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2023-03-10linux-user/sparc: Fix sparc64_{get, set}_context trapsRichard Henderson
These traps are present for sparc64 with ilp32, aka sparc32plus. Enabling them means adjusting the defines over in signal.c, and fixing an incorrect usage of abi_ulong when we really meant the full register, target_ulong. Signed-off-by: Richard Henderson <richard.henderson@linaro.org> Message-Id: <20230216054516.1267305-7-richard.henderson@linaro.org> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user/sparc: Tidy window spill/fill trapsRichard Henderson
Add some macros to localize the hw difference between v9 and pre-v9. Signed-off-by: Richard Henderson <richard.henderson@linaro.org> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Message-Id: <20230216054516.1267305-6-richard.henderson@linaro.org> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user/sparc: Use TT_TRAP for flush windowsRichard Henderson
The v9 and pre-v9 code can be unified with this macro. Signed-off-by: Richard Henderson <richard.henderson@linaro.org> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Message-Id: <20230216054516.1267305-5-richard.henderson@linaro.org> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user/sparc: Tidy syscall error returnRichard Henderson
Reduce ifdefs with #define syscall_cc. Signed-off-by: Richard Henderson <richard.henderson@linaro.org> Message-Id: <20230216054516.1267305-4-richard.henderson@linaro.org> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user/sparc: Tidy syscall trapRichard Henderson
Use TT_TRAP. For sparc32, 0x88 is the "Slowaris" system call, currently BAD_TRAP in the kernel's ttable_32.S. For sparc64, 0x110 is tl0_linux32, the sparc32 trap, now folded into the TARGET_ABI32 case via TT_TRAP. For sparc64, there does still exist trap 0x111 as tl0_oldlinux64, which was replaced by 0x16d as tl0_linux64 in 1998. Since no one has noticed, don't bother implementing it now. Signed-off-by: Richard Henderson <richard.henderson@linaro.org> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Message-Id: <20230216054516.1267305-3-richard.henderson@linaro.org> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: Emulate CLONE_PIDFD flag in clone()Helge Deller
Add emulation for the CLONE_PIDFD flag of the clone() syscall. This flag was added in Linux kernel 5.2. Successfully tested on a x86-64 Linux host with hppa-linux target. Can be verified by running the testsuite of the qcoro debian package, which breaks hard and kills the currently logged-in user without this patch. Signed-off-by: Helge Deller <deller@gmx.de> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <Y4XoJCpvUA1JD7Sj@p100> [lv: define CLONE_PIDFD if it is not] Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: Add translation for argument of msync()Helge Deller
msync() uses the flags MS_ASYNC, MS_INVALIDATE and MS_SYNC, which differ between platforms, specifcally on alpha and hppa. Add a target to host translation for those and wire up a nicer strace output. This fixes the testsuite of the macaulay2 debian package with a hppa-linux guest on a x86-64 host. Signed-off-by: Helge Deller <deller@gmx.de> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <Y5rMcts4qe15RaVN@p100> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: handle netlink flag NLA_F_NESTEDMathis Marion
Newer kernel versions require this flag to be present contrary to older ones. Depending on the libnl version it is added or not. Typically when using rtnl_link_inet6_set_addr_gen_mode, the netlink packet generated may contain the following attribute: with libnl 3.4 {nla_len=16, nla_type=IFLA_AF_SPEC}, [ {nla_len=12, nla_type=AF_INET6}, [{nla_len=5, nla_type=IFLA_INET6_ADDR_GEN_MODE}, IN6_ADDR_GEN_MODE_NONE] ] with libnl 3.7 {nla_len=16, nla_type=NLA_F_NESTED|IFLA_AF_SPEC}, [ {nla_len=12, nla_type=NLA_F_NESTED|AF_INET6}, [{nla_len=5, nla_type=IFLA_INET6_ADDR_GEN_MODE}, IN6_ADDR_GEN_MODE_NONE]] ] Masking the type is likely needed in other places. Only the above cases are implemented in this patch. Signed-off-by: Mathis Marion <mathis.marion@silabs.com> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <20230307154256.101528-3-Mathis.Marion@silabs.com> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: fix sockaddr_in6 endiannessMathis Marion
