November 2022 - gerrit-log - lists.osmocom.org

Change in osmocom-bb[master]: mobile: split gsm_send_voice() -> gsm_send_voice_{msg,frame}()

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30336 ) Change subject: mobile: split gsm_send_voice() -> gsm_send_voice_{msg,frame}() ...................................................................... mobile: split gsm_send_voice() -> gsm_send_voice_{msg,frame}() This is needed to avoid sending voice frames to the L1PHY without having to use MNCC specific struct gsm_data_frame. Change-Id: I37241555cd648a8e2b57fa072c708f93cd1ba5a9 Related: OS#5599 --- M src/host/layer23/include/osmocom/bb/mobile/voice.h M src/host/layer23/src/mobile/gsm48_cc.c M src/host/layer23/src/mobile/voice.c 3 files changed, 17 insertions(+), 7 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/36/30336/1 diff --git a/src/host/layer23/include/osmocom/bb/mobile/voice.h b/src/host/layer23/include/osmocom/bb/mobile/voice.h index 623f6c8..8c4c5b6 100644 --- a/src/host/layer23/include/osmocom/bb/mobile/voice.h +++ b/src/host/layer23/include/osmocom/bb/mobile/voice.h @@ -2,6 +2,8 @@ struct osmocom_ms; struct gsm_data_frame; +struct msgb; int gsm_voice_init(struct osmocom_ms *ms); -int gsm_send_voice(struct osmocom_ms *ms, struct gsm_data_frame *data); +int gsm_send_voice_msg(struct osmocom_ms *ms, struct msgb *msg); +int gsm_send_voice_frame(struct osmocom_ms *ms, const struct gsm_data_frame *frame); diff --git a/src/host/layer23/src/mobile/gsm48_cc.c b/src/host/layer23/src/mobile/gsm48_cc.c index a8d7a65..503c9fd 100644 --- a/src/host/layer23/src/mobile/gsm48_cc.c +++ b/src/host/layer23/src/mobile/gsm48_cc.c @@ -1966,7 +1966,7 @@ switch (msg_type) { case GSM_TCHF_FRAME: - return gsm_send_voice(ms, arg); + return gsm_send_voice_frame(ms, arg); case MNCC_LCHAN_MODIFY: return 0; case MNCC_FRAME_RECV: diff --git a/src/host/layer23/src/mobile/voice.c b/src/host/layer23/src/mobile/voice.c index 7bcf8a6..5f569c7 100644 --- a/src/host/layer23/src/mobile/voice.c +++ b/src/host/layer23/src/mobile/voice.c @@ -56,7 +56,7 @@ /* send voice frame back, if appropriate */ if (ms->settings.audio.io_handler == AUDIO_IOH_LOOPBACK) - gsm_send_voice(ms, mncc); + gsm_send_voice_frame(ms, mncc); /* distribute and then free */ if (ms->mncc_entity.mncc_recv && ms->mncc_entity.ref) { @@ -71,7 +71,13 @@ /* * send voice */ -int gsm_send_voice(struct osmocom_ms *ms, struct gsm_data_frame *data) +int gsm_send_voice_msg(struct osmocom_ms *ms, struct msgb *msg) +{ + /* Forward to RR */ + return gsm48_rr_tx_voice(ms, msg); +} + +int gsm_send_voice_frame(struct osmocom_ms *ms, const struct gsm_data_frame *frame) { struct msgb *nmsg; int len; @@ -85,7 +91,9 @@ len = GSM_EFR_BYTES; break; default: - LOGP(DL1C, LOGL_ERROR, "gsm_send_voice, msg_type=0x%02x: not implemented\n", data->msg_type); + LOGP(DL1C, LOGL_ERROR, + "gsm_send_voice, msg_type=0x%02x: not implemented\n", + frame->msg_type); return -EINVAL; } @@ -93,9 +101,9 @@ if (!nmsg) return -ENOMEM; nmsg->l2h = msgb_put(nmsg, len); - memcpy(nmsg->l2h, data->data, len); + memcpy(nmsg->l2h, frame->data, len); - return gsm48_rr_tx_voice(ms, nmsg); + return gsm_send_voice_msg(ms, nmsg); } /* -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30336 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: I37241555cd648a8e2b57fa072c708f93cd1ba5a9 Gerrit-Change-Number: 30336 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-MessageType: newchange

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Change in osmocom-bb[master]: mobile: integrate GAPK based audio (voice) I/O support

