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 | 91 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 ++++++ doc/aaf.txt | 18 ++++++++++++ 5 files changed, 130 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..7890e10 --- /dev/null +++ b/aaf/pcm_aaf.c @@ -0,0 +1,91 @@ +/* + * 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) +{ + snd_pcm_aaf_t *aaf = io->private_data; + + if (!aaf) + return -EBADFD; + + free(aaf); + aaf = NULL; + 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 3e2d233..1524cba 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 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.

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 ---

A quick word about CLOCK_TAI: even though it is supported since kernel v3.10, it is not mentioned in clock_gettime() manpage. We'll submit a patch to fix this.

aaf/pcm_aaf.c | 623 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ doc/aaf.txt | 72 +++++++ 2 files changed, 695 insertions(+)

diff --git a/aaf/pcm_aaf.c b/aaf/pcm_aaf.c index ac0b971..5d98cd3 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.h> #include <linux/if_ether.h> +#include <linux/if_packet.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_SEC 1000000000ULL #define NSEC_PER_USEC 1000ULL +#define TAI_OFFSET (37 * 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 +57,73 @@ 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; + + char *audiobuf; + + struct avtp_stream_pdu *pdu; + int pdu_size; + uint8_t pdu_seq; + + uint64_t mclk_start_time; + uint64_t mclk_period; + uint64_t mclk_ticks; + + 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; @@ -147,6 +232,368 @@ 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_audio_buffer(snd_pcm_aaf_t *aaf) +{ + char *audiobuf; + ssize_t frame_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; + + audiobuf = calloc(io->buffer_size, frame_size); + if (!audiobuf) + return -ENOMEM; + + aaf->audiobuf = audiobuf; + return 0; +} + +static int aaf_init_areas(snd_pcm_aaf_t *aaf) +{ + snd_pcm_channel_area_t *audiobuf_areas, *payload_areas; + 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; + + audiobuf_areas = calloc(io->channels, sizeof(snd_pcm_channel_area_t)); + if (!audiobuf_areas) + return -ENOMEM; + + payload_areas = calloc(io->channels, sizeof(snd_pcm_channel_area_t)); + if (!payload_areas) { + free(audiobuf_areas); + return -ENOMEM; + } + + for (unsigned int i = 0; i < io->channels; i++) { + audiobuf_areas[i].addr = aaf->audiobuf; + audiobuf_areas[i].first = i * sample_size * 8; + audiobuf_areas[i].step = frame_size * 8; + + payload_areas[i].addr = aaf->pdu->avtp_payload; + payload_areas[i].first = i * sample_size * 8; + payload_areas[i].step = frame_size * 8; + } + + aaf->audiobuf_areas = audiobuf_areas; + aaf->payload_areas = payload_areas; + return 0; +} + +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->mclk_period = (NSEC_PER_SEC * aaf->frames_per_pdu) / io->rate; + aaf->mclk_ticks = 0; + aaf->mclk_start_time = now.tv_sec * NSEC_PER_SEC + now.tv_nsec + + aaf->mclk_period; + + time_utc = TAI_TO_UTC(aaf->mclk_start_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; + itspec.it_interval.tv_nsec = aaf->mclk_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->mclk_start_time = 0; + aaf->mclk_period = 0; + aaf->mclk_ticks = 0; + return 0; +} + +static uint64_t aaf_mclk_gettime(snd_pcm_aaf_t *aaf) +{ + return aaf->mclk_start_time + aaf->mclk_period * aaf->mclk_ticks; +} + +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 == 0) { + /* If there is no frames available for transmission, we reached + * an underrun state. + */ + return -EPIPE; + } + if (hw_avail < aaf->frames_per_pdu) { + /* If there isn't enough frames to fill the AVTPDU, we drop + * them. This behavior is suggested by IEEE 1722-2016 spec, + * section 7.3.5. + */ + aaf_inc_hw_ptr(aaf, hw_avail); + return 0; + } + + 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--) { + res = aaf_tx_pdu(aaf); + if (res < 0) + return res; + aaf->mclk_ticks++; + } + + return 0; +} + +static int aaf_set_hw_constraint(snd_pcm_aaf_t *aaf) +{ + int res; + snd_pcm_ioplug_t *io = &aaf->io; + const unsigned int accesses[] = { + SND_PCM_ACCESS_RW_INTERLEAVED, + }; + const unsigned int formats[] = { + SND_PCM_FORMAT_S16_BE, + SND_PCM_FORMAT_S24_3BE, + SND_PCM_FORMAT_S32_BE, + SND_PCM_FORMAT_FLOAT_BE, + }; + 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) { snd_pcm_aaf_t *aaf = io->private_data; @@ -159,26 +606,185 @@ static int aaf_close(snd_pcm_ioplug_t *io) 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_pdu(aaf); + if (res < 0) + return res; + + res = aaf_init_audio_buffer(aaf); + if (res < 0) + goto err_free_pdu; + + res = aaf_init_areas(aaf); + if (res < 0) + goto err_free_audiobuf; + + res = aaf_init_socket(aaf); + if (res < 0) + goto err_free_areas; + + res = aaf_init_timer(aaf); + if (res < 0) + goto err_close_sk; + + return 0; + +err_close_sk: + close(aaf->sk_fd); +err_free_areas: + free(aaf->audiobuf_areas); + free(aaf->payload_areas); +err_free_audiobuf: + free(aaf->audiobuf); +err_free_pdu: + free(aaf->pdu); + return res; +} + +static int aaf_hw_free(snd_pcm_ioplug_t *io) +{ + snd_pcm_aaf_t *aaf = io->private_data; + + close(aaf->timer_fd); + close(aaf->sk_fd); + free(aaf->audiobuf_areas); + free(aaf->payload_areas); + free(aaf->audiobuf); + free(aaf->pdu); + 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) @@ -186,12 +792,21 @@ 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; @@ -200,12 +815,20 @@ 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/doc/aaf.txt b/doc/aaf.txt index d817249..ac9dd9d 100644 --- a/doc/aaf.txt +++ b/doc/aaf.txt @@ -9,6 +9,78 @@ 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 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 -------------------