The sin6_scope_id field uses the host byte order, so there is a conversion to be made when host and target endianness differ. Signed-off-by: Mathis Marion <mathis.marion@silabs.com> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Message-Id: <20230307154256.101528-2-Mathis.Marion@silabs.com> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: Add strace for prlimit64() syscallHelge Deller
Add proper prlimit64() strace output. Signed-off-by: Helge Deller <deller@gmx.de> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <20221222190639.124078-1-deller@gmx.de> [lvivier: use print_raw_param64()] Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: Provide print_raw_param64() for 64-bit valuesHelge Deller
Add a new function print_raw_param64() to print 64-bit values in the same way as print_raw_param(). This prevents that qemu_log() is used to work around the problem that print_raw_param() can only print 32-bit values when compiled for 32-bit targets. Additionally convert the existing 64-bit users in print_timespec64(), print_rlimit64() and print_preadwrite64() over to this new function and drop some unneccessary spaces. Suggested-by: Laurent Vivier <laurent@vivier.eu> Signed-off-by: Helge Deller <deller@gmx.de> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <Y9lNbFNyRSUhhrHa@p100> [lvivier: remove print_preadwrite64 and print_rlimit64 part] Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: Fix brk() to release pagesHelge Deller
The current brk() implementation does not de-allocate pages if a lower address is given compared to earlier brk() calls. But according to the manpage, brk() shall deallocate memory in this case and currently it breaks a real-world application, specifically building the debian gcl package in qemu-user. Fix this issue by reworking the qemu brk() implementation. Tested with the C-code testcase included in qemu commit 4d1de87c750, and by building debian package of gcl in a hppa-linux guest on a x86-64 host. Signed-off-by: Helge Deller <deller@gmx.de> Message-Id: <Y6gId80ek49TK1xB@p100> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: fill out task state in /proc/self/statAndreas Schwab
Some programs want to match an actual task state character. Signed-off-by: Andreas Schwab <schwab@suse.de> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <mvmedq2kxoe.fsf@suse.de> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: add support for xtensa FDPICMax Filippov
Define xtensa-specific info_is_fdpic and fill in FDPIC-specific registers in the xtensa version of init_thread. Signed-off-by: Max Filippov <jcmvbkbc@gmail.com> Message-Id: <20230205061230.544451-1-jcmvbkbc@gmail.com> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: Fix unaligned memory access in prlimit64 syscallIlya Leoshkevich
target_rlimit64 contains uint64_t fields, so it's 8-byte aligned on some hosts, while some guests may align their respective type on a 4-byte boundary. This may lead to an unaligned access, which is an UB. Fix by defining the fields as abi_ullong. This makes the host alignment match that of the guest, and lets the compiler know that it should emit code that can deal with the guest alignment. While at it, also use __get_user() and __put_user() instead of tswap64(). Fixes: 163a05a8398b ("linux-user: Implement prlimit64 syscall") Reported-by: Richard Henderson <richard.henderson@linaro.org> Signed-off-by: Ilya Leoshkevich <iii@linux.ibm.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <20230224003907.263914-2-iii@linux.ibm.com> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: add target to host netlink conversionsMathis Marion
Added conversions for: - IFLA_MTU - IFLA_TXQLEN - IFLA_AF_SPEC AF_INET6 IFLA_INET6_ADDR_GEN_MODE These relate to the libnl functions rtnl_link_set_mtu, rtnl_link_set_txqlen, and rtnl_link_inet6_set_addr_gen_mode. Signed-off-by: Mathis Marion <mathis.marion@silabs.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <20230220085822.626798-4-Mathis.Marion@silabs.com> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: fix timerfd read endianness conversionMathis Marion
When reading the expiration count from a timerfd, the endianness of the 64bit value read is the one of the host, just as for eventfds. Signed-off-by: Mathis Marion <mathis.marion@silabs.com> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <20230220085822.626798-2-Mathis.Marion@silabs.com> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-10linux-user: Fix access to /proc/self/exeHelge Deller