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30337 ) Change subject: mobile: integrate GAPK based audio (voice) I/O support ...................................................................... mobile: integrate GAPK based audio (voice) I/O support This change introduces a new feature to the mobile application - audio I/O support, which allows the user to speak right from the host side running mobile through its ordinary mic and speakers. The audio I/O is based on libosmo-gapk, which offers ALSA sound sysyem support. The libosmo-gapk's API implies to use the processing chains and queue, which basically consist of a source block, several processing blocks, and a sink block. So, there are two voice processing chains: - 'pq_audio_source' (voice capture -> frame encoding), - 'pq_audio_sink' (frame decoding -> voice playback). Both of them exchange frames from two dedicated buffers: - 'tch_fb_ul' - a buffer for to be played DL TCH frames, - 'tch_fb_dl' - a buffer for encoded UL TCH frames. In its turn, both buffers are served by a new gapk_io_dequeue() function, which is being called within the mobile_work() loop. Change-Id: Ib86b0746606c191573cc773f01172afbb52f33a9 Related: OS#5599 --- M doc/examples/mobile/default.cfg M doc/examples/mobile/multi_ms.cfg M src/host/layer23/configure.ac M src/host/layer23/include/osmocom/bb/common/logging.h M src/host/layer23/include/osmocom/bb/common/osmocom_data.h A src/host/layer23/include/osmocom/bb/mobile/gapk_io.h M src/host/layer23/include/osmocom/bb/mobile/settings.h M src/host/layer23/src/common/logging.c M src/host/layer23/src/mobile/Makefile.am M src/host/layer23/src/mobile/app_mobile.c A src/host/layer23/src/mobile/gapk_io.c M src/host/layer23/src/mobile/gsm48_rr.c M src/host/layer23/src/mobile/main.c M src/host/layer23/src/mobile/settings.c M src/host/layer23/src/mobile/voice.c M src/host/layer23/src/mobile/vty_interface.c 16 files changed, 715 insertions(+), 40 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/37/30337/1 diff --git a/doc/examples/mobile/default.cfg b/doc/examples/mobile/default.cfg index e24e07f..841a356 100644 --- a/doc/examples/mobile/default.cfg +++ b/doc/examples/mobile/default.cfg @@ -63,4 +63,6 @@ rplmn 001 01 audio io-handler none + alsa-output-dev default + alsa-input-dev default no shutdown diff --git a/doc/examples/mobile/multi_ms.cfg b/doc/examples/mobile/multi_ms.cfg index 86a9840..c7d4097 100644 --- a/doc/examples/mobile/multi_ms.cfg +++ b/doc/examples/mobile/multi_ms.cfg @@ -63,6 +63,8 @@ rplmn 001 01 audio io-handler none + alsa-output-dev default + alsa-input-dev default no shutdown ! ms two @@ -117,4 +119,6 @@ rplmn 001 01 audio io-handler none + alsa-output-dev default + alsa-input-dev default no shutdown diff --git a/src/host/layer23/configure.ac b/src/host/layer23/configure.ac index f74efa2..b75fefe 100644 --- a/src/host/layer23/configure.ac +++ b/src/host/layer23/configure.ac @@ -44,6 +44,7 @@ PKG_CHECK_MODULES(LIBOSMOVTY, libosmovty >= 0.10.0) PKG_CHECK_MODULES(LIBOSMOGSM, libosmogsm) PKG_CHECK_MODULES(LIBOSMOCODEC, libosmocodec) +PKG_CHECK_MODULES(LIBOSMOGAPK, libosmogapk) AC_CHECK_LIB(gps, gps_waiting, LIBGPS_CFLAGS=" -D_HAVE_GPSD" LIBGPS_LIBS=" -lgps ",,) AC_SUBST([LIBGPS_CFLAGS]) AC_SUBST([LIBGPS_LIBS]) diff --git a/src/host/layer23/include/osmocom/bb/common/logging.h b/src/host/layer23/include/osmocom/bb/common/logging.h index bf6e6aa..e968528 100644 --- a/src/host/layer23/include/osmocom/bb/common/logging.h +++ b/src/host/layer23/include/osmocom/bb/common/logging.h @@ -25,6 +25,7 @@ DMOB, DPRIM, DLUA, + DGAPK, }; extern const struct log_info log_info; diff --git a/src/host/layer23/include/osmocom/bb/common/osmocom_data.h b/src/host/layer23/include/osmocom/bb/common/osmocom_data.h index 14e594c..1b26533 100644 --- a/src/host/layer23/include/osmocom/bb/common/osmocom_data.h +++ b/src/host/layer23/include/osmocom/bb/common/osmocom_data.h @@ -8,6 +8,7 @@ struct osmocom_ms; /* FIXME no 'mobile' specific stuff should be here */ +#include <osmocom/bb/mobile/gapk_io.h> #include <osmocom/bb/mobile/support.h> #include <osmocom/bb/mobile/settings.h> #include <osmocom/bb/mobile/subscriber.h> @@ -94,6 +95,9 @@ struct osmomncc_entity mncc_entity; struct llist_head trans_list; + /* Audio I/O */ + struct gapk_io_state *gapk_io; + void *lua_state; int lua_cb_ref; char *lua_script; diff --git a/src/host/layer23/include/osmocom/bb/mobile/gapk_io.h b/src/host/layer23/include/osmocom/bb/mobile/gapk_io.h new file mode 100644 index 0000000..1b3ffa7 --- /dev/null +++ b/src/host/layer23/include/osmocom/bb/mobile/gapk_io.h @@ -0,0 +1,32 @@ +#pragma once + +#include <osmocom/gapk/procqueue.h> +#include <osmocom/gapk/codecs.h> + +/* Forward declarations */ +struct osmocom_ms; + +struct gapk_io_state { + /* src/alsa -> proc/codec -> sink/tch_fb */ + struct osmo_gapk_pq *pq_source; + /* src/tch_fb -> proc/codec -> sink/alsa */ + struct osmo_gapk_pq *pq_sink; + + /* Description of currently used codec / format */ + const struct osmo_gapk_format_desc *phy_fmt_desc; + const struct osmo_gapk_codec_desc *codec_desc; + + /* Buffer for to be played TCH frames (from DL) */ + struct llist_head tch_fb_dl; + /* Buffer for encoded TCH frames (for UL) */ + struct llist_head tch_fb_ul; +}; + +void gapk_io_init(void); +int gapk_io_dequeue(struct osmocom_ms *ms); + +int gapk_io_init_ms_chan(struct osmocom_ms *ms, + uint8_t ch_type, uint8_t ch_mode); +int gapk_io_init_ms(struct osmocom_ms *ms, + enum osmo_gapk_codec_type codec); +int gapk_io_clean_up_ms(struct osmocom_ms *ms); diff --git a/src/host/layer23/include/osmocom/bb/mobile/settings.h b/src/host/layer23/include/osmocom/bb/mobile/settings.h index 8edde53..e2e30d8 100644 --- a/src/host/layer23/include/osmocom/bb/mobile/settings.h +++ b/src/host/layer23/include/osmocom/bb/mobile/settings.h @@ -17,6 +17,8 @@ enum audio_io_handler { /* No handler, drop frames */ AUDIO_IOH_NONE = 0, + /* libosmo-gapk based handler */ + AUDIO_IOH_GAPK, /* L1 PHY (e.g. Calypso DSP) */ AUDIO_IOH_L1PHY, /* Return to sender */ @@ -29,6 +31,8 @@ struct audio_settings { enum audio_io_handler io_handler; + char alsa_output_dev[128]; + char alsa_input_dev[128]; }; struct gsm_settings { diff --git a/src/host/layer23/src/common/logging.c b/src/host/layer23/src/common/logging.c index 96decf9..56f0322 100644 --- a/src/host/layer23/src/common/logging.c +++ b/src/host/layer23/src/common/logging.c @@ -141,6 +141,12 @@ .color = "\033[1;32m", .enabled = 1, .loglevel = LOGL_DEBUG, }, + [DGAPK] = { + .name = "DGAPK", + .description = "GAPK audio", + .color = "\033[0;36m", + .enabled = 1, .loglevel = LOGL_DEBUG, + }, }; const struct log_info log_info = { diff --git a/src/host/layer23/src/mobile/Makefile.am b/src/host/layer23/src/mobile/Makefile.am index 3db27b7..95039ad 100644 --- a/src/host/layer23/src/mobile/Makefile.am +++ b/src/host/layer23/src/mobile/Makefile.am @@ -9,6 +9,7 @@ $(LIBOSMOVTY_CFLAGS) \ $(LIBOSMOGSM_CFLAGS) \ $(LIBOSMOCODEC_CFLAGS) \ + $(LIBOSMOGAPK_CFLAGS) \ $(LIBGPS_CFLAGS) \ $(LIBLUA_CFLAGS) \ $(NULL) @@ -30,6 +31,7 @@ support.c \ transaction.c \ vty_interface.c \ + gapk_io.c \ voice.c \ $(NULL) @@ -43,6 +45,7 @@ $(LIBOSMOVTY_LIBS) \ $(LIBOSMOGSM_LIBS) \ $(LIBOSMOCODEC_LIBS) \ + $(LIBOSMOGAPK_LIBS) \ $(LIBGPS_LIBS) \ $(LIBLUA_LIBS) \ $(NULL) diff --git a/src/host/layer23/src/mobile/app_mobile.c b/src/host/layer23/src/mobile/app_mobile.c index dd67e70..af3fa61 100644 --- a/src/host/layer23/src/mobile/app_mobile.c +++ b/src/host/layer23/src/mobile/app_mobile.c @@ -33,6 +33,7 @@ #include <osmocom/bb/mobile/app_mobile.h> #include <osmocom/bb/mobile/mncc.h> #include <osmocom/bb/mobile/voice.h> +#include <osmocom/bb/mobile/gapk_io.h> #include <osmocom/bb/mobile/primitives.h> #include <osmocom/bb/common/sap_interface.h> @@ -72,6 +73,7 @@ w |= gsm322_cs_dequeue(ms); w |= gsm_sim_job_dequeue(ms); w |= mncc_dequeue(ms); + w |= gapk_io_dequeue(ms); if (w) work = 1; } while (w); @@ -156,6 +158,10 @@ return -EBUSY; } + /* Clean up GAPK state, if preset */ + if (ms->gapk_io != NULL) + gapk_io_clean_up_ms(ms); + gsm322_exit(ms); gsm48_mm_exit(ms); gsm48_rr_exit(ms); @@ -448,6 +454,9 @@ osmo_gps_init(); + /* Init GAPK audio I/O */ + gapk_io_init(); + vty_info.tall_ctx = l23_ctx; vty_init(&vty_info); logging_vty_add_cmds(); diff --git a/src/host/layer23/src/mobile/gapk_io.c b/src/host/layer23/src/mobile/gapk_io.c new file mode 100644 index 0000000..c92b799 --- /dev/null +++ b/src/host/layer23/src/mobile/gapk_io.c @@ -0,0 +1,565 @@ +/* + * GAPK (GSM Audio Pocket Knife) based audio I/O + * + * (C) 2017-2022 by Vadim Yanitskiy <axilirator(a)gmail.com> + * Contributions by sysmocom - s.f.m.c. GmbH <info(a)sysmocom.de> + * + * All Rights Reserved + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 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 General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + * + */ + +#include <string.h> +#include <errno.h> + +#include <osmocom/core/msgb.h> +#include <osmocom/core/utils.h> + +#include <osmocom/gsm/protocol/gsm_04_08.h> +#include <osmocom/gsm/protocol/gsm_08_58.h> + +#include <osmocom/gapk/procqueue.h> +#include <osmocom/gapk/formats.h> +#include <osmocom/gapk/codecs.h> +#include <osmocom/gapk/common.h> + +#include <osmocom/bb/common/osmocom_data.h> +#include <osmocom/bb/common/logging.h> + +#include <osmocom/bb/mobile/voice.h> + +/* The RAW PCM format is common for both audio source and sink */ +static const struct osmo_gapk_format_desc *rawpcm_fmt; + +static int pq_queue_tch_fb_recv(void *_state, uint8_t *out, + const uint8_t *in, unsigned int in_len) +{ + struct gapk_io_state *state = (struct gapk_io_state *)_state; + struct msgb *tch_msg; + size_t frame_len; + + /* Obtain one TCH frame from the DL buffer */ + tch_msg = msgb_dequeue(&state->tch_fb_dl); + + /* Make sure we've got a frame */ + if (tch_msg == NULL) + return -EIO; + + /* Calculate received frame length */ + frame_len = msgb_l3len(tch_msg); + + /* Copy the frame bytes from message */ + memcpy(out, tch_msg->l3h, frame_len); + + /* Release memory */ + msgb_free(tch_msg); + + return frame_len; +} + +static int pq_queue_tch_fb_send(void *_state, uint8_t *out, + const uint8_t *in, unsigned int in_len) +{ + struct gapk_io_state *state = (struct gapk_io_state *)_state; + struct msgb *tch_msg; + + /* Allocate a new message for the lower layers */ + tch_msg = msgb_alloc_headroom(in_len + 64, 64, "TCH frame"); + if (tch_msg == NULL) + return -ENOMEM; + + /* Copy the frame bytes to a new message */ + tch_msg->l2h = msgb_put(tch_msg, in_len); + memcpy(tch_msg->l2h, in, in_len); + + /* Put encoded TCH frame to the UL buffer */ + msgb_enqueue(&state->tch_fb_ul, tch_msg); + + return 0; +} + +/** + * A custom TCH frame buffer block, which actually + * handles incoming frames from DL buffer and puts + * outgoing frames to UL buffer... + */ +static int pq_queue_tch_fb(struct osmo_gapk_pq *pq, + struct gapk_io_state *io_state, + bool is_src) +{ + struct osmo_gapk_pq_item *item; + unsigned int frame_len; + + LOGP(DGAPK, LOGL_DEBUG, "PQ '%s': Adding TCH frame buffer %s\n", + pq->name, is_src ? "input" : "output"); + + /* Allocate and add a new queue item */ + item = osmo_gapk_pq_add_item(pq); + if (item == NULL) + return -ENOMEM; + + /* General item type and description */ + item->type = is_src ? OSMO_GAPK_ITEM_TYPE_SOURCE : OSMO_GAPK_ITEM_TYPE_SINK; + item->cat_name = is_src ? "source" : "sink"; + item->sub_name = "tch_fb"; + + /* I/O length */ + frame_len = io_state->phy_fmt_desc->frame_len; + item->len_in = is_src ? 0 : frame_len; + item->len_out = is_src ? frame_len : 0; + + /* Handler and it's state */ + item->proc = is_src ? &pq_queue_tch_fb_recv : &pq_queue_tch_fb_send; + item->state = io_state; + + return 0; +} + +/** + * Auxiliary wrapper around format conversion block. + * Is used to perform either a conversion from the format, + * produced by encoder, to canonical, or a conversion + * from canonical format to the format expected by decoder. + */ +static int pq_queue_codec_fmt_conv(struct osmo_gapk_pq *pq, + const struct osmo_gapk_codec_desc *codec, + bool is_src) +{ + const struct osmo_gapk_format_desc *codec_fmt_desc; + + /* Get format description */ + codec_fmt_desc = osmo_gapk_fmt_get_from_type(is_src ? + codec->codec_enc_format_type : codec->codec_dec_format_type); + if (codec_fmt_desc == NULL) + return -ENOTSUP; + + /* Put format conversion block */ + return osmo_gapk_pq_queue_fmt_convert(pq, codec_fmt_desc, !is_src); +} + +/** + * Prepares the following queue (source is mic): + * + * source/alsa -> proc/codec -> proc/format -> + * -> proc/format -> sink/tch_fb + * + * The two format conversion blocks are aimed to + * convert an encoder specific format + * to a PHY specific format. + */ +static int prepare_audio_source(struct gapk_io_state *gapk_io, + const char *alsa_input_dev) +{ + struct osmo_gapk_pq *pq; + char *pq_desc; + int rc; + + LOGP(DGAPK, LOGL_DEBUG, "Prepare audio input (capture) chain\n"); + + /* Allocate a processing queue */ + pq = osmo_gapk_pq_create("pq_audio_source"); + if (pq == NULL) + return -ENOMEM; + + /* ALSA audio source */ + rc = osmo_gapk_pq_queue_alsa_input(pq, alsa_input_dev, rawpcm_fmt->frame_len); + if (rc) + goto error; + + /* Frame encoder */ + rc = osmo_gapk_pq_queue_codec(pq, gapk_io->codec_desc, 1); + if (rc) + goto error; + + /* Encoder specific format -> canonical */ + rc = pq_queue_codec_fmt_conv(pq, gapk_io->codec_desc, true); + if (rc) + goto error; + + /* Canonical -> PHY specific format */ + rc = osmo_gapk_pq_queue_fmt_convert(pq, gapk_io->phy_fmt_desc, 1); + if (rc) + goto error; + + /* TCH frame buffer sink */ + rc = pq_queue_tch_fb(pq, gapk_io, false); + if (rc) + goto error; + + /* Check composed queue in strict mode */ + rc = osmo_gapk_pq_check(pq, 1); + if (rc) + goto error; + + /* Prepare queue (allocate buffers, etc.) */ + rc = osmo_gapk_pq_prepare(pq); + if (rc) + goto error; + + /* Save pointer within MS GAPK state */ + gapk_io->pq_source = pq; + + /* Describe prepared chain */ + pq_desc = osmo_gapk_pq_describe(pq); + LOGP(DGAPK, LOGL_DEBUG, "PQ '%s': chain '%s' prepared\n", pq->name, pq_desc); + talloc_free(pq_desc); + + return 0; + +error: + talloc_free(pq); + return rc; +} + +/** + * Prepares the following queue (sink is speaker): + * + * src/tch_fb -> proc/format -> [proc/ecu] -> + * proc/format -> proc/codec -> sink/alsa + * + * The two format conversion blocks (proc/format) + * are aimed to convert a PHY specific format + * to an encoder specific format. + * + * A ECU (Error Concealment Unit) block is optionally + * added if implemented for a given codec. + */ +static int prepare_audio_sink(struct gapk_io_state *gapk_io, + const char *alsa_output_dev) +{ + struct osmo_gapk_pq *pq; + char *pq_desc; + int rc; + + LOGP(DGAPK, LOGL_DEBUG, "Prepare audio output (playback) chain\n"); + + /* Allocate a processing queue */ + pq = osmo_gapk_pq_create("pq_audio_sink"); + if (pq == NULL) + return -ENOMEM; + + /* TCH frame buffer source */ + rc = pq_queue_tch_fb(pq, gapk_io, true); + if (rc) + goto error; + +#if 0 + /* TODO: PHY specific format -> canonical */ + rc = osmo_gapk_pq_queue_fmt_convert(pq, gapk_io->phy_fmt_desc, 0); + if (rc) + goto error; +#endif + + /* Optional ECU (Error Concealment Unit) */ + osmo_gapk_pq_queue_ecu(pq, gapk_io->codec_desc); + +#if 0 + /* TODO: canonical -> decoder specific format */ + rc = pq_queue_codec_fmt_conv(pq, gapk_io->codec_desc, false); + if (rc) + goto error; +#endif + + /* Frame decoder */ + rc = osmo_gapk_pq_queue_codec(pq, gapk_io->codec_desc, 0); + if (rc) + goto error; + + /* ALSA audio sink */ + rc = osmo_gapk_pq_queue_alsa_output(pq, alsa_output_dev, rawpcm_fmt->frame_len); + if (rc) + goto error; + + /* Check composed queue in strict mode */ + rc = osmo_gapk_pq_check(pq, 1); + if (rc) + goto error; + + /* Prepare queue (allocate buffers, etc.) */ + rc = osmo_gapk_pq_prepare(pq); + if (rc) + goto error; + + /* Save pointer within MS GAPK state */ + gapk_io->pq_sink = pq; + + /* Describe prepared chain */ + pq_desc = osmo_gapk_pq_describe(pq); + LOGP(DGAPK, LOGL_DEBUG, "PQ '%s': chain '%s' prepared\n", pq->name, pq_desc); + talloc_free(pq_desc); + + return 0; + +error: + talloc_free(pq); + return rc; +} + +/** + * Cleans up both TCH frame I/O buffers, destroys both + * processing queues (chains), and deallocates the memory. + * Should be called when a voice call is finished... + */ +int gapk_io_clean_up_ms(struct osmocom_ms *ms) +{ + struct msgb *msg; + + if (ms->gapk_io == NULL) + return 0; + + /* Flush TCH frame I/O buffers */ + while ((msg = msgb_dequeue(&ms->gapk_io->tch_fb_dl))) + msgb_free(msg); + while ((msg = msgb_dequeue(&ms->gapk_io->tch_fb_ul))) + msgb_free(msg); + + /* Destroy both audio I/O chains */ + if (ms->gapk_io->pq_source) + osmo_gapk_pq_destroy(ms->gapk_io->pq_source); + if (ms->gapk_io->pq_sink) + osmo_gapk_pq_destroy(ms->gapk_io->pq_sink); + + talloc_free(ms->gapk_io); + ms->gapk_io = NULL; + + return 0; +} + +/** + * Picks the corresponding PHY's frame format for a given codec. + * To be used with PHYs that produce audio frames in RTP format, + * such as trxcon (GSM 05.03 libosmocoding API). + */ +static enum osmo_gapk_format_type phy_fmt_pick_rtp(enum osmo_gapk_codec_type codec) +{ + switch (codec) { + case CODEC_HR: + return FMT_RTP_HR_IETF; + case CODEC_FR: + return FMT_GSM; + case CODEC_EFR: + return FMT_RTP_EFR; + case CODEC_AMR: + return FMT_RTP_AMR; + default: + return FMT_INVALID; + } +} + +/** + * Allocates both TCH frame I/O buffers + * and prepares both processing queues (chains). + * Should be called when a voice call is initiated... + */ +int gapk_io_init_ms(struct osmocom_ms *ms, enum osmo_gapk_codec_type codec) +{ + const struct osmo_gapk_format_desc *phy_fmt_desc; + const struct osmo_gapk_codec_desc *codec_desc; + struct gsm_settings *set = &ms->settings; + enum osmo_gapk_format_type phy_fmt; + struct gapk_io_state *gapk_io; + int rc = 0; + + LOGP(DGAPK, LOGL_NOTICE, "Initialize GAPK I/O\n"); + + OSMO_ASSERT(ms->gapk_io == NULL); + + /* Make sure that the chosen codec has description */ + codec_desc = osmo_gapk_codec_get_from_type(codec); + if (codec_desc == NULL) { + LOGP(DGAPK, LOGL_ERROR, "Invalid codec type 0x%02x\n", codec); + return -EINVAL; + } + + /* Make sure that the chosen codec is supported */ + if (codec_desc->codec_encode == NULL || codec_desc->codec_decode == NULL) { + LOGP(DGAPK, LOGL_ERROR, + "Codec '%s' is not supported by GAPK\n", codec_desc->name); + return -ENOTSUP; + } + + /** + * Pick the corresponding PHY's frame format + * TODO: ask PHY, which format is supported? + * FIXME: RTP (valid for trxcon) is used for now + */ + phy_fmt = phy_fmt_pick_rtp(codec); + phy_fmt_desc = osmo_gapk_fmt_get_from_type(phy_fmt); + if (phy_fmt_desc == NULL) { + LOGP(DGAPK, LOGL_ERROR, "Failed to pick the PHY specific " + "frame format for codec '%s'\n", codec_desc->name); + return -EINVAL; + } + + gapk_io = talloc_zero(ms, struct gapk_io_state); + if (gapk_io == NULL) { + LOGP(DGAPK, LOGL_ERROR, "Failed to allocate memory\n"); + return -ENOMEM; + } + + /* Init TCH frame I/O buffers */ + INIT_LLIST_HEAD(&gapk_io->tch_fb_dl); + INIT_LLIST_HEAD(&gapk_io->tch_fb_ul); + + /* Store the codec / format description */ + gapk_io->codec_desc = codec_desc; + gapk_io->phy_fmt_desc = phy_fmt_desc; + + /* Use gapk_io_state as talloc context for both chains */ + osmo_gapk_set_talloc_ctx(gapk_io); + + /* Prepare both source and sink chains */ + rc |= prepare_audio_source(gapk_io, set->audio.alsa_input_dev); + rc |= prepare_audio_sink(gapk_io, set->audio.alsa_output_dev); + + /* Fall back to ms instance */ + osmo_gapk_set_talloc_ctx(ms); + + /* If at lease one chain constructor failed */ + if (rc) { + /* Destroy both audio I/O chains */ + if (gapk_io->pq_source) + osmo_gapk_pq_destroy(gapk_io->pq_source); + if (gapk_io->pq_sink) + osmo_gapk_pq_destroy(gapk_io->pq_sink); + + /* Release the memory and return */ + talloc_free(gapk_io); + + LOGP(DGAPK, LOGL_ERROR, "Failed to initialize GAPK I/O\n"); + return rc; + } + + /* Init pointers */ + ms->gapk_io = gapk_io; + + LOGP(DGAPK, LOGL_NOTICE, + "GAPK I/O initialized for MS '%s', codec '%s'\n", + ms->name, codec_desc->name); + + return 0; +} + +/** + * Wrapper around gapk_io_init_ms(), that maps both + * given GSM 04.08 channel type (HR/FR) and channel + * mode to a codec from 'osmo_gapk_codec_type' enum, + * checks if a mapped codec is supported by GAPK, + * and finally calls the wrapped function. + */ +int gapk_io_init_ms_chan(struct osmocom_ms *ms, + uint8_t ch_type, uint8_t ch_mode) +{ + enum osmo_gapk_codec_type codec; + + /* Map GSM 04.08 channel mode to GAPK codec type */ + switch (ch_mode) { + case GSM48_CMODE_SPEECH_V1: /* FR or HR */ + if (ch_type == RSL_CHAN_Bm_ACCHs) + codec = CODEC_FR; + else + codec = CODEC_HR; + break; + + case GSM48_CMODE_SPEECH_EFR: + codec = CODEC_EFR; + break; + + case GSM48_CMODE_SPEECH_AMR: + codec = CODEC_AMR; + break; + + /* Signalling or CSD, do nothing */ + case GSM48_CMODE_DATA_14k5: + case GSM48_CMODE_DATA_12k0: + case GSM48_CMODE_DATA_6k0: + case GSM48_CMODE_DATA_3k6: + case GSM48_CMODE_SIGN: + return 0; + default: + LOGP(DGAPK, LOGL_ERROR, "Unhandled channel mode 0x%02x (%s)\n", + ch_mode, get_value_string(gsm48_chan_mode_names, ch_mode)); + return -EINVAL; + } + + return gapk_io_init_ms(ms, codec); +} + +/** + * Performs basic initialization of GAPK library, + * setting the talloc root context and a logging category. + * Should be called during the application initialization... + */ +void gapk_io_init(void) +{ + /* Init logging subsystem */ + osmo_gapk_log_init(DGAPK); + + /* Make RAWPCM format info easy to access */ + rawpcm_fmt = osmo_gapk_fmt_get_from_type(FMT_RAWPCM_S16LE); +} + +/* Serves both TCH frame I/O buffers */ +int gapk_io_dequeue(struct osmocom_ms *ms) +{ + struct gapk_io_state *gapk_io = ms->gapk_io; + struct llist_head *entry; + size_t frame_count = 0; + int work = 0; + + /* There is no active call, nothing to do */ + if (gapk_io == NULL) + return 0; + + /** + * Make sure we have at least two frames + * to prevent discontinuous playback. + */ + llist_for_each(entry, &gapk_io->tch_fb_dl) + if (++frame_count > 2) + break; + if (frame_count < 2) + return 0; + + /** + * TODO: if there is an active call, but no TCH frames + * in DL buffer, put silence frames using the upcoming + * ECU (Error Concealment Unit) of libosmocodec. + */ + while (!llist_empty(&gapk_io->tch_fb_dl)) { + /* Decode and play received DL TCH frame */ + osmo_gapk_pq_execute(gapk_io->pq_sink); + + /* Record and encode an UL TCH frame back */ + osmo_gapk_pq_execute(gapk_io->pq_source); + + work |= 1; + } + + while (!llist_empty(&gapk_io->tch_fb_ul)) { + struct msgb *tch_msg; + + /* Obtain one TCH frame from the UL buffer */ + tch_msg = msgb_dequeue(&gapk_io->tch_fb_ul); + + /* Push a voice frame to the lower layers */ + gsm_send_voice_msg(ms, tch_msg); + + work |= 1; + } + + return work; +} diff --git a/src/host/layer23/src/mobile/gsm48_rr.c b/src/host/layer23/src/mobile/gsm48_rr.c index 8eaf0f6..01fac62 100644 --- a/src/host/layer23/src/mobile/gsm48_rr.c +++ b/src/host/layer23/src/mobile/gsm48_rr.c @@ -75,6 +75,8 @@ #include <osmocom/bb/common/logging.h> #include <osmocom/bb/common/networks.h> #include <osmocom/bb/common/l1ctl.h> + +#include <osmocom/bb/mobile/gapk_io.h> #include <osmocom/bb/mobile/vty.h> #include <osmocom/bb/common/utils.h> @@ -3443,6 +3445,13 @@ return -ENOTSUP; } + /* Poke GAPK audio back-end, if it is chosen */ + if (ms->settings.audio.io_handler == AUDIO_IOH_GAPK) { + rc = gapk_io_init_ms_chan(ms, ch_type, mode); + if (rc) + return rc; + } + /* Apply indicated channel mode */ LOGP(DRR, LOGL_INFO, "setting TCH mode to %s, audio mode to %d\n", get_value_string(gsm48_chan_mode_names, mode), rr->audio_mode); @@ -3969,6 +3978,10 @@ if (cause) return gsm48_rr_tx_ass_fail(ms, cause, RSL_MT_DATA_REQ); + /* Poke GAPK audio back-end, if it is chosen */ + if (ms->settings.audio.io_handler == AUDIO_IOH_GAPK) + gapk_io_init_ms_chan(ms, ch_type, cda->mode); + #ifdef TEST_FREQUENCY_MOD LOGP(DRR, LOGL_INFO, " TESTING: frequency modify ASS.CMD\n"); before_time = 1; @@ -5544,6 +5557,7 @@ case AUDIO_IOH_L1PHY: rr->audio_mode = AUDIO_RX_SPEAKER | AUDIO_TX_MICROPHONE; break; + case AUDIO_IOH_GAPK: case AUDIO_IOH_LOOPBACK: rr->audio_mode = AUDIO_RX_TRAFFIC_IND | AUDIO_TX_TRAFFIC_REQ; break; diff --git a/src/host/layer23/src/mobile/main.c b/src/host/layer23/src/mobile/main.c index 926358e..088e6c9 100644 --- a/src/host/layer23/src/mobile/main.c +++ b/src/host/layer23/src/mobile/main.c @@ -58,7 +58,7 @@ const char *debug_default = - "DCS:DNB:DPLMN:DRR:DMM:DSIM:DCC:DMNCC:DSS:DLSMS:DPAG:DSUM:DSAP:DGPS:DMOB:DPRIM:DLUA"; + "DCS:DNB:DPLMN:DRR:DMM:DSIM:DCC:DMNCC:DSS:DLSMS:DPAG:DSUM:DSAP:DGPS:DMOB:DPRIM:DLUA:DGAPK"; const char *openbsc_copyright = "Copyright (C) 2010-2015 Andreas Eversberg, Sylvain Munaut, Holger Freyther, Harald Welte\n" diff --git a/src/host/layer23/src/mobile/settings.c b/src/host/layer23/src/mobile/settings.c index 10e9984..c471ba6 100644 --- a/src/host/layer23/src/mobile/settings.c +++ b/src/host/layer23/src/mobile/settings.c @@ -29,6 +29,7 @@ static char *layer2_socket_path = "/tmp/osmocom_l2"; static char *sap_socket_path = "/tmp/osmocom_sap"; static char *mncc_socket_path = "/tmp/ms_mncc"; +static char *alsa_dev_default = "default"; int gsm_settings_init(struct osmocom_ms *ms) { @@ -44,6 +45,8 @@ /* Audio settings: drop TCH frames by default */ set->audio.io_handler = AUDIO_IOH_NONE; + OSMO_STRLCPY_ARRAY(set->audio.alsa_output_dev, alsa_dev_default); + OSMO_STRLCPY_ARRAY(set->audio.alsa_input_dev, alsa_dev_default); /* Built-in MNCC handler */ set->mncc_handler = MNCC_HANDLER_INTERNAL; @@ -203,6 +206,7 @@ const struct value_string audio_io_handler_names[] = { { AUDIO_IOH_NONE, "none" }, + { AUDIO_IOH_GAPK, "gapk" }, { AUDIO_IOH_L1PHY, "l1phy" }, { AUDIO_IOH_LOOPBACK, "loopback" }, { 0x00, NULL} diff --git a/src/host/layer23/src/mobile/voice.c b/src/host/layer23/src/mobile/voice.c index 5f569c7..0cd8a96 100644 --- a/src/host/layer23/src/mobile/voice.c +++ b/src/host/layer23/src/mobile/voice.c @@ -1,5 +1,6 @@ /* * (C) 2010 by Andreas Eversberg <jolly(a)eversberg.eu> + * (C) 2017-2018 by Vadim Yanitskiy <axilirator(a)gmail.com> * * All Rights Reserved * @@ -15,61 +16,47 @@ * */ -#include <stdlib.h> +#include <string.h> +#include <errno.h> #include <osmocom/core/msgb.h> #include <osmocom/codec/codec.h> #include <osmocom/bb/common/logging.h> #include <osmocom/bb/common/osmocom_data.h> +#include <osmocom/bb/mobile/settings.h> +#include <osmocom/bb/mobile/gapk_io.h> #include <osmocom/bb/mobile/mncc.h> #include <osmocom/bb/mobile/voice.h> - /* - * receive voice + * TCH frame (voice) router */ - static int gsm_recv_voice(struct osmocom_ms *ms, struct msgb *msg) { - struct gsm_data_frame *mncc; - - /* Drop the l1ctl_info_dl header */ - msgb_pull_to_l2(msg); - /* push mncc header in front of data */ - mncc = (struct gsm_data_frame *) - msgb_push(msg, sizeof(struct gsm_data_frame)); - mncc->callref = ms->mncc_entity.ref; - - /* FIXME: FR, EFR only! */ - switch (ms->rrlayer.cd_now.mode) { - case GSM48_CMODE_SPEECH_V1: - mncc->msg_type = GSM_TCHF_FRAME; + switch (ms->settings.audio.io_handler) { + case AUDIO_IOH_LOOPBACK: + /* Send voice frame back, if appropriate */ + gsm_send_voice_msg(ms, msg); break; - case GSM48_CMODE_SPEECH_EFR: - mncc->msg_type = GSM_TCHF_FRAME_EFR; + case AUDIO_IOH_GAPK: + /* Prevent null pointer dereference */ + OSMO_ASSERT(ms->gapk_io != NULL); + + /* Push a frame to the DL frame buffer */ + msgb_enqueue(&ms->gapk_io->tch_fb_dl, msg); break; + case AUDIO_IOH_L1PHY: + case AUDIO_IOH_NONE: default: - /* TODO: print error message here */ - goto exit_free; + msgb_free(msg); } - /* send voice frame back, if appropriate */ - if (ms->settings.audio.io_handler == AUDIO_IOH_LOOPBACK) - gsm_send_voice_frame(ms, mncc); - - /* distribute and then free */ - if (ms->mncc_entity.mncc_recv && ms->mncc_entity.ref) { - ms->mncc_entity.mncc_recv(ms, mncc->msg_type, mncc); - } - -exit_free: - msgb_free(msg); return 0; } /* - * send voice + * Send voice to the lower layers */ int gsm_send_voice_msg(struct osmocom_ms *ms, struct msgb *msg) { @@ -107,12 +94,10 @@ } /* - * init + * Init TCH frame (voice) router */ - int gsm_voice_init(struct osmocom_ms *ms) { ms->l1_entity.l1_traffic_ind = gsm_recv_voice; - return 0; } diff --git a/src/host/layer23/src/mobile/vty_interface.c b/src/host/layer23/src/mobile/vty_interface.c index 0abd79c..3bfe532 100644 --- a/src/host/layer23/src/mobile/vty_interface.c +++ b/src/host/layer23/src/mobile/vty_interface.c @@ -1543,8 +1543,12 @@ set->any_timeout, VTY_NEWLINE); vty_out(vty, " audio%s", VTY_NEWLINE); - if (!hide_default || set->audio.io_handler != AUDIO_IOH_NONE) - vty_out(vty, " io-handler %s%s", audio_io_handler_name(set->audio.io_handler), VTY_NEWLINE); + vty_out(vty, " io-handler %s%s", + audio_io_handler_name(set->audio.io_handler), VTY_NEWLINE); + vty_out(vty, " alsa-output-dev %s%s", + set->audio.alsa_output_dev, VTY_NEWLINE); + vty_out(vty, " alsa-input-dev %s%s", + set->audio.alsa_input_dev, VTY_NEWLINE); /* no shutdown must be written to config, because shutdown is default */ vty_out(vty, " %sshutdown%s", (ms->shutdown != MS_SHUTDOWN_NONE) ? "" : "no ", @@ -2831,9 +2835,10 @@ } DEFUN(cfg_ms_audio_io_handler, cfg_ms_audio_io_handler_cmd, - "io-handler (none|l1phy|loopback)", + "io-handler (none|gapk|l1phy|loopback)", "Set TCH frame I/O handler\n" "No handler, drop TCH frames (default)\n" + "libosmo-gapk based I/O handler (requires ALSA)\n" "L1 PHY (e.g. Calypso DSP in Mot C1xx phones)\n" "Return TCH frame payload back to sender\n") { @@ -2847,6 +2852,40 @@ return set_audio_io_handler(vty, AUDIO_IOH_NONE); } +DEFUN(cfg_ms_audio_alsa_out_dev, cfg_ms_audio_alsa_out_dev_cmd, + "alsa-output-dev (default|NAME)", + "Set ALSA playback (i.e. speakers) device name\n" + "Default system playback device (default)\n" + "Name of a custom playback device") +{ + struct osmocom_ms *ms = vty->index; + struct gsm_settings *set = &ms->settings; + const char *dev_name = argv[0]; + + /* Just copy device name */ + strncpy(set->audio.alsa_output_dev, dev_name, + sizeof(set->audio.alsa_output_dev) - 1); + + return CMD_SUCCESS; +} + +DEFUN(cfg_ms_audio_alsa_in_dev, cfg_ms_audio_alsa_in_dev_cmd, + "alsa-input-dev (default|NAME)", + "Set ALSA recording (i.e. mic) device name\n" + "Default system recording device (default)\n" + "Name of a custom recording device") +{ + struct osmocom_ms *ms = vty->index; + struct gsm_settings *set = &ms->settings; + const char *dev_name = argv[0]; + + /* Just copy device name */ + strncpy(set->audio.alsa_input_dev, dev_name, + sizeof(set->audio.alsa_input_dev) - 1); + + return CMD_SUCCESS; +} + DEFUN(cfg_no_shutdown, cfg_ms_no_shutdown_cmd, "no shutdown", NO_STR "Activate and run MS") { @@ -3123,6 +3162,8 @@ install_node(&audio_node, config_write_dummy); install_element(AUDIO_NODE, &cfg_ms_audio_io_handler_cmd); install_element(AUDIO_NODE, &cfg_ms_audio_no_io_handler_cmd); + install_element(AUDIO_NODE, &cfg_ms_audio_alsa_out_dev_cmd); + install_element(AUDIO_NODE, &cfg_ms_audio_alsa_in_dev_cmd); /* Register the talloc context introspection command */ osmo_talloc_vty_add_cmds(); -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30337 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: Ib86b0746606c191573cc773f01172afbb52f33a9 Gerrit-Change-Number: 30337 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-CC: fixeria <axilirator(a)gmail.com> Gerrit-MessageType: newchange

