configure, the plugin isn't
built.
+
+Plugin Configuration
+--------------------
+
+The plugin parameters are passed via ALSA configuration file. They are defined
+as follows:
+
+ * ifname: Network interface used to transmit/receive AVTP packets.
+
+ * addr: Stream destination MAC address.
+
+ * prio: Priority used by the plugin to transmit AVTP traffic. This
+ option is relevant only when operating in playback mode.
+
+ * streamid: Stream ID associated with the AAF stream transmitted or
+ received by the plugin.
+
+ * mtt: Maximum Transit Time (in microseconds) as defined in AVTP spec
+ section 4.3.3. This option is relevant only when operating in
+ playback mode.
+
+ * time_uncertainty: Maximum Time Uncertainty (in microseconds) as
+ defined by AVTP spec section 4.3.3. This option is relevant only when
+ operating in playback mode.
+
+ * frames_per_pdu: Number of audio frames transmitted in one AVTPDU.
--
2.14.4
^ permalink raw reply related [flat|nested] 12+ messages in thread* [PATCH v2 - AAF PCM plugin 3/7] aaf: Implement Playback mode support
2018-10-25 1:11 [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Andre Guedes
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 1/7] aaf: Introduce plugin skeleton Andre Guedes
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 2/7] aaf: Load configuration parameters Andre Guedes
@ 2018-10-25 1:11 ` Andre Guedes
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 4/7] aaf: Prepare for Capture " Andre Guedes
` (4 subsequent siblings)
7 siblings, 0 replies; 12+ messages in thread
From: Andre Guedes @ 2018-10-25 1:11 UTC (permalink / raw)
To: alsa-devel; +Cc: tiwai, pierre-louis.bossart, liam.r.girdwood
This patch implements the playback mode support from the AAF plugin.
Simply put, this mode works as follows: PCM samples provided by alsa-lib
layer are encapsulated into AVTPDUs and transmitted through the network.
In summary, the playback mode implements a typical AVTP Talker.
When the AAF device is put in running state, its media clock is started.
At every tick from the media clock, audio frames are consumed from the
audio buffer, encapsulated into an AVTPDU, and transmitted to the
network. The presentation time from each AVTPDU is calculated taking in
consideration the maximum transit time and time uncertainty values
configured by the user.
Below follows some discussion about implementation details:
AVTP protocol doesn't support all formats and rates available in ALSA so
the plugin sets some constraints to ensure only supported configurations
are used (see aaf_set_hw_constraint function).
The plugin implements a media clock which is the source from AVTP
timestamps. The AVTP timestamp is based on PTP time which uses
International Atomic Time (TAI) coordinate system. The media clock is
implemented through a periodic timer using timerfd infrastructure so the
plugin requires that system clock and PTP clock are synchronized
(instructions on how to sync these clocks are provided in doc/aaf.txt).
CLOCK_TAI clockid isn't currently supported by timerfd so the timer fd
is created using CLOCK_REALTIME and the start time is converted from TAI
to UTC.
Even though only one file descriptor is used to implement the playback
mode, this patch doesn't leverage ioplug->poll_fd but defines poll
callbacks instead. The reason is these callbacks will be required to
support capture mode (to be implemented by upcoming patch).
The TSN data plane interface is the AF_PACKET socket family so the
plugin uses an AF_PACKET socket to send/receive AVTPDUs. Linux requires
CAP_NET_RAW capability in order to open an AF_PACKET socket so the
application that instantiates the plugin must have it. For further info
about AF_PACKET socket family see packet(7).
Signed-off-by: Andre Guedes <andre.guedes@intel.com>
---
aaf/pcm_aaf.c | 644 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
configure.ac | 2 +-
doc/aaf.txt | 99 +++++++++
3 files changed, 744 insertions(+), 1 deletion(-)
diff --git a/aaf/pcm_aaf.c b/aaf/pcm_aaf.c
index c9d4b70..86bfe50 100644
--- a/aaf/pcm_aaf.c
+++ b/aaf/pcm_aaf.c
@@ -20,12 +20,32 @@
#include <alsa/asoundlib.h>
#include <alsa/pcm_external.h>
+#include <arpa/inet.h>
+#include <avtp.h>
+#include <avtp_aaf.h>
+#include <limits.h>
#include <linux/if_ether.h>
+#include <linux/if_packet.h>
#include <net/if.h>
#include <string.h>
#include <stdint.h>
+#include <sys/ioctl.h>
+#include <sys/timerfd.h>
+
+#ifdef AAF_DEBUG
+#define pr_debug(...) SNDERR(__VA_ARGS__)
+#else
+#define pr_debug(...) (void)0
+#endif
+
+#define ARRAY_SIZE(a) (sizeof(a)/sizeof((a)[0]))
#define NSEC_PER_USEC 1000
+#define NSEC_PER_SEC 1000000000
+#define TAI_OFFSET (37ULL * NSEC_PER_SEC)
+#define TAI_TO_UTC(t) (t - TAI_OFFSET)
+
+#define FD_COUNT_PLAYBACK 1
typedef struct {
snd_pcm_ioplug_t io;
@@ -37,8 +57,71 @@ typedef struct {
int mtt;
int t_uncertainty;
snd_pcm_uframes_t frames_per_pdu;
+
+ int sk_fd;
+ int timer_fd;
+
+ struct sockaddr_ll sk_addr;
+
+ struct avtp_stream_pdu *pdu;
+ int pdu_size;
+ uint8_t pdu_seq;
+
+ uint64_t timer_starttime;
+ uint64_t timer_period;
+ uint64_t timer_expirations;
+
+ snd_pcm_channel_area_t *audiobuf_areas;
+ snd_pcm_channel_area_t *payload_areas;
+
+ snd_pcm_uframes_t hw_ptr;
+ snd_pcm_uframes_t boundary;
} snd_pcm_aaf_t;
+static unsigned int alsa_to_avtp_format(snd_pcm_format_t format)
+{
+ switch (format) {
+ case SND_PCM_FORMAT_S16_BE:
+ return AVTP_AAF_FORMAT_INT_16BIT;
+ case SND_PCM_FORMAT_S24_3BE:
+ return AVTP_AAF_FORMAT_INT_24BIT;
+ case SND_PCM_FORMAT_S32_BE:
+ return AVTP_AAF_FORMAT_INT_32BIT;
+ case SND_PCM_FORMAT_FLOAT_BE:
+ return AVTP_AAF_FORMAT_FLOAT_32BIT;
+ default:
+ return AVTP_AAF_FORMAT_USER;
+ }
+}
+
+static unsigned int alsa_to_avtp_rate(unsigned int rate)
+{
+ switch (rate) {
+ case 8000:
+ return AVTP_AAF_PCM_NSR_8KHZ;
+ case 16000:
+ return AVTP_AAF_PCM_NSR_16KHZ;
+ case 24000:
+ return AVTP_AAF_PCM_NSR_24KHZ;
+ case 32000:
+ return AVTP_AAF_PCM_NSR_32KHZ;
+ case 44100:
+ return AVTP_AAF_PCM_NSR_44_1KHZ;
+ case 48000:
+ return AVTP_AAF_PCM_NSR_48KHZ;
+ case 88200:
+ return AVTP_AAF_PCM_NSR_88_2KHZ;
+ case 96000:
+ return AVTP_AAF_PCM_NSR_96KHZ;
+ case 176400:
+ return AVTP_AAF_PCM_NSR_176_4KHZ;
+ case 192000:
+ return AVTP_AAF_PCM_NSR_192KHZ;
+ default:
+ return AVTP_AAF_PCM_NSR_USER;
+ }
+}
+
static int aaf_load_config(snd_pcm_aaf_t *aaf, snd_config_t *conf)
{
snd_config_iterator_t cur, next;
@@ -152,32 +235,576 @@ err:
return -EINVAL;
}
+static int aaf_init_socket(snd_pcm_aaf_t *aaf)
+{
+ int fd, res;
+ struct ifreq req;
+
+ fd = socket(AF_PACKET, SOCK_DGRAM|SOCK_NONBLOCK, htons(ETH_P_TSN));
+ if (fd < 0) {
+ SNDERR("Failed to open AF_PACKET socket");
+ return -errno;
+ }
+
+ snprintf(req.ifr_name, sizeof(req.ifr_name), "%s", aaf->ifname);
+ res = ioctl(fd, SIOCGIFINDEX, &req);
+ if (res < 0) {
+ SNDERR("Failed to get network interface index");
+ res = -errno;
+ goto err;
+ }
+
+ aaf->sk_addr.sll_family = AF_PACKET;
+ aaf->sk_addr.sll_protocol = htons(ETH_P_TSN);
+ aaf->sk_addr.sll_halen = ETH_ALEN;
+ aaf->sk_addr.sll_ifindex = req.ifr_ifindex;
+ memcpy(&aaf->sk_addr.sll_addr, aaf->addr, ETH_ALEN);
+
+ res = setsockopt(fd, SOL_SOCKET, SO_PRIORITY, &aaf->prio,
+ sizeof(aaf->prio));
+ if (res < 0) {
+ SNDERR("Failed to set socket priority");
+ res = -errno;
+ goto err;
+ }
+
+ aaf->sk_fd = fd;
+ return 0;
+
+err:
+ close(fd);
+ return res;
+}
+
+static int aaf_init_timer(snd_pcm_aaf_t *aaf)
+{
+ int fd;
+
+ fd = timerfd_create(CLOCK_REALTIME, TFD_NONBLOCK);
+ if (fd < 0)
+ return -errno;
+
+ aaf->timer_fd = fd;
+ return 0;
+}
+
+static int aaf_init_pdu(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ struct avtp_stream_pdu *pdu;
