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author | Peter Maydell <peter.maydell@linaro.org> | 2019-09-24 13:51:51 +0100 |
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committer | Peter Maydell <peter.maydell@linaro.org> | 2019-09-24 13:51:51 +0100 |
commit | 860d9048c78ce59c5903c3d5209df56f38400986 (patch) | |
tree | bf8e0e0b752cdeee65e762c6a7d1800813b5310b /audio/sdlaudio.c | |
parent | 8dc57281b80f1de11a7c114a0acb6f192b3e1a9a (diff) | |
parent | 571a8c522e0095239598347ac0add93337c1e0bf (diff) | |
download | qemu-860d9048c78ce59c5903c3d5209df56f38400986.zip qemu-860d9048c78ce59c5903c3d5209df56f38400986.tar.gz qemu-860d9048c78ce59c5903c3d5209df56f38400986.tar.bz2 |
Merge remote-tracking branch 'remotes/kraxel/tags/audio-20190924-pull-request' into staging
audio: documentation fixes.
audio: new backend api (first part of the surround sound patch series).
# gpg: Signature made Tue 24 Sep 2019 07:19:31 BST
# gpg: using RSA key 4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>" [full]
# gpg: aka "Gerd Hoffmann <gerd@kraxel.org>" [full]
# gpg: aka "Gerd Hoffmann (private) <kraxel@gmail.com>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901 FE7D 4CB6 D8EE D3E8 7138
* remotes/kraxel/tags/audio-20190924-pull-request:
audio: split ctl_* functions into enable_* and volume_*
audio: common rate control code for timer based outputs
audio: unify input and output mixeng buffer management
audio: remove remains of the old backend api
wavaudio: port to the new audio backend api
spiceaudio: port to the new audio backend api
sdlaudio: port to the new audio backend api
paaudio: port to the new audio backend api
ossaudio: port to the new audio backend api
noaudio: port to the new audio backend api
dsoundaudio: port to the new audio backend api
coreaudio: port to the new audio backend api
alsaaudio: port to the new audio backend api
audio: api for mixeng code free backends
audio: fix ALSA period-length typo in documentation
audio: fix buffer-length typo in documentation
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Diffstat (limited to 'audio/sdlaudio.c')
-rw-r--r-- | audio/sdlaudio.c | 104 |
1 files changed, 45 insertions, 59 deletions
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c index 14b11f0..5c6bcfc 100644 --- a/audio/sdlaudio.c +++ b/audio/sdlaudio.c @@ -41,8 +41,6 @@ typedef struct SDLVoiceOut { HWVoiceOut hw; - size_t live; - size_t decr; } SDLVoiceOut; static struct SDLAudioState { @@ -184,62 +182,59 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len) SDLVoiceOut *sdl = opaque; SDLAudioState *s = &glob_sdl; HWVoiceOut *hw = &sdl->hw; - size_t samples = len >> hw->info.shift; - size_t to_mix, decr; - if (s->exit || !sdl->live) { + if (s->exit) { return; } /* dolog ("in callback samples=%zu live=%zu\n", samples, sdl->live); */ - to_mix = MIN(samples, sdl->live); - decr = to_mix; - while (to_mix) { - size_t chunk = MIN(to_mix, hw->samples - hw->rpos); - struct st_sample *src = hw->mix_buf + hw->rpos; - - /* dolog ("in callback to_mix %zu, chunk %zu\n", to_mix, chunk); */ - hw->clip(buf, src, chunk); - hw->rpos = (hw->rpos + chunk) % hw->samples; - to_mix -= chunk; - buf += chunk << hw->info.shift; + while (hw->pending_emul && len) { + size_t write_len; + ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul; + if (start < 0) { + start += hw->size_emul; + } + assert(start >= 0 && start < hw->size_emul); + + write_len = MIN(MIN(hw->pending_emul, len), + hw->size_emul - start); + + memcpy(buf, hw->buf_emul + start, write_len); + hw->pending_emul -= write_len; + len -= write_len; + buf += write_len; } - samples -= decr; - sdl->live -= decr; - sdl->decr += decr; - /* dolog ("done len=%zu\n", len); */ - - /* SDL2 does not clear the remaining buffer for us, so do it on our own */ - if (samples) { - memset(buf, 0, samples << hw->info.shift); + /* clear remaining buffer that we couldn't fill with data */ + if (len) { + memset(buf, 0, len); } } -static size_t sdl_run_out(HWVoiceOut *hw, size_t live) -{ - size_t decr; - SDLVoiceOut *sdl = (SDLVoiceOut *) hw; - - SDL_LockAudio(); - - if (sdl->decr > live) { - ldebug ("sdl->decr %d live %d sdl->live %d\n", - sdl->decr, - live, - sdl->live); +#define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args, fail, unlock) \ + static ret_type glue(sdl_, name)args_decl \ + { \ + ret_type ret; \ + \ + SDL_LockAudio(); \ + \ + ret = glue(audio_generic_, name)args; \ + \ + SDL_UnlockAudio(); \ + return ret; \ } - decr = MIN (sdl->decr, live); - sdl->decr -= decr; - - sdl->live = live; +SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size), + (hw, size), *size = 0, sdl_unlock) +SDL_WRAPPER_FUNC(put_buffer_out_nowrite, size_t, + (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size), + /*nothing*/, sdl_unlock_and_post) +SDL_WRAPPER_FUNC(write, size_t, + (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size), + /*nothing*/, sdl_unlock_and_post) - SDL_UnlockAudio(); - - return decr; -} +#undef SDL_WRAPPER_FUNC static void sdl_fini_out (HWVoiceOut *hw) { @@ -290,20 +285,9 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as, return 0; } -static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...) +static void sdl_enable_out(HWVoiceOut *hw, bool enable) { - (void) hw; - - switch (cmd) { - case VOICE_ENABLE: - SDL_PauseAudio (0); - break; - - case VOICE_DISABLE: - SDL_PauseAudio (1); - break; - } - return 0; + SDL_PauseAudio(!enable); } static void *sdl_audio_init(Audiodev *dev) @@ -336,8 +320,10 @@ static void sdl_audio_fini (void *opaque) static struct audio_pcm_ops sdl_pcm_ops = { .init_out = sdl_init_out, .fini_out = sdl_fini_out, - .run_out = sdl_run_out, - .ctl_out = sdl_ctl_out, + .write = sdl_write, + .get_buffer_out = sdl_get_buffer_out, + .put_buffer_out = sdl_put_buffer_out_nowrite, + .enable_out = sdl_enable_out, }; static struct audio_driver sdl_audio_driver = { |