When accsssing /proc/self/exe from a userspace program, linux-user tries to resolve the name via realpath(), which may fail if the process changed the working directory in the meantime. An example: - a userspace program ist started with ./testprogram - the program runs chdir("/tmp") - then the program calls readlink("/proc/self/exe") - linux-user tries to run realpath("./testprogram") which fails because ./testprogram isn't in /tmp - readlink() will return -ENOENT back to the program Avoid this issue by resolving the full path name of the started process at startup of linux-user and store it in real_exec_path[]. This then simplifies the emulation of readlink() and readlinkat() as well, because they can simply copy the path string to userspace. I noticed this bug because the testsuite of the debian package "pandoc" failed on linux-user while it succeeded on real hardware. The full log is here: https://buildd.debian.org/status/fetch.php?pkg=pandoc&arch=hppa&ver=2.17.1.1-1.1%2Bb1&stamp=1670153210&raw=0 Signed-off-by: Helge Deller <deller@gmx.de> Reviewed-by: Laurent Vivier <laurent@vivier.eu> Message-Id: <20221205113825.20615-1-deller@gmx.de> Signed-off-by: Laurent Vivier <laurent@vivier.eu>
2023-03-06Merge tag 'audio-pull-request' of https://gitlab.com/marcandre.lureau/qemu ↵Peter Maydell
into staging Audio patches for QEMU 8.0 Cleanups and improvements from Volker Rümelin. # -----BEGIN PGP SIGNATURE----- # # iQJQBAABCAA6FiEEh6m9kz+HxgbSdvYt2ujhCXWWnOUFAmQFjUccHG1hcmNhbmRy # ZS5sdXJlYXVAcmVkaGF0LmNvbQAKCRDa6OEJdZac5ZLMD/sFWmVYccRqvOfOJvuC # dJ7EtABmC6b4MxLkMyP5WLMoCezSha0vgcYwAL3oZX1dIBoGfaUXXdxIM8IK8rAv # J2Fe2p9D+6tqzqc8jkbx0DmY040A3Gb18P60jL3NeQfmqCt7KmzuQoCMn6htPyyy # NMgSRG3wy+esglmeU2majhxQZh2Vmwsi6i37A5Fg8b27oWLYHxeS6Lgp3PNAwJsA # k/RKJeZDMjDh0f0wSHZN+8w0VVFQa3BjqoNV4UAgqSjsPaRlnQxppG2FEXNNrtLb # bJ4rbXgSO68wrcweHgC2L7qWhLnYwdcdQFAPotmt2V1jtH4xx6f+ovwCdf0vZ5Fo # YhHixpP41gFvF7sJWSZO0wUFKXTyLSdWIPaUKP9iMTZfXpBxo5H6rkPsZ6v5yyG2 # tLFlL0Rv/h3liEkngmRbw81gB9te92ZGXOjkB2Q69tGQl/tXRkhrn5VT2mufb7BV # WUsdtSFl6P231lDuzQkFp85PkoRtr8eDPkkZW8/xgRx5h8j5I2bc6LRMd63w2tE6 # TGdUfiaqH396r1T87ynBdBWSc4H6YuImTouy27rAv/50x8w4cgk66wY+UGB7D8FX # Eg4rU5Dzco2D+1RY50zSqZgPxIWjnC4xtYQXLaJURe0Way1HQqt/Pvp/aVFKUOef # snYV1FLDvHg2Cf9LCGqvpW8b/w== # =d4LY # -----END PGP SIGNATURE----- # gpg: Signature made Mon 06 Mar 2023 06:50:47 GMT # gpg: using RSA key 87A9BD933F87C606D276F62DDAE8E10975969CE5 # gpg: issuer "marcandre.lureau@redhat.com" # gpg: Good signature from "Marc-André Lureau <marcandre.lureau@redhat.com>" [full] # gpg: aka "Marc-André Lureau <marcandre.lureau@gmail.com>" [full] # Primary key fingerprint: 87A9 BD93 3F87 C606 D276 F62D DAE8 E109 7596 9CE5 * tag 'audio-pull-request' of https://gitlab.com/marcandre.lureau/qemu: (27 commits) audio: remove sw->ratio audio/audio_template: substitute sw->hw with hw audio: handle leftover audio frame from upsampling audio: make recording packet length calculation exact audio: rename variables in audio_pcm_sw_read() audio: replace the resampling loop in audio_pcm_sw_read() audio: make playback packet length calculation exact audio: remove unused noop_conv() function audio: don't misuse audio_pcm_sw_write() audio: rename variables in audio_pcm_sw_write() audio: remove sw == NULL check audio: replace the resampling loop in audio_pcm_sw_write() audio: make the resampling code greedy audio: change type and name of the resample buffer audio: change type of mix_buf and conv_buf alsaaudio: reintroduce default recording settings alsaaudio: change default playback settings audio: remove audio_calloc() function audio/audio_template: use g_new0() to replace audio_calloc() audio/audio_template: use g_malloc0() to replace audio_calloc() ... Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
2023-03-06Merge tag 'pull-tcg-20230305' of https://gitlab.com/rth7680/qemu into stagingPeter Maydell
tcg: Merge two sequential labels accel/tcg: Retain prot flags from tlb_fill accel/tcg: Honor TLB_DISCARD_WRITE in atomic_mmu_lookup accel/tcg: Honor TLB_WATCHPOINTS in atomic_mmu_lookup target/sparc: Use tlb_set_page_full include/qemu/cpuid: Introduce xgetbv_low tcg/i386: Mark Win64 call-saved vector regs as reserved tcg: Decode the operand to INDEX_op_mb in dumps Portion of the target/ patchset which eliminates use of tcg_temp_free* Portion of the target/ patchset which eliminates use of tcg_const* # -----BEGIN PGP SIGNATURE----- # # iQFRBAABCgA7FiEEekgeeIaLTbaoWgXAZN846K9+IV8FAmQFNegdHHJpY2hhcmQu # aGVuZGVyc29uQGxpbmFyby5vcmcACgkQZN846K9+IV9WsQf+Ljs3WA5lvMPlpaSn # Li35ay/A1f2cU6FYspl81su4/c7Ft9Q8rkPF4K1n1rwuvqR91G25WTQIrw8NFPXZ # VU9GNGQc1qIVYO/hAH3fvgDmPxUF+tJDgT/BTNc1ldy6/v7QM3GWcEy8+O3H9S+K # uj6vIuWke0ukq6ZGmSAZnXEaJFq3HU26mcP4KxDxfIUcezMtDVp6QevqzVxM65aa # pUDh3qtsLGOxIYwthvu6avMQXORBhSB75awCuYH4QPJRpr3ahigcGsCr2gdVAQ8p # R7BbpUUdK5Huos971oouJrt5FwwbVgGEx78eF27sl0H8QMoNhsfyn6PcN8nPENLJ # MZYd+w== # =8goQ # -----END PGP SIGNATURE----- # gpg: Signature made Mon 06 Mar 2023 00:38:00 GMT # gpg: using RSA key 7A481E78868B4DB6A85A05C064DF38E8AF7E215F # gpg: issuer "richard.henderson@linaro.org" # gpg: Good signature from "Richard Henderson <richard.henderson@linaro.org>" [full] # Primary key fingerprint: 7A48 1E78 868B 4DB6 A85A 05C0 64DF 38E8 AF7E 215F * tag 'pull-tcg-20230305' of https://gitlab.com/rth7680/qemu: (84 commits) target/xtensa: Avoid tcg_const_i32 target/xtensa: Split constant in bit shift target/xtensa: Use tcg_gen_subfi_i32 in translate_sll target/xtensa: Avoid tcg_const_i32 in translate_l32r target/xtensa: Tidy translate_clamps target/xtensa: Tidy translate_bb target/sparc: Avoid tcg_const_{tl,i32} target/s390x: Split out gen_ri2 target/riscv: Avoid tcg_const_* target/microblaze: Avoid tcg_const_* throughout target/i386: Simplify POPF target/hexagon/idef-parser: Use gen_constant for gen_extend_tcg_width_op target/hexagon/idef-parser: Use gen_tmp for gen_rvalue_pred target/hexagon/idef-parser: Use gen_tmp for gen_pred_assign target/hexagon/idef-parser: Use gen_tmp for LPCFG target/hexagon: Use tcg_constant_* for gen_constant_from_imm docs/devel/tcg-ops: Drop recommendation to free temps tracing: remove transform.py include/exec/gen-icount: Drop tcg_temp_free in gen_tb_start target/tricore: Drop tcg_temp_free ... Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
2023-03-06audio: remove sw->ratioVolker Rümelin
Simplify the resample buffer size calculation. For audio playback we have sw->ratio = ((int64_t)sw->hw->info.freq << 32) / sw->info.freq; samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; This can be simplified to samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq); For audio recording we have sw->ratio = ((int64_t)sw->info.freq << 32) / sw->hw->info.freq; samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; This can be simplified to samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq); With hw = sw->hw this becomes in both cases samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq); Now that sw->ratio is no longer needed, remove sw->ratio. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-15-vr_qemu@t-online.de>
2023-03-06audio/audio_template: substitute sw->hw with hwVolker Rümelin
Substitute sw->hw with hw in the audio_pcm_sw_alloc_resources_* functions. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-14-vr_qemu@t-online.de>
2023-03-06audio: handle leftover audio frame from upsamplingVolker Rümelin
Upsampling may leave one remaining audio frame in the input buffer. The emulated audio playback devices are currently resposible to write this audio frame again in the next write cycle. Push that task down to audio_pcm_sw_write. This is another step towards an audio callback interface that guarantees that when audio frontends are told they can write n audio frames, they can actually do so. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-13-vr_qemu@t-online.de>
2023-03-06audio: make recording packet length calculation exactVolker Rümelin
Introduce the new function st_rate_frames_out() to calculate the exact number of audio output frames the resampling code can generate from a given number of audio input frames. When upsampling, this function returns the maximum number of output frames. This new function replaces the audio_frontend_frames_in() function, which calculated the average number of output frames rounded down to the nearest integer. The audio_frontend_frames_in() function was additionally used to limit the number of output frames to the resample buffer size. In audio_pcm_sw_read() the variable resample_buf.size replaces the open coded audio_frontend_frames_in() function. In audio_run_in() an additional MIN() function is necessary. After this patch the audio packet length calculation for audio recording is exact. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-12-vr_qemu@t-online.de>
2023-03-06audio: rename variables in audio_pcm_sw_read()Volker Rümelin
The audio_pcm_sw_read() function uses a few very unspecific variable names. Rename them for better readability. ret => total_out total => total_in size => buf_len samples => frames_out_max Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-11-vr_qemu@t-online.de>
2023-03-06audio: replace the resampling loop in audio_pcm_sw_read()Volker Rümelin
