diff options
author | V3n3RiX <venerix@koprulu.sector> | 2023-10-06 17:57:58 +0100 |
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committer | V3n3RiX <venerix@koprulu.sector> | 2023-10-06 17:57:58 +0100 |
commit | 8368c614e6a1232f34a14cd6495cf2f68933030d (patch) | |
tree | fb445ed1110e0361c9c25136c00c4fb948bb46c4 /media-libs/libpulse/files | |
parent | 2b03b0d9c5ca3890c219ebd24721133e766e9bb9 (diff) |
gentoo auto-resync : 06:10:2023 - 17:57:58
Diffstat (limited to 'media-libs/libpulse/files')
4 files changed, 210 insertions, 0 deletions
diff --git a/media-libs/libpulse/files/pulseaudio-16.1-add-more-standard-samplerates.patch b/media-libs/libpulse/files/pulseaudio-16.1-add-more-standard-samplerates.patch new file mode 100644 index 000000000000..e1ea200c4c4c --- /dev/null +++ b/media-libs/libpulse/files/pulseaudio-16.1-add-more-standard-samplerates.patch @@ -0,0 +1,35 @@ +commit 86c5fbab5778685e19b5a4a9b8eb04ca90dff780 +Author: Igor V. Kovalenko <igor.v.kovalenko@gmail.com> +Date: Sun Feb 5 19:49:10 2023 +0300 + + alsa-util: Add more standard sample rates. + + Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/776> + +diff --git a/src/modules/alsa/alsa-util.c b/src/modules/alsa/alsa-util.c +index 9f35cb20f..fd30f18bd 100644 +--- a/src/modules/alsa/alsa-util.c ++++ b/src/modules/alsa/alsa-util.c +@@ -1430,7 +1430,8 @@ unsigned int *pa_alsa_get_supported_rates(snd_pcm_t *pcm, unsigned int fallback_ + 32000, 44100, 48000, + 64000, 88200, 96000, + 128000, 176400, 192000, +- 384000 }; ++ 352800, 384000, ++ 705600, 768000 }; + bool supported[PA_ELEMENTSOF(all_rates)] = { false, }; + snd_pcm_hw_params_t *hwparams; + unsigned int i, j, n, *rates = NULL; +diff --git a/src/pulse/sample.h b/src/pulse/sample.h +index 35346a865..65c0c5d6b 100644 +--- a/src/pulse/sample.h ++++ b/src/pulse/sample.h +@@ -128,7 +128,7 @@ PA_C_DECL_BEGIN + #define PA_CHANNELS_MAX 32U + + /** Maximum allowed sample rate */ +-#define PA_RATE_MAX (48000U*8U) ++#define PA_RATE_MAX (48000U*16U) + + /** Sample format */ + typedef enum pa_sample_format { diff --git a/media-libs/libpulse/files/pulseaudio-16.1-fix-memblock-alignment.patch b/media-libs/libpulse/files/pulseaudio-16.1-fix-memblock-alignment.patch new file mode 100644 index 000000000000..b05e530ac89d --- /dev/null +++ b/media-libs/libpulse/files/pulseaudio-16.1-fix-memblock-alignment.patch @@ -0,0 +1,122 @@ +commit 300db779224625144d6279d230c2daa857c967d8 +Author: Igor V. Kovalenko <igor.v.kovalenko@gmail.com> +Date: Thu Feb 9 13:28:29 2023 +0300 + + pstream: Pass frame size to keep split memblock parts aligned + + `pa_pstream_send_memblock()` would split incoming memblock into parts not + exceeding maximum pool block size. + + To make sure split parts of memblock are still frame-aligned add new `align` arg + to `pa_pstream_send_memblock`, find out required alignment from stream sample + format and pass it there. Bump default alignment to 256 which is good up to + 32bit 64ch frames. + + Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/780> + +diff --git a/src/modules/module-tunnel.c b/src/modules/module-tunnel.c +index 61f427bd3..ab094ba4e 100644 +--- a/src/modules/module-tunnel.c ++++ b/src/modules/module-tunnel.c +@@ -676,7 +676,7 @@ static int sink_process_msg(pa_msgobject *o, int code, void *data, int64_t offse + * IO thread context where the rest of the messages are + * dispatched. Yeah, ugly, but I am a lazy bastard. */ + +- pa_pstream_send_memblock(u->pstream, u->channel, 0, PA_SEEK_RELATIVE, chunk); ++ pa_pstream_send_memblock(u->pstream, u->channel, 