+ ssize_t frame_size, payload_size, pdu_size;
+ snd_pcm_ioplug_t *io = &aaf->io;
+
+ frame_size = snd_pcm_format_size(io->format, io->channels);
+ if (frame_size < 0)
+ return frame_size;
+
+ payload_size = frame_size * aaf->frames_per_pdu;
+ pdu_size = sizeof(*pdu) + payload_size;
+ pdu = calloc(1, pdu_size);
+ if (!pdu)
+ return -ENOMEM;
+
+ res = avtp_aaf_pdu_init(pdu);
+ if (res < 0)
+ goto err;
+
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_TV, 1);
+ if (res < 0)
+ goto err;
+
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_STREAM_ID, aaf->streamid);
+ if (res < 0)
+ goto err;
+
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_FORMAT,
+ alsa_to_avtp_format(io->format));
+ if (res < 0)
+ goto err;
+
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_NSR,
+ alsa_to_avtp_rate(io->rate));
+ if (res < 0)
+ goto err;
+
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_CHAN_PER_FRAME,
+ io->channels);
+ if (res < 0)
+ goto err;
+
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_BIT_DEPTH,
+ snd_pcm_format_width(io->format));
+ if (res < 0)
+ goto err;
+
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_STREAM_DATA_LEN,
+ payload_size);
+ if (res < 0)
+ goto err;
+
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_SP, AVTP_AAF_PCM_SP_NORMAL);
+ if (res < 0)
+ goto err;
+
+ aaf->pdu = pdu;
+ aaf->pdu_size = pdu_size;
+ return 0;
+
+err:
+ free(pdu);
+ return res;
+}
+
+static int aaf_init_areas(snd_pcm_aaf_t *aaf, snd_pcm_channel_area_t *areas,
+ void *buf)
+{
+ ssize_t sample_size, frame_size;
+ snd_pcm_ioplug_t *io = &aaf->io;
+
+ sample_size = snd_pcm_format_size(io->format, 1);
+ if (sample_size < 0)
+ return sample_size;
+
+ frame_size = sample_size * io->channels;
+
+ for (unsigned int i = 0; i < io->channels; i++) {
+ areas[i].addr = buf;
+ areas[i].first = i * sample_size * 8;
+ areas[i].step = frame_size * 8;
+ }
+
+ return 0;
+}
+
+static int aaf_init_payload_areas(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ snd_pcm_channel_area_t *areas;
+ snd_pcm_ioplug_t *io = &aaf->io;
+
+ areas = calloc(io->channels, sizeof(snd_pcm_channel_area_t));
+ if (!areas)
+ return -ENOMEM;
+
+ res = aaf_init_areas(aaf, areas, aaf->pdu->avtp_payload);
+ if (res < 0)
+ goto err;
+
+ aaf->payload_areas = areas;
+ return 0;
+
+err:
+ free(areas);
+ return res;
+}
+
+static int aaf_init_audiobuf_areas(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ char *buf;
+ ssize_t frame_size;
+ snd_pcm_channel_area_t *areas;
+ snd_pcm_ioplug_t *io = &aaf->io;
+
+ frame_size = snd_pcm_format_size(io->format, io->channels);
+ if (frame_size < 0)
+ return frame_size;
+
+ buf = calloc(io->buffer_size, frame_size);
+ if (!buf)
+ return -ENOMEM;
+
+ areas = calloc(io->channels, sizeof(snd_pcm_channel_area_t));
+ if (!areas) {
+ res = -ENOMEM;
+ goto err_free_buf;
+ }
+
+ res = aaf_init_areas(aaf, areas, buf);
+ if (res < 0)
+ goto err_free_areas;
+
+ aaf->audiobuf_areas = areas;
+ return 0;
+
+err_free_areas:
+ free(areas);
+err_free_buf:
+ free(buf);
+ return res;
+}
+
+static void aaf_inc_hw_ptr(snd_pcm_aaf_t *aaf, snd_pcm_uframes_t val)
+{
+ aaf->hw_ptr += val;
+
+ if (aaf->hw_ptr >= aaf->boundary)
+ aaf->hw_ptr -= aaf->boundary;
+}
+
+static int aaf_mclk_start_playback(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ struct timespec now;
+ struct itimerspec itspec;
+ uint64_t time_utc;
+ snd_pcm_ioplug_t *io = &aaf->io;
+
+ res = clock_gettime(CLOCK_TAI, &now);
+ if (res < 0) {
+ SNDERR("Failed to get time from clock");
+ return -errno;
+ }
+
+ aaf->timer_expirations = 0;
+ aaf->timer_period = (uint64_t)NSEC_PER_SEC * aaf->frames_per_pdu /
+ io->rate;
+ aaf->timer_starttime = now.tv_sec * NSEC_PER_SEC + now.tv_nsec +
+ aaf->timer_period;
+
+ time_utc = TAI_TO_UTC(aaf->timer_starttime);
+ itspec.it_value.tv_sec = time_utc / NSEC_PER_SEC;
+ itspec.it_value.tv_nsec = time_utc % NSEC_PER_SEC;
+ itspec.it_interval.tv_sec = 0;
+ itspec.it_interval.tv_nsec = aaf->timer_period;
+ res = timerfd_settime(aaf->timer_fd, TFD_TIMER_ABSTIME, &itspec, NULL);
+ if (res < 0)
+ return -errno;
+
+ return 0;
+}
+
+static int aaf_mclk_reset(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ struct itimerspec itspec = { 0 };
+
+ res = timerfd_settime(aaf->timer_fd, 0, &itspec, NULL);
+ if (res < 0) {
+ SNDERR("Failed to stop media clock");
+ return res;
+ }
+
+ aaf->timer_starttime = 0;
+ aaf->timer_period = 0;
+ aaf->timer_expirations = 0;
+ return 0;
+}
+
+static uint64_t aaf_mclk_gettime(snd_pcm_aaf_t *aaf)
+{
+ if (aaf->timer_expirations == 0)
+ return 0;
+
+ return aaf->timer_starttime + aaf->timer_period *
+ (aaf->timer_expirations - 1);
+}
+
+static int aaf_tx_pdu(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ uint64_t ptime;
+ ssize_t n;
+ snd_pcm_uframes_t hw_avail;
+ snd_pcm_ioplug_t *io = &aaf->io;
+ struct avtp_stream_pdu *pdu = aaf->pdu;
+
+ hw_avail = snd_pcm_ioplug_hw_avail(io, aaf->hw_ptr, io->appl_ptr);
+ if (hw_avail < aaf->frames_per_pdu) {
+ /* If the number of available frames is less than number of
+ * frames needed to fill an AVTPDU, we reached an underrun
+ * state.
+ */
+ return -EPIPE;
+ }
+
+ res = snd_pcm_areas_copy_wrap(aaf->payload_areas, 0,
+ aaf->frames_per_pdu,
+ aaf->audiobuf_areas,
+ (aaf->hw_ptr % io->buffer_size),
+ io->buffer_size, io->channels,
+ aaf->frames_per_pdu, io->format);
+ if (res < 0) {
+ SNDERR("Failed to copy data to AVTP payload");
+ return res;
+ }
+
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_SEQ_NUM, aaf->pdu_seq++);
+ if (res < 0)
+ return res;
+
+ ptime = aaf_mclk_gettime(aaf) + aaf->mtt + aaf->t_uncertainty;
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_TIMESTAMP, ptime);
+ if (res < 0)
+ return res;
+
+ n = sendto(aaf->sk_fd, aaf->pdu, aaf->pdu_size, 0,
+ (struct sockaddr *) &aaf->sk_addr,
+ sizeof(aaf->sk_addr));
+ if (n < 0 || n != aaf->pdu_size) {
+ SNDERR("Failed to send AAF PDU");
+ return -EIO;
+ }
+
+ aaf_inc_hw_ptr(aaf, aaf->frames_per_pdu);
+ return 0;
+}
+
+static int aaf_mclk_timeout_playback(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ ssize_t n;
+ uint64_t expirations;
+
+ n = read(aaf->timer_fd, &expirations, sizeof(uint64_t));
+ if (n < 0) {
+ SNDERR("Failed to read() timer");
+ return -errno;
+ }
+
+ if (expirations != 1)
+ pr_debug("Missed %llu tx interval(s) ", expirations - 1);
+
+ while (expirations--) {
+ aaf->timer_expirations++;
+
+ res = aaf_tx_pdu(aaf);
+ if (res < 0)
+ return res;
+ }
+
+ return 0;
+}
+
+static int aaf_set_hw_constraint(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ snd_pcm_ioplug_t *io = &aaf->io;
+ static const unsigned int accesses[] = {
+ SND_PCM_ACCESS_RW_INTERLEAVED,
+ };
+ static const unsigned int formats[] = {
+ SND_PCM_FORMAT_S16_BE,
+ SND_PCM_FORMAT_S24_3BE,
+ SND_PCM_FORMAT_S32_BE,
+ SND_PCM_FORMAT_FLOAT_BE,
+ };
+ static const unsigned int rates[] = {
+ 8000,
+ 16000,
+ 24000,
+ 32000,
+ 44100,
+ 48000,
+ 88200,
+ 96000,
+ 176400,
+ 192000,
+ };
+
+ res = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS,
+ ARRAY_SIZE(accesses), accesses);
+ if (res < 0)
+ return res;
+
+ res = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT,
+ ARRAY_SIZE(formats), formats);
+ if (res < 0)
+ return res;
+
+ res = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_RATE,
+ ARRAY_SIZE(rates), rates);
+ if (res < 0)
+ return res;
+
+ return 0;
+}
+
static int aaf_close(snd_pcm_ioplug_t *io)
{
free(io->private_data);
return 0;
}
+static int aaf_hw_params(snd_pcm_ioplug_t *io,
+ snd_pcm_hw_params_t *params ATTRIBUTE_UNUSED)
+{
+ int res;