2 years, 7 months

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Change in osmocom-bb[master]: mobile: gsm48_rr_tx_voice(): drop TCH/F channel limitation

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30333 ) Change subject: mobile: gsm48_rr_tx_voice(): drop TCH/F channel limitation ...................................................................... mobile: gsm48_rr_tx_voice(): drop TCH/F channel limitation Change-Id: I06e84ad47bafd4676af0e136b825e77471587b23 Related: OS#5599 --- M src/host/layer23/src/mobile/gsm48_rr.c 1 file changed, 0 insertions(+), 7 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/33/30333/1 diff --git a/src/host/layer23/src/mobile/gsm48_rr.c b/src/host/layer23/src/mobile/gsm48_rr.c index 223188d..8eaf0f6 100644 --- a/src/host/layer23/src/mobile/gsm48_rr.c +++ b/src/host/layer23/src/mobile/gsm48_rr.c @@ -5711,7 +5711,6 @@ int gsm48_rr_tx_voice(struct osmocom_ms *ms, struct msgb *msg) { struct gsm48_rrlayer *rr = &ms->rrlayer; - uint8_t ch_type, ch_subch, ch_ts; struct l1ctl_traffic_req *tr; if (!rr->dm_est) { @@ -5719,12 +5718,6 @@ goto error; } - rsl_dec_chan_nr(rr->cd_now.chan_nr, &ch_type, &ch_subch, &ch_ts); - if (ch_type != RSL_CHAN_Bm_ACCHs) { - LOGP(DRR, LOGL_INFO, "Current channel is not (yet) TCH/F\n"); - goto error; - } - tr = (struct l1ctl_traffic_req *)msg->l2h; if (voice_frame_verify(rr->cd_now.mode, tr->data, msgb_l2len(msg)) < 0) goto error; -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30333 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: I06e84ad47bafd4676af0e136b825e77471587b23 Gerrit-Change-Number: 30333 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-MessageType: newchange

2 years, 7 months

1
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Change in osmocom-bb[master]: mobile: voice.h: use '#pragma once' include guard

