From: Marc-André Lureau <[email protected]> The endianness field used an int to represent a boolean concept, with 0 meaning little-endian and 1 meaning big-endian. This required runtime validation to reject invalid values and made the code less readable.
Replace with a bool big_endian field that is self-documenting and type-safe. The compiler now enforces valid values, eliminating the need for the validation check in audio_validate_settings(). Reviewed-by: Thomas Huth <[email protected]> Reviewed-by: Mark Cave-Ayland <[email protected]> Reviewed-by: Akihiko Odaki <[email protected]> Signed-off-by: Marc-André Lureau <[email protected]> --- audio/dsound_template.h | 2 +- include/qemu/audio.h | 2 +- audio/alsaaudio.c | 8 ++++---- audio/audio-mixeng-be.c | 21 ++++----------------- audio/jackaudio.c | 4 ++-- audio/ossaudio.c | 22 +++++++++++----------- audio/paaudio.c | 22 +++++++++++----------- audio/pwaudio.c | 30 +++++++++++++++--------------- audio/sdlaudio.c | 26 +++++++++++++------------- audio/sndioaudio.c | 2 +- audio/spiceaudio.c | 4 ++-- audio/wavaudio.c | 2 +- audio/wavcapture.c | 2 +- hw/audio/ac97.c | 2 +- hw/audio/adlib.c | 2 +- hw/audio/asc.c | 2 +- hw/audio/cs4231a.c | 6 +++--- hw/audio/es1370.c | 2 +- hw/audio/gus.c | 2 +- hw/audio/lm4549.c | 6 +++--- hw/audio/sb16.c | 8 ++++---- hw/audio/via-ac97.c | 2 +- hw/audio/virtio-snd.c | 2 +- hw/audio/wm8750.c | 4 ++-- hw/display/xlnx_dp.c | 2 +- hw/usb/dev-audio.c | 2 +- tests/audio/test-audio.c | 2 +- ui/vnc.c | 2 +- 28 files changed, 90 insertions(+), 103 deletions(-) diff --git a/audio/dsound_template.h b/audio/dsound_template.h index af4019bcb34..f9761120875 100644 --- a/audio/dsound_template.h +++ b/audio/dsound_template.h @@ -244,7 +244,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as) } ds->first_time = true; - obt_as.endianness = 0; + obt_as.big_endian = false; audio_pcm_init_info (&hw->info, &obt_as); if (bc.dwBufferBytes % hw->info.bytes_per_frame) { diff --git a/include/qemu/audio.h b/include/qemu/audio.h index 9e4566d60aa..cff8a334f36 100644 --- a/include/qemu/audio.h +++ b/include/qemu/audio.h @@ -38,7 +38,7 @@ typedef struct audsettings { int freq; int nchannels; AudioFormat fmt; - int endianness; + bool big_endian; } audsettings; typedef struct SWVoiceOut SWVoiceOut; diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 889240ad0c7..e2290ea814a 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -666,7 +666,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as) struct audsettings obt_as; Audiodev *dev = hw->s->dev; - req.fmt = aud_to_alsafmt (as->fmt, as->endianness); + req.fmt = aud_to_alsafmt (as->fmt, as->big_endian); req.freq = as->freq; req.nchannels = as->nchannels; @@ -677,7 +677,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as) obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; obt_as.fmt = obt.fmt; - obt_as.endianness = obt.endianness; + obt_as.big_endian = obt.endianness; audio_pcm_init_info (&hw->info, &obt_as); hw->samples = obt.samples; @@ -752,7 +752,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as) struct audsettings obt_as; Audiodev *dev = hw->s->dev; - req.fmt = aud_to_alsafmt (as->fmt, as->endianness); + req.fmt = aud_to_alsafmt (as->fmt, as->big_endian); req.freq = as->freq; req.nchannels = as->nchannels; @@ -763,7 +763,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as) obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; obt_as.fmt = obt.fmt; - obt_as.endianness = obt.endianness; + obt_as.big_endian = obt.endianness; audio_pcm_init_info (&hw->info, &obt_as); hw->samples = obt.samples; diff --git a/audio/audio-mixeng-be.c b/audio/audio-mixeng-be.c index 931d5b206f2..d05202aabae 100644 --- a/audio/audio-mixeng-be.c +++ b/audio/audio-mixeng-be.c @@ -135,19 +135,7 @@ static void audio_print_settings (const struct audsettings *as) break; } - AUD_log (NULL, " endianness="); - switch (as->endianness) { - case 0: - AUD_log (NULL, "little"); - break; - case 1: - AUD_log (NULL, "big"); - break; - default: - AUD_log (NULL, "invalid"); - break; - } - AUD_log (NULL, "\n"); + AUD_log (NULL, " endianness=%s\n", as->big_endian ? "big" : "little"); } static int audio_validate_settings (const struct audsettings *as) @@ -155,7 +143,6 @@ static int audio_validate_settings (const struct audsettings *as) int invalid; invalid = as->nchannels < 1; - invalid |= as->endianness != 0 && as->endianness != 1; switch (as->fmt) { case AUDIO_FORMAT_S8: @@ -180,7 +167,7 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, const struct audsetti return info->af == as->fmt && info->freq == as->freq && info->nchannels == as->nchannels - && info->swap_endianness == (as->endianness != HOST_BIG_ENDIAN); + && info->swap_endianness == (as->big_endian != HOST_BIG_ENDIAN); } void audio_pcm_init_info (struct audio_pcm_info *info, const struct audsettings *as) @@ -190,7 +177,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, const struct audsettings info->nchannels = as->nchannels; info->bytes_per_frame = as->nchannels * audio_format_bits(as->fmt) / 8; info->bytes_per_second = info->freq * info->bytes_per_frame; - info->swap_endianness = (as->endianness != HOST_BIG_ENDIAN); + info->swap_endianness = (as->big_endian != HOST_BIG_ENDIAN); } void audio_pcm_info_clear_buf(struct audio_pcm_info *info, void *buf, int len) @@ -1797,7 +1784,7 @@ audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo) .freq = pdo->frequency, .nchannels = pdo->channels, .fmt = pdo->format, - .endianness = HOST_BIG_ENDIAN, + .big_endian = HOST_BIG_ENDIAN, }; } diff --git a/audio/jackaudio.c b/audio/jackaudio.c index a2f725a3232..d9ae8edc0e0 100644 --- a/audio/jackaudio.c +++ b/audio/jackaudio.c @@ -531,7 +531,7 @@ static int qjack_init_out(HWVoiceOut *hw, struct audsettings *as) .freq = jo->c.freq, .nchannels = jo->c.nchannels, .fmt = AUDIO_FORMAT_F32, - .endianness = 0 + .big_endian = false }; audio_pcm_init_info(&hw->info, &os); @@ -566,7 +566,7 @@ static int qjack_init_in(HWVoiceIn *hw, struct audsettings *as) .freq = ji->c.freq, .nchannels = ji->c.nchannels, .fmt = AUDIO_FORMAT_F32, - .endianness = 0 + .big_endian = false }; audio_pcm_init_info(&hw->info, &is); diff --git a/audio/ossaudio.c b/audio/ossaudio.c index 81f49f9a7e7..157979e84b4 100644 --- a/audio/ossaudio.c +++ b/audio/ossaudio.c @@ -160,36 +160,36 @@ static int aud_to_ossfmt(AudioFormat fmt, bool big_endian) } } -static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness) +static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, bool *big_endian) { switch (ossfmt) { case AFMT_S8: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_S8; break; case AFMT_U8: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_U8; break; case