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ res = aaf_init_socket(aaf);
+ if (res < 0)
+ return res;
+
+ res = aaf_init_timer(aaf);
+ if (res < 0)
+ goto err_close_sk;
+
+ res = aaf_init_pdu(aaf);
+ if (res < 0)
+ goto err_close_timer;
+
+ res = aaf_init_payload_areas(aaf);
+ if (res < 0)
+ goto err_free_pdu;
+
+ res = aaf_init_audiobuf_areas(aaf);
+ if (res < 0)
+ goto err_free_payload_areas;
+
+ return 0;
+
+err_free_payload_areas:
+ free(aaf->payload_areas);
+err_free_pdu:
+ free(aaf->pdu);
+err_close_timer:
+ close(aaf->timer_fd);
+err_close_sk:
+ close(aaf->sk_fd);
+ return res;
+}
+
+static int aaf_hw_free(snd_pcm_ioplug_t *io)
+{
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ close(aaf->sk_fd);
+ close(aaf->timer_fd);
+ free(aaf->pdu);
+ free(aaf->payload_areas);
+ free(aaf->audiobuf_areas->addr);
+ free(aaf->audiobuf_areas);
+ return 0;
+}
+
+static int aaf_sw_params(snd_pcm_ioplug_t *io, snd_pcm_sw_params_t *params)
+{
+ int res;
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ res = snd_pcm_sw_params_get_boundary(params, &aaf->boundary);
+ if (res < 0)
+ return res;
+
+ return 0;
+}
+
static snd_pcm_sframes_t aaf_pointer(snd_pcm_ioplug_t *io)
{
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ return aaf->hw_ptr;
+}
+
+static int aaf_poll_descriptors_count(snd_pcm_ioplug_t *io ATTRIBUTE_UNUSED)
+{
+ return FD_COUNT_PLAYBACK;
+}
+
+static int aaf_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd,
+ unsigned int space)
+{
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ if (space != FD_COUNT_PLAYBACK)
+ return -EINVAL;
+
+ pfd[0].fd = aaf->timer_fd;
+ pfd[0].events = POLLIN;
+ return space;
+}
+
+static int aaf_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfd,
+ unsigned int nfds, unsigned short *revents)
+{
+ int res;
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ if (nfds != FD_COUNT_PLAYBACK)
+ return -EINVAL;
+
+ if (pfd[0].revents & POLLIN) {
+ res = aaf_mclk_timeout_playback(aaf);
+ if (res < 0)
+ return res;
+
+ *revents = POLLIN;
+ }
+
+ return 0;
+}
+
+static int aaf_prepare(snd_pcm_ioplug_t *io)
+{
+ int res;
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ aaf->pdu_seq = 0;
+ aaf->hw_ptr = 0;
+
+ res = aaf_mclk_reset(aaf);
+ if (res < 0)
+ return res;
+
return 0;
}
static int aaf_start(snd_pcm_ioplug_t *io)
{
+ int res;
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ res = aaf_mclk_start_playback(aaf);
+ if (res < 0)
+ return res;
+
return 0;
}
static int aaf_stop(snd_pcm_ioplug_t *io)
{
+ int res;
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ res = aaf_mclk_reset(aaf);
+ if (res < 0)
+ return res;
+
return 0;
}
+static snd_pcm_sframes_t aaf_transfer(snd_pcm_ioplug_t *io,
+ const snd_pcm_channel_area_t *areas,
+ snd_pcm_uframes_t offset,
+ snd_pcm_uframes_t size)
+{
+ int res;
+ snd_pcm_aaf_t *aaf = io->private_data;
+
+ res = snd_pcm_areas_copy_wrap(aaf->audiobuf_areas,
+ (io->appl_ptr % io->buffer_size),
+ io->buffer_size, areas, offset, size,
+ io->channels, size, io->format);
+ if (res < 0)
+ return res;
+
+ return size;
+}
+
static const snd_pcm_ioplug_callback_t aaf_callback = {
.close = aaf_close,
+ .hw_params = aaf_hw_params,
+ .hw_free = aaf_hw_free,
+ .sw_params = aaf_sw_params,
.pointer = aaf_pointer,
+ .poll_descriptors_count = aaf_poll_descriptors_count,
+ .poll_descriptors = aaf_poll_descriptors,
+ .poll_revents = aaf_poll_revents,
+ .prepare = aaf_prepare,
.start = aaf_start,
.stop = aaf_stop,
+ .transfer = aaf_transfer,
};
SND_PCM_PLUGIN_DEFINE_FUNC(aaf)
@@ -185,12 +812,21 @@ SND_PCM_PLUGIN_DEFINE_FUNC(aaf)
snd_pcm_aaf_t *aaf;
int res;
+ /* For now the plugin only supports Playback mode i.e. AAF Talker
+ * functionality.
+ */
+ if (stream != SND_PCM_STREAM_PLAYBACK)
+ return -EINVAL;
+
aaf = calloc(1, sizeof(*aaf));
if (!aaf) {
SNDERR("Failed to allocate memory");
return -ENOMEM;
}
+ aaf->sk_fd = -1;
+ aaf->timer_fd = -1;
+
res = aaf_load_config(aaf, conf);
if (res < 0)
goto err;
@@ -199,12 +835,20 @@ SND_PCM_PLUGIN_DEFINE_FUNC(aaf)
aaf->io.name = "AVTP Audio Format (AAF) Plugin";
aaf->io.callback = &aaf_callback;
aaf->io.private_data = aaf;
+ aaf->io.flags = SND_PCM_IOPLUG_FLAG_BOUNDARY_WA;
res = snd_pcm_ioplug_create(&aaf->io, name, stream, mode);
if (res < 0) {
SNDERR("Failed to create ioplug instance");
goto err;
}
+ res = aaf_set_hw_constraint(aaf);
+ if (res < 0) {
+ SNDERR("Failed to set hw constraints");
+ snd_pcm_ioplug_delete(&aaf->io);
+ goto err;
+ }
+
*pcmp = aaf->io.pcm;
return 0;
diff --git a/configure.ac b/configure.ac
index eadaed6..1eb9d65 100644
--- a/configure.ac
+++ b/configure.ac
@@ -181,7 +181,7 @@ AC_ARG_ENABLE([aaf],
if test "x$enable_aaf" != "xno"; then
PKG_CHECK_MODULES(AVTP, avtp >= 0.1, [HAVE_AAF=yes], [HAVE_AAF=no])
- AC_CHECK_HEADERS([linux/if_ether.h], [], [HAVE_AAF=no])
+ AC_CHECK_HEADERS([linux/if_ether.h linux/if_packet.h], [], [HAVE_AAF=no])
fi
AM_CONDITIONAL(HAVE_AAF, test x$HAVE_AAF = xyes)
diff --git a/doc/aaf.txt b/doc/aaf.txt
index afc794b..c278166 100644
--- a/doc/aaf.txt
+++ b/doc/aaf.txt
@@ -9,6 +9,79 @@ to transmit/receive audio samples through a Time-Sensitive Network (TSN)
capable network. The plugin enables media applications to easily implement AVTP
Talker and Listener functionalities.
+AVTP is designed to take advantage of generalized Precision Time Protocol
+(gPTP) and Forwarding and Queuing Enhancements for Time-Sensitive Streams
+(FQTSS). gPTP ensures AVTP talkers and listeners share the same time reference
+so the presentation time from AVTP can be used to inform when PCM samples
+should be presented to the application layer. FQTSS provides bandwidth
+reservation and traffic prioritization for the AVTP stream.
+
+gPTP functionality is provided by the Linuxptp project while FQTSS
+functionality is provided by Linux Traffic Control system since kernel version
+4.15.
+
+gPTP Setup
+----------
+
+The Linuxptp project provides the ptp4l daemon, which synchronizes the PTP
+clock from NIC, and the pmc tool which communicates with ptp4l to get/set
+some runtime settings. The project also provides the phc2sys daemon which
+synchronizes the PTP clock and system clock.
+
+The AAF Plugin requires system clock is synchronized with PTP clock and TAI
+offset is properly set in the kernel. ptp4l and phc2sys can be set up in many
+different ways, below we provide an example that fullfils the plugin
+requirements. For further information check ptp4l(8) and phc2sys(8).
+
+In the following instructions, replace $IFNAME by your PTP capable NIC
+interface. The gPTP.cfg file mentioned below can be found in /usr/share/
+doc/linuxptp/ (depending on your distro).
+
+Synchronize PTP clock with PTP time:
+
+ $ ptp4l -f gPTP.cfg -i $IFNAME
+
+Enable TAI offset to be automatically set by phc2sys:
+
+ $ pmc -u -t 1 -b 0 'SET GRANDMASTER_SETTINGS_NP \
+ clockClass 248 clockAccuracy 0xfe \
+ offsetScaledLogVariance 0xffff \
+ currentUtcOffset 37 leap61 0 leap59 0 \
+ currentUtcOffsetValid 1 p pTimescale 1 \
+ timeTraceable 1 frequencyTraceable 0 timeSource 0xa0'
+
+Synchronize system clock with PTP clock:
+
+ $ phc2sys -f gPTP.cfg -s $IFNAME -c CLOCK_REALTIME -w
+
+The commands above should be run on both AVTP Talker and Listener hosts.
+
+FQTSS Setup
+-----------
+
+The Linux Traffic Control system provides the mqprio and cbs qdiscs which
+enable FQTSS on Linux. Below we provide an example to configure those qdiscs in
+order to transmit an AAF stream with the following features: class A, 6 audio
+frames per AVTPDU, 48 kHz sampling rate, 16-bit sample size, stereo. For
+further information on how to configure these qdiscs check tc-mqprio(8) and
+tc-cbs(8) man pages.