Replace the resampling loop in audio_pcm_sw_read() with the new function audio_pcm_sw_resample_in(). Unlike the old resample loop the new function will try to consume input frames even if the output buffer is full. This is necessary when downsampling to avoid reading less audio frames than calculated in advance. The loop was unrolled to avoid complicated loop control conditions in this case. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-10-vr_qemu@t-online.de>
2023-03-06audio: make playback packet length calculation exactVolker Rümelin
Introduce the new function st_rate_frames_in() to calculate the exact number of audio input frames needed to get a given number of audio output frames. The exact number of frames depends only on the difference of opos - ipos and the number of output frames. When downsampling, this function returns the maximum number of input frames needed. This new function replaces the audio_frontend_frames_out() function, which calculated the average number of input frames rounded down to the nearest integer. Because audio_frontend_frames_out() also limited the number of input frames to the size of the resample buffer, st_rate_frames_in() is not a direct replacement and two additional MIN() functions are needed. One to prevent resample buffer overflows and one to limit the available bytes for the audio frontends. After this patch the audio packet length calculation for playback is exact. When upsampling, it's still possible that the audio frontends can't write the last audio frame. This will be fixed later. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-9-vr_qemu@t-online.de>
2023-03-06audio: remove unused noop_conv() functionVolker Rümelin
The function audio_capture_mix_and_clear() no longer uses audio_pcm_sw_write() to resample audio frames from one internal buffer to another. For this reason, the noop_conv() function is now unused. Remove it. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-8-vr_qemu@t-online.de>
2023-03-06audio: don't misuse audio_pcm_sw_write()Volker Rümelin
The audio_pcm_sw_write() function is intended to convert a PCM audio stream to the internal representation, adjust the volume, and then mix it with the other audio streams with a possibly changed sample rate in mix_buf. In order for the audio_capture_mix_and_clear() function to use audio_pcm_sw_write(), it must bypass the first two tasks of audio_pcm_sw_write(). Since patch "audio: split out the resampling loop in audio_pcm_sw_write()" this is no longer necessary, because now the audio_pcm_sw_resample_out() function can be used instead of audio_pcm_sw_write(). Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-7-vr_qemu@t-online.de>
2023-03-06audio: rename variables in audio_pcm_sw_write()Volker Rümelin
The audio_pcm_sw_write() function uses a lot of very unspecific variable names. Rename them for better readability. ret => total_in total => total_out size => buf_len hwsamples => hw->mix_buf.size samples => frames_in_max Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-6-vr_qemu@t-online.de>
2023-03-06audio: remove sw == NULL checkVolker Rümelin
All call sites of audio_pcm_sw_write() guarantee that sw is not NULL. Remove the unnecessary NULL check. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-5-vr_qemu@t-online.de>
2023-03-06audio: replace the resampling loop in audio_pcm_sw_write()Volker Rümelin
Replace the resampling loop in audio_pcm_sw_write() with the new function audio_pcm_sw_resample_out(). Unlike the old resample loop the new function will try to consume input frames even if the output buffer is full. This is necessary when downsampling to avoid reading less audio frames than calculated in advance. The loop was unrolled to avoid complicated loop control conditions in this case. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-4-vr_qemu@t-online.de>
2023-03-06audio: make the resampling code greedyVolker Rümelin
Read the maximum possible number of audio frames instead of the minimum necessary number of frames when the audio stream is downsampled and the output buffer is limited. This makes the function symmetrical to upsampling when the input buffer is limited. The maximum possible number of frames is written here. With this change it's easier to calculate the exact number of audio frames the resample function will read or write. These two functions will be introduced later. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-3-vr_qemu@t-online.de>