AFMT_S16_LE: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_S16; break; case AFMT_U16_LE: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_U16; break; case AFMT_S16_BE: - *endianness = 1; + *big_endian = true; *fmt = AUDIO_FORMAT_S16; break; case AFMT_U16_BE: - *endianness = 1; + *big_endian = true; *fmt = AUDIO_FORMAT_U16; break; @@ -475,7 +475,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as) oss->fd = -1; - req.fmt = aud_to_ossfmt (as->fmt, as->endianness); + req.fmt = aud_to_ossfmt (as->fmt, as->big_endian); req.freq = as->freq; req.nchannels = as->nchannels; @@ -483,7 +483,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as) return -1; } - err = oss_to_audfmt(obt.fmt, &obt_as.fmt, &obt_as.endianness); + err = oss_to_audfmt(obt.fmt, &obt_as.fmt, &obt_as.big_endian); if (err) { oss_anal_close (&fd); return -1; @@ -601,14 +601,14 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as) oss->fd = -1; - req.fmt = aud_to_ossfmt (as->fmt, as->endianness); + req.fmt = aud_to_ossfmt (as->fmt, as->big_endian); req.freq = as->freq; req.nchannels = as->nchannels; if (oss_open(1, &req, as, &obt, &fd, dev)) { return -1; } - err = oss_to_audfmt(obt.fmt, &obt_as.fmt, &obt_as.endianness); + err = oss_to_audfmt(obt.fmt, &obt_as.fmt, &obt_as.big_endian); if (err) { oss_anal_close (&fd); return -1; diff --git a/audio/paaudio.c b/audio/paaudio.c index 1501b26386a..c9957105543 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -352,28 +352,28 @@ static pa_sample_format_t audfmt_to_pa(AudioFormat afmt, bool big_endian) return format; } -static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness) +static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, bool *big_endian) { switch (fmt) { case PA_SAMPLE_U8: return AUDIO_FORMAT_U8; case PA_SAMPLE_S16BE: - *endianness = 1; + *big_endian = true; return AUDIO_FORMAT_S16; case PA_SAMPLE_S16LE: - *endianness = 0; + *big_endian = false; return AUDIO_FORMAT_S16; case PA_SAMPLE_S32BE: - *endianness = 1; + *big_endian = true; return AUDIO_FORMAT_S32; case PA_SAMPLE_S32LE: - *endianness = 0; + *big_endian = false; return AUDIO_FORMAT_S32; case PA_SAMPLE_FLOAT32BE: - *endianness = 1; + *big_endian = true; return AUDIO_FORMAT_F32; case PA_SAMPLE_FLOAT32LE: - *endianness = 0; + *big_endian = false; return AUDIO_FORMAT_F32; default: error_report("pulseaudio: Internal logic error: Bad pa_sample_format %d", fmt); @@ -531,7 +531,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as) PAConnection *c = apa->conn; pa->g = apa; - ss.format = audfmt_to_pa (as->fmt, as->endianness); + ss.format = audfmt_to_pa (as->fmt, as->big_endian); ss.channels = as->nchannels; ss.rate = as->freq; @@ -541,7 +541,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as) ba.maxlength = -1; ba.prebuf = -1; - obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness); + obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.big_endian); pa->stream = qpa_simple_new ( c, @@ -582,7 +582,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as) PAConnection *c = apa->conn; pa->g = apa; - ss.format = audfmt_to_pa (as->fmt, as->endianness); + ss.format = audfmt_to_pa (as->fmt, as->big_endian); ss.channels = as->nchannels; ss.rate = as->freq; @@ -592,7 +592,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as) ba.minreq = -1; ba.prebuf = -1; - obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness); + obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.big_endian); pa->stream = qpa_simple_new ( c, diff --git a/audio/pwaudio.c b/audio/pwaudio.c index 6a80e5a333e..b70bf0c1353 100644 --- a/audio/pwaudio.c +++ b/audio/pwaudio.c @@ -365,7 +365,7 @@ audfmt_to_pw(AudioFormat fmt, bool big_endian) } static AudioFormat -pw_to_audfmt(enum spa_audio_format fmt, int *endianness, +pw_to_audfmt(enum spa_audio_format fmt, bool *big_endian, uint32_t *sample_size) { switch (fmt) { @@ -377,43 +377,43 @@ pw_to_audfmt(enum spa_audio_format fmt, int *endianness, return AUDIO_FORMAT_U8; case SPA_AUDIO_FORMAT_S16_BE: *sample_size = 2; - *endianness = 1; + *big_endian = true; return AUDIO_FORMAT_S16; case SPA_AUDIO_FORMAT_S16_LE: *sample_size = 2; - *endianness = 0; + *big_endian = false; return AUDIO_FORMAT_S16; case SPA_AUDIO_FORMAT_U16_BE: *sample_size = 2; - *endianness = 1; + *big_endian = true; return AUDIO_FORMAT_U16; case SPA_AUDIO_FORMAT_U16_LE: *sample_size = 2; - *endianness = 0; + *big_endian = false; return AUDIO_FORMAT_U16; case SPA_AUDIO_FORMAT_S32_BE: *sample_size = 4; - *endianness = 1; + *big_endian = true; return AUDIO_FORMAT_S32; case SPA_AUDIO_FORMAT_S32_LE: *sample_size = 4; - *endianness = 0; + *big_endian = false; return AUDIO_FORMAT_S32; case SPA_AUDIO_FORMAT_U32_BE: *sample_size = 4; - *endianness = 1; + *big_endian = true; return AUDIO_FORMAT_U32; case SPA_AUDIO_FORMAT_U32_LE: *sample_size = 4; - *endianness = 0; + *big_endian = false; return AUDIO_FORMAT_U32; case SPA_AUDIO_FORMAT_F32_BE: *sample_size = 4; - *endianness = 1; + *big_endian = true; return AUDIO_FORMAT_F32; case SPA_AUDIO_FORMAT_F32_LE: *sample_size = 4; - *endianness = 0; + *big_endian = false; return AUDIO_FORMAT_F32; default: *sample_size = 1; @@ -534,13 +534,13 @@ qpw_init_out(HWVoiceOut *hw, struct audsettings *as) pw_thread_loop_lock(c->thread_loop); - v->info.format = audfmt_to_pw(as->fmt, as->endianness); + v->info.format = audfmt_to_pw(as->fmt, as->big_endian); v->info.channels = as->nchannels; qpw_set_position(as->nchannels, v->info.position); v->info.rate = as->freq; obt_as.fmt = - pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size); + pw_to_audfmt(v->info.format, &obt_as.big_endian, &v->frame_size); v->frame_size *= as->nchannels; v->req = (uint64_t)AUDIO_MIXENG_BACKEND(c)->dev->timer_period * v->info.rate @@ -581,13 +581,13 @@ qpw_init_in(HWVoiceIn *hw, struct audsettings *as) pw_thread_loop_lock(c->thread_loop); - v->info.format = audfmt_to_pw(as->fmt, as->endianness); + v->info.format = audfmt_to_pw(as->fmt, as->big_endian); v->info.channels = as->nchannels; qpw_set_position(as->nchannels, v->info.position); v->info.rate = as->freq; obt_as.fmt = - pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size); + pw_to_audfmt(v->info.format, &obt_as.big_endian, &v->frame_size); v->frame_size *= as->nchannels; /* call the function that creates a new stream for recording */ diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c index 5f332678209..936a3ed076d 100644 --- a/audio/sdlaudio.c +++ b/audio/sdlaudio.c @@ -96,56 +96,56 @@ static int aud_to_sdlfmt (AudioFormat fmt) } } -static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness) +static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, bool *big_endian) { switch (sdlfmt) { case AUDIO_S8: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_S8; break; case