From: Kővágó, Zoltán <dirty.ice.hu@gmail.com> I had to include an enum for audio sampling formats into qapi, but that meant duplicating the audfmt_e enum. This patch replaces audfmt_e and associated values with the qapi generated AudioFormat enum. This patch is mostly a search-and-replace, except for switches where the qapi generated AUDIO_FORMAT_MAX caused problems. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Reviewed-by: Thomas Huth <thuth@redhat.com> Message-id: 01251b2758a1679c66842120b77c0fb46d7d0eaf.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> --- audio/audio.h | 12 +---- audio/alsaaudio.c | 53 +++++++++++---------- audio/audio.c | 97 +++++++++++++++++++++------------------ audio/audio_win_int.c | 18 ++++---- audio/ossaudio.c | 30 ++++++------ audio/paaudio.c | 28 +++++------ audio/sdlaudio.c | 26 +++++------ audio/spiceaudio.c | 4 +- audio/wavaudio.c | 17 ++++--- audio/wavcapture.c | 2 +- hw/arm/omap2.c | 2 +- hw/audio/ac97.c | 2 +- hw/audio/adlib.c | 2 +- hw/audio/cs4231a.c | 6 +-- hw/audio/es1370.c | 4 +- hw/audio/gus.c | 2 +- hw/audio/hda-codec.c | 18 ++++---- hw/audio/lm4549.c | 6 +-- hw/audio/milkymist-ac97.c | 2 +- hw/audio/pcspk.c | 2 +- hw/audio/sb16.c | 14 +++--- hw/audio/wm8750.c | 6 +-- hw/display/xlnx_dp.c | 2 +- hw/input/tsc210x.c | 2 +- hw/usb/dev-audio.c | 2 +- ui/vnc.c | 26 +++++------ 26 files changed, 196 insertions(+), 189 deletions(-) diff --git a/audio/audio.h b/audio/audio.h index f4339a185e4e..02f29a3b3ef7 100644 --- a/audio/audio.h +++ b/audio/audio.h @@ -26,18 +26,10 @@ #define QEMU_AUDIO_H #include "qemu/queue.h" +#include "qapi/qapi-types-audio.h" typedef void (*audio_callback_fn) (void *opaque, int avail); -typedef enum { - AUD_FMT_U8, - AUD_FMT_S8, - AUD_FMT_U16, - AUD_FMT_S16, - AUD_FMT_U32, - AUD_FMT_S32 -} audfmt_e; - #ifdef HOST_WORDS_BIGENDIAN #define AUDIO_HOST_ENDIANNESS 1 #else @@ -47,7 +39,7 @@ typedef enum { struct audsettings { int freq; int nchannels; - audfmt_e fmt; + AudioFormat fmt; int endianness; }; diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 635be73bf495..5bd034267fce 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -87,7 +87,7 @@ struct alsa_params_req { struct alsa_params_obt { int freq; - audfmt_e fmt; + AudioFormat fmt; int endianness; int nchannels; snd_pcm_uframes_t samples; @@ -294,16 +294,16 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len) return audio_pcm_sw_write (sw, buf, len); } -static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness) +static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness) { switch (fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: return SND_PCM_FORMAT_S8; - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: return SND_PCM_FORMAT_U8; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: if (endianness) { return SND_PCM_FORMAT_S16_BE; } @@ -311,7 +311,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness) return SND_PCM_FORMAT_S16_LE; } - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: if (endianness) { return SND_PCM_FORMAT_U16_BE; } @@ -319,7 +319,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness) return SND_PCM_FORMAT_U16_LE; } - case AUD_FMT_S32: + case AUDIO_FORMAT_S32: if (endianness) { return SND_PCM_FORMAT_S32_BE; } @@ -327,7 +327,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness) return SND_PCM_FORMAT_S32_LE; } - case AUD_FMT_U32: + case AUDIO_FORMAT_U32: if (endianness) { return SND_PCM_FORMAT_U32_BE; } @@ -344,58 +344,58 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness) } } -static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, +static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt, int *endianness) { switch (alsafmt) { case SND_PCM_FORMAT_S8: *endianness = 0; - *fmt = AUD_FMT_S8; + *fmt = AUDIO_FORMAT_S8; break; case SND_PCM_FORMAT_U8: *endianness = 0; - *fmt = AUD_FMT_U8; + *fmt = AUDIO_FORMAT_U8; break; case SND_PCM_FORMAT_S16_LE: *endianness = 0; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case SND_PCM_FORMAT_U16_LE: *endianness = 0; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; case SND_PCM_FORMAT_S16_BE: *endianness = 1; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case SND_PCM_FORMAT_U16_BE: *endianness = 1; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; case SND_PCM_FORMAT_S32_LE: *endianness = 0; - *fmt = AUD_FMT_S32; + *fmt = AUDIO_FORMAT_S32; break; case SND_PCM_FORMAT_U32_LE: *endianness = 0; - *fmt = AUD_FMT_U32; + *fmt = AUDIO_FORMAT_U32; break; case SND_PCM_FORMAT_S32_BE: *endianness = 1; - *fmt = AUD_FMT_S32; + *fmt = AUDIO_FORMAT_S32; break; case SND_PCM_FORMAT_U32_BE: *endianness = 1; - *fmt = AUD_FMT_U32; + *fmt = AUDIO_FORMAT_U32; break; default: @@ -638,19 +638,22 @@ static int alsa_open (int in, struct alsa_params_req *req, bytes_per_sec = freq << (nchannels == 2); switch (obt->fmt) { - case AUD_FMT_S8: - case AUD_FMT_U8: + case AUDIO_FORMAT_S8: + case AUDIO_FORMAT_U8: break; - case AUD_FMT_S16: - case AUD_FMT_U16: + case AUDIO_FORMAT_S16: + case AUDIO_FORMAT_U16: bytes_per_sec <<= 1; break; - case AUD_FMT_S32: - case AUD_FMT_U32: + case AUDIO_FORMAT_S32: + case AUDIO_FORMAT_U32: bytes_per_sec <<= 2; break; + + default: + abort(); } threshold = (conf->threshold * bytes_per_sec) / 1000; diff --git a/audio/audio.c b/audio/audio.c index 909c817103c8..77216e50102e 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -113,7 +113,7 @@ static struct { .settings = { .freq = 44100, .nchannels = 2, - .fmt = AUD_FMT_S16, + .fmt = AUDIO_FORMAT_S16, .endianness = AUDIO_HOST_ENDIANNESS, } }, @@ -125,7 +125,7 @@ static struct { .settings = { .freq = 44100, .nchannels = 2, - .fmt = AUD_FMT_S16, + .fmt = AUDIO_FORMAT_S16, .endianness = AUDIO_HOST_ENDIANNESS, } }, @@ -257,58 +257,61 @@ static char *audio_alloc_prefix (const char *s) return r; } -static const char *audio_audfmt_to_string (audfmt_e fmt) +static const char *audio_audfmt_to_string (AudioFormat fmt) { switch (fmt) { - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: return "U8"; - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: return "U16"; - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: return "S8"; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: return "S16"; - case AUD_FMT_U32: + case AUDIO_FORMAT_U32: return "U32"; - case AUD_FMT_S32: + case AUDIO_FORMAT_S32: return "S32"; + + default: + abort(); } dolog ("Bogus audfmt %d returning S16\n", fmt); return "S16"; } -static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, +static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval, int *defaultp) { if (!strcasecmp (s, "u8")) { *defaultp = 0; - return AUD_FMT_U8; + return AUDIO_FORMAT_U8; } else if (!strcasecmp (s, "u16")) { *defaultp = 0; - return AUD_FMT_U16; + return AUDIO_FORMAT_U16; } else if (!strcasecmp (s, "u32")) { *defaultp = 0; - return AUD_FMT_U32; + return AUDIO_FORMAT_U32; } else if (!strcasecmp (s, "s8")) { *defaultp = 0; - return AUD_FMT_S8; + return AUDIO_FORMAT_S8; } else if (!strcasecmp (s, "s16")) { *defaultp = 0; - return AUD_FMT_S16; + return AUDIO_FORMAT_S16; } else if (!strcasecmp (s, "s32")) { *defaultp = 0; - return AUD_FMT_S32; + return AUDIO_FORMAT_S32; } else { dolog ("Bogus audio format `%s' using %s\n", @@ -318,8 +321,8 @@ static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, } } -static audfmt_e audio_get_conf_fmt (const char *envname, - audfmt_e defval, +static AudioFormat audio_get_conf_fmt (const char *envname, + AudioFormat defval, int *defaultp) { const char *var = getenv (envname); @@ -421,7 +424,7 @@ static void audio_print_options (const char *prefix, case AUD_OPT_FMT: { - audfmt_e *fmtp = opt->valp; + AudioFormat *fmtp = opt->valp; printf ( "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n", state, @@ -492,7 +495,7 @@ static void audio_process_options (const char *prefix, case AUD_OPT_FMT: { - audfmt_e *fmtp = opt->valp; + AudioFormat *fmtp = opt->valp; *fmtp = audio_get_conf_fmt (optname, *fmtp, &def); } break; @@ -524,22 +527,22 @@ static void audio_print_settings (struct audsettings *as) dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels); switch (as->fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: AUD_log (NULL, "S8"); break; - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: AUD_log (NULL, "U8"); break; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: AUD_log (NULL, "S16"); break; - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: AUD_log (NULL, "U16"); break; - case AUD_FMT_S32: + case AUDIO_FORMAT_S32: AUD_log (NULL, "S32"); break; - case AUD_FMT_U32: + case AUDIO_FORMAT_U32: AUD_log (NULL, "U32"); break; default: @@ -570,12 +573,12 @@ static int audio_validate_settings (struct audsettings *as) invalid |= as->endianness != 0 && as->endianness != 1; switch (as->fmt) { - case AUD_FMT_S8: - case AUD_FMT_U8: - case AUD_FMT_S16: - case AUD_FMT_U16: - case AUD_FMT_S32: - case AUD_FMT_U32: + case AUDIO_FORMAT_S8: + case AUDIO_FORMAT_U8: + case AUDIO_FORMAT_S16: + case AUDIO_FORMAT_U16: + case AUDIO_FORMAT_S32: + case AUDIO_FORMAT_U32: break; default: invalid = 1; @@ -591,25 +594,28 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a int bits = 8, sign = 0; switch (as->fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: sign = 1; /* fall through */ - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: break; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: sign = 1; /* fall through */ - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: bits = 16; break; - case AUD_FMT_S32: + case AUDIO_FORMAT_S32: sign = 1; /* fall through */ - case AUD_FMT_U32: + case AUDIO_FORMAT_U32: bits = 32; break; + + default: + abort(); } return info->freq == as->freq && info->nchannels == as->nchannels @@ -623,24 +629,27 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) int bits = 8, sign = 0, shift = 0; switch (as->fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: sign = 1; - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: break; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: sign = 1; - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: bits = 16; shift = 1; break; - case AUD_FMT_S32: + case AUDIO_FORMAT_S32: sign = 1; - case AUD_FMT_U32: + case AUDIO_FORMAT_U32: bits = 32; shift = 2; break; + + default: + abort(); } info->freq = as->freq; diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c index 6900008d0c47..b938fd667bae 100644 --- a/audio/audio_win_int.c +++ b/audio/audio_win_int.c @@ -24,20 +24,20 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx, wfx->cbSize = 0; switch (as->fmt) { - case AUD_FMT_S8: - case AUD_FMT_U8: + case AUDIO_FORMAT_S8: + case AUDIO_FORMAT_U8: wfx->wBitsPerSample = 8; break; - case AUD_FMT_S16: - case AUD_FMT_U16: + case AUDIO_FORMAT_S16: + case AUDIO_FORMAT_U16: wfx->wBitsPerSample = 16; wfx->nAvgBytesPerSec <<= 1; wfx->nBlockAlign <<= 1; break; - case