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30334 ) Change subject: mobile: voice.h: use '#pragma once' include guard ...................................................................... mobile: voice.h: use '#pragma once' include guard Change-Id: I024e13c0f66ae1e42b0c8e2ff77874fcb883e85a Related: OS#5599 --- M src/host/layer23/include/osmocom/bb/mobile/voice.h 1 file changed, 1 insertion(+), 4 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/34/30334/1 diff --git a/src/host/layer23/include/osmocom/bb/mobile/voice.h b/src/host/layer23/include/osmocom/bb/mobile/voice.h index a052418..b9817f8 100644 --- a/src/host/layer23/include/osmocom/bb/mobile/voice.h +++ b/src/host/layer23/include/osmocom/bb/mobile/voice.h @@ -1,7 +1,4 @@ -#ifndef _voice_h -#define _voice_h +#pragma once int gsm_voice_init(struct osmocom_ms *ms); int gsm_send_voice(struct osmocom_ms *ms, struct gsm_data_frame *data); - -#endif /* _voice_h */ -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30334 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: I024e13c0f66ae1e42b0c8e2ff77874fcb883e85a Gerrit-Change-Number: 30334 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-CC: fixeria <axilirator(a)gmail.com> Gerrit-MessageType: newchange

2 years, 7 months

1
0
0 0

Change in osmocom-bb[master]: mobile: voice.h: add missing forward declarations

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30335 ) Change subject: mobile: voice.h: add missing forward declarations ...................................................................... mobile: voice.h: add missing forward declarations Change-Id: I854abdcde5bc324f6d0cf309792b36fc64601070 Related: OS#5599 --- M src/host/layer23/include/osmocom/bb/mobile/voice.h 1 file changed, 3 insertions(+), 0 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/35/30335/1 diff --git a/src/host/layer23/include/osmocom/bb/mobile/voice.h b/src/host/layer23/include/osmocom/bb/mobile/voice.h index b9817f8..623f6c8 100644 --- a/src/host/layer23/include/osmocom/bb/mobile/voice.h +++ b/src/host/layer23/include/osmocom/bb/mobile/voice.h @@ -1,4 +1,7 @@ #pragma once +struct osmocom_ms; +struct gsm_data_frame; + int gsm_voice_init(struct osmocom_ms *ms); int gsm_send_voice(struct osmocom_ms *ms, struct gsm_data_frame *data); -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30335 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: I854abdcde5bc324f6d0cf309792b36fc64601070 Gerrit-Change-Number: 30335 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-MessageType: newchange

2 years, 7 months

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Change in osmocom-bb[master]: mobile: gsm48_rr_set_mode(): fix copy-paste in comment

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30331 ) Change subject: mobile: gsm48_rr_set_mode(): fix copy-paste in comment ...................................................................... mobile: gsm48_rr_set_mode(): fix copy-paste in comment Change-Id: I45212cd02ac50d0d363918709720e997500e45a9 Related: OS#5599 --- M src/host/layer23/src/mobile/gsm48_rr.c 1 file changed, 1 insertion(+), 1 deletion(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/31/30331/1 diff --git a/src/host/layer23/src/mobile/gsm48_rr.c b/src/host/layer23/src/mobile/gsm48_rr.c index e8c998d..2ddc130 100644 --- a/src/host/layer23/src/mobile/gsm48_rr.c +++ b/src/host/layer23/src/mobile/gsm48_rr.c @@ -3443,7 +3443,7 @@ return -ENOTSUP; } - /* setting (new) timing advance */ + /* Apply indicated channel mode */ LOGP(DRR, LOGL_INFO, "setting TCH mode to %s, audio mode to %d\n", get_value_string(gsm48_chan_mode_names, mode), rr->audio_mode); l1ctl_tx_tch_mode_req(ms, mode, rr->audio_mode, rr->tch_loop_mode); -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30331 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: I45212cd02ac50d0d363918709720e997500e45a9 Gerrit-Change-Number: 30331 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-CC: fixeria <axilirator(a)gmail.com> Gerrit-MessageType: newchange

2 years, 7 months

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Change in osmocom-bb[master]: mobile: properly handle RR CHANNEL MODE MODIFY message

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30332 ) Change subject: mobile: properly handle RR CHANNEL MODE MODIFY message ...................................................................... mobile: properly handle RR CHANNEL MODE MODIFY message According to 3GPP TS 04.08, section 3.4.6.1.3 "Abnormal cases" of "channel mode modify procedure", if the MS doesn't support the indicated channel mode, it shall retain the old mode and return the associated channel mode information in the ACKNOWLEDGE message. Previously, if an indicated mode is not supported, we used to indicate the 'CHAN_MODE_UNACCT' RR case without sending the ACKNOWLEDGE message. Also, the result of gsm48_rr_set_mode() was ignored. Let's fix this! Change-Id: I952436ec796273e56341f9d3492b4a3b3a5dc410 Related: OS#5599 --- M src/host/layer23/src/mobile/gsm48_rr.c 1 file changed, 24 insertions(+), 8 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/32/30332/1 diff --git a/src/host/layer23/src/mobile/gsm48_rr.c b/src/host/layer23/src/mobile/gsm48_rr.c index 2ddc130..223188d 100644 --- a/src/host/layer23/src/mobile/gsm48_rr.c +++ b/src/host/layer23/src/mobile/gsm48_rr.c @@ -3573,7 +3573,7 @@ int payload_len = msgb_l3len(msg) - sizeof(*gh) - sizeof(*cm); struct gsm48_rr_cd *cd = &rr->cd_now; uint8_t ch_type, ch_subch, ch_ts; - uint8_t cause; + int rc; LOGP(DRR, LOGL_INFO, "CHANNEL MODE MODIFY\n"); @@ -3605,14 +3605,30 @@ gsm_print_arfcn(cd->arfcn), ch_ts, ch_subch, cd->tsc, cm->mode); } - /* mode */ - cause = gsm48_rr_check_mode(ms, cd->chan_nr, cm->mode); - if (cause) - return gsm48_rr_tx_rr_status(ms, cause); - cd->mode = cm->mode; - gsm48_rr_set_mode(ms, cd->chan_nr, cd->mode); - return gsm48_rr_tx_chan_modify_ack(ms, &cm->chan_desc, cm->mode); + /** + * According to 3GPP TS 04.08, section 3.4.6.1.3 + * "Abnormal cases" of "channel mode modify procedure", + * if the MS doesn't support the indicated channel mode, + * it shall retain the old mode and return the associated + * channel mode information in the ACKNOWLEDGE message. + */ + + /* Check if we support this channel mode */ + rc = gsm48_rr_check_mode(ms, cd->chan_nr, cm->mode); + if (rc) + goto ack; + + /* Attempt to apply this mode */ + rc = gsm48_rr_set_mode(ms, cd->chan_nr, cm->mode); + if (rc) + goto ack; + + /* Finally set (a new) mode */ + cd->mode = cm->mode; + +ack: + return gsm48_rr_tx_chan_modify_ack(ms, &cm->chan_desc, cd->mode); } /* 9.1.3 sending ASSIGNMENT COMPLETE */ -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30332 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: I952436ec796273e56341f9d3492b4a3b3a5dc410 Gerrit-Change-Number: 30332 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-CC: fixeria <axilirator(a)gmail.com> Gerrit-MessageType: newchange

2 years, 7 months

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Change in osmocom-bb[master]: mobile: gsm48_rr_set_mode(): check rc of rsl_dec_chan_nr()