+
+On the host that will run as AVTP Talker (i.e. plugin in playback mode), run
+the following commands:
+
+Configure mpqrio qdisc (replace $HANDLE_ID by an unused handle ID):
+
+ $ tc qdisc add dev $IFNAME parent root handle $HANDLE_ID mqprio \
+ num_tc 3 map 2 2 1 0 2 2 2 2 2 2 2 2 2 2 2 2 \
+ queues 1@0 1@1 2@2 hw 0
+
+Configure cbs qdisc:
+
+ $ tc qdisc replace dev $IFNAME parent $HANDLE_ID:1 cbs idleslope 5760 \
+ sendslope -994240 hicredit 9 locredit -89 offload 1
+
+No FQTSS configuration is required at the host running as AVTP Listener.
+
Plugin Dependencies
-------------------
@@ -47,3 +120,29 @@ as follows:
operating in playback mode.
* frames_per_pdu: Number of audio frames transmitted in one AVTPDU.
+
+Plugin Usage
+------------
+
+The plugin provides the PCM type "aaf". Configure an AAF PCM virtual device
+according to the AAF stream you want to transmit or receive. A hypothetical
+configuration for the stream described in the 'FQTSS Setup' section is shown
+below:
+
+ pcm.aaf0 {
+ type aaf
+ ifname eth0
+ addr 01:AA:AA:AA:AA:AA
+ prio 3
+ streamid AA:BB:CC:DD:EE:FF:0000
+ mtt 2000
+ time_uncertainty 125
+ frames_per_pdu 6
+ }
+
+Put the above to ~/.asoundrc (or /etc/asound.conf), and use the AAF PCM virtual
+device 'aaf0' with your favorite alsa-utils tool.
+
+For example, to stream the pink noise generated by 'speaker-test', run:
+
+ $ speaker-test -F S16_BE -c 2 -r 48000 -D aaf0
--
2.14.4
^ permalink raw reply related [flat|nested] 12+ messages in thread* [PATCH v2 - AAF PCM plugin 4/7] aaf: Prepare for Capture mode support
2018-10-25 1:11 [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Andre Guedes
` (2 preceding siblings ...)
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 3/7] aaf: Implement Playback mode support Andre Guedes
@ 2018-10-25 1:11 ` Andre Guedes
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 5/7] aaf: Implement " Andre Guedes
` (3 subsequent siblings)
7 siblings, 0 replies; 12+ messages in thread
From: Andre Guedes @ 2018-10-25 1:11 UTC (permalink / raw)
To: alsa-devel; +Cc: tiwai, pierre-louis.bossart, liam.r.girdwood
The plugin code assumes only Playback mode is supported. This patch
prepares the code to support both Playback and Capture mode. Capture
mode support is implemented by a follow-up patch.
Signed-off-by: Andre Guedes <andre.guedes@intel.com>
---
aaf/pcm_aaf.c | 159 ++++++++++++++++++++++++++++++++++------------------------
1 file changed, 94 insertions(+), 65 deletions(-)
diff --git a/aaf/pcm_aaf.c b/aaf/pcm_aaf.c
index 86bfe50..fba0752 100644
--- a/aaf/pcm_aaf.c
+++ b/aaf/pcm_aaf.c
@@ -239,6 +239,7 @@ static int aaf_init_socket(snd_pcm_aaf_t *aaf)
{
int fd, res;
struct ifreq req;
+ snd_pcm_ioplug_t *io = &aaf->io;
fd = socket(AF_PACKET, SOCK_DGRAM|SOCK_NONBLOCK, htons(ETH_P_TSN));
if (fd < 0) {
@@ -260,12 +261,17 @@ static int aaf_init_socket(snd_pcm_aaf_t *aaf)
aaf->sk_addr.sll_ifindex = req.ifr_ifindex;
memcpy(&aaf->sk_addr.sll_addr, aaf->addr, ETH_ALEN);
- res = setsockopt(fd, SOL_SOCKET, SO_PRIORITY, &aaf->prio,
- sizeof(aaf->prio));
- if (res < 0) {
- SNDERR("Failed to set socket priority");
- res = -errno;
- goto err;
+ if (io->stream == SND_PCM_STREAM_PLAYBACK) {
+ res = setsockopt(fd, SOL_SOCKET, SO_PRIORITY, &aaf->prio,
+ sizeof(aaf->prio));
+ if (res < 0) {
+ SNDERR("Failed to set socket priority");
+ res = -errno;
+ goto err;
+ }
+ } else {
+ /* TODO: Implement Capture mode support. */
+ return -ENOTSUP;
}
aaf->sk_fd = fd;
@@ -305,46 +311,50 @@ static int aaf_init_pdu(snd_pcm_aaf_t *aaf)
if (!pdu)
return -ENOMEM;
- res = avtp_aaf_pdu_init(pdu);
- if (res < 0)
- goto err;
+ if (io->stream == SND_PCM_STREAM_PLAYBACK) {
+ res = avtp_aaf_pdu_init(pdu);
+ if (res < 0)
+ goto err;
- res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_TV, 1);
- if (res < 0)
- goto err;
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_TV, 1);
+ if (res < 0)
+ goto err;
- res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_STREAM_ID, aaf->streamid);
- if (res < 0)
- goto err;
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_STREAM_ID,
+ aaf->streamid);
+ if (res < 0)
+ goto err;
- res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_FORMAT,
- alsa_to_avtp_format(io->format));
- if (res < 0)
- goto err;
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_FORMAT,
+ alsa_to_avtp_format(io->format));
+ if (res < 0)
+ goto err;
- res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_NSR,
- alsa_to_avtp_rate(io->rate));
- if (res < 0)
- goto err;
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_NSR,
+ alsa_to_avtp_rate(io->rate));
+ if (res < 0)
+ goto err;
- res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_CHAN_PER_FRAME,
- io->channels);
- if (res < 0)
- goto err;
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_CHAN_PER_FRAME,
+ io->channels);
+ if (res < 0)
+ goto err;
- res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_BIT_DEPTH,
- snd_pcm_format_width(io->format));
- if (res < 0)
- goto err;
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_BIT_DEPTH,
+ snd_pcm_format_width(io->format));
+ if (res < 0)
+ goto err;
- res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_STREAM_DATA_LEN,
- payload_size);
- if (res < 0)
- goto err;
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_STREAM_DATA_LEN,
+ payload_size);
+ if (res < 0)
+ goto err;
- res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_SP, AVTP_AAF_PCM_SP_NORMAL);
- if (res < 0)
- goto err;
+ res = avtp_aaf_pdu_set(pdu, AVTP_AAF_FIELD_SP,
+ AVTP_AAF_PCM_SP_NORMAL);
+ if (res < 0)
+ goto err;
+ }
aaf->pdu = pdu;
aaf->pdu_size = pdu_size;
@@ -699,7 +709,10 @@ static snd_pcm_sframes_t aaf_pointer(snd_pcm_ioplug_t *io)
static int aaf_poll_descriptors_count(snd_pcm_ioplug_t *io ATTRIBUTE_UNUSED)
{
- return FD_COUNT_PLAYBACK;
+ if (io->stream == SND_PCM_STREAM_PLAYBACK)
+ return FD_COUNT_PLAYBACK;
+ else
+ return -ENOTSUP;
}
static int aaf_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd,
@@ -707,11 +720,17 @@ static int aaf_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd,
{
snd_pcm_aaf_t *aaf = io->private_data;
- if (space != FD_COUNT_PLAYBACK)
- return -EINVAL;
+ if (io->stream == SND_PCM_STREAM_PLAYBACK) {
+ if (space != FD_COUNT_PLAYBACK)
+ return -EINVAL;
+
+ pfd[0].fd = aaf->timer_fd;
+ pfd[0].events = POLLIN;
+ } else {
+ /* TODO: Implement Capture mode support. */
+ return -ENOTSUP;
+ }
- pfd[0].fd = aaf->timer_fd;
- pfd[0].events = POLLIN;
return space;
}
@@ -721,15 +740,20 @@ static int aaf_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfd,
int res;
snd_pcm_aaf_t *aaf = io->private_data;
- if (nfds != FD_COUNT_PLAYBACK)
- return -EINVAL;
+ if (io->stream == SND_PCM_STREAM_PLAYBACK) {
+ if (nfds != FD_COUNT_PLAYBACK)
+ return -EINVAL;
- if (pfd[0].revents & POLLIN) {
- res = aaf_mclk_timeout_playback(aaf);
- if (res < 0)
- return res;
+ if (pfd[0].revents & POLLIN) {
+ res = aaf_mclk_timeout_playback(aaf);
+ if (res < 0)
+ return res;
- *revents = POLLIN;
+ *revents = POLLIN;
+ }
+ } else {
+ /* TODO: Implement Capture mode support. */
+ return -ENOTSUP;
}
return 0;
@@ -755,9 +779,14 @@ static int aaf_start(snd_pcm_ioplug_t *io)
int res;
snd_pcm_aaf_t *aaf = io->private_data;
- res = aaf_mclk_start_playback(aaf);
- if (res < 0)
- return res;
+ if (io->stream == SND_PCM_STREAM_PLAYBACK) {
+ res = aaf_mclk_start_playback(aaf);
+ if (res < 0)
+ return res;
+ } else {
+ /* TODO: Implement Capture mode support. */
+ return -ENOTSUP;
+ }
return 0;
}
@@ -782,12 +811,18 @@ static snd_pcm_sframes_t aaf_transfer(snd_pcm_ioplug_t *io,
int res;
snd_pcm_aaf_t *aaf = io->private_data;
- res = snd_pcm_areas_copy_wrap(aaf->audiobuf_areas,
- (io->appl_ptr % io->buffer_size),
- io->buffer_size, areas, offset, size,
- io->channels, size, io->format);
- if (res < 0)
- return res;
+ if (io->stream == SND_PCM_STREAM_PLAYBACK) {
+ res = snd_pcm_areas_copy_wrap(aaf->audiobuf_areas,
+ (io->appl_ptr % io->buffer_size),
+ io->buffer_size, areas, offset,
+ size, io->channels, size,
+ io->format);
+ if (res < 0)
+ return res;
+ } else {
+ /* TODO: Implement Capture mode support. */
+ return -ENOTSUP;
+ }
return size;
}
@@ -812,12 +847,6 @@ SND_PCM_PLUGIN_DEFINE_FUNC(aaf)
snd_pcm_aaf_t *aaf;
int res;
- /* For now the plugin only supports Playback mode i.e. AAF Talker
- * functionality.