AUD_FMT_S32: - case AUD_FMT_U32: + case AUDIO_FORMAT_S32: + case AUDIO_FORMAT_U32: wfx->wBitsPerSample = 32; wfx->nAvgBytesPerSec <<= 2; wfx->nBlockAlign <<= 2; @@ -85,15 +85,15 @@ int waveformat_to_audio_settings (WAVEFORMATEX *wfx, switch (wfx->wBitsPerSample) { case 8: - as->fmt = AUD_FMT_U8; + as->fmt = AUDIO_FORMAT_U8; break; case 16: - as->fmt = AUD_FMT_S16; + as->fmt = AUDIO_FORMAT_S16; break; case 32: - as->fmt = AUD_FMT_S32; + as->fmt = AUDIO_FORMAT_S32; break; default: diff --git a/audio/ossaudio.c b/audio/ossaudio.c index 6c69622b4c7e..355e8fbda519 100644 --- a/audio/ossaudio.c +++ b/audio/ossaudio.c @@ -70,7 +70,7 @@ typedef struct OSSVoiceIn { struct oss_params { int freq; - audfmt_e fmt; + AudioFormat fmt; int nchannels; int nfrags; int fragsize; @@ -148,16 +148,16 @@ static int oss_write (SWVoiceOut *sw, void *buf, int len) return audio_pcm_sw_write (sw, buf, len); } -static int aud_to_ossfmt (audfmt_e fmt, int endianness) +static int aud_to_ossfmt (AudioFormat fmt, int endianness) { switch (fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: return AFMT_S8; - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: return AFMT_U8; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: if (endianness) { return AFMT_S16_BE; } @@ -165,7 +165,7 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness) return AFMT_S16_LE; } - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: if (endianness) { return AFMT_U16_BE; } @@ -182,37 +182,37 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness) } } -static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness) +static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness) { switch (ossfmt) { case AFMT_S8: *endianness = 0; - *fmt = AUD_FMT_S8; + *fmt = AUDIO_FORMAT_S8; break; case AFMT_U8: *endianness = 0; - *fmt = AUD_FMT_U8; + *fmt = AUDIO_FORMAT_U8; break; case AFMT_S16_LE: *endianness = 0; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case AFMT_U16_LE: *endianness = 0; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; case AFMT_S16_BE: *endianness = 1; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case AFMT_U16_BE: *endianness = 1; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; default: @@ -500,7 +500,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as, int endianness; int err; int fd; - audfmt_e effective_fmt; + AudioFormat effective_fmt; struct audsettings obt_as; OSSConf *conf = drv_opaque; @@ -667,7 +667,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) int endianness; int err; int fd; - audfmt_e effective_fmt; + AudioFormat effective_fmt; struct audsettings obt_as; OSSConf *conf = drv_opaque; diff --git a/audio/paaudio.c b/audio/paaudio.c index 6153b908dae1..8246f260a8ac 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -385,21 +385,21 @@ static int qpa_read (SWVoiceIn *sw, void *buf, int len) return audio_pcm_sw_read (sw, buf, len); } -static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness) +static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness) { int format; switch (afmt) { - case AUD_FMT_S8: - case AUD_FMT_U8: + case AUDIO_FORMAT_S8: + case AUDIO_FORMAT_U8: format = PA_SAMPLE_U8; break; - case AUD_FMT_S16: - case AUD_FMT_U16: + case AUDIO_FORMAT_S16: + case AUDIO_FORMAT_U16: format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE; break; - case AUD_FMT_S32: - case AUD_FMT_U32: + case AUDIO_FORMAT_S32: + case AUDIO_FORMAT_U32: format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE; break; default: @@ -410,26 +410,26 @@ static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness) return format; } -static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness) +static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness) { switch (fmt) { case PA_SAMPLE_U8: - return AUD_FMT_U8; + return AUDIO_FORMAT_U8; case PA_SAMPLE_S16BE: *endianness = 1; - return AUD_FMT_S16; + return AUDIO_FORMAT_S16; case PA_SAMPLE_S16LE: *endianness = 0; - return AUD_FMT_S16; + return AUDIO_FORMAT_S16; case PA_SAMPLE_S32BE: *endianness = 1; - return AUD_FMT_S32; + return AUDIO_FORMAT_S32; case PA_SAMPLE_S32LE: *endianness = 0; - return AUD_FMT_S32; + return AUDIO_FORMAT_S32; default: dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt); - return AUD_FMT_U8; + return AUDIO_FORMAT_U8; } } diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c index f7ee70b15347..4cd4cbaf0043 100644 --- a/audio/sdlaudio.c +++ b/audio/sdlaudio.c @@ -68,19 +68,19 @@ static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...) AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ()); } -static int aud_to_sdlfmt (audfmt_e fmt) +static int aud_to_sdlfmt (AudioFormat fmt) { switch (fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: return AUDIO_S8; - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: return AUDIO_U8; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: return AUDIO_S16LSB; - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: return AUDIO_U16LSB; default: @@ -92,37 +92,37 @@ static int aud_to_sdlfmt (audfmt_e fmt) } } -static int sdl_to_audfmt(int sdlfmt, audfmt_e *fmt, int *endianness) +static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness) { switch (sdlfmt) { case AUDIO_S8: *endianness = 0; - *fmt = AUD_FMT_S8; + *fmt = AUDIO_FORMAT_S8; break; case AUDIO_U8: *endianness = 0; - *fmt = AUD_FMT_U8; + *fmt = AUDIO_FORMAT_U8; break; case AUDIO_S16LSB: *endianness = 0; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case AUDIO_U16LSB: *endianness = 0; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; case AUDIO_S16MSB: *endianness = 1; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case AUDIO_U16MSB: *endianness = 1; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; default: @@ -265,7 +265,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as, SDL_AudioSpec req, obt; int endianness; int err; - audfmt_e effective_fmt; + AudioFormat effective_fmt; struct audsettings obt_as; req.freq = as->freq; diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c index 6ad0eafbc67f..3aeb0cb35722 100644 --- a/audio/spiceaudio.c +++ b/audio/spiceaudio.c @@ -130,7 +130,7 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as, settings.freq = SPICE_INTERFACE_PLAYBACK_FREQ; #endif settings.nchannels = SPICE_INTERFACE_PLAYBACK_CHAN; - settings.fmt = AUD_FMT_S16; + settings.fmt = AUDIO_FORMAT_S16; settings.endianness = AUDIO_HOST_ENDIANNESS; audio_pcm_init_info (&hw->info, &settings); @@ -258,7 +258,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) settings.freq = SPICE_INTERFACE_RECORD_FREQ; #endif settings.nchannels = SPICE_INTERFACE_RECORD_CHAN; - settings.fmt = AUD_FMT_S16; + settings.fmt = AUDIO_FORMAT_S16; settings.endianness = AUDIO_HOST_ENDIANNESS; audio_pcm_init_info (&hw->info, &settings); diff --git a/audio/wavaudio.c b/audio/wavaudio.c index 40adfa30c33a..35a614785ece 100644 --- a/audio/wavaudio.c +++ b/audio/wavaudio.c @@ -117,20 +117,23 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as, stereo = wav_as.nchannels == 2; switch (wav_as.fmt) { - case AUD_FMT_S8: - case AUD_FMT_U8: + case AUDIO_FORMAT_S8: + case AUDIO_FORMAT_U8: bits16 = 0; break; - case AUD_FMT_S16: - case AUD_FMT_U16: + case AUDIO_FORMAT_S16: + case AUDIO_FORMAT_U16: bits16 = 1; break; - case AUD_FMT_S32: - case AUD_FMT_U32: + case AUDIO_FORMAT_S32: + case AUDIO_FORMAT_U32: dolog ("WAVE files can not handle 32bit formats\n"); return -1; + + default: + abort(); } hdr[34] = bits16 ? 0x10 : 0x08; @@ -225,7 +228,7 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...) static WAVConf glob_conf = { .settings.freq = 44100, .settings.nchannels = 2, - .settings.fmt = AUD_FMT_S16, + .settings.fmt = AUDIO_FORMAT_S16, .wav_path = "qemu.wav" }; diff --git a/audio/wavcapture.c b/audio/wavcapture.c index cd24570aa70c..74320dfecc76 100644 --- a/audio/wavcapture.c +++ b/audio/wavcapture.c @@ -136,7 +136,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq, as.freq = freq; as.nchannels = 1 << stereo; - as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8; + as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8; as.endianness = 0; ops.notify = wav_notify; diff --git a/hw/arm/omap2.c b/hw/arm/omap2.c index 94dffb2f577c..446223906e47 100644 --- a/hw/arm/omap2.c +++ b/hw/arm/omap2.c @@ -273,7 +273,7 @@ static void omap_eac_format_update(struct omap_eac_s *s) * does I2S specify it? */ /* All register writes are 16 bits so we we store 16-bit samples * in the buffers regardless of AGCFR[B8_16] value. */ - fmt.fmt = AUD_FMT_U16; + fmt.fmt = AUDIO_FORMAT_U16; s->codec.in_voice = AUD_open_in(&s->codec.card, s->codec.in_voice, "eac.codec.in", s, omap_eac_in_cb, &fmt); diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c index d799533aa991..2265622d44a2 100644 --- a/hw/audio/ac97.c +++ b/hw/audio/ac97.c @@ -365,7 +365,7 @@ static void open_voice (AC97LinkState *s, int index, int freq) as.freq = freq; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; if (freq > 0) { diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c index 97b876c7e08c..0957780a3dc0 100644 --- a/hw/audio/adlib.c +++ b/hw/audio/adlib.c @@ -269,7 +269,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp) as.freq = s->freq; as.nchannels = SHIFT; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = AUDIO_HOST_ENDIANNESS; AUD_register_card ("adlib", &s->card); diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c index 9089dcb47e84..62da75eefebb 100644 --- a/hw/audio/cs4231a.c +++ b/hw/audio/cs4231a.c @@ -288,7 +288,7 @@ static void cs_reset_voices (CSState *s, uint32_t val) switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) { case 0: - as.fmt = AUD_FMT_U8; + as.fmt = AUDIO_FORMAT_U8; s->shift = as.nchannels == 2; break; @@ -298,7 +298,7 @@ static void cs_reset_voices (CSState *s, uint32_t val) case 3: s->tab = ALawDecompressTable; x_law: - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = AUDIO_HOST_ENDIANNESS; s->shift = as.nchannels == 2; break; @@ -307,7 +307,7 @@ static void cs_reset_voices (CSState *s, uint32_t val) as.endianness = 1; /* fall through */ case 2: - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; s->shift = as.nchannels; break; diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c index 97789a077196..a5314d66fdc2 100644 --- a/hw/audio/es1370.c +++ b/hw/audio/es1370.c @@ -414,14 +414,14 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl) i, new_freq, 1 << (new_fmt & 1), - (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8, + (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8, d->shift); if (new_freq) { struct audsettings as; as.freq = new_freq; as.nchannels = 1 << (new_fmt & 1); - as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8; + as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8; as.endianness = 0; if (i == ADC_CHANNEL) { diff --git a/hw/audio/gus.c b/hw/audio/gus.c