* 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>
Peter Maydell [Mon, 6 Mar 2023 10:20:04 +0000 (10:20 +0000)]
Merge tag 'pull-tcg-20230305' of https://gitlab.com/rth7680/qemu into staging
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*
* 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>
Volker Rümelin [Fri, 24 Feb 2023 19:05:53 +0000 (20:05 +0100)]
audio: handle leftover audio frame from upsampling
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.
Volker Rümelin [Fri, 24 Feb 2023 19:05:52 +0000 (20:05 +0100)]
audio: make recording packet length calculation exact
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>
Volker Rümelin [Fri, 24 Feb 2023 19:05:50 +0000 (20:05 +0100)]
audio: replace the resampling loop in audio_pcm_sw_read()
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.
Volker Rümelin [Fri, 24 Feb 2023 19:05:49 +0000 (20:05 +0100)]
audio: make playback packet length calculation exact
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>
Volker Rümelin [Fri, 24 Feb 2023 19:05:48 +0000 (20:05 +0100)]
audio: remove unused noop_conv() function
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.
Volker Rümelin [Fri, 24 Feb 2023 19:05:47 +0000 (20:05 +0100)]
audio: don't misuse audio_pcm_sw_write()
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().
Volker Rümelin [Fri, 24 Feb 2023 19:05:44 +0000 (20:05 +0100)]
audio: replace the resampling loop in audio_pcm_sw_write()
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.
Volker Rümelin [Fri, 24 Feb 2023 19:05:43 +0000 (20:05 +0100)]
audio: make the resampling code greedy
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.
Volker Rümelin [Fri, 24 Feb 2023 19:05:42 +0000 (20:05 +0100)]
audio: change type and name of the resample buffer
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.
Volker Rümelin [Fri, 24 Feb 2023 19:05:41 +0000 (20:05 +0100)]
audio: change type of mix_buf and conv_buf
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.
Volker Rümelin [Sat, 21 Jan 2023 09:47:35 +0000 (10:47 +0100)]
alsaaudio: reintroduce default recording settings
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.
Volker Rümelin [Sat, 21 Jan 2023 09:47:34 +0000 (10:47 +0100)]
alsaaudio: change default playback settings
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
Volker Rümelin [Sat, 21 Jan 2023 09:47:31 +0000 (10:47 +0100)]
audio/audio_template: use g_malloc0() to replace audio_calloc()
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>
Volker Rümelin [Sat, 21 Jan 2023 09:47:26 +0000 (10:47 +0100)]
audio: don't show unnecessary error messages
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.
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.
target/hexagon/idef-parser: Use gen_constant for gen_extend_tcg_width_op
We already have a temporary, res, which we can use for the intermediate
shift result. Simplify the constant to -1 instead of 0xf*f.
This was the last use of gen_tmp_value, so remove it.
Reviewed-by: Taylor Simpson <tsimpson@quicinc.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
target/hexagon/idef-parser: Use gen_tmp for gen_rvalue_pred
The allocation is immediately followed by either tcg_gen_mov_i32
or gen_read_preg (which contains tcg_gen_mov_i32), so the zero
initialization is immediately discarded.
Reviewed-by: Taylor Simpson <tsimpson@quicinc.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
The GET_USR_FIELD macro initializes the output, so the initial assignment
of zero is discarded. This is the only use of get_tmp_value outside of
parser-helper.c, so make it static.
Reviewed-by: Taylor Simpson <tsimpson@quicinc.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
This file, and a couple of uses, got left behind when the
tcg stuff was removed from tracetool.
Fixes: 126d4123c50a ("tracing: excise the tcg related from tracetool") Reviewed-by: Alex Bennée <alex.bennee@linaro.org> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries.
Reviewed-by: Peter Maydell <peter.maydell@linaro.org> Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries.
Remove the g1 and g2 members of DisasCompare, as they were
used to track which temps needed to be freed.
Reviewed-by: Peter Maydell <peter.maydell@linaro.org> Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
target/sparc: Remove egress label in disas_sparc_context
Reviewed-by: Peter Maydell <peter.maydell@linaro.org> Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries,
therefore there's no need to record temps for later freeing.
Replace the few uses with tcg_temp_new_i32.
Reviewed-by: Peter Maydell <peter.maydell@linaro.org> Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries,
therefore there's no need to record temps for later freeing.
Replace the few uses with tcg_temp_new.
Reviewed-by: Peter Maydell <peter.maydell@linaro.org> Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries.
Reviewed-by: Weiwei Li <liweiwei@iscas.ac.cn> Reviewed-by: Daniel Henrique Barboza <dbarboza@ventanamicro.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries,
therefore there's no need to record temps for later freeing.
Replace the few uses with tcg_temp_new.
Reviewed-by: Weiwei Li <liweiwei@iscas.ac.cn> Reviewed-by: Daniel Henrique Barboza <dbarboza@ventanamicro.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries,
therefore there's no need to record temps for later freeing.
Replace the few uses with tcg_temp_new_i64.
Reviewed-by: Weiwei Li <liweiwei@iscas.ac.cn> Reviewed-by: Daniel Henrique Barboza <dbarboza@ventanamicro.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries.
Remove the g1 and g2 members of DisasCompare, as they were
used to track which temps needed to be freed.
Reviewed-by: Peter Maydell <peter.maydell@linaro.org> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries,
therefore there's no need to record temps for later freeing.
Replace the few uses with tcg_temp_new.
Reviewed-by: Song Gao <gaosong@loongson.cn> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Translators are no longer required to free tcg temporaries.
This removes gen_rvalue_free, gen_rvalue_free_manual and
free_variables, whose only purpose was to emit tcg_temp_free.
Reviewed-by: Taylor Simpson <tsimpson@quicinc.com> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
Only the use within cpu_reg requires a writable temp,
so inline new_tmp_a64_zero there. All other uses are
fine with a constant temp, so use tcg_constant_i64(0).
Reviewed-by: Peter Maydell <peter.maydell@linaro.org> Signed-off-by: Richard Henderson <richard.henderson@linaro.org>
projects / qemu-xen.git / log