- */
- if (stream != SND_PCM_STREAM_PLAYBACK)
- return -EINVAL;
-
aaf = calloc(1, sizeof(*aaf));
if (!aaf) {
SNDERR("Failed to allocate memory");
--
2.14.4
^ permalink raw reply related [flat|nested] 12+ messages in thread* [PATCH v2 - AAF PCM plugin 5/7] aaf: Implement Capture mode support
2018-10-25 1:11 [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Andre Guedes
` (3 preceding siblings ...)
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 4/7] aaf: Prepare for Capture " Andre Guedes
@ 2018-10-25 1:11 ` Andre Guedes
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 6/7] aaf: Implement dump() ioplug callback Andre Guedes
` (2 subsequent siblings)
7 siblings, 0 replies; 12+ messages in thread
From: Andre Guedes @ 2018-10-25 1:11 UTC (permalink / raw)
To: alsa-devel; +Cc: tiwai, pierre-louis.bossart, liam.r.girdwood
This patch implements the capture mode support from the AAF plugin.
Simply put, this mode works as follows: AVTPDUs are received from the
network, the PCM samples are retrieved and presented to the alsa-lib
layer at the presentation time. In summary, the capture mode implements
a typical AVTP Listener.
Once the AAF device is put in running state, packet reception is
started. Every time an AVTPDU is received, the plugin checks if it is
valid (according to the stream configuration provided by the user) and
copies the PCM samples to the audio buffer. Note that at this moment,
the samples are not presented to the alsa-lib layer yet (i.e. hw_ptr is
not incremented).
The media clock starts at the presentation time from the first AVTPDU.
At every tick from the media clock, PCM samples are presented to the
alsa-lib layer.
Below follows some secondary discussion about this patch:
The functions aaf_mclk_start_playback() and aaf_mclk_start_capture()
have some common code. This patch does some code refactoring so the
common code is put in a new function (aaf_mclk_start) which is called
by both aaf_mclk_start_playback() and aaf_mclk_start_capture().
Also, the helper function aaf_inc_hw_ptr() is refactored so it can be
used to increment both aaf->hw_ptr and aaf->hw_virt_ptr.
Signed-off-by: Andre Guedes <andre.guedes@intel.com>
---
aaf/pcm_aaf.c | 425 +++++++++++++++++++++++++++++++++++++++++++++++++++++-----
doc/aaf.txt | 9 ++
2 files changed, 400 insertions(+), 34 deletions(-)
diff --git a/aaf/pcm_aaf.c b/aaf/pcm_aaf.c
index fba0752..147e5b0 100644
--- a/aaf/pcm_aaf.c
+++ b/aaf/pcm_aaf.c
@@ -28,6 +28,7 @@
#include <linux/if_packet.h>
#include <net/if.h>
#include <string.h>
+#include <stdbool.h>
#include <stdint.h>
#include <sys/ioctl.h>
#include <sys/timerfd.h>
@@ -46,6 +47,7 @@
#define TAI_TO_UTC(t) (t - TAI_OFFSET)
#define FD_COUNT_PLAYBACK 1
+#define FD_COUNT_CAPTURE 2
typedef struct {
snd_pcm_ioplug_t io;
@@ -75,7 +77,10 @@ typedef struct {
snd_pcm_channel_area_t *payload_areas;
snd_pcm_uframes_t hw_ptr;
+ snd_pcm_uframes_t hw_virt_ptr;
snd_pcm_uframes_t boundary;
+
+ uint64_t prev_ptime;
} snd_pcm_aaf_t;
static unsigned int alsa_to_avtp_format(snd_pcm_format_t format)
@@ -122,6 +127,106 @@ static unsigned int alsa_to_avtp_rate(unsigned int rate)
}
}
+static bool is_pdu_valid(struct avtp_stream_pdu *pdu, uint64_t streamid,
+ unsigned int data_len, unsigned int format,
+ unsigned int nsr, unsigned int channels,
+ unsigned int depth)
+{
+ int res;
+ uint64_t val64;
+ uint32_t val32;
+ struct avtp_common_pdu *common = (struct avtp_common_pdu *) pdu;
+
+ res = avtp_pdu_get(common, AVTP_FIELD_SUBTYPE, &val32);
+ if (res < 0)
+ return false;
+ if (val32 != AVTP_SUBTYPE_AAF) {
+ pr_debug("Subtype mismatch: expected %u, got %u",
+ AVTP_SUBTYPE_AAF, val32);
+ return false;
+ }
+
+ res = avtp_pdu_get(common, AVTP_FIELD_VERSION, &val32);
+ if (res < 0)
+ return false;
+ if (val32 != 0) {
+ pr_debug("Version mismatch: expected %u, got %u", 0, val32);
+ return false;
+ }
+
+ res = avtp_aaf_pdu_get(pdu, AVTP_AAF_FIELD_STREAM_ID, &val64);
+ if (res < 0)
+ return false;
+ if (val64 != streamid) {
+ pr_debug("Streamid mismatch: expected %lu, got %lu", streamid,
+ val64);
+ return false;
+ }
+
+ res = avtp_aaf_pdu_get(pdu, AVTP_AAF_FIELD_TV, &val64);
+ if (res < 0)
+ return false;
+ if (val64 != 1) {
+ pr_debug("TV mismatch: expected %u, got %lu", 1, val64);
+ return false;
+ }
+
+ res = avtp_aaf_pdu_get(pdu, AVTP_AAF_FIELD_SP, &val64);
+ if (res < 0)
+ return false;
+ if (val64 != AVTP_AAF_PCM_SP_NORMAL) {
+ pr_debug("SP mismatch: expected %u, got %lu",
+ AVTP_AAF_PCM_SP_NORMAL, val64);
+ return false;
+ }
+
+ res = avtp_aaf_pdu_get(pdu, AVTP_AAF_FIELD_FORMAT, &val64);
+ if (res < 0)
+ return false;
+ if (val64 != format) {
+ pr_debug("Format mismatch: expected %u, got %lu", format,
+ val64);
+ return false;
+ }
+
+ res = avtp_aaf_pdu_get(pdu, AVTP_AAF_FIELD_NSR, &val64);
+ if (res < 0)
+ return false;
+ if (val64 != nsr) {
+ pr_debug("NSR mismatch: expected %u, got %lu", nsr, val64);
+ return false;
+ }
+
+ res = avtp_aaf_pdu_get(pdu, AVTP_AAF_FIELD_CHAN_PER_FRAME, &val64);
+ if (res < 0)
+ return false;
+ if (val64 != channels) {
+ pr_debug("Channels mismatch: expected %u, got %lu", channels,
+ val64);
+ return false;
+ }
+
+ res = avtp_aaf_pdu_get(pdu, AVTP_AAF_FIELD_BIT_DEPTH, &val64);
+ if (res < 0)
+ return false;
+ if (val64 != depth) {
+ pr_debug("Bit depth mismatch: expected %u, got %lu", depth,
+ val64);
+ return false;
+ }
+
+ res = avtp_aaf_pdu_get(pdu, AVTP_AAF_FIELD_STREAM_DATA_LEN, &val64);
+ if (res < 0)
+ return false;
+ if (val64 != data_len) {
+ pr_debug("Data len mismatch: expected %u, got %lu",
+ data_len, val64);
+ return false;
+ }
+
+ return true;
+}
+
static int aaf_load_config(snd_pcm_aaf_t *aaf, snd_config_t *conf)
{
snd_config_iterator_t cur, next;
@@ -270,8 +375,27 @@ static int aaf_init_socket(snd_pcm_aaf_t *aaf)
goto err;
}
} else {
- /* TODO: Implement Capture mode support. */
- return -ENOTSUP;
+ struct packet_mreq mreq = { 0 };
+
+ res = bind(fd, (struct sockaddr *) &aaf->sk_addr,
+ sizeof(aaf->sk_addr));
+ if (res < 0) {
+ SNDERR("Failed to bind socket");
+ res = -errno;
+ goto err;
+ }
+
+ mreq.mr_ifindex = req.ifr_ifindex;
+ mreq.mr_type = PACKET_MR_MULTICAST;
+ mreq.mr_alen = ETH_ALEN;
+ memcpy(&mreq.mr_address, aaf->addr, ETH_ALEN);
+ res = setsockopt(fd, SOL_PACKET, PACKET_ADD_MEMBERSHIP,
+ &mreq, sizeof(struct packet_mreq));
+ if (res < 0) {
+ SNDERR("Failed to add multicast address");
+ res = -errno;
+ goto err;
+ }
}
aaf->sk_fd = fd;
@@ -444,35 +568,26 @@ err_free_buf:
return res;
}
-static void aaf_inc_hw_ptr(snd_pcm_aaf_t *aaf, snd_pcm_uframes_t val)
+static void aaf_inc_ptr(snd_pcm_uframes_t *ptr, snd_pcm_uframes_t val,
+ snd_pcm_uframes_t boundary)
{
- aaf->hw_ptr += val;
+ *ptr += val;
- if (aaf->hw_ptr >= aaf->boundary)
- aaf->hw_ptr -= aaf->boundary;
+ if (*ptr > boundary)
+ *ptr -= boundary;
}
-static int aaf_mclk_start_playback(snd_pcm_aaf_t *aaf)
+static int aaf_mclk_start(snd_pcm_aaf_t *aaf, uint64_t time, uint64_t period)
{
int res;
- struct timespec now;
struct itimerspec itspec;
uint64_t time_utc;
- snd_pcm_ioplug_t *io = &aaf->io;
-
- res = clock_gettime(CLOCK_TAI, &now);
- if (res < 0) {
- SNDERR("Failed to get time from clock");
- return -errno;
- }
aaf->timer_expirations = 0;
- aaf->timer_period = (uint64_t)NSEC_PER_SEC * aaf->frames_per_pdu /
- io->rate;
- aaf->timer_starttime = now.tv_sec * NSEC_PER_SEC + now.tv_nsec +
- aaf->timer_period;
+ aaf->timer_period = period;
+ aaf->timer_starttime = time;
- time_utc = TAI_TO_UTC(aaf->timer_starttime);
+ time_utc = TAI_TO_UTC(time);
itspec.it_value.tv_sec = time_utc / NSEC_PER_SEC;
itspec.it_value.tv_nsec = time_utc % NSEC_PER_SEC;
itspec.it_interval.tv_sec = 0;
@@ -484,6 +599,65 @@ static int aaf_mclk_start_playback(snd_pcm_aaf_t *aaf)
return 0;
}
+static int aaf_mclk_start_playback(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ struct timespec now;
+ uint64_t time, period;
+ snd_pcm_ioplug_t *io = &aaf->io;
+
+ res = clock_gettime(CLOCK_TAI, &now);
+ if (res < 0) {
+ SNDERR("Failed to get time from clock");
+ return -errno;
+ }
+
+ period = (uint64_t)NSEC_PER_SEC * aaf->frames_per_pdu / io->rate;
+ time = now.tv_sec * NSEC_PER_SEC + now.tv_nsec + period;
+ res = aaf_mclk_start(aaf, time, period);
+ if (res < 0)
+ return res;
+
+ return 0;
+}
+
+static int aaf_mclk_start_capture(snd_pcm_aaf_t *aaf, uint32_t avtp_time)
+{
+ int res;
+ struct timespec tspec;
+ uint64_t now, ptime, period;
+ snd_pcm_ioplug_t *io = &aaf->io;
+
+ res = clock_gettime(CLOCK_TAI, &tspec);
+ if (res < 0) {
+ SNDERR("Failed to get time from clock");
+ return -errno;
+ }
+
+ now = (uint64_t)tspec.tv_sec * NSEC_PER_SEC + tspec.tv_nsec;
+
+ /* The avtp_timestamp within AAF packet is the lower part (32
+ * less-significant bits) from presentation time calculated by the
+ * talker.