index 8e0b27e0f237..b3e2a7fdd504 100644 --- a/hw/audio/gus.c +++ b/hw/audio/gus.c @@ -251,7 +251,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp) as.freq = s->freq; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = GUS_ENDIANNESS; s->voice = AUD_open_out ( diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c index 617a1c1016b8..c25bfa38b143 100644 --- a/hw/audio/hda-codec.c +++ b/hw/audio/hda-codec.c @@ -99,9 +99,9 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as) } switch (format & AC_FMT_BITS_MASK) { - case AC_FMT_BITS_8: as->fmt = AUD_FMT_S8; break; - case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break; - case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break; + case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break; + case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break; + case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break; } as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1; @@ -134,12 +134,12 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as) /* -------------------------------------------------------------------------- */ static const char *fmt2name[] = { - [ AUD_FMT_U8 ] = "PCM-U8", - [ AUD_FMT_S8 ] = "PCM-S8", - [ AUD_FMT_U16 ] = "PCM-U16", - [ AUD_FMT_S16 ] = "PCM-S16", - [ AUD_FMT_U32 ] = "PCM-U32", - [ AUD_FMT_S32 ] = "PCM-S32", + [ AUDIO_FORMAT_U8 ] = "PCM-U8", + [ AUDIO_FORMAT_S8 ] = "PCM-S8", + [ AUDIO_FORMAT_U16 ] = "PCM-U16", + [ AUDIO_FORMAT_S16 ] = "PCM-S16", + [ AUDIO_FORMAT_U32 ] = "PCM-U32", + [ AUDIO_FORMAT_S32 ] = "PCM-S32", }; typedef struct HDAAudioState HDAAudioState; diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c index a46f2301af82..af8b22b541e9 100644 --- a/hw/audio/lm4549.c +++ b/hw/audio/lm4549.c @@ -185,7 +185,7 @@ void lm4549_write(lm4549_state *s, struct audsettings as; as.freq = value; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; s->voice = AUD_open_out( @@ -255,7 +255,7 @@ static int lm4549_post_load(void *opaque, int version_id) struct audsettings as; as.freq = freq; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; s->voice = AUD_open_out( @@ -292,7 +292,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback data_req_cb, void* opaque) /* Open a default voice */ as.freq = 48000; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; s->voice = AUD_open_out( diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c index bc8db71ae0dd..90cce1e6eded 100644 --- a/hw/audio/milkymist-ac97.c +++ b/hw/audio/milkymist-ac97.c @@ -308,7 +308,7 @@ static void milkymist_ac97_realize(DeviceState *dev, Error **errp) as.freq = 48000; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 1; s->voice_in = AUD_open_in(&s->card, s->voice_in, diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c index b80a62ce90a4..fdbb4b6e9923 100644 --- a/hw/audio/pcspk.c +++ b/hw/audio/pcspk.c @@ -162,7 +162,7 @@ static void pcspk_initfn(Object *obj) static void pcspk_realizefn(DeviceState *dev, Error **errp) { - struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0}; + struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUDIO_FORMAT_U8, 0}; ISADevice *isadev = ISA_DEVICE(dev); PCSpkState *s = PC_SPEAKER(dev); diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c index c5b9bf79e819..65ea0cd93832 100644 --- a/hw/audio/sb16.c +++ b/hw/audio/sb16.c @@ -66,7 +66,7 @@ typedef struct SB16State { int fmt_stereo; int fmt_signed; int fmt_bits; - audfmt_e fmt; + AudioFormat fmt; int dma_auto; int block_size; int fifo; @@ -224,7 +224,7 @@ static void continue_dma8 (SB16State *s) static void dma_cmd8 (SB16State *s, int mask, int dma_len) { - s->fmt = AUD_FMT_U8; + s->fmt = AUDIO_FORMAT_U8; s->use_hdma = 0; s->fmt_bits = 8; s->fmt_signed = 0; @@ -319,18 +319,18 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len) if (16 == s->fmt_bits) { if (s->fmt_signed) { - s->fmt = AUD_FMT_S16; + s->fmt = AUDIO_FORMAT_S16; } else { - s->fmt = AUD_FMT_U16; + s->fmt = AUDIO_FORMAT_U16; } } else { if (s->fmt_signed) { - s->fmt = AUD_FMT_S8; + s->fmt = AUDIO_FORMAT_S8; } else { - s->fmt = AUD_FMT_U8; + s->fmt = AUDIO_FORMAT_U8; } } @@ -852,7 +852,7 @@ static void legacy_reset (SB16State *s) as.freq = s->freq; as.nchannels = 1; - as.fmt = AUD_FMT_U8; + as.fmt = AUDIO_FORMAT_U8; as.endianness = 0; s->voice = AUD_open_out ( diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c index 169b006ade2d..ca0ad73caf7d 100644 --- a/hw/audio/wm8750.c +++ b/hw/audio/wm8750.c @@ -201,7 +201,7 @@ static void wm8750_set_format(WM8750State *s) in_fmt.endianness = 0; in_fmt.nchannels = 2; in_fmt.freq = s->adc_hz; - in_fmt.fmt = AUD_FMT_S16; + in_fmt.fmt = AUDIO_FORMAT_S16; s->adc_voice[0] = AUD_open_in(&s->card, s->adc_voice[0], CODEC ".input1", s, wm8750_audio_in_cb, &in_fmt); @@ -214,7 +214,7 @@ static void wm8750_set_format(WM8750State *s) out_fmt.endianness = 0; out_fmt.nchannels = 2; out_fmt.freq = s->dac_hz; - out_fmt.fmt = AUD_FMT_S16; + out_fmt.fmt = AUDIO_FORMAT_S16; s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0], CODEC ".speaker", s, wm8750_audio_out_cb, &out_fmt); @@ -681,7 +681,7 @@ uint32_t wm8750_adc_dat(void *opaque) if (s->idx_in >= sizeof(s->data_in)) { wm8750_in_load(s); if (s->idx_in >= sizeof(s->data_in)) { - return 0x80008000; /* silence in AUD_FMT_S16 sample format */ + return 0x80008000; /* silence in AUDIO_FORMAT_S16 sample format */ } } diff --git a/hw/display/xlnx_dp.c b/hw/display/xlnx_dp.c index cc0f9bc9cc8b..11b09bd18c19 100644 --- a/hw/display/xlnx_dp.c +++ b/hw/display/xlnx_dp.c @@ -1260,7 +1260,7 @@ static void xlnx_dp_realize(DeviceState *dev, Error **errp) as.freq = 44100; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; AUD_register_card("xlnx_dp.audio", &s->aud_card); diff --git a/hw/input/tsc210x.c b/hw/input/tsc210x.c index 2eb3cb951878..41731619bb6a 100644 --- a/hw/input/tsc210x.c +++ b/hw/input/tsc210x.c @@ -318,7 +318,7 @@ static void tsc2102_audio_output_update(TSC210xState *s) fmt.endianness = 0; fmt.nchannels = 2; fmt.freq = s->codec.tx_rate; - fmt.fmt = AUD_FMT_S16; + fmt.fmt = AUDIO_FORMAT_S16; s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0], "tsc2102.sink", s, (void *) tsc210x_audio_out_cb, &fmt); diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c index 28ac7c5165ff..c46d5eeb79a3 100644 --- a/hw/usb/dev-audio.c +++ b/hw/usb/dev-audio.c @@ -650,7 +650,7 @@ static void usb_audio_realize(USBDevice *dev, Error **errp) s->out.vol[1] = 240; /* 0 dB */ s->out.as.freq = USBAUDIO_SAMPLE_RATE; s->out.as.nchannels = 2; - s->out.as.fmt = AUD_FMT_S16; + s->out.as.fmt = AUDIO_FORMAT_S16; s->out.as.endianness = 0; streambuf_init(&s->out.buf, s->buffer); diff --git a/ui/vnc.c b/ui/vnc.c index da4a21d4ce94..b724e998bab6 100644 --- a/ui/vnc.c +++ b/ui/vnc.c @@ -1013,16 +1013,16 @@ static void vnc_update_throttle_offset(VncState *vs) int bps; switch (vs->as.fmt) { default: - case AUD_FMT_U8: - case AUD_FMT_S8: + case AUDIO_FORMAT_U8: + case AUDIO_FORMAT_S8: bps = 1; break; - case AUD_FMT_U16: - case AUD_FMT_S16: + case AUDIO_FORMAT_U16: + case AUDIO_FORMAT_S16: bps = 2; break; - case AUD_FMT_U32: - case AUD_FMT_S32: + case AUDIO_FORMAT_U32: + case AUDIO_FORMAT_S32: bps = 4; break; } @@ -2369,12 +2369,12 @@ static int protocol_client_msg(VncState *vs, uint8_t *data, size_t len) if (len == 4) return 10; switch (read_u8(data, 4)) { - case 0: vs->as.fmt = AUD_FMT_U8; break; - case 1: vs->as.fmt = AUD_FMT_S8; break; - case 2: vs->as.fmt = AUD_FMT_U16; break; - case 3: vs->as.fmt = AUD_FMT_S16; break; - case 4: vs->as.fmt = AUD_FMT_U32; break; - case 5: vs->as.fmt = AUD_FMT_S32; break; + case 0: vs->as.fmt = AUDIO_FORMAT_U8; break; + case 1: vs->as.fmt = AUDIO_FORMAT_S8; break; + case 2: vs->as.fmt = AUDIO_FORMAT_U16; break; + case 3: vs->as.fmt = AUDIO_FORMAT_S16; break; + case 4: vs->as.fmt = AUDIO_FORMAT_U32; break; + case 5: vs->as.fmt = AUDIO_FORMAT_S32; break; default: VNC_DEBUG("Invalid audio format %d\n", read_u8(data, 4)); vnc_client_error(vs); @@ -3105,7 +3105,7 @@ static void vnc_connect(VncDisplay *vd, QIOChannelSocket *sioc, vs->as.freq = 44100; vs->as.nchannels = 2; - vs->as.fmt = AUD_FMT_S16; + vs->as.fmt = AUDIO_FORMAT_S16; vs->as.endianness = 0; qemu_mutex_init(&vs->output_mutex); -- 2.18.1

From: Kővágó, Zoltán <dirty.ice.hu@gmail.com> Audio drivers now get an Audiodev * as config paramters, instead of the global audio_option structs. There is some code in audio/audio_legacy.c that converts the old environment variables to audiodev options (this way backends do not have to worry about legacy options). It also contains a replacement of -audio-help, which prints out the equivalent -audiodev based config of the currently specified environment variables. Note that backends are not updated and still rely on environment variables. Also note that (due to moving try-poll from global to backend specific option) currently ALSA and OSS will always try poll mode, regardless of environment variables or -audiodev options. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: e99a7cbdac0d13512743880660b2032024703e4c.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> --- audio/audio.h | 18 +- audio/audio_int.h | 20 +- audio/audio_template.h | 42 ++- audio/alsaaudio.c | 2 +- audio/audio.c | 630 ++++++++++++++++++----------------------- audio/audio_legacy.c | 293 +++++++++++++++++++ audio/coreaudio.c | 2 +- audio/dsoundaudio.c | 2 +- audio/noaudio.c | 2 +- audio/ossaudio.c | 2 +- audio/paaudio.c | 2 +- audio/sdlaudio.c | 2 +- audio/spiceaudio.c | 2 +- audio/wavaudio.c | 2 +- vl.c | 7 +- audio/Makefile.objs | 2 +- 16 files changed, 650 insertions(+), 380 deletions(-) create mode 100644 audio/audio_legacy.c diff --git a/audio/audio.h b/audio/audio.h index 02f29a3b3ef7..64b0f761bcaa 100644 --- a/audio/audio.h +++ b/audio/audio.h @@ -36,12 +36,21 @@ typedef void (*audio_callback_fn) (void *opaque, int avail); #define AUDIO_HOST_ENDIANNESS 0 #endif -struct audsettings { +typedef struct audsettings { int freq; int nchannels; AudioFormat fmt; int endianness; -}; +} audsettings; + +audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo); +int audioformat_bytes_per_sample(AudioFormat fmt); +int audio_buffer_frames(AudiodevPerDirectionOptions *pdo, + audsettings *as, int def_usecs); +int audio_buffer_samples(AudiodevPerDirectionOptions *pdo, + audsettings *as, int def_usecs); +int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo, + audsettings *as, int def_usecs); typedef enum { AUD_CNOTIFY_ENABLE, @@ -81,7 +90,6 @@ typedef struct QEMUAudioTimeStamp { void AUD_vlog (const char *cap, const char *fmt, va_list ap) GCC_FMT_ATTR(2, 0); void AUD_log (const char *cap, const char *fmt, ...) GCC_FMT_ATTR(2, 3); -void AUD_help (void); void AUD_register_card (const char *name, QEMUSoundCard *card); void AUD_remove_card (QEMUSoundCard *card); CaptureVoiceOut *AUD_add_capture ( @@ -163,4 +171,8 @@ void