2023-03-06audio: change type and name of the resample bufferVolker Rümelin
Change the type of the resample buffer from struct st_sample * to STSampleBuffer. Also change the name from buf to resample_buf for better readability. The new variables resample_buf.size and resample_buf.pos will be used after the next patches. There is no functional change. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-2-vr_qemu@t-online.de>
2023-03-06audio: change type of mix_buf and conv_bufVolker Rümelin
Change the type of mix_buf in struct HWVoiceOut and conv_buf in struct HWVoiceIn from STSampleBuffer * to STSampleBuffer. However, a buffer pointer is still needed. For this reason in struct STSampleBuffer samples[] is changed to *buffer. This is a preparation for the next patch. The next patch will add this line, which is not possible with the current struct STSampleBuffer definition. + sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2; There are no functional changes. Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20230224190555.7409-1-vr_qemu@t-online.de>
2023-03-06alsaaudio: reintroduce default recording settingsVolker Rümelin
Audio recording with ALSA default settings currently doesn't work. The debug log shows updates every 0.75s and 1.5s. audio: Elapsed since last alsa run (running): 0.743030 audio: Elapsed since last alsa run (running): 1.486048 audio: Elapsed since last alsa run (running): 0.743008 audio: Elapsed since last alsa run (running): 1.485878 audio: Elapsed since last alsa run (running): 1.486040 audio: Elapsed since last alsa run (running): 1.485886 The time between updates should be in the 10ms range. Audio recording with ALSA has the same timing contraints as playback. Reintroduce the default recording settings and use the same default settings for recording as for playback. The term "reintroduce" is correct because commit a93f328177 ("alsaaudio: port to -audiodev config") removed the default settings for recording. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-11-vr_qemu@t-online.de>
2023-03-06alsaaudio: change default playback settingsVolker Rümelin
The currently used default playback settings in the ALSA audio backend are a bit unfortunate. With a few emulated audio devices, audio playback does not work properly. Here is a short part of the debug log while audio is playing (elapsed time in seconds). audio: Elapsed since last alsa run (running): 0.046244 audio: Elapsed since last alsa run (running): 0.023137 audio: Elapsed since last alsa run (running): 0.023170 audio: Elapsed since last alsa run (running): 0.023650 audio: Elapsed since last alsa run (running): 0.060802 audio: Elapsed since last alsa run (running): 0.031931 For some audio devices the time of more than 23ms between updates is too long. Set the period time to 5.8ms so that the maximum time between two updates typically does not exceed 11ms. This roughly matches the 10ms period time when doing playback with the audio timer. After this patch the debug log looks like this. audio: Elapsed since last alsa run (running): 0.011919 audio: Elapsed since last alsa run (running): 0.005788 audio: Elapsed since last alsa run (running): 0.005995 audio: Elapsed since last alsa run (running): 0.011069 audio: Elapsed since last alsa run (running): 0.005901 audio: Elapsed since last alsa run (running): 0.006084 Acked-by: Christian Schoenebeck <qemu_oss@crudebyte.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-10-vr_qemu@t-online.de>
2023-03-06audio: remove audio_calloc() functionVolker Rümelin
Now that the last call site of audio_calloc() was removed, remove the unused audio_calloc() function. Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-9-vr_qemu@t-online.de>
2023-03-06audio/audio_template: use g_new0() to replace audio_calloc()Volker Rümelin
Replace audio_calloc() with the equivalent g_new0(). With a n_structs argument >= 1, g_new0() never returns NULL. Also remove the unnecessary NULL checks. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Daniel P. Berrangé <berrange@redhat.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-8-vr_qemu@t-online.de>
2023-03-06audio/audio_template: use g_malloc0() to replace audio_calloc()Volker Rümelin
Use g_malloc0() as a direct replacement for audio_calloc(). Since the type of the parameter n_bytes of the function g_malloc0() is unsigned, the type of the variables voice_size_out and voice_size_in has been changed to size_t. This means that the function argument no longer has to be checked for negative values. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Daniel P. Berrangé <berrange@redhat.com> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-7-vr_qemu@t-online.de>