+ */
+ ptime = (now & 0xFFFFFFFF00000000ULL) | avtp_time;
+
+ /* If 'ptime' is less than the 'now', it means the higher part
+ * from 'ptime' needs to be incremented by 1 in order to recover the
+ * presentation time set by the talker.
+ */
+ if (ptime < now)
+ ptime += (1ULL << 32);
+
+ period = (uint64_t)NSEC_PER_SEC * aaf->frames_per_pdu / io->rate;
+ res = aaf_mclk_start(aaf, ptime, period);
+ if (res < 0)
+ return res;
+
+ aaf->prev_ptime = ptime;
+ return 0;
+}
+
static int aaf_mclk_reset(snd_pcm_aaf_t *aaf)
{
int res;
@@ -556,7 +730,124 @@ static int aaf_tx_pdu(snd_pcm_aaf_t *aaf)
return -EIO;
}
- aaf_inc_hw_ptr(aaf, aaf->frames_per_pdu);
+ aaf_inc_ptr(&aaf->hw_ptr, aaf->frames_per_pdu, aaf->boundary);
+ return 0;
+}
+
+static int aaf_rx_pdu(snd_pcm_aaf_t *aaf)
+{
+ int res;
+ ssize_t n;
+ uint64_t seq, avtp_time;
+ snd_pcm_uframes_t hw_avail;
+ snd_pcm_ioplug_t *io = &aaf->io;
+ snd_pcm_t *pcm = io->pcm;
+
+ n = recv(aaf->sk_fd, aaf->pdu, aaf->pdu_size, 0);
+ if (n < 0) {
+ SNDERR("Failed to receive data");
+ return -errno;
+ }
+ if (n != aaf->pdu_size) {
+ pr_debug("AVTPDU dropped: Invalid size");
+ return 0;
+ }
+
+ if (io->state == SND_PCM_STATE_DRAINING) {
+ /* If device is in DRAIN state, we shouldn't copy any more data
+ * to audio buffer. So we are done here.
+ */
+ return 0;
+ }
+
+ if (!is_pdu_valid(aaf->pdu, aaf->streamid,
+ snd_pcm_frames_to_bytes(pcm, aaf->frames_per_pdu),
+ alsa_to_avtp_format(io->format),
+ alsa_to_avtp_rate(io->rate),
+ io->channels, snd_pcm_format_width(io->format))) {
+ pr_debug("AVTPDU dropped: Bad field(s)");
+ return 0;
+ }
+
+ res = avtp_aaf_pdu_get(aaf->pdu, AVTP_AAF_FIELD_SEQ_NUM, &seq);
+ if (res < 0)
+ return res;
+ if (seq != aaf->pdu_seq) {
+ pr_debug("Sequence mismatch: expected %u, got %lu",
+ aaf->pdu_seq, seq);
+ aaf->pdu_seq = seq;
+ }
+ aaf->pdu_seq++;
+
+ res = avtp_aaf_pdu_get(aaf->pdu, AVTP_AAF_FIELD_TIMESTAMP, &avtp_time);
+ if (res < 0)
+ return res;
+
+ if (aaf->timer_starttime == 0) {
+ res = aaf_mclk_start_capture(aaf, avtp_time);
+ if (res < 0)
+ return res;
+ } else {
+ uint64_t ptime = aaf->prev_ptime + aaf->timer_period;
+
+ if (avtp_time != ptime % (1ULL << 32)) {
+ pr_debug("Packet dropped: PT not expected");
+ return 0;
+ }
+
+ if (ptime < aaf_mclk_gettime(aaf)) {
+ pr_debug("Packet dropped: PT in the past");
+ return 0;
+ }
+
+ aaf->prev_ptime = ptime;
+ }
+
+ hw_avail = snd_pcm_ioplug_hw_avail(io, aaf->hw_virt_ptr, io->appl_ptr);
+ if (hw_avail < aaf->frames_per_pdu) {
+ /* If there isn't enough space available on buffer to copy the
+ * samples from AVTPDU, it means we've reached an overrun
+ * state.
+ */
+ return -EPIPE;
+ }
+
+ res = snd_pcm_areas_copy_wrap(aaf->audiobuf_areas,
+ (aaf->hw_virt_ptr % io->buffer_size),
+ io->buffer_size, aaf->payload_areas,
+ 0, aaf->frames_per_pdu, io->channels,
+ aaf->frames_per_pdu, io->format);
+ if (res < 0) {
+ SNDERR("Failed to copy data from AVTP payload");
+ return res;
+ }
+
+ aaf_inc_ptr(&aaf->hw_virt_ptr, aaf->frames_per_pdu, aaf->boundary);
+ return 0;
+}
+
+static int aaf_flush_rx_buf(snd_pcm_aaf_t *aaf)
+{
+ char *tmp;
+ ssize_t n;
+
+ tmp = malloc(aaf->pdu_size);
+ if (!tmp)
+ return -ENOMEM;
+
+ do {
+ n = recv(aaf->sk_fd, tmp, aaf->pdu_size, 0);
+ } while (n != -1);
+
+ if (errno != EAGAIN && errno != EWOULDBLOCK) {
+ /* Something unexpected has happened while flushing the socket
+ * rx buffer so we return error.
+ */
+ free(tmp);
+ return -errno;
+ }
+
+ free(tmp);
return 0;
}
@@ -586,6 +877,43 @@ static int aaf_mclk_timeout_playback(snd_pcm_aaf_t *aaf)
return 0;
}
+static int aaf_mclk_timeout_capture(snd_pcm_aaf_t *aaf)
+{
+ ssize_t n;
+ uint64_t expirations;
+ snd_pcm_sframes_t len;
+ snd_pcm_ioplug_t *io = &aaf->io;
+
+ n = read(aaf->timer_fd, &expirations, sizeof(uint64_t));
+ if (n < 0) {
+ SNDERR("Failed to read() timer");
+ return -errno;
+ }
+
+ if (expirations != 1)
+ pr_debug("Missed %llu presentation time(s) ", expirations - 1);
+
+ while (expirations--) {
+ aaf->timer_expirations++;
+
+ len = aaf->hw_virt_ptr - aaf->hw_ptr;
+ if (len < 0)
+ len += aaf->boundary;
+
+ if ((snd_pcm_uframes_t) len > io->buffer_size) {
+ /* If the distance between hw virtual pointer and hw
+ * pointer is greater than the buffer size, it means we
+ * had an overrun error so -EPIPE is returned.