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.

Signed-off-by: Andre Guedes andre.guedes@intel.com --- aaf/pcm_aaf.c | 387 ++++++++++++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 366 insertions(+), 21 deletions(-)

diff --git a/aaf/pcm_aaf.c b/aaf/pcm_aaf.c index 4d48612..dfe9013 100644 --- a/aaf/pcm_aaf.c +++ b/aaf/pcm_aaf.c @@ -28,6 +28,7 @@ #include <linux/if_ether.h> #include <linux/if_packet.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; @@ -68,6 +70,7 @@ typedef struct { struct avtp_stream_pdu *pdu; int pdu_size; uint8_t pdu_seq; + uint64_t pdu_count;

uint64_t mclk_start_time; uint64_t mclk_period; @@ -77,6 +80,7 @@ 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; } snd_pcm_aaf_t;

@@ -124,6 +128,106 @@ 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; @@ -267,8 +371,27 @@ 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; @@ -417,12 +540,13 @@ static int aaf_init_areas(snd_pcm_aaf_t *aaf) return 0; }

-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) @@ -456,6 +580,53 @@ static int aaf_mclk_start_playback(snd_pcm_aaf_t *aaf) return 0; }

+static int aaf_mclk_start_capture(snd_pcm_aaf_t *aaf, uint32_t avtp_time) +{ + int res; + struct timespec tspec; + struct itimerspec itspec; + uint64_t ptime, now, time_utc; + 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 = (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); + + aaf->mclk_period = (NSEC_PER_SEC * aaf->frames_per_pdu) / io->rate; + aaf->mclk_ticks = 0; + aaf->mclk_start_time = ptime; + + time_utc = TAI_TO_UTC(ptime); + 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->mclk_period; + res = timerfd_settime(aaf->timer_fd, TFD_TIMER_ABSTIME, &itspec, NULL); + if (res < 0) { + SNDERR("Failed to set timer"); + return -errno; + } + + return 0; +} + static int aaf_mclk_reset(snd_pcm_aaf_t *aaf) { int res; @@ -499,7 +670,7 @@ static int aaf_tx_pdu(snd_pcm_aaf_t *aaf) * them. This behavior is suggested by IEEE 1722-2016 spec, * section 7.3.5. */ - aaf_inc_hw_ptr(aaf, hw_avail); + aaf_inc_ptr(&aaf->hw_ptr, hw_avail, aaf->boundary); return 0; }

@@ -531,7 +702,117 @@ 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 || n != aaf->pdu_size) { + SNDERR("Failed to receive data"); + return -EIO; + } + + 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))) + 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->mclk_start_time == 0) { + res = aaf_mclk_start_capture(aaf, avtp_time); + if (res < 0) + return res; + } else { + uint64_t ptime = aaf->mclk_start_time + aaf->mclk_period * + aaf->pdu_count; + + 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; + } + } + + 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->pdu_count++; + 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; }

@@ -560,6 +841,41 @@ 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_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--) { + snd_pcm_sframes_t 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); + aaf->mclk_ticks++; + } + + return 0; +} + static int aaf_set_hw_constraint(snd_pcm_aaf_t *aaf) { int res; @@ -693,7 +1009,7 @@ 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, @@ -708,8 +1024,13 @@ 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; @@ -733,8 +1054,22 @@ 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; @@ -746,7 +1081,9 @@ static int aaf_prepare(snd_pcm_ioplug_t *io) snd_pcm_aaf_t *aaf = io->private_data;

aaf->pdu_seq = 0; + aaf->pdu_count = 0; aaf->hw_ptr = 0; + aaf->hw_virt_ptr = 0; res = aaf_mclk_reset(aaf); if (res < 0) return res; @@ -761,13 +1098,17 @@ 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; }

@@ -797,13 +1138,17 @@ 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; }