audio_sample_to_uint64(void *samples, int pos, void audio_sample_from_uint64(void *samples, int pos, uint64_t left, uint64_t right); +void audio_parse_option(const char *opt); +void audio_init_audiodevs(void); +void audio_legacy_help(void); + #endif /* QEMU_AUDIO_H */ diff --git a/audio/audio_int.h b/audio/audio_int.h index 6c451b995ccc..7bf5dfc0b5f6 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -146,7 +146,7 @@ struct audio_driver { const char *name; const char *descr; struct audio_option *options; - void *(*init) (void); + void *(*init) (Audiodev *); void (*fini) (void *); struct audio_pcm_ops *pcm_ops; int can_be_default; @@ -193,6 +193,7 @@ struct SWVoiceCap { typedef struct AudioState { struct audio_driver *drv; + Audiodev *dev; void *drv_opaque; QEMUTimer *ts; @@ -203,10 +204,13 @@ typedef struct AudioState { int nb_hw_voices_out; int nb_hw_voices_in; int vm_running; + int64_t period_ticks; } AudioState; extern const struct mixeng_volume nominal_volume; +extern const char *audio_prio_list[]; + void audio_driver_register(audio_driver *drv); audio_driver *audio_driver_lookup(const char *name); @@ -248,4 +252,18 @@ static inline int audio_ring_dist (int dst, int src, int len) #define AUDIO_STRINGIFY_(n) #n #define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n) +typedef struct AudiodevListEntry { + Audiodev *dev; + QSIMPLEQ_ENTRY(AudiodevListEntry) next; +} AudiodevListEntry; + +typedef QSIMPLEQ_HEAD(, AudiodevListEntry) AudiodevListHead; +AudiodevListHead audio_handle_legacy_opts(void); + +void audio_free_audiodev_list(AudiodevListHead *head); + +void audio_create_pdos(Audiodev *dev); +AudiodevPerDirectionOptions *audio_get_pdo_in(Audiodev *dev); +AudiodevPerDirectionOptions *audio_get_pdo_out(Audiodev *dev); + #endif /* QEMU_AUDIO_INT_H */ diff --git a/audio/audio_template.h b/audio/audio_template.h index 7de227d2d10c..1232bb54db0e 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -299,11 +299,42 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as) return NULL; } +AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev) +{ + switch (dev->driver) { + case AUDIODEV_DRIVER_NONE: + return dev->u.none.TYPE; + case AUDIODEV_DRIVER_ALSA: + return qapi_AudiodevAlsaPerDirectionOptions_base(dev->u.alsa.TYPE); + case AUDIODEV_DRIVER_COREAUDIO: + return qapi_AudiodevCoreaudioPerDirectionOptions_base( + dev->u.coreaudio.TYPE); + case AUDIODEV_DRIVER_DSOUND: + return dev->u.dsound.TYPE; + case AUDIODEV_DRIVER_OSS: + return qapi_AudiodevOssPerDirectionOptions_base(dev->u.oss.TYPE); + case AUDIODEV_DRIVER_PA: + return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE); + case AUDIODEV_DRIVER_SDL: + return dev->u.sdl.TYPE; + case AUDIODEV_DRIVER_SPICE: + return dev->u.spice.TYPE; + case AUDIODEV_DRIVER_WAV: + return dev->u.wav.TYPE; + + case AUDIODEV_DRIVER__MAX: + break; + } + abort(); +} + static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as) { HW *hw; + AudioState *s = &glob_audio_state; + AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev); - if (glue (conf.fixed_, TYPE).enabled && glue (conf.fixed_, TYPE).greedy) { + if (pdo->fixed_settings) { hw = glue (audio_pcm_hw_add_new_, TYPE) (as); if (hw) { return hw; @@ -331,9 +362,11 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) ( SW *sw; HW *hw; struct audsettings hw_as; + AudioState *s = &glob_audio_state; + AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev); - if (glue (conf.fixed_, TYPE).enabled) { - hw_as = glue (conf.fixed_, TYPE).settings; + if (pdo->fixed_settings) { + hw_as = audiodev_to_audsettings(pdo); } else { hw_as = *as; @@ -398,6 +431,7 @@ SW *glue (AUD_open_, TYPE) ( ) { AudioState *s = &glob_audio_state; + AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev); if (audio_bug(__func__, !card || !name || !callback_fn || !as)) { dolog ("card=%p name=%p callback_fn=%p as=%p\n", @@ -422,7 +456,7 @@ SW *glue (AUD_open_, TYPE) ( return sw; } - if (!glue (conf.fixed_, TYPE).enabled && sw) { + if (!pdo->fixed_settings && sw) { glue (AUD_close_, TYPE) (card, sw); sw = NULL; } diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 5bd034267fce..8302f3e882d7 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -1125,7 +1125,7 @@ static ALSAConf glob_conf = { .pcm_name_in = "default", }; -static void *alsa_audio_init (void) +static void *alsa_audio_init(Audiodev *dev) { ALSAConf *conf = g_malloc(sizeof(ALSAConf)); *conf = glob_conf; diff --git a/audio/audio.c b/audio/audio.c index 77216e50102e..79ed36066604 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -26,6 +26,9 @@ #include "audio.h" #include "monitor/monitor.h" #include "qemu/timer.h" +#include "qapi/error.h" +#include "qapi/qobject-input-visitor.h" +#include "qapi/qapi-visit-audio.h" #include "sysemu/sysemu.h" #include "qemu/cutils.h" #include "sysemu/replay.h" @@ -46,14 +49,16 @@ The 1st one is the one used by default, that is the reason that we generate the list. */ -static const char *audio_prio_list[] = { +const char *audio_prio_list[] = { "spice", CONFIG_AUDIO_DRIVERS "none", "wav", + NULL }; static QLIST_HEAD(, audio_driver) audio_drivers; +static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs); void audio_driver_register(audio_driver *drv) { @@ -80,61 +85,6 @@ audio_driver *audio_driver_lookup(const char *name) return NULL; } -static void audio_module_load_all(void) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(audio_prio_list); i++) { - audio_driver_lookup(audio_prio_list[i]); - } -} - -struct fixed_settings { - int enabled; - int nb_voices; - int greedy; - struct audsettings settings; -}; - -static struct { - struct fixed_settings fixed_out; - struct fixed_settings fixed_in; - union { - int hertz; - int64_t ticks; - } period; - int try_poll_in; - int try_poll_out; -} conf = { - .fixed_out = { /* DAC fixed settings */ - .enabled = 1, - .nb_voices = 1, - .greedy = 1, - .settings = { - .freq = 44100, - .nchannels = 2, - .fmt = AUDIO_FORMAT_S16, - .endianness = AUDIO_HOST_ENDIANNESS, - } - }, - - .fixed_in = { /* ADC fixed settings */ - .enabled = 1, - .nb_voices = 1, - .greedy = 1, - .settings = { - .freq = 44100, - .nchannels = 2, - .fmt = AUDIO_FORMAT_S16, - .endianness = AUDIO_HOST_ENDIANNESS, - } - }, - - .period = { .hertz = 100 }, - .try_poll_in = 1, - .try_poll_out = 1, -}; - static AudioState glob_audio_state; const struct mixeng_volume nominal_volume = { @@ -151,9 +101,6 @@ const struct mixeng_volume nominal_volume = { #ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED #error No its not #else -static void audio_print_options (const char *prefix, - struct audio_option *opt); - int audio_bug (const char *funcname, int cond) { if (cond) { @@ -161,16 +108,9 @@ int audio_bug (const char *funcname, int cond) AUD_log (NULL, "A bug was just triggered in %s\n", funcname); if (!shown) { - struct audio_driver *d; - shown = 1; AUD_log (NULL, "Save all your work and restart without audio\n"); - AUD_log (NULL, "Please send bug report to av1474@comtv.ru\n"); AUD_log (NULL, "I am sorry\n"); - d = glob_audio_state.drv; - if (d) { - audio_print_options (d->name, d->options); - } } AUD_log (NULL, "Context:\n"); @@ -232,31 +172,6 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size) return g_malloc0 (len); } -static char *audio_alloc_prefix (const char *s) -{ - const char qemu_prefix[] = "QEMU_"; - size_t len, i; - char *r, *u; - - if (!s) { - return NULL; - } - - len = strlen (s); - r = g_malloc (len + sizeof (qemu_prefix)); - - u = r + sizeof (qemu_prefix) - 1; - - pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix); - pstrcat (r, len + sizeof (qemu_prefix), s); - - for (i = 0; i < len; ++i) { - u[i] = qemu_toupper(u[i]); - } - - return r; -} - static const char *audio_audfmt_to_string (AudioFormat fmt) { switch (fmt) { @@ -382,78 +297,6 @@ void AUD_log (const char *cap, const char *fmt, ...) va_end (ap); } -static void audio_print_options (const char *prefix, - struct audio_option *opt) -{ - char *uprefix; - - if (!prefix) { - dolog ("No prefix specified\n"); - return; - } - - if (!opt) { - dolog ("No options\n"); - return; - } - - uprefix = audio_alloc_prefix (prefix); - - for (; opt->name; opt++) { - const char *state = "default"; - printf (" %s_%s: ", uprefix, opt->name); - - if (opt->overriddenp && *opt->overriddenp) { - state = "current"; - } - - switch (opt->tag) { - case AUD_OPT_BOOL: - { - int *intp = opt->valp; - printf ("boolean, %s = %d\n", state, *intp ? 1 : 0); - } - break; - - case AUD_OPT_INT: - { - int *intp = opt->valp; - printf ("integer, %s = %d\n", state, *intp); - } - break; - - case AUD_OPT_FMT: - { - AudioFormat *fmtp = opt->valp; - printf ( - "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n", - state, - audio_audfmt_to_string (*fmtp) - ); - } - break; - - case AUD_OPT_STR: - { - const char **strp = opt->valp; - printf ("string, %s = %s\n", - state, - *strp ? *strp : "(not set)"); - } - break; - - default: - printf ("???\n"); - dolog ("Bad value tag for option %s_%s %d\n", - uprefix, opt->name, opt->tag); - break; - } - printf (" %s\n", opt->descr); - } - - g_free (uprefix); -} - static void audio_process_options (const char *prefix, struct audio_option *opt) { @@ -1141,11 +984,11 @@ static void audio_reset_timer (AudioState *s) { if (audio_is_timer_needed ()) { timer_mod_anticipate_ns(s->ts, - qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks); + qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks); if (!audio_timer_running) { audio_timer_running = true; audio_timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); - trace_audio_timer_start(conf.period.ticks / SCALE_MS); + trace_audio_timer_start(s->period_ticks / SCALE_MS); } } else { timer_del(s->ts); @@ -1159,16 +1002,17 @@ static void audio_reset_timer (AudioState *s) static void audio_timer (void *opaque) { int64_t now, diff; + AudioState *s = opaque; now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); diff = now - audio_timer_last; - if (diff > conf.period.ticks * 3 / 2) { + if (diff > s->period_ticks * 3 / 2) { trace_audio_timer_delayed(diff / SCALE_MS); } audio_timer_last = now; - audio_run ("timer"); - audio_reset_timer (opaque); + audio_run("timer"); + audio_reset_timer(s); } /* @@ -1228,7 +1072,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on) if (!hw->enabled) { hw->enabled = 1; if (s->vm_running) { - hw->pcm_ops->ctl_out (hw, VOICE_ENABLE, conf.try_poll_out); + hw->pcm_ops->ctl_out(hw, VOICE_ENABLE, true); audio_reset_timer (s); } } @@ -1273,7 +1117,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on) if (!hw->enabled) { hw->enabled = 1; if (s->vm_running) { - hw->pcm_ops->ctl_in (hw, VOICE_ENABLE, conf.try_poll_in); + hw->pcm_ops->ctl_in(hw, VOICE_ENABLE, true); audio_reset_timer (s); } } @@ -1594,169 +1438,13 @@ void audio_run (const char *msg) #endif } -static struct audio_option audio_options[] = { - /* DAC */ - { - .name = "DAC_FIXED_SETTINGS", - .tag = AUD_OPT_BOOL, - .valp = &conf.fixed_out.enabled, - .descr = "Use fixed settings for host DAC" - }, - { - .name = "DAC_FIXED_FREQ", - .tag = AUD_OPT_INT, - .valp = &conf.fixed_out.settings.freq, - .descr = "Frequency for fixed host DAC" - }, - { - .name = "DAC_FIXED_FMT", - .tag = AUD_OPT_FMT, - .valp = &conf.fixed_out.settings.fmt, - .descr = "Format for fixed host DAC" - }, - { - .name = "DAC_FIXED_CHANNELS", - .tag = AUD_OPT_INT, - .valp = &conf.fixed_out.settings.nchannels, - .descr = "Number of channels for fixed DAC (1 - mono, 2 - stereo)" - }, - { - .name = "DAC_VOICES", - .tag = AUD_OPT_INT, - .valp = &conf.fixed_out.nb_voices, - .descr = "Number of voices for DAC" - }, - { - .name = "DAC_TRY_POLL", - .tag = AUD_OPT_BOOL, - .valp = &conf.try_poll_out, - .descr = "Attempt using poll mode for DAC" - }, - /* ADC */ - { - .name = "ADC_FIXED_SETTINGS", - .tag = AUD_OPT_BOOL, - .valp = &conf.fixed_in.enabled, - .descr = "Use fixed settings for host ADC" - }, - { - .name = "ADC_FIXED_FREQ", - .tag = AUD_OPT_INT, - .valp = &conf.fixed_in.settings.freq, - .descr = "Frequency for fixed host ADC" - }, - { - .name = "ADC_FIXED_FMT", - .tag = AUD_OPT_FMT, - .valp = &conf.fixed_in.settings.fmt, - .descr = "Format for fixed host ADC" - }, - { - .name = "ADC_FIXED_CHANNELS", - .tag = AUD_OPT_INT, - .valp = &conf.fixed_in.settings.nchannels, - .descr = "Number of channels for fixed ADC (1 - mono, 2 - stereo)" - }, - { - .name = "ADC_VOICES", - .tag = AUD_OPT_INT, - .valp = &conf.fixed_in.nb_voices, - .descr = "Number of voices for ADC" - }, - { - .name = "ADC_TRY_POLL", - .tag = AUD_OPT_BOOL, - .valp = &conf.try_poll_in, - .descr = "Attempt using poll mode for ADC" - }, - /* Misc */ - { - .name = "TIMER_PERIOD", - .tag = AUD_OPT_INT, - .valp = &conf.period.hertz, - .descr = "Timer period in HZ (0 - use lowest possible)" - }, - { /* End of list */ } -}; - -static void audio_pp_nb_voices (const char *typ, int nb) -{ - switch (nb) { - case 0: - printf ("Does not support %s\n", typ); - break; - case 1: - printf ("One %s voice\n", typ); - break; - case INT_MAX: - printf ("Theoretically supports many %s voices\n", typ); - break; - default: - printf ("Theoretically supports up to %d %s voices\n", nb, typ); - break; - } - -} - -void AUD_help (void) -{ - struct audio_driver *d; - - /* make sure we print the help text for modular drivers too */ - audio_module_load_all(); - - audio_process_options ("AUDIO", audio_options); - QLIST_FOREACH(d, &audio_drivers, next) { - if (d->options) { - audio_process_options (d->name, d->options); - } - } - - printf ("Audio options:\n"); - audio_print_options ("AUDIO", audio_options); - printf ("\n"); - - printf ("Available drivers:\n"); - - QLIST_FOREACH(d, &audio_drivers, next) { - - printf ("Name: %s\n", d->name); - printf ("Description: %s\n", d->descr); - - audio_pp_nb_voices ("playback", d->max_voices_out); - audio_pp_nb_voices ("capture", d->max_voices_in); - - if (d->options) { - printf ("Options:\n"); - audio_print_options (d->name, d->options); - } - else { - printf ("No options\n"); - } - printf ("\n"); - } - - printf ( - "Options are settable through environment variables.\n" - "Example:\n" -#ifdef _WIN32 - " set QEMU_AUDIO_DRV=wav\n" - " set QEMU_WAV_PATH=c:\\tune.wav\n" -#else - " export QEMU_AUDIO_DRV=wav\n" - " export QEMU_WAV_PATH=$HOME/tune.wav\n" - "(for csh replace export with setenv in the above)\n" -#endif - " qemu ...\n\n" - ); -} - -static int audio_driver_init(AudioState *s, struct audio_driver *drv, bool msg) +static int audio_driver_init(AudioState *s, struct audio_driver *drv, + bool msg, Audiodev *dev) { if (drv->options) { audio_process_options (drv->name, drv->options); } - s->drv_opaque = drv->init (); + s->drv_opaque = drv->init(dev); if (s->drv_opaque) { audio_init_nb_voices_out (drv); @@ -1782,11 +1470,11 @@ static void audio_vm_change_state_handler (void *opaque, int running, s->vm_running = running; while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) { - hwo->pcm_ops->ctl_out (hwo, op, conf.try_poll_out); + hwo->pcm_ops->ctl_out(hwo, op, true); } while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) { - hwi->pcm_ops->ctl_in (hwi, op, conf.try_poll_in); + hwi->pcm_ops->ctl_in(hwi, op, true); } audio_reset_timer (s); } @@ -1836,6 +1524,11 @@ void audio_cleanup(void) s->drv->fini (s->drv_opaque); s->drv = NULL; } + + if (s->dev) { + qapi_free_Audiodev(s->dev); + s->dev = NULL; + } } static const VMStateDescription vmstate_audio = { @@ -1847,19 +1540,58 @@ static const VMStateDescription vmstate_audio = { } }; -static void audio_init (void) +static void audio_validate_opts(Audiodev *dev, Error **errp); + +static AudiodevListEntry *audiodev_find( + AudiodevListHead *head, const char *drvname) +{ + AudiodevListEntry *e; + QSIMPLEQ_FOREACH(e, head, next) { + if (strcmp(AudiodevDriver_str(e->dev->driver), drvname) == 0) { + return e; + } + } + + return NULL; +} + +static int audio_init(Audiodev *dev) { size_t i; int done = 0; - const char *drvname; + const char *drvname = NULL; VMChangeStateEntry *e; AudioState *s = &glob_audio_state; struct audio_driver *driver; + /* silence gcc warning about uninitialized variable */ + AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head); if (s->drv) { - return; + if (dev) { + dolog("Cannot create more than one audio backend, sorry\n"); + qapi_free_Audiodev(dev); + } + return -1; } + if (dev) { + /* -audiodev option */ + drvname = AudiodevDriver_str(dev->driver); + } else { + /* legacy implicit initialization */ + head = audio_handle_legacy_opts(); + /* + * In case of legacy initialization, all Audiodevs in the list will have + * the same configuration (except the driver), so it does't matter which + * one we chose. We need an Audiodev to set up AudioState before we can + * init a driver. Also note that dev at this point is still in the + * list. + */ + dev = QSIMPLEQ_FIRST(&head)->dev; + audio_validate_opts(dev, &error_abort); + } + s->dev = dev; + QLIST_INIT (&s->hw_head_out); QLIST_INIT (&s->hw_head_in); QLIST_INIT (&s->cap_head); @@ -1867,10 +1599,8 @@ static void audio_init (void) s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s); - audio_process_options ("AUDIO", audio_options); - - s->nb_hw_voices_out = conf.fixed_out.nb_voices; - s->nb_hw_voices_in = conf.fixed_in.nb_voices; + s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices; + s->nb_hw_voices_in = audio_get_pdo_in(dev)->voices; if (s->nb_hw_voices_out <= 0) { dolog ("Bogus number of playback voices %d, setting to 1\n", @@ -1884,46 +1614,42 @@ static void audio_init (void) s->nb_hw_voices_in = 0; } - { - int def; - drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def); - } - if (drvname) { driver = audio_driver_lookup(drvname); if (driver) { - done = !audio_driver_init(s, driver, true); + done = !audio_driver_init(s, driver, true, dev); } else { dolog ("Unknown audio driver `%s'\n", drvname); - dolog ("Run with -audio-help to list available drivers\n"); } - } - - if (!done) { - for (i = 0; !done && i < ARRAY_SIZE(audio_prio_list); i++) { + } else { + for (i = 0; audio_prio_list[i]; i++) { + AudiodevListEntry *e = audiodev_find(&head, audio_prio_list[i]); driver = audio_driver_lookup(audio_prio_list[i]); - if (driver && driver->can_be_default) { - done = !audio_driver_init(s, driver, false); + + if (e && driver) { + s->dev = dev = e->dev; + audio_validate_opts(dev, &error_abort); + done = !audio_driver_init(s, driver, false, dev); + if (done) { + e->dev = NULL; + break; + } } } } + audio_free_audiodev_list(&head); if (!done) { driver = audio_driver_lookup("none"); - done = !audio_driver_init(s, driver, false); + done = !audio_driver_init(s, driver, false, dev); assert(done); dolog("warning: Using timer based audio emulation\n"); } - if (conf.period.hertz <= 0) { - if (conf.period.hertz < 0) { - dolog ("warning: Timer period is negative - %d " - "treating as zero\n", - conf.period.hertz); - } - conf.period.ticks = 1; + if (dev->timer_period <= 0) { + s->period_ticks = 1; } else { - conf.period.ticks = NANOSECONDS_PER_SECOND / conf.period.hertz; + s->period_ticks = NANOSECONDS_PER_SECOND / dev->timer_period; } e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s); @@ -1934,11 +1660,22 @@ static void audio_init (void) QLIST_INIT (&s->card_head); vmstate_register (NULL, 0, &vmstate_audio, s); + return 0; +} + +void audio_free_audiodev_list(AudiodevListHead *head) +{ + AudiodevListEntry *e; + while ((e = QSIMPLEQ_FIRST(head))) { + QSIMPLEQ_REMOVE_HEAD(head, next); + qapi_free_Audiodev(e->dev); + g_free(e); + } } void AUD_register_card (const char *name, QEMUSoundCard *card) { - audio_init (); + audio_init(NULL); card->name = g_strdup (name); memset (&card->entries, 0, sizeof (card->entries)); QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries); @@ -2078,3 +1815,174 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol) } } } + +void audio_create_pdos(Audiodev *dev) +{ + switch (dev->driver) { +#define CASE(DRIVER, driver, pdo_name) \ + case AUDIODEV_DRIVER_##DRIVER: \ + if (!dev->u.driver.has_in) { \ + dev->u.driver.in = g_malloc0( \ + sizeof(Audiodev##pdo_name##PerDirectionOptions)); \ + dev->u.driver.has_in = true; \ + } \ + if (!dev->u.driver.has_out) { \ + dev->u.driver.out = g_malloc0( \ + sizeof(AudiodevAlsaPerDirectionOptions)); \ + dev->u.driver.has_out = true; \ + } \ + break + + CASE(NONE, none, ); + CASE(ALSA, alsa, Alsa); + CASE(COREAUDIO, coreaudio, Coreaudio); + CASE(DSOUND, dsound, ); + CASE(OSS, oss, Oss); + CASE(PA, pa, Pa); + CASE(SDL, sdl, ); + CASE(SPICE, spice, ); + CASE(WAV, wav, ); + + case AUDIODEV_DRIVER__MAX: + abort(); + }; +} + +static void audio_validate_per_direction_opts( + AudiodevPerDirectionOptions *pdo, Error **errp) +{ + if (!pdo->has_fixed_settings) { + pdo->has_fixed_settings = true; + pdo->fixed_settings = true; + } + if (!pdo->fixed_settings && + (pdo->has_frequency || pdo->has_channels || pdo->has_format)) { + error_setg(errp, + "You can't use frequency, channels or format with fixed-settings=off"); + return; + } + + if (!pdo->has_frequency) { + pdo->has_frequency = true; + pdo->frequency = 44100; + } + if (!pdo->has_channels) { + pdo->has_channels = true; + pdo->channels = 2; + } + if (!pdo->has_voices) { + pdo->has_voices = true; + pdo->voices = 1; + } + if (!pdo->has_format) { + pdo->has_format = true; + pdo->format = AUDIO_FORMAT_S16; + } +} + +static void audio_validate_opts(Audiodev *dev, Error **errp) +{ + Error *err = NULL; + + audio_create_pdos(dev); + + audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err); + if (err) { + error_propagate(errp, err); + return; + } + + audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err); + if (err) { + error_propagate(errp, err); + return; + } + + if (!dev->has_timer_period) { + dev->has_timer_period = true; + dev->timer_period = 10000; /* 100Hz -> 10ms */ + } +} + +void audio_parse_option(const char *opt) +{ + AudiodevListEntry *e; + Audiodev *dev = NULL; + + Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal); + visit_type_Audiodev(v, NULL, &dev, &error_fatal); + visit_free(v); + + audio_validate_opts(dev, &error_fatal); + + e = g_malloc0(sizeof(AudiodevListEntry)); + e->dev = dev; + QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next); +} + +void audio_init_audiodevs(void) +{ + AudiodevListEntry *e; + + QSIMPLEQ_FOREACH(e, &audiodevs, next) { + audio_init(e->dev); + } +} + +audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo) +{ + return (audsettings) { + .freq = pdo->frequency, + .nchannels = pdo->channels, + .fmt = pdo->format, + .endianness = AUDIO_HOST_ENDIANNESS, + }; +} + +int audioformat_bytes_per_sample(AudioFormat fmt) +{ + switch (fmt) { + case AUDIO_FORMAT_U8: + case AUDIO_FORMAT_S8: + return 1; + + case AUDIO_FORMAT_U16: + case AUDIO_FORMAT_S16: + return 2; + + case AUDIO_FORMAT_U32: + case AUDIO_FORMAT_S32: + return 4; + + case AUDIO_FORMAT__MAX: + ; + } + abort(); +} + + +/* frames = freq * usec / 1e6 */ +int audio_buffer_frames(AudiodevPerDirectionOptions *pdo, + audsettings *as, int def_usecs) +{ + uint64_t usecs = pdo->has_buffer_length ? pdo->buffer_length : def_usecs; + return (as->freq * usecs + 500000) / 1000000; +} + +/* samples = channels * frames = channels * freq * usec / 1e6 */ +int audio_buffer_samples(AudiodevPerDirectionOptions *pdo, + audsettings *as, int def_usecs) +{ + return as->nchannels * audio_buffer_frames(pdo, as, def_usecs); +} + +/* + * bytes = bytes_per_sample * samples = + * bytes_per_sample * channels * freq * usec / 1e6 + */ +int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo, + audsettings *as, int def_usecs) +{ + return audio_buffer_samples(pdo, as, def_usecs) * + audioformat_bytes_per_sample(as->fmt); +} diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c new file mode 100644 index 000000000000..ad3b781addf8 --- /dev/null +++ b/audio/audio_legacy.c @@ -0,0 +1,293 @@ +/* + * QEMU Audio subsystem: legacy configuration handling + * + * Copyright (c) 2015-2019 Zoltán Kővágó <DirtY.iCE.hu@gmail.com> + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ +#include "qemu/osdep.h" +#include "audio.h" +#include "audio_int.h" +#include "qemu-common.h" +#include "qemu/cutils.h" +#include "qapi/error.h" +#include "qapi/qapi-visit-audio.h" +#include "qapi/visitor-impl.h" + +#define AUDIO_CAP "audio-legacy" +#include "audio_int.h" + +static uint32_t toui32(const char *str) +{ + unsigned long long ret; + if (parse_uint_full(str, &ret, 10) || ret > UINT32_MAX) { + dolog("Invalid integer value `%s'\n", str); + exit(1); + } + return ret; +} + +/* helper functions to convert env variables */ +static void get_bool(const char *env, bool *dst, bool *has_dst) +{ + const char *val = getenv(env); + if (val) { + *dst = toui32(val) != 0; + *has_dst = true; + } +} + +static void get_int(const char *env, uint32_t *dst, bool *has_dst) +{ + const char *val = getenv(env); + if (val) { + *dst = toui32(val); + *has_dst = true; + } +} + +static void get_fmt(const char *env, AudioFormat *dst, bool *has_dst) +{ + const char *val = getenv(env); + if (val) { + size_t i; + for (i = 0; AudioFormat_lookup.size; ++i) { + if (strcasecmp(val, AudioFormat_lookup.array[i]) == 0) { + *dst = i; + *has_dst = true; + return; + } + } + + dolog("Invalid audio format `%s'\n", val); + exit(1); + } +} + +/* backend specific functions */ +/* todo */ + +/* general */ +static void handle_per_direction( + AudiodevPerDirectionOptions *pdo, const char *prefix) +{ + char buf[64]; + size_t len = strlen(prefix); + + memcpy(buf, prefix, len); + strcpy(buf + len, "FIXED_SETTINGS"); + get_bool(buf, &pdo->fixed_settings, &pdo->has_fixed_settings); + + strcpy(buf + len, "FIXED_FREQ"); + get_int(buf, &pdo->frequency, &pdo->has_frequency); + + strcpy(buf + len, "FIXED_FMT"); + get_fmt(buf, &pdo->format, &pdo->has_format); + + strcpy(buf + len, "FIXED_CHANNELS"); + get_int(buf, &pdo->channels, &pdo->has_channels); + + strcpy(buf + len, "VOICES"); + get_int(buf, &pdo->voices, &pdo->has_voices); +} + +static AudiodevListEntry *legacy_opt(const char *drvname) +{ + AudiodevListEntry *e = g_malloc0(sizeof(AudiodevListEntry)); + e->dev = g_malloc0(sizeof(Audiodev)); + e->dev->id = g_strdup(drvname); + e->dev->driver = qapi_enum_parse( + &AudiodevDriver_lookup, drvname, -1, &error_abort); + + audio_create_pdos(e->dev); + + handle_per_direction(audio_get_pdo_in(e->dev), "QEMU_AUDIO_ADC_"); + handle_per_direction(audio_get_pdo_out(e->dev), "QEMU_AUDIO_DAC_"); + + get_int("QEMU_AUDIO_TIMER_PERIOD", + &e->dev->timer_period, &e->dev->has_timer_period); + + return e; +} + +AudiodevListHead audio_handle_legacy_opts(void) +{ + const char *drvname = getenv("QEMU_AUDIO_DRV"); + AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head); + + if (drvname) { + AudiodevListEntry *e; + audio_driver *driver = audio_driver_lookup(drvname); + if (!driver) { + dolog("Unknown audio driver `%s'\n", drvname); + exit(1); + } + e = legacy_opt(drvname); + QSIMPLEQ_INSERT_TAIL(&head, e, next); + } else { + for (int i = 0; audio_prio_list[i]; i++) { + audio_driver *driver = audio_driver_lookup(audio_prio_list[i]); + if (driver && driver->can_be_default) { + AudiodevListEntry *e = legacy_opt(driver->name); + QSIMPLEQ_INSERT_TAIL(&head, e, next); + } + } + if (QSIMPLEQ_EMPTY(&head)) { + dolog("Internal error: no default audio driver available\n"); + exit(1); + } + } + + return head; +} + +/* visitor to print -audiodev option */ +typedef struct { + Visitor visitor; + + bool comma; + GList *path; +} LegacyPrintVisitor; + +static void lv_start_struct(Visitor *v, const char *name, void **obj, + size_t size, Error **errp) +{ + LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v; + lv->path = g_list_append(lv->path, g_strdup(name)); +} + +static void lv_end_struct(Visitor *v, void **obj) +{ + LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v; + lv->path = g_list_delete_link(lv->path, g_list_last(lv->path)); +} + +static void lv_print_key(Visitor *v, const char *name) +{ + GList *e; + LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v; + if (lv->comma) { + putchar(','); + } else { + lv->comma = true; + } + + for (e = lv->path; e; e = e->next) { + if (e->data) { + printf("%s.", (const char *) e->data); + } + } + + printf("%s=", name); +} + +static void lv_type_int64(Visitor *v, const char *name, int64_t *obj, + Error **errp) +{ + lv_print_key(v, name); + printf("%" PRIi64, *obj); +} + +static void lv_type_uint64(Visitor *v, const char *name, uint64_t *obj, + Error **errp) +{ + lv_print_key(v, name); + printf("%" PRIu64, *obj); +} + +static void lv_type_bool(Visitor *v, const char *name, bool *obj, Error **errp) +{ + lv_print_key(v, name); + printf("%s", *obj ? "on" : "off"); +} + +static void lv_type_str(Visitor *v, const char *name, char **obj, Error **errp) +{ + const char *str = *obj; + lv_print_key(v, name); + + while (*str) { + if (*str == ',') { + putchar(','); + } + putchar(*str++); + } +} + +static void lv_complete(Visitor *v, void *opaque) +{ + LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v; + assert(lv->path == NULL); +} + +static void lv_free(Visitor *v) +{ + LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v; + + g_list_free_full(lv->path, g_free); + g_free(lv); +} + +static Visitor *legacy_visitor_new(void) +{ + LegacyPrintVisitor *lv = g_malloc0(sizeof(LegacyPrintVisitor)); + + lv->visitor.start_struct = lv_start_struct; + lv->visitor.end_struct = lv_end_struct; + /* lists not supported */ + lv->visitor.type_int64 = lv_type_int64; + lv->visitor.type_uint64 = lv_type_uint64; + lv->visitor.type_bool = lv_type_bool; + lv->visitor.type_str = lv_type_str; + + lv->visitor.type = VISITOR_OUTPUT; + lv->visitor.complete = lv_complete; + lv->visitor.free = lv_free; + + return &lv->visitor; +} + +void audio_legacy_help(void) +{ + AudiodevListHead head; + AudiodevListEntry *e; + + printf("Environment variable based configuration deprecated.\n"); + printf("Please use the new -audiodev option.\n"); + + head = audio_handle_legacy_opts(); + printf("\nEquivalent -audiodev to your current environment variables:\n"); + if (!getenv("QEMU_AUDIO_DRV")) { + printf("(Since you didn't specify QEMU_AUDIO_DRV, I'll list all " + "possibilities)\n"); + } + + QSIMPLEQ_FOREACH(e, &head, next) { + Visitor *v; + Audiodev *dev = e->dev; + printf("-audiodev "); + + v = legacy_visitor_new(); + visit_type_Audiodev(v, NULL, &dev, &error_abort); + visit_free(v); + + printf("\n"); + } + audio_free_audiodev_list(&head); +} diff --git a/audio/coreaudio.c b/audio/coreaudio.c index 638c60b300b2..7d4225dbeeff 100644 --- a/audio/coreaudio.c +++ b/audio/coreaudio.c @@ -685,7 +685,7 @@ static CoreaudioConf glob_conf = { .nbuffers = 4, }; -static void *coreaudio_audio_init (void) +static void *coreaudio_audio_init(Audiodev *dev) { CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf)); *conf = glob_conf; diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c index 3ed73a30d1e3..02fe777cba75 100644 --- a/audio/dsoundaudio.c +++ b/audio/dsoundaudio.c @@ -783,7 +783,7 @@ static void dsound_audio_fini (void *opaque) g_free(s); } -static void *dsound_audio_init (void) +static void *dsound_audio_init(Audiodev *dev) { int err; HRESULT hr; diff --git a/audio/noaudio.c b/audio/noaudio.c index 1bfebeca7da9..79690af1eabd 100644 --- a/audio/noaudio.c +++ b/audio/noaudio.c @@ -136,7 +136,7 @@ static int no_ctl_in (HWVoiceIn *hw, int cmd, ...) return 0; } -static void *no_audio_init (void) +static void *no_audio_init(Audiodev *dev) { return &no_audio_init; } diff --git a/audio/ossaudio.c b/audio/ossaudio.c index 355e8fbda519..e0cadbef29df 100644 --- a/audio/ossaudio.c +++ b/audio/ossaudio.c @@ -842,7 +842,7 @@ static OSSConf glob_conf = { .policy = 5 }; -static void *oss_audio_init (void) +static void *oss_audio_init(Audiodev *dev) { OSSConf *conf = g_malloc(sizeof(OSSConf)); *conf = glob_conf; diff --git a/audio/paaudio.c b/audio/paaudio.c index 8246f260a8ac..d649c58e3d4c 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -812,7 +812,7 @@ static PAConf glob_conf = { .samples = 4096, }; -static void *qpa_audio_init (void) +static void *qpa_audio_init(Audiodev *dev) { if (glob_conf.server == NULL) { char pidfile[64]; diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c index 4cd4cbaf0043..b9ae506a5651 100644 --- a/audio/sdlaudio.c +++ b/audio/sdlaudio.c @@ -315,7 +315,7 @@ static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...) return 0; } -static void *sdl_audio_init (void) +static void *sdl_audio_init(Audiodev *dev) { SDLAudioState *s = &glob_sdl; if (s->driver_created) { diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c index 3aeb0cb35722..affc3df17f90 