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30329 ) Change subject: mobile: gsm48_rr_set_mode(): check rc of rsl_dec_chan_nr() ...................................................................... mobile: gsm48_rr_set_mode(): check rc of rsl_dec_chan_nr() The rsl_dec_chan_nr() may fail to decode RSL channel number, so the 'ch_type' variable would be uninitialized. Let's check rc. Change-Id: I9ab18bdaf41a29fcd32a7060668ef9db07b8cf7e Related: OS#5599 --- M src/host/layer23/src/mobile/gsm48_rr.c 1 file changed, 8 insertions(+), 1 deletion(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/29/30329/1 diff --git a/src/host/layer23/src/mobile/gsm48_rr.c b/src/host/layer23/src/mobile/gsm48_rr.c index 6092835..e7722f1 100644 --- a/src/host/layer23/src/mobile/gsm48_rr.c +++ b/src/host/layer23/src/mobile/gsm48_rr.c @@ -3427,9 +3427,16 @@ { struct gsm48_rrlayer *rr = &ms->rrlayer; uint8_t ch_type, ch_subch, ch_ts; + int rc; + + /* Decode RSL channel number */ + rc = rsl_dec_chan_nr(chan_nr, &ch_type, &ch_subch, &ch_ts); + if (rc) { + LOGP(DRR, LOGL_ERROR, "Couldn't decode RSL channel number\n"); + return -EINVAL; + } /* only apply mode to TCH/F or TCH/H */ - rsl_dec_chan_nr(chan_nr, &ch_type, &ch_subch, &ch_ts); if (ch_type != RSL_CHAN_Bm_ACCHs && ch_type != RSL_CHAN_Lm_ACCHs) return -ENOTSUP; -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30329 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: I9ab18bdaf41a29fcd32a7060668ef9db07b8cf7e Gerrit-Change-Number: 30329 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-CC: fixeria <axilirator(a)gmail.com> Gerrit-MessageType: newchange

2 years, 7 months

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Change in osmocom-bb[master]: mobile: gsm48_rr_set_mode(): print error if ch_type is not TCH

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30330 ) Change subject: mobile: gsm48_rr_set_mode(): print error if ch_type is not TCH ...................................................................... mobile: gsm48_rr_set_mode(): print error if ch_type is not TCH The CHANNEL MODE MODIFY message only applies to TCH channels, and in general should not arrive on non-traffic channels. Let's print error message if this happens in order to facilitate finding possible bugs. Change-Id: I4ab63c4ae4262c8166de37e4873fc3f1b8ec6fe7 Related: OS#5599 --- M src/host/layer23/src/mobile/gsm48_rr.c 1 file changed, 4 insertions(+), 2 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/30/30330/1 diff --git a/src/host/layer23/src/mobile/gsm48_rr.c b/src/host/layer23/src/mobile/gsm48_rr.c index e7722f1..e8c998d 100644 --- a/src/host/layer23/src/mobile/gsm48_rr.c +++ b/src/host/layer23/src/mobile/gsm48_rr.c @@ -3437,9 +3437,11 @@ } /* only apply mode to TCH/F or TCH/H */ - if (ch_type != RSL_CHAN_Bm_ACCHs - && ch_type != RSL_CHAN_Lm_ACCHs) + if (ch_type != RSL_CHAN_Bm_ACCHs && ch_type != RSL_CHAN_Lm_ACCHs) { + LOGP(DRR, LOGL_ERROR, "CHANNEL MODE MODIFY only applies " + "to TCH channels, ignoring...\n"); return -ENOTSUP; + } /* setting (new) timing advance */ LOGP(DRR, LOGL_INFO, "setting TCH mode to %s, audio mode to %d\n", -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30330 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: I4ab63c4ae4262c8166de37e4873fc3f1b8ec6fe7 Gerrit-Change-Number: 30330 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-CC: fixeria <axilirator(a)gmail.com> Gerrit-MessageType: newchange

2 years, 7 months

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Change in osmocom-bb[master]: mobile: allow enabling/disabling handling of voice in the L1PHY

by fixeria

fixeria has uploaded this change for review. ( https://gerrit.osmocom.org/c/osmocom-bb/+/30327 ) Change subject: mobile: allow enabling/disabling handling of voice in the L1PHY ...................................................................... mobile: allow enabling/disabling handling of voice in the L1PHY Some L1PHY targets (e.g. Calypso based Mot C1xx phones) have built in microphone and speaker. Some targets do not have them. Currently we unconditionally instruct the L1PHY to handle TCH frames internally. Make this behavior configurable via the VTY interface. Change-Id: I131f213ef7c2736f7310f0183b83f3bc3064cd98 Related: OS#5599 --- M src/host/layer23/include/osmocom/bb/mobile/settings.h M src/host/layer23/src/mobile/gsm48_rr.c M src/host/layer23/src/mobile/settings.c M src/host/layer23/src/mobile/vty_interface.c 4 files changed, 19 insertions(+), 4 deletions(-) git pull ssh://gerrit.osmocom.org:29418/osmocom-bb refs/changes/27/30327/1 diff --git a/src/host/layer23/include/osmocom/bb/mobile/settings.h b/src/host/layer23/include/osmocom/bb/mobile/settings.h index d5620ff..8edde53 100644 --- a/src/host/layer23/include/osmocom/bb/mobile/settings.h +++ b/src/host/layer23/include/osmocom/bb/mobile/settings.h @@ -17,6 +17,8 @@ enum audio_io_handler { /* No handler, drop frames */ AUDIO_IOH_NONE = 0, + /* L1 PHY (e.g. Calypso DSP) */ + AUDIO_IOH_L1PHY, /* Return to sender */ AUDIO_IOH_LOOPBACK, }; diff --git a/src/host/layer23/src/mobile/gsm48_rr.c b/src/host/layer23/src/mobile/gsm48_rr.c index fa11001..f54c945 100644 --- a/src/host/layer23/src/mobile/gsm48_rr.c +++ b/src/host/layer23/src/mobile/gsm48_rr.c @@ -5512,9 +5512,20 @@ start_rr_t_meas(rr, 1, 0); - rr->audio_mode = AUDIO_TX_MICROPHONE | AUDIO_RX_SPEAKER; rr->tch_loop_mode = L1CTL_TCH_LOOP_OPEN; + /* Configure audio handling in the L1PHY */ + switch (ms->settings.audio.io_handler) { + case AUDIO_IOH_L1PHY: + rr->audio_mode = AUDIO_RX_SPEAKER | AUDIO_TX_MICROPHONE; + break; + case AUDIO_IOH_LOOPBACK: + rr->audio_mode = AUDIO_RX_TRAFFIC_IND | AUDIO_TX_TRAFFIC_REQ; + break; + case AUDIO_IOH_NONE: + rr->audio_mode = 0x00; + } + return 0; } diff --git a/src/host/layer23/src/mobile/settings.c b/src/host/layer23/src/mobile/settings.c index 3444d66..10e9984 100644 --- a/src/host/layer23/src/mobile/settings.c +++ b/src/host/layer23/src/mobile/settings.c @@ -203,6 +203,7 @@ const struct value_string audio_io_handler_names[] = { { AUDIO_IOH_NONE, "none" }, + { AUDIO_IOH_L1PHY, "l1phy" }, { AUDIO_IOH_LOOPBACK, "loopback" }, { 0x00, NULL} }; diff --git a/src/host/layer23/src/mobile/vty_interface.c b/src/host/layer23/src/mobile/vty_interface.c index 9c3e429..0abd79c 100644 --- a/src/host/layer23/src/mobile/vty_interface.c +++ b/src/host/layer23/src/mobile/vty_interface.c @@ -2831,10 +2831,11 @@ } DEFUN(cfg_ms_audio_io_handler, cfg_ms_audio_io_handler_cmd, - "io-handler (loopback|none)", + "io-handler (none|l1phy|loopback)", "Set TCH frame I/O handler\n" - "Return TCH frame payload back to sender\n" - "No handler, drop TCH frames (default)") + "No handler, drop TCH frames (default)\n" + "L1 PHY (e.g. Calypso DSP in Mot C1xx phones)\n" + "Return TCH frame payload back to sender\n") { int val = get_string_value(audio_io_handler_names, argv[0]); return set_audio_io_handler(vty, val); -- To view, visit https://gerrit.osmocom.org/c/osmocom-bb/+/30327 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmocom-bb Gerrit-Branch: master Gerrit-Change-Id: I131f213ef7c2736f7310f0183b83f3bc3064cd98 Gerrit-Change-Number: 30327 Gerrit-PatchSet: 1 Gerrit-Owner: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-MessageType: newchange

2 years, 7 months

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