0, PA_SEEK_RELATIVE, chunk, pa_frame_size(&u->sink->sample_spec)); + + u->receive_counter += chunk->length; + +diff --git a/src/pulse/stream.c b/src/pulse/stream.c +index 0aa627396..3585b27e8 100644 +--- a/src/pulse/stream.c ++++ b/src/pulse/stream.c +@@ -1535,7 +1535,7 @@ int pa_stream_write_ext_free( + s->write_memblock = NULL; + s->write_data = NULL; + +- pa_pstream_send_memblock(s->context->pstream, s->channel, offset, seek, &chunk); ++ pa_pstream_send_memblock(s->context->pstream, s->channel, offset, seek, &chunk, pa_frame_size(&s->sample_spec)); + pa_memblock_unref(chunk.memblock); + + } else { +@@ -1569,7 +1569,7 @@ int pa_stream_write_ext_free( + pa_memblock_release(chunk.memblock); + } + +- pa_pstream_send_memblock(s->context->pstream, s->channel, t_offset, t_seek, &chunk); ++ pa_pstream_send_memblock(s->context->pstream, s->channel, t_offset, t_seek, &chunk, pa_frame_size(&s->sample_spec)); + + t_offset = 0; + t_seek = PA_SEEK_RELATIVE; +diff --git a/src/pulsecore/protocol-native.c b/src/pulsecore/protocol-native.c +index 672182fbc..1342dee10 100644 +--- a/src/pulsecore/protocol-native.c ++++ b/src/pulsecore/protocol-native.c +@@ -1260,7 +1260,7 @@ static void native_connection_send_memblock(pa_native_connection *c) { + if (schunk.length > r->buffer_attr.fragsize) + schunk.length = r->buffer_attr.fragsize; + +- pa_pstream_send_memblock(c->pstream, r->index, 0, PA_SEEK_RELATIVE, &schunk); ++ pa_pstream_send_memblock(c->pstream, r->index, 0, PA_SEEK_RELATIVE, &schunk, pa_memblockq_get_base(r->memblockq)); + + pa_memblockq_drop(r->memblockq, schunk.length); + pa_memblock_unref(schunk.memblock); +@@ -2535,7 +2535,7 @@ static void setup_srbchannel(pa_native_connection *c, pa_mem_type_t shm_type) { + mc.memblock = srbt.memblock; + mc.index = 0; + mc.length = pa_memblock_get_length(srbt.memblock); +- pa_pstream_send_memblock(c->pstream, 0, 0, 0, &mc); ++ pa_pstream_send_memblock(c->pstream, 0, 0, 0, &mc, 0); + + c->srbpending = srb; + return; +diff --git a/src/pulsecore/pstream.c b/src/pulsecore/pstream.c +index 7147b776a..ff62f464b 100644 +--- a/src/pulsecore/pstream.c ++++ b/src/pulsecore/pstream.c +@@ -82,6 +82,10 @@ typedef uint32_t pa_pstream_descriptor[PA_PSTREAM_DESCRIPTOR_MAX]; + */ + #define FRAME_SIZE_MAX_ALLOW (1024*1024*16) + ++/* Default memblock alignment used with pa_pstream_send_memblock() ++ */ ++#define DEFAULT_PSTREAM_MEMBLOCK_ALIGN (256) ++ + PA_STATIC_FLIST_DECLARE(items, 0, pa_xfree); + + struct item_info { +@@ -475,7 +479,7 @@ void pa_pstream_send_packet(pa_pstream*p, pa_packet *packet, pa_cmsg_ancil_data + p->mainloop->defer_enable(p->defer_event, 1); + } + +-void pa_pstream_send_memblock(pa_pstream*p, uint32_t channel, int64_t offset, pa_seek_mode_t seek_mode, const pa_memchunk *chunk) { ++void pa_pstream_send_memblock(pa_pstream*p, uint32_t channel, int64_t offset, pa_seek_mode_t seek_mode, const pa_memchunk *chunk, size_t align) { + size_t length, idx; + size_t bsm; + +@@ -492,6 +496,11 @@ void pa_pstream_send_memblock(pa_pstream*p, uint32_t channel, int64_t offset, pa + + bsm = pa_mempool_block_size_max(p->mempool); + ++ if (align == 0) ++ align = DEFAULT_PSTREAM_MEMBLOCK_ALIGN; ++ ++ bsm = (bsm / align) * align; ++ + while (length > 0) { + struct item_info *i; + size_t n; +diff --git a/src/pulsecore/pstream.h b/src/pulsecore/pstream.h +index 2bff270ad..88bdca4cc 100644 +--- a/src/pulsecore/pstream.h ++++ b/src/pulsecore/pstream.h +@@ -51,7 +51,7 @@ void pa_pstream_unlink(pa_pstream *p); + int pa_pstream_attach_memfd_shmid(pa_pstream *p, unsigned shm_id, int memfd_fd); + + void pa_pstream_send_packet(pa_pstream*p, pa_packet *packet, pa_cmsg_ancil_data *ancil_data); +-void pa_pstream_send_memblock(pa_pstream*p, uint32_t channel, int64_t offset, pa_seek_mode_t seek, const pa_memchunk *chunk); ++void pa_pstream_send_memblock(pa_pstream*p, uint32_t channel, int64_t offset, pa_seek_mode_t seek, const pa_memchunk *chunk, size_t align); + void pa_pstream_send_release(pa_pstream *p, uint32_t block_id); + void pa_pstream_send_revoke(pa_pstream *p, uint32_t block_id); + diff --git a/media-libs/libpulse/files/pulseaudio-16.1-smoother-start-paused.patch b/media-libs/libpulse/files/pulseaudio-16.1-smoother-start-paused.patch new file mode 100644 index 000000000000..3f0acbc8f95e --- /dev/null +++ b/media-libs/libpulse/files/pulseaudio-16.1-smoother-start-paused.patch @@ -0,0 +1,26 @@ +commit 8fe50bbc31e11abf2f30864f1e2dbdaa16d0e1c3 +Author: Georg Chini <georg@chini.tk> +Date: Thu Aug 25 08:11:04 2022 +0200 + + time-smoother-2: Fix stream time when stream starts paused + + When a stream is started but has not yet called smoother_2_put(), pa_smoother_2_get() + returns the time since the start of the stream even if the stream was started paused. + When the stream is started paused, pa_smoother_2_get() should return 0 instead. This + patch fixes the problem. + + Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/745> + +diff --git a/src/pulsecore/time-smoother_2.c b/src/pulsecore/time-smoother_2.c +index e14b52f72..ea7ec1b36 100644 +--- a/src/pulsecore/time-smoother_2.c ++++ b/src/pulsecore/time-smoother_2.c +@@ -295,7 +295,7 @@ pa_usec_t pa_smoother_2_get(pa_smoother_2 *s, pa_usec_t time_stamp) { + + /* If the smoother has not started, just return system time since resume */ + if (!s->start_time) { +- if (time_stamp >= s->resume_time) ++ if (time_stamp >= s->resume_time && !s->paused) + current_time = time_stamp - s->resume_time; + else + current_time = 0; diff --git a/media-libs/libpulse/files/pulseaudio-16.1-smoother-time-calculation.patch b/media-libs/libpulse/files/pulseaudio-16.1-smoother-time-calculation.patch new file mode 100644 index 000000000000..9d976e5be984 --- /dev/null +++ b/media-libs/libpulse/files/pulseaudio-16.1-smoother-time-calculation.patch @@ -0,0 +1,27 @@ +commit c3eae5d00cb79bd897049483126e75bb48a69cd1 +Author: flyingOwl <ofenfisch@googlemail.com> +Date: Fri Dec 30 00:16:03 2022 +0100 + + time-smoother-2: Fix time calculation by comparing timestamps + + This fixes the rare case of resume_time being bigger than time_stamp. Which + happens sometimes when a gstreamer client is quickly seeking through a + media file. The resulting integer underflow then causes a huge value in + current_time which will break the playback. + + Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/764> + +diff --git a/src/pulsecore/time-smoother_2.c b/src/pulsecore/time-smoother_2.c +index ea7ec1b36..46cc5e9cc 100644 +--- a/src/pulsecore/time-smoother_2.c ++++ b/src/pulsecore/time-smoother_2.c +@@ -307,7 +307,8 @@ pa_usec_t pa_smoother_2_get(pa_smoother_2 *s, pa_usec_t time_stamp) { + /* If we are initializing, add the time since resume to the card time at pause_time */ + else if (s->init) { + current_time += (s->pause_time - s->start_time - s->time_offset - s->fixup_time) * s->time_factor; +- current_time += (time_stamp - s->resume_time) * s->time_factor; ++ if (time_stamp > s->resume_time) ++ current_time += (time_stamp - s->resume_time) * s->time_factor; + + /* Smoother is running, calculate current sound card time */ + } else |