+ */
+ return -EPIPE;
+ }
+
+ aaf_inc_ptr(&aaf->hw_ptr, aaf->frames_per_pdu, aaf->boundary);
+ }
+
+ return 0;
+}
+
static int aaf_set_hw_constraint(snd_pcm_aaf_t *aaf)
{
int res;
@@ -712,7 +1040,7 @@ static int aaf_poll_descriptors_count(snd_pcm_ioplug_t *io ATTRIBUTE_UNUSED)
if (io->stream == SND_PCM_STREAM_PLAYBACK)
return FD_COUNT_PLAYBACK;
else
- return -ENOTSUP;
+ return FD_COUNT_CAPTURE;
}
static int aaf_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd,
@@ -727,8 +1055,13 @@ static int aaf_poll_descriptors(snd_pcm_ioplug_t *io, struct pollfd *pfd,
pfd[0].fd = aaf->timer_fd;
pfd[0].events = POLLIN;
} else {
- /* TODO: Implement Capture mode support. */
- return -ENOTSUP;
+ if (space != FD_COUNT_CAPTURE)
+ return -EINVAL;
+
+ pfd[0].fd = aaf->timer_fd;
+ pfd[0].events = POLLIN;
+ pfd[1].fd = aaf->sk_fd;
+ pfd[1].events = POLLIN;
}
return space;
@@ -752,8 +1085,22 @@ static int aaf_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfd,
*revents = POLLIN;
}
} else {
- /* TODO: Implement Capture mode support. */
- return -ENOTSUP;
+ if (nfds != FD_COUNT_CAPTURE)
+ return -EINVAL;
+
+ if (pfd[0].revents & POLLIN) {
+ res = aaf_mclk_timeout_capture(aaf);
+ if (res < 0)
+ return res;
+
+ *revents = POLLIN;
+ }
+
+ if (pfd[1].revents & POLLIN) {
+ res = aaf_rx_pdu(aaf);
+ if (res < 0)
+ return res;
+ }
}
return 0;
@@ -766,6 +1113,8 @@ static int aaf_prepare(snd_pcm_ioplug_t *io)
aaf->pdu_seq = 0;
aaf->hw_ptr = 0;
+ aaf->hw_virt_ptr = 0;
+ aaf->prev_ptime = 0;
res = aaf_mclk_reset(aaf);
if (res < 0)
@@ -781,13 +1130,17 @@ static int aaf_start(snd_pcm_ioplug_t *io)
if (io->stream == SND_PCM_STREAM_PLAYBACK) {
res = aaf_mclk_start_playback(aaf);
- if (res < 0)
- return res;
} else {
- /* TODO: Implement Capture mode support. */
- return -ENOTSUP;
+ /* Discard any packet on socket buffer to ensure the plugin
+ * process only packets that arrived after the device has
+ * started.
+ */
+ res = aaf_flush_rx_buf(aaf);
}
+ if (res < 0)
+ return res;
+
return 0;
}
@@ -817,13 +1170,17 @@ static snd_pcm_sframes_t aaf_transfer(snd_pcm_ioplug_t *io,
io->buffer_size, areas, offset,
size, io->channels, size,
io->format);
- if (res < 0)
- return res;
} else {
- /* TODO: Implement Capture mode support. */
- return -ENOTSUP;
+ res = snd_pcm_areas_copy_wrap(areas, offset, (offset + size),
+ aaf->audiobuf_areas,
+ (io->appl_ptr % io->buffer_size),
+ io->buffer_size, io->channels,
+ size, io->format);
}
+ if (res < 0)
+ return res;
+
return size;
}
diff --git a/doc/aaf.txt b/doc/aaf.txt
index c278166..58d9c02 100644
--- a/doc/aaf.txt
+++ b/doc/aaf.txt
@@ -146,3 +146,12 @@ device 'aaf0' with your favorite alsa-utils tool.
For example, to stream the pink noise generated by 'speaker-test', run:
$ speaker-test -F S16_BE -c 2 -r 48000 -D aaf0
+
+To receive the AAF stream generated by the command above, run the following
+command in another host:
+
+ $ arecord -t raw -f S16_BE -c 2 -r 48000 -D aaf0 -vv /dev/null
+
+If you want to playback the contents of the AAF stream, change the command-line
+above so the output from 'arecord' is redirected to 'aplay', or simply use the
+'alsaloop' tool.
--
2.14.4
^ permalink raw reply related [flat|nested] 12+ messages in thread* [PATCH v2 - AAF PCM plugin 6/7] aaf: Implement dump() ioplug callback
2018-10-25 1:11 [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Andre Guedes
` (4 preceding siblings ...)
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 5/7] aaf: Implement " Andre Guedes
@ 2018-10-25 1:11 ` Andre Guedes
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 7/7] aaf: Add support for direct read/write transfers Andre Guedes
2018-10-25 6:41 ` [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Jaroslav Kysela
7 siblings, 0 replies; 12+ messages in thread
From: Andre Guedes @ 2018-10-25 1:11 UTC (permalink / raw)
To: alsa-devel; +Cc: tiwai, pierre-louis.bossart, liam.r.girdwood
This patch introduces the aaf_dump() function which prints information
about both PCM and AVTP setup.
Signed-off-by: Andre Guedes <andre.guedes@intel.com>
---
aaf/pcm_aaf.c | 24 ++++++++++++++++++++++++
1 file changed, 24 insertions(+)
diff --git a/aaf/pcm_aaf.c b/aaf/pcm_aaf.c
index 147e5b0..b022575 100644
--- a/aaf/pcm_aaf.c
+++ b/aaf/pcm_aaf.c
@@ -23,6 +23,7 @@
#include <arpa/inet.h>
#include <avtp.h>
#include <avtp_aaf.h>
+#include <inttypes.h>
#include <limits.h>
#include <linux/if_ether.h>
#include <linux/if_packet.h>
@@ -964,6 +965,28 @@ static int aaf_close(snd_pcm_ioplug_t *io)
return 0;
}
+static void aaf_dump(snd_pcm_ioplug_t *io, snd_output_t *out)
+{
+ snd_pcm_aaf_t *aaf = io->private_data;
+ snd_pcm_t *pcm = io->pcm;
+
+ snd_output_printf(out, "%s\n", io->name);
+ snd_output_printf(out, "PCM setup is:\n");
+ snd_pcm_dump_setup(pcm, out);
+ snd_output_printf(out, "AVTP setup is:\n");
+ snd_output_printf(out, " ifname: %s\n", aaf->ifname);
+ snd_output_printf(out, " macaddr: %.2X:%.2X:%.2X:%.2X:%.2X:%.2X\n",
+ aaf->addr[0], aaf->addr[1], aaf->addr[2],
+ aaf->addr[3], aaf->addr[4], aaf->addr[5]);
+ snd_output_printf(out, " priority: %d\n", aaf->prio);
+ snd_output_printf(out, " streamid: %"PRIX64"\n", aaf->streamid);
+ snd_output_printf(out, " mtt: %d\n", aaf->mtt / NSEC_PER_USEC);
+ snd_output_printf(out, " time uncertainty: %d\n",
+ aaf->t_uncertainty / NSEC_PER_USEC);
+ snd_output_printf(out, " frames per AVTPDU: %lu\n",
+ aaf->frames_per_pdu);
+}
+
static int aaf_hw_params(snd_pcm_ioplug_t *io,
snd_pcm_hw_params_t *params ATTRIBUTE_UNUSED)
{
@@ -1186,6 +1209,7 @@ static snd_pcm_sframes_t aaf_transfer(snd_pcm_ioplug_t *io,
static const snd_pcm_ioplug_callback_t aaf_callback = {
.close = aaf_close,
+ .dump = aaf_dump,
.hw_params = aaf_hw_params,
.hw_free = aaf_hw_free,
.sw_params = aaf_sw_params,
--
2.14.4
^ permalink raw reply related [flat|nested] 12+ messages in thread* [PATCH v2 - AAF PCM plugin 7/7] aaf: Add support for direct read/write transfers
2018-10-25 1:11 [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Andre Guedes
` (5 preceding siblings ...)
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 6/7] aaf: Implement dump() ioplug callback Andre Guedes
@ 2018-10-25 1:11 ` Andre Guedes
2018-10-25 6:41 ` [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Jaroslav Kysela
7 siblings, 0 replies; 12+ messages in thread
From: Andre Guedes @ 2018-10-25 1:11 UTC (permalink / raw)
To: alsa-devel; +Cc: tiwai, pierre-louis.bossart, liam.r.girdwood
This patch adds support for direct read/write transfers (i.e. mmap
access mode) to the AAF plugin.
In order to enable direct read/write transfers, the AAF plugin is
required to implement ioplug's pseudo mmap mode. In this mode, the audio
buffer management (e.g. areas allocation, clean up, and data copy) is
handled at upper layers, making the AAF plugin simpler. So this patch
removes all code related to audio buffer management as well as the
transfer() callback from the AAF plugin.