2023-03-06audio/alsaaudio: use g_new0() instead of audio_calloc()Volker Rümelin
Replace audio_calloc() with the equivalent g_new0(). The value of the g_new0() argument count is >= 1, which means g_new0() will never return NULL. Also remove the unnecessary NULL check. Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Reviewed-by: Richard Henderson <richard.henderson@linaro.org> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-6-vr_qemu@t-online.de>
2023-03-06audio/mixeng: use g_new0() instead of audio_calloc()Volker Rümelin
Replace audio_calloc() with the equivalent g_new0(). With a n_structs argument of 1, g_new0() never returns NULL. Also remove the unnecessary NULL checks. Reviewed-by: Richard Henderson <richard.henderson@linaro.org> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-5-vr_qemu@t-online.de>
2023-03-06audio: remove unused #define AUDIO_STRINGIFYVolker Rümelin
Remove the unused #define AUDIO_STRINGIFY. It was last used before commit 470bcabd8f ("audio: Replace AUDIO_FUNC with __func__"). Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Reviewed-by: Thomas Huth <thuth@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-4-vr_qemu@t-online.de>
2023-03-06audio: rename hardware store to backendVolker Rümelin
Use a consistent friendly name for the HWVoiceOut and HWVoiceIn structures. Reviewed-by: Thomas Huth <thuth@redhat.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Message-Id: <20230121094735.11644-3-vr_qemu@t-online.de>
2023-03-06audio: don't show unnecessary error messagesVolker Rümelin
Let the audio_pcm_create_voice_pair_* functions handle error reporting. This avoids an additional error message in case the guest selected an unimplemented sample rate. Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-2-vr_qemu@t-online.de>
2023-03-06audio: log unimplemented audio device sample ratesVolker Rümelin
Some emulated audio devices allow guests to select very low sample rates that the audio subsystem doesn't support. The lowest supported sample rate depends on the audio backend used and in most cases can be changed with various -audiodev arguments. Until now, the audio_bug function emits an error message similar to the following error message A bug was just triggered in audio_calloc Save all your work and restart without audio I am sorry Context: audio_pcm_sw_alloc_resources_out passed invalid arguments to audio_calloc nmemb=0 size=16 (len=0) audio: Could not allocate buffer for `ac97.po' (0 samples) and the audio subsystem continues without sound for the affected device. The fact that the selected sample rate is not supported is not a guest error. Instead of displaying an error message, the missing audio support is now logged. Simply continuing without sound is correct, since the audio stream won't transport anything reasonable at such high resample ratios anyway. The AUD_open_* functions return NULL like before. The opened audio device will not be registered in the audio subsystem and consequently the audio frontend callback functions will not be called. The AUD_read and AUD_write functions return early in this case. This is necessary because, for example, the Sound Blaster 16 emulation calls AUD_write from the DMA callback function. Acked-by: Christian Schoenebeck <qemu_oss@crudebyte.com> Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> Message-Id: <20230121094735.11644-1-vr_qemu@t-online.de>
2023-03-06MAINTAINERS: add myself to ui/ and audio/Marc-André Lureau
Helping out with patch review & queue handling. Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com> Reviewed-by: Gerd Hoffmann <kraxel@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <philmd@linaro.org> Message-Id: <20230207085610.1033536-1-marcandre.lureau@redhat.com>
2023-03-05target/xtensa: Avoid tcg_const_i32Richard Henderson
All remaining uses are strictly read-only. Reviewed-by: Max Filippov <jcmvbkbc@gmail.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
2023-03-05target/xtensa: Split constant in bit shiftRichard Henderson
Reviewed-by: Max Filippov <jcmvbkbc@gmail.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
2023-03-05target/xtensa: Use tcg_gen_subfi_i32 in translate_sllRichard Henderson
Reviewed-by: Max Filippov <jcmvbkbc@gmail.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
generated by cgit v1.2.3 (git 2.26.2) at 2025-03-01 20:01:31 +0000