100644 --- a/audio/spiceaudio.c +++ b/audio/spiceaudio.c @@ -77,7 +77,7 @@ static const SpiceRecordInterface record_sif = { .base.minor_version = SPICE_INTERFACE_RECORD_MINOR, }; -static void *spice_audio_init (void) +static void *spice_audio_init(Audiodev *dev) { if (!using_spice) { return NULL; diff --git a/audio/wavaudio.c b/audio/wavaudio.c index 35a614785ece..9eff3555b35f 100644 --- a/audio/wavaudio.c +++ b/audio/wavaudio.c @@ -232,7 +232,7 @@ static WAVConf glob_conf = { .wav_path = "qemu.wav" }; -static void *wav_audio_init (void) +static void *wav_audio_init(Audiodev *dev) { WAVConf *conf = g_malloc(sizeof(WAVConf)); *conf = glob_conf; diff --git a/vl.c b/vl.c index 4a350de5cd19..db04aea7f8c9 100644 --- a/vl.c +++ b/vl.c @@ -3253,9 +3253,12 @@ int main(int argc, char **argv, char **envp) add_device_config(DEV_BT, optarg); break; case QEMU_OPTION_audio_help: - AUD_help (); + audio_legacy_help(); exit (0); break; + case QEMU_OPTION_audiodev: + audio_parse_option(optarg); + break; case QEMU_OPTION_soundhw: select_soundhw (optarg); break; @@ -4447,6 +4450,8 @@ int main(int argc, char **argv, char **envp) /* do monitor/qmp handling at preconfig state if requested */ main_loop(); + audio_init_audiodevs(); + /* from here on runstate is RUN_STATE_PRELAUNCH */ machine_run_board_init(current_machine); diff --git a/audio/Makefile.objs b/audio/Makefile.objs index db4fa7f18f25..dca87f63470d 100644 --- a/audio/Makefile.objs +++ b/audio/Makefile.objs @@ -1,4 +1,4 @@ -common-obj-y = audio.o noaudio.o wavaudio.o mixeng.o +common-obj-y = audio.o audio_legacy.o noaudio.o wavaudio.o mixeng.o common-obj-$(CONFIG_SPICE) += spiceaudio.o common-obj-$(CONFIG_AUDIO_COREAUDIO) += coreaudio.o common-obj-$(CONFIG_AUDIO_DSOUND) += dsoundaudio.o -- 2.18.1

From: Kővágó, Zoltán <dirty.ice.hu@gmail.com> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: ba58f7e2501b1454b5e0f91c03aad416a404fc6a.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> --- audio/audio_legacy.c | 28 +++++++++++++++++++++++++ audio/coreaudio.c | 49 ++++++++++---------------------------------- 2 files changed, 39 insertions(+), 38 deletions(-) diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c index f58c8b6aee62..5f557a2de9b7 100644 --- a/audio/audio_legacy.c +++ b/audio/audio_legacy.c @@ -109,6 +109,17 @@ static uint32_t frames_to_usecs(uint32_t frames, return (frames * 1000000 + freq / 2) / freq; } + +static void get_frames_to_usecs(const char *env, uint32_t *dst, bool *has_dst, + AudiodevPerDirectionOptions *pdo) +{ + const char *val = getenv(env); + if (val) { + *dst = frames_to_usecs(toui32(val), pdo); + *has_dst = true; + } +} + /* backend specific functions */ /* ALSA */ static void handle_alsa_per_direction( @@ -156,6 +167,19 @@ static void handle_alsa(Audiodev *dev) &aopt->threshold, &aopt->has_threshold); } +/* coreaudio */ +static void handle_coreaudio(Audiodev *dev) +{ + get_frames_to_usecs( + "QEMU_COREAUDIO_BUFFER_SIZE", + &dev->u.coreaudio.out->buffer_length, + &dev->u.coreaudio.out->has_buffer_length, + qapi_AudiodevCoreaudioPerDirectionOptions_base(dev->u.coreaudio.out)); + get_int("QEMU_COREAUDIO_BUFFER_COUNT", + &dev->u.coreaudio.out->buffer_count, + &dev->u.coreaudio.out->has_buffer_count); +} + /* general */ static void handle_per_direction( AudiodevPerDirectionOptions *pdo, const char *prefix) @@ -201,6 +225,10 @@ static AudiodevListEntry *legacy_opt(const char *drvname) handle_alsa(e->dev); break; + case AUDIODEV_DRIVER_COREAUDIO: + handle_coreaudio(e->dev); + break; + default: break; } diff --git a/audio/coreaudio.c b/audio/coreaudio.c index 7d4225dbeeff..1ee43b7d5f12 100644 --- a/audio/coreaudio.c +++ b/audio/coreaudio.c @@ -36,11 +36,6 @@ #define MAC_OS_X_VERSION_10_6 1060 #endif -typedef struct { - int buffer_frames; - int nbuffers; -} CoreaudioConf; - typedef struct coreaudioVoiceOut { HWVoiceOut hw; pthread_mutex_t mutex; @@ -507,7 +502,9 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as, int err; const char *typ = "playback"; AudioValueRange frameRange; - CoreaudioConf *conf = drv_opaque; + Audiodev *dev = drv_opaque; + AudiodevCoreaudioPerDirectionOptions *cpdo = dev->u.coreaudio.out; + int frames; /* create mutex */ err = pthread_mutex_init(&core->mutex, NULL); @@ -538,16 +535,17 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as, return -1; } - if (frameRange.mMinimum > conf->buffer_frames) { + frames = audio_buffer_frames( + qapi_AudiodevCoreaudioPerDirectionOptions_base(cpdo), as, 11610); + if (frameRange.mMinimum > frames) { core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum; dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum); - } - else if (frameRange.mMaximum < conf->buffer_frames) { + } else if (frameRange.mMaximum < frames) { core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum; dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum); } else { - core->audioDevicePropertyBufferFrameSize = conf->buffer_frames; + core->audioDevicePropertyBufferFrameSize = frames; } /* set Buffer Frame Size */ @@ -568,7 +566,8 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as, "Could not get device buffer frame size\n"); return -1; } - hw->samples = conf->nbuffers * core->audioDevicePropertyBufferFrameSize; + hw->samples = (cpdo->has_buffer_count ? cpdo->buffer_count : 4) * + core->audioDevicePropertyBufferFrameSize; /* get StreamFormat */ status = coreaudio_get_streamformat(core->outputDeviceID, @@ -680,40 +679,15 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...) return 0; } -static CoreaudioConf glob_conf = { - .buffer_frames = 512, - .nbuffers = 4, -}; - static void *coreaudio_audio_init(Audiodev *dev) { - CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf)); - *conf = glob_conf; - - return conf; + return dev; } static void coreaudio_audio_fini (void *opaque) { - g_free(opaque); } -static struct audio_option coreaudio_options[] = { - { - .name = "BUFFER_SIZE", - .tag = AUD_OPT_INT, - .valp = &glob_conf.buffer_frames, - .descr = "Size of the buffer in frames" - }, - { - .name = "BUFFER_COUNT", - .tag = AUD_OPT_INT, - .valp = &glob_conf.nbuffers, - .descr = "Number of buffers" - }, - { /* End of list */ } -}; - static struct audio_pcm_ops coreaudio_pcm_ops = { .init_out = coreaudio_init_out, .fini_out = coreaudio_fini_out, @@ -725,7 +699,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = { static struct audio_driver coreaudio_audio_driver = { .name = "coreaudio", .descr = "CoreAudio http://developer.apple.com/audio/coreaudio.html", - .options = coreaudio_options, .init = coreaudio_audio_init, .fini = coreaudio_audio_fini, .pcm_ops = &coreaudio_pcm_ops, -- 2.18.1

From: Kővágó, Zoltán <dirty.ice.hu@gmail.com> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 0c35b1956ed1105244309a774c038816c040108c.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> --- audio/noaudio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/audio/noaudio.c b/audio/noaudio.c index 79690af1eabd..ccc611fc8470 100644 --- a/audio/noaudio.c +++ b/audio/noaudio.c @@ -163,7 +163,6 @@ static struct audio_pcm_ops no_pcm_ops = { static struct audio_driver no_audio_driver = { .name = "none", .descr = "Timer based audio emulation", - .options = NULL, .init = no_audio_init, .fini = no_audio_fini, .pcm_ops = &no_pcm_ops, -- 2.18.1

From: Kővágó, Zoltán <dirty.ice.hu@gmail.com> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: c74dc9c282075fba6928c40b2deae057fa0d4049.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> --- audio/audio_legacy.c | 38 +++++++++++++++++++++ audio/paaudio.c | 81 ++++++++++++++++---------------------------- 2 files changed, 67 insertions(+), 52 deletions(-) diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c index 94f95bbc7318..8d7f1475b56a 100644 --- a/audio/audio_legacy.c +++ b/audio/audio_legacy.c @@ -127,6 +127,16 @@ static uint32_t samples_to_usecs(uint32_t samples, return frames_to_usecs(samples / channels, pdo); } +static void get_samples_to_usecs(const char *env, uint32_t *dst, bool *has_dst, + AudiodevPerDirectionOptions *pdo) +{ + const char *val = getenv(env); + if (val) { + *dst = samples_to_usecs(toui32(val), pdo); + *has_dst = true; + } +} + static uint32_t bytes_to_usecs(uint32_t bytes, AudiodevPerDirectionOptions *pdo) { AudioFormat fmt = pdo->has_format ? pdo->format : AUDIO_FORMAT_S16; @@ -247,6 +257,30 @@ static void handle_oss(Audiodev *dev) get_int("QEMU_OSS_POLICY", &oopt->dsp_policy, &oopt->has_dsp_policy); } +/* pulseaudio */ +static void handle_pa_per_direction( + AudiodevPaPerDirectionOptions *ppdo, const char *env) +{ + get_str(env, &ppdo->name, &ppdo->has_name); +} + +static void handle_pa(Audiodev *dev) +{ + handle_pa_per_direction(dev->u.pa.in, "QEMU_PA_SOURCE"); + handle_pa_per_direction(dev->u.pa.out, "QEMU_PA_SINK"); + + get_samples_to_usecs( + "QEMU_PA_SAMPLES", &dev->u.pa.in->buffer_length, + &dev->u.pa.in->has_buffer_length, + qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.in)); + get_samples_to_usecs( + "QEMU_PA_SAMPLES", &dev->u.pa.out->buffer_length, + &dev->u.pa.out->has_buffer_length, + qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.out)); + + get_str("QEMU_PA_SERVER", &dev->u.pa.server, &dev->u.pa.has_server); +} + /* general */ static void handle_per_direction( AudiodevPerDirectionOptions *pdo, const char *prefix) @@ -304,6 +338,10 @@ static AudiodevListEntry *legacy_opt(const char *drvname) handle_oss(e->dev); break; + case AUDIODEV_DRIVER_PA: + handle_pa(e->dev); + break; + default: break; } diff --git a/audio/paaudio.c b/audio/paaudio.c index d649c58e3d4c..5d410ed73f50 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -2,6 +2,7 @@ #include "qemu/osdep.h" #include "qemu-common.h" #include "audio.h" +#include "qapi/opts-visitor.h" #include <pulse/pulseaudio.h> @@ -10,14 +11,7 @@ #include "audio_pt_int.h" typedef struct { - int samples; - char *server; - char *sink; - char *source; -} PAConf; - -typedef struct { - PAConf conf; + Audiodev *dev; pa_threaded_mainloop *mainloop; pa_context *context; } paaudio; @@ -32,6 +26,7 @@ typedef struct { void *pcm_buf; struct audio_pt pt; paaudio *g; + int samples; } PAVoiceOut; typedef struct { @@ -46,6 +41,7 @@ typedef struct { const void *read_data; size_t read_index, read_length; paaudio *g; + int samples; } PAVoiceIn; static void qpa_audio_fini(void *opaque); @@ -227,7 +223,7 @@ static void *qpa_thread_out (void *arg) } } - decr = to_mix = audio_MIN(pa->live, pa->g->conf.samples >> 5); + decr = to_mix = audio_MIN(pa->live, pa->samples >> 5); rpos = pa->rpos; if (audio_pt_unlock(&pa->pt, __func__)) { @@ -319,7 +315,7 @@ static void *qpa_thread_in (void *arg) } } - incr = to_grab = audio_MIN(pa->dead, pa->g->conf.samples >> 5); + incr = to_grab = audio_MIN(pa->dead, pa->samples >> 5); wpos = pa->wpos; if (audio_pt_unlock(&pa->pt, __func__)) { @@ -546,6 +542,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as, struct audsettings obt_as = *as; PAVoiceOut *pa = (PAVoiceOut *) hw; paaudio *g = pa->g = drv_opaque; + AudiodevPaOptions *popts = &g->dev->u.pa; + AudiodevPaPerDirectionOptions *ppdo = popts->out; ss.format = audfmt_to_pa (as->fmt, as->endianness); ss.channels = as->nchannels; @@ -566,7 +564,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as, g, "qemu", PA_STREAM_PLAYBACK, - g->conf.sink, + ppdo->has_name ? ppdo->name : NULL, &ss, NULL, /* channel map */ &ba, /* buffering attributes */ @@ -578,7 +576,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as, } audio_pcm_init_info (&hw->info, &obt_as); - hw->samples = g->conf.samples; + hw->samples = pa->samples = audio_buffer_samples( + qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440); pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift); pa->rpos = hw->rpos; if (!pa->pcm_buf) { @@ -612,6 +611,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) struct audsettings obt_as = *as; PAVoiceIn *pa = (PAVoiceIn *) hw; paaudio *g = pa->g = drv_opaque; + AudiodevPaOptions *popts = &g->dev->u.pa; + AudiodevPaPerDirectionOptions *ppdo = popts->in; ss.format = audfmt_to_pa (as->fmt, as->endianness); ss.channels = as->nchannels; @@ -623,7 +624,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) g, "qemu", PA_STREAM_RECORD, - g->conf.source, + ppdo->has_name ? ppdo->name : NULL, &ss, NULL, /* channel map */ NULL, /* buffering attributes */ @@ -635,7 +636,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) } audio_pcm_init_info (&hw->info, &obt_as); - hw->samples = g->conf.samples; + hw->samples = pa->samples = audio_buffer_samples( + qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440); pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift); pa->wpos = hw->wpos; if (!pa->pcm_buf) { @@ -808,13 +810,13 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...) } /* common */ -static PAConf glob_conf = { - .samples = 4096, -}; - static void *qpa_audio_init(Audiodev *dev) { - if (glob_conf.server == NULL) { + paaudio *g; + AudiodevPaOptions *popts = &dev->u.pa; + const char *server; + + if (!popts->has_server) { char pidfile[64]; char *runtime; struct stat st; @@ -829,8 +831,12 @@ static void *qpa_audio_init(Audiodev *dev) } } - paaudio *g = g_malloc(sizeof(paaudio)); - g->conf = glob_conf; + assert(dev->driver == AUDIODEV_DRIVER_PA); + + g = g_malloc(sizeof(paaudio)); + server = popts->has_server ? popts->server : NULL; + + g->dev = dev; g->mainloop = NULL; g->context = NULL; @@ -840,14 +846,14 @@ static void *qpa_audio_init(Audiodev *dev) } g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop), - g->conf.server); + server); if (!g->context) { goto fail; } pa_context_set_state_callback (g->context, context_state_cb, g); - if (pa_context_connect (g->context, g->conf.server, 0, NULL) < 0) { + if (pa_context_connect(g->context, server, 0, NULL) < 0) { qpa_logerr (pa_context_errno (g->context), "pa_context_connect() failed\n"); goto fail; @@ -910,34 +916,6 @@ static void qpa_audio_fini (void *opaque) g_free(g); } -struct audio_option qpa_options[] = { - { - .name = "SAMPLES", - .tag = AUD_OPT_INT, - .valp = &glob_conf.samples, - .descr = "buffer size in samples" - }, - { - .name = "SERVER", - .tag = AUD_OPT_STR, - .valp = &glob_conf.server, - .descr = "server address" - }, - { - .name = "SINK", - .tag = AUD_OPT_STR, - .valp = &glob_conf.sink, - .descr = "sink device name" - }, - { - .name = "SOURCE", - .tag = AUD_OPT_STR, - .valp = &glob_conf.source, - .descr = "source device name" - }, - { /* End of list */ } -}; - static struct audio_pcm_ops qpa_pcm_ops = { .init_out = qpa_init_out, .fini_out = qpa_fini_out, @@ -955,7 +933,6 @@ static struct audio_pcm_ops qpa_pcm_ops = { static struct audio_driver pa_audio_driver = { .name = "pa", .descr = "http://www.pulseaudio.org/", - .options = qpa_options, .init = qpa_audio_init, .fini = qpa_audio_fini, .pcm_ops = &qpa_pcm_ops, -- 2.18.1

From: Kővágó, Zoltán <dirty.ice.hu@gmail.com> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 040d1241f29104ecfeecb26b0a14f0d042484fc5.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> --- audio/spiceaudio.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c index affc3df17f90..4f7873af5afd 100644 --- a/audio/spiceaudio.c +++ b/audio/spiceaudio.c @@ -373,10 +373,6 @@ static int line_in_ctl (HWVoiceIn *hw, int cmd, ...) return 0; } -static struct audio_option audio_options[] = { - { /* end of list */ }, -}; - static struct audio_pcm_ops audio_callbacks = { .init_out = line_out_init, .fini_out = line_out_fini, @@ -394,7 +390,6 @@ static struct audio_pcm_ops audio_callbacks = { static struct audio_driver spice_audio_driver = { .name = "spice", .descr = "spice audio driver", - .options = audio_options, .init = spice_audio_init, .fini = spice_audio_fini, .pcm_ops = &audio_callbacks, -- 2.18.1

From: Kővágó, Zoltán <dirty.ice.hu@gmail.com> Remove no longer needed code. Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: 82c176a5222a56e4fde45d2b423ada797caddd5d.1552083282.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com> --- audio/audio_int.h | 17 ----- audio/audio.c | 186 +--------------------------------------------- 2 files changed, 4 insertions(+), 199 deletions(-) diff --git a/audio/audio_int.h b/audio/audio_int.h index 7bf5dfc0b5f6..3f14842709bf 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -33,22 +33,6 @@ struct audio_pcm_ops; -typedef enum { - AUD_OPT_INT, - AUD_OPT_FMT, - AUD_OPT_STR, - AUD_OPT_BOOL -} audio_option_tag_e; - -struct audio_option { - const char *name; - audio_option_tag_e tag; - void *valp; - const char *descr; - int *overriddenp; - int overridden; -}; - struct audio_callback { void *opaque; audio_callback_fn fn; @@ -145,7 +129,6 @@ typedef struct audio_driver audio_driver; struct audio_driver { const char *name; const char *descr; - struct audio_option *options; void *(*init) (Audiodev *); void (*fini) (void *); struct audio_pcm_ops *pcm_ops; diff --git a/audio/audio.c b/audio/audio.c index 79ed36066604..5fd9a58a808f 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -172,113 +172,6 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size) return g_malloc0 (len); } -static const char *audio_audfmt_to_string (AudioFormat fmt) -{ - switch (fmt) { - case AUDIO_FORMAT_U8: - return "U8"; - - case AUDIO_FORMAT_U16: - return "U16"; - - case AUDIO_FORMAT_S8: - return "S8"; - - case AUDIO_FORMAT_S16: - return "S16"; - - case AUDIO_FORMAT_U32: - return "U32"; - - case AUDIO_FORMAT_S32: - return "S32"; - - default: - abort(); - } - - dolog ("Bogus audfmt %d returning S16\n", fmt); - return "S16"; -} - -static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval, - int *defaultp) -{ - if (!strcasecmp (s, "u8")) { - *defaultp = 0; - return AUDIO_FORMAT_U8; - } - else if (!strcasecmp (s, "u16")) { - *defaultp = 0; - return AUDIO_FORMAT_U16; - } - else if (!strcasecmp (s, "u32")) { - *defaultp = 0; - return AUDIO_FORMAT_U32; - } - else if (!strcasecmp (s, "s8")) { - *defaultp = 0; - return AUDIO_FORMAT_S8; - } - else if (!strcasecmp (s, "s16")) { - *defaultp = 0; - return AUDIO_FORMAT_S16; - } - else if (!strcasecmp (s, "s32")) { - *defaultp = 0; - return AUDIO_FORMAT_S32; - } - else { - dolog ("Bogus audio format `%s' using %s\n", - s, audio_audfmt_to_string (defval)); - *defaultp = 1; - return defval; - } -} - -static AudioFormat audio_get_conf_fmt (const char *envname, - AudioFormat defval, - int *defaultp) -{ - const char *var = getenv (envname); - if (!var) { - *defaultp = 1; - return defval; - } - return audio_string_to_audfmt (var, defval, defaultp); -} - -static int audio_get_conf_int (const char *key, int defval, int *defaultp) -{ - int val; - char *strval; - - strval = getenv (key); - if (strval && !qemu_strtoi(strval, NULL, 10, &val)) { - *defaultp = 0; - return val; - } - else { - *defaultp = 1; - return defval; - } -} - -static const char *audio_get_conf_str (const char *key, - const char *defval, - int *defaultp) -{ - const char *val = getenv (key); - if (!val) { - *defaultp = 1; - return defval; - } - else { - *defaultp = 0; - return val; - } -} - void AUD_vlog (const char *cap, const char *fmt, va_list ap) { if (cap) { @@ -297,74 +190,6 @@ void AUD_log (const char *cap, const char *fmt, ...) va_end (ap); } -static void audio_process_options (const char *prefix, - struct audio_option *opt) -{ - gchar *prefix_upper; - - if (audio_bug(__func__, !prefix)) { - dolog ("prefix = NULL\n"); - return; - } - - if (audio_bug(__func__, !opt)) { - dolog ("opt = NULL\n"); - return; - } - - prefix_upper = g_utf8_strup(prefix, -1); - - for (; opt->name; opt++) { - char *optname; - int def; - - if (!opt->valp) { - dolog ("Option value pointer for `%s' is not set\n", - opt->name); - continue; - } - - optname = g_strdup_printf("QEMU_%s_%s", prefix_upper, opt->name); - - def = 1; - switch (opt->tag) { - case AUD_OPT_BOOL: - case AUD_OPT_INT: - { - int *intp = opt->valp; - *intp = audio_get_conf_int (optname, *intp, &def); - } - break; - - case AUD_OPT_FMT: - { - AudioFormat *fmtp = opt->valp; - *fmtp = audio_get_conf_fmt (optname, *fmtp, &def); - } - break; - - case AUD_OPT_STR: - { - const char **strp = opt->valp; - *strp = audio_get_conf_str (optname, *strp, &def); - } - break; - - default: - dolog ("Bad value tag for option `%s' - %d\n", - optname, opt->tag); - break; - } - - if (!opt->overriddenp) { - opt->overriddenp = &opt->overridden; - } - *opt->overriddenp = !def; - g_free (optname); - } - g_free(prefix_upper); -} - static void audio_print_settings (struct audsettings *as) { dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels); @@ -1072,7 +897,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on) if (!hw->enabled) { hw->enabled = 1; if (s->vm_running) { - hw->pcm_ops->ctl_out(hw, VOICE_ENABLE, true); + hw->pcm_ops->ctl_out(hw, VOICE_ENABLE); audio_reset_timer (s); } } @@ -1117,7 +942,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on) if (!hw->enabled) { hw->enabled = 1; if (s->vm_running) { - hw->pcm_ops->ctl_in(hw, VOICE_ENABLE, true); + hw->pcm_ops->ctl_in(hw, VOICE_ENABLE); audio_reset_timer (s); } } @@ -1441,9 +1266,6 @@ void audio_run (const char *msg) static int audio_driver_init(AudioState *s, struct audio_driver *drv, bool msg, Audiodev *dev) { - if (drv->options) { - audio_process_options (drv->name, drv->options); - } s->drv_opaque = drv->init(dev); if (s->drv_opaque) { @@ -1470,11 +1292,11 @@ static void audio_vm_change_state_handler (void *opaque, int running, s->vm_running = running; while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) { - hwo->pcm_ops->ctl_out(hwo, op, true); + hwo->pcm_ops->ctl_out(hwo, op); } while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) { - hwi->pcm_ops->ctl_in(hwi, op, true); + hwi->pcm_ops->ctl_in(hwi, op); } audio_reset_timer (s); } -- 2.18.1