Signed-off-by: Andre Guedes <andre.guedes@intel.com>
---
aaf/pcm_aaf.c | 78 +++--------------------------------------------------------
1 file changed, 4 insertions(+), 74 deletions(-)
diff --git a/aaf/pcm_aaf.c b/aaf/pcm_aaf.c
index b022575..65fa0fb 100644
--- a/aaf/pcm_aaf.c
+++ b/aaf/pcm_aaf.c
@@ -74,7 +74,7 @@ typedef struct {
uint64_t timer_period;
uint64_t timer_expirations;
- snd_pcm_channel_area_t *audiobuf_areas;
+ const snd_pcm_channel_area_t *audiobuf_areas;
snd_pcm_channel_area_t *payload_areas;
snd_pcm_uframes_t hw_ptr;
@@ -533,42 +533,6 @@ err:
return res;
}
-static int aaf_init_audiobuf_areas(snd_pcm_aaf_t *aaf)
-{
- int res;
- char *buf;
- ssize_t frame_size;
- snd_pcm_channel_area_t *areas;
- snd_pcm_ioplug_t *io = &aaf->io;
-
- frame_size = snd_pcm_format_size(io->format, io->channels);
- if (frame_size < 0)
- return frame_size;
-
- buf = calloc(io->buffer_size, frame_size);
- if (!buf)
- return -ENOMEM;
-
- areas = calloc(io->channels, sizeof(snd_pcm_channel_area_t));
- if (!areas) {
- res = -ENOMEM;
- goto err_free_buf;
- }
-
- res = aaf_init_areas(aaf, areas, buf);
- if (res < 0)
- goto err_free_areas;
-
- aaf->audiobuf_areas = areas;
- return 0;
-
-err_free_areas:
- free(areas);
-err_free_buf:
- free(buf);
- return res;
-}
-
static void aaf_inc_ptr(snd_pcm_uframes_t *ptr, snd_pcm_uframes_t val,
snd_pcm_uframes_t boundary)
{
@@ -921,6 +885,7 @@ static int aaf_set_hw_constraint(snd_pcm_aaf_t *aaf)
snd_pcm_ioplug_t *io = &aaf->io;
static const unsigned int accesses[] = {
SND_PCM_ACCESS_RW_INTERLEAVED,
+ SND_PCM_ACCESS_MMAP_INTERLEAVED,
};
static const unsigned int formats[] = {
SND_PCM_FORMAT_S16_BE,
@@ -1009,14 +974,8 @@ static int aaf_hw_params(snd_pcm_ioplug_t *io,
if (res < 0)
goto err_free_pdu;
- res = aaf_init_audiobuf_areas(aaf);
- if (res < 0)
- goto err_free_payload_areas;
-
return 0;
-err_free_payload_areas:
- free(aaf->payload_areas);
err_free_pdu:
free(aaf->pdu);
err_close_timer:
@@ -1034,8 +993,6 @@ static int aaf_hw_free(snd_pcm_ioplug_t *io)
close(aaf->timer_fd);
free(aaf->pdu);
free(aaf->payload_areas);
- free(aaf->audiobuf_areas->addr);
- free(aaf->audiobuf_areas);
return 0;
}
@@ -1134,6 +1091,7 @@ static int aaf_prepare(snd_pcm_ioplug_t *io)
int res;
snd_pcm_aaf_t *aaf = io->private_data;
+ aaf->audiobuf_areas = snd_pcm_ioplug_mmap_areas(io);
aaf->pdu_seq = 0;
aaf->hw_ptr = 0;
aaf->hw_virt_ptr = 0;
@@ -1179,34 +1137,6 @@ static int aaf_stop(snd_pcm_ioplug_t *io)
return 0;
}
-static snd_pcm_sframes_t aaf_transfer(snd_pcm_ioplug_t *io,
- const snd_pcm_channel_area_t *areas,
- snd_pcm_uframes_t offset,
- snd_pcm_uframes_t size)
-{
- int res;
- snd_pcm_aaf_t *aaf = io->private_data;
-
- if (io->stream == SND_PCM_STREAM_PLAYBACK) {
- res = snd_pcm_areas_copy_wrap(aaf->audiobuf_areas,
- (io->appl_ptr % io->buffer_size),
- io->buffer_size, areas, offset,
- size, io->channels, size,
- io->format);
- } else {
- res = snd_pcm_areas_copy_wrap(areas, offset, (offset + size),
- aaf->audiobuf_areas,
- (io->appl_ptr % io->buffer_size),
- io->buffer_size, io->channels,
- size, io->format);
- }
-
- if (res < 0)
- return res;
-
- return size;
-}
-
static const snd_pcm_ioplug_callback_t aaf_callback = {
.close = aaf_close,
.dump = aaf_dump,
@@ -1220,7 +1150,6 @@ static const snd_pcm_ioplug_callback_t aaf_callback = {
.prepare = aaf_prepare,
.start = aaf_start,
.stop = aaf_stop,
- .transfer = aaf_transfer,
};
SND_PCM_PLUGIN_DEFINE_FUNC(aaf)
@@ -1246,6 +1175,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(aaf)
aaf->io.callback = &aaf_callback;
aaf->io.private_data = aaf;
aaf->io.flags = SND_PCM_IOPLUG_FLAG_BOUNDARY_WA;
+ aaf->io.mmap_rw = 1;
res = snd_pcm_ioplug_create(&aaf->io, name, stream, mode);
if (res < 0) {
SNDERR("Failed to create ioplug instance");
--
2.14.4
^ permalink raw reply related [flat|nested] 12+ messages in thread* Re: [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin
2018-10-25 1:11 [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Andre Guedes
` (6 preceding siblings ...)
2018-10-25 1:11 ` [PATCH v2 - AAF PCM plugin 7/7] aaf: Add support for direct read/write transfers Andre Guedes
@ 2018-10-25 6:41 ` Jaroslav Kysela
2018-10-25 6:51 ` github repo (Re: [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin) Takashi Iwai
7 siblings, 1 reply; 12+ messages in thread
From: Jaroslav Kysela @ 2018-10-25 6:41 UTC (permalink / raw)
To: Andre Guedes, alsa-devel; +Cc: tiwai, pierre-louis.bossart, liam.r.girdwood
Dne 25.10.2018 v 03:11 Andre Guedes napsal(a):
> Hi all,
>
> This new version addresses the feedback from v1 and fixes a bug on capture
> mode. Code-wise, the changes are very trivial, no major diffs. Below follows
> the highlights:
>
> * Standard headers like linux/if_ether.h are now checked by configure script.
> * The 'static' modifier was added to array variables in aaf_hw_constraint().
> * aaf_close() callback was simplified.
> * aaf_rx_pdu() was modified so it doesn't return an error in case the number
> of bytes read by recv() is different from aaf->pdu_size. This is an
> expected situation and it means the received AVTPDU doesn't belong to the
> AAF stream the plugin is interested in so it should simply drop that
> AVTPDU. This fixes a bug when running multiple streams (with different
> AVPTU sizes) concurrently.
>
> Additionally, this new version improves the instructions in doc/aaf.txt in
> order to make it easier to test the plugin. Anyone should be able to test it
> just by copying and pasting the instructions.
Thanks for your code. I did a quick review and a compilation test and it
seems fine to include this improved code to alsa-plugins. I committed
your changes.
Our repos are on github now, too : https://github.com/alsa-project
Jaroslav
--
Jaroslav Kysela <perex@perex.cz>
Linux Sound Maintainer; ALSA Project; Red Hat, Inc.
^ permalink raw reply [flat|nested] 12+ messages in thread* github repo (Re: [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin)
2018-10-25 6:41 ` [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin Jaroslav Kysela
@ 2018-10-25 6:51 ` Takashi Iwai
2018-10-25 7:36 ` Jaroslav Kysela
0 siblings, 1 reply; 12+ messages in thread
From: Takashi Iwai @ 2018-10-25 6:51 UTC (permalink / raw)
To: Jaroslav Kysela
Cc: Andre Guedes, alsa-devel, liam.r.girdwood, pierre-louis.bossart
On Thu, 25 Oct 2018 08:41:15 +0200,
Jaroslav Kysela wrote:
>
> Our repos are on github now, too : https://github.com/alsa-project
I've seen your recent actions, and wondered whether this is a
permanent move or a testing. Will you keep hosting the same git repos
on git.alsa-project.org?
thanks,
Takashi
^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: github repo (Re: [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin)
2018-10-25 6:51 ` github repo (Re: [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin) Takashi Iwai
@ 2018-10-25 7:36 ` Jaroslav Kysela
2018-10-25 7:42 ` Takashi Iwai
0 siblings, 1 reply; 12+ messages in thread
From: Jaroslav Kysela @ 2018-10-25 7:36 UTC (permalink / raw)
To: Takashi Iwai
Cc: Andre Guedes, alsa-devel, liam.r.girdwood, pierre-louis.bossart
Dne 25.10.2018 v 08:51 Takashi Iwai napsal(a):
> On Thu, 25 Oct 2018 08:41:15 +0200,
> Jaroslav Kysela wrote:
>>
>> Our repos are on github now, too : https://github.com/alsa-project
>
> I've seen your recent actions, and wondered whether this is a
> permanent move or a testing. Will you keep hosting the same git repos
> on git.alsa-project.org?
I will do a proper github announce later today. And, it's one of the
question for the discussion. For me, it would be probably better to do a
mirror (and allow the push requests only to one primary repository -
github?).
Jaroslav
--
Jaroslav Kysela <perex@perex.cz>
Linux Sound Maintainer; ALSA Project; Red Hat, Inc.
^ permalink raw reply [flat|nested] 12+ messages in thread
* Re: github repo (Re: [PATCH v2 - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin)
2018-10-25 7:36 ` Jaroslav Kysela
@ 2018-10-25 7:42 ` Takashi Iwai
0 siblings, 0 replies; 12+ messages in thread
From: Takashi Iwai @ 2018-10-25 7:42 UTC (permalink / raw)
To: Jaroslav Kysela
Cc: Andre Guedes, alsa-devel, liam.r.girdwood, pierre-louis.bossart
On Thu, 25 Oct 2018 09:36:54 +0200,
Jaroslav Kysela wrote:
>
> Dne 25.10.2018 v 08:51 Takashi Iwai napsal(a):
> > On Thu, 25 Oct 2018 08:41:15 +0200,
> > Jaroslav Kysela wrote:
> >>
> >> Our repos are on github now, too : https://github.com/alsa-project
> >
> > I've seen your recent actions, and wondered whether this is a
> > permanent move or a testing. Will you keep hosting the same git repos
> > on git.alsa-project.org?
>
> I will do a proper github announce later today. And, it's one of the
> question for the discussion. For me, it would be probably better to do a
> mirror (and allow the push requests only to one primary repository -
> github?).
I don't mind either way, but limiting the push to only the primary
repo makes sense.
IMO, a bigger question is whether we allow github PR (supposing moving
to github as the primary repo). Usually the changes are submitted to
alsa-devel ML for review, so far. But with github PR, it may be
missing, and may be bad for the workflow of people including me.
I'm not sure whether hooking up the PR notification to ML would work.
thanks,
Takashi
^ permalink raw reply [flat|nested] 12+ messages in thread