AUDIO_U8: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_U8; break; case AUDIO_S16LSB: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_S16; break; case AUDIO_U16LSB: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_U16; break; case AUDIO_S16MSB: - *endianness = 1; + *big_endian = true; *fmt = AUDIO_FORMAT_S16; break; case AUDIO_U16MSB: - *endianness = 1; + *big_endian = true; *fmt = AUDIO_FORMAT_U16; break; case AUDIO_S32LSB: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_S32; break; case AUDIO_S32MSB: - *endianness = 1; + *big_endian = true; *fmt = AUDIO_FORMAT_S32; break; case AUDIO_F32LSB: - *endianness = 0; + *big_endian = false; *fmt = AUDIO_FORMAT_F32; break; case AUDIO_F32MSB: - *endianness = 1; + *big_endian = true; *fmt = AUDIO_FORMAT_F32; break; @@ -350,7 +350,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as) return -1; } - err = sdl_to_audfmt(obt.format, &obt_as.fmt, &obt_as.endianness); + err = sdl_to_audfmt(obt.format, &obt_as.fmt, &obt_as.big_endian); if (err) { sdl_close_out(sdl); return -1; @@ -406,7 +406,7 @@ static int sdl_init_in(HWVoiceIn *hw, audsettings *as) return -1; } - err = sdl_to_audfmt(obt.format, &obt_as.fmt, &obt_as.endianness); + err = sdl_to_audfmt(obt.format, &obt_as.fmt, &obt_as.big_endian); if (err) { sdl_close_in(sdl); return -1; diff --git a/audio/sndioaudio.c b/audio/sndioaudio.c index 766eceeeefe..761f23f8273 100644 --- a/audio/sndioaudio.c +++ b/audio/sndioaudio.c @@ -387,7 +387,7 @@ static int sndio_init(SndioVoice *self, } if (req.bits > 8) { - req.le = as->endianness ? 0 : 1; + req.le = !as->big_endian; } req.rate = as->freq; diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c index 81b897e13e0..f903ef77e72 100644 --- a/audio/spiceaudio.c +++ b/audio/spiceaudio.c @@ -106,7 +106,7 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as) #endif settings.nchannels = SPICE_INTERFACE_PLAYBACK_CHAN; settings.fmt = AUDIO_FORMAT_S16; - settings.endianness = HOST_BIG_ENDIAN; + settings.big_endian = HOST_BIG_ENDIAN; audio_pcm_init_info (&hw->info, &settings); hw->samples = LINE_OUT_SAMPLES; @@ -222,7 +222,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as) #endif settings.nchannels = SPICE_INTERFACE_RECORD_CHAN; settings.fmt = AUDIO_FORMAT_S16; - settings.endianness = HOST_BIG_ENDIAN; + settings.big_endian = HOST_BIG_ENDIAN; audio_pcm_init_info (&hw->info, &settings); hw->samples = LINE_IN_SAMPLES; diff --git a/audio/wavaudio.c b/audio/wavaudio.c index eed1e501af0..55d12d6cdfd 100644 --- a/audio/wavaudio.c +++ b/audio/wavaudio.c @@ -111,7 +111,7 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as) hdr[34] = bits16 ? 0x10 : 0x08; - wav_as.endianness = 0; + wav_as.big_endian = false; audio_pcm_init_info (&hw->info, &wav_as); hw->samples = 1024; diff --git a/audio/wavcapture.c b/audio/wavcapture.c index 69aa91e35f6..2dac9461710 100644 --- a/audio/wavcapture.c +++ b/audio/wavcapture.c @@ -137,7 +137,7 @@ int wav_start_capture(AudioBackend *state, CaptureState *s, const char *path, as.freq = freq; as.nchannels = 1 << stereo; as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8; - as.endianness = 0; + as.big_endian = false; ops.notify = wav_notify; ops.capture = wav_capture; diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c index 5d442b7e067..fd74c249949 100644 --- a/hw/audio/ac97.c +++ b/hw/audio/ac97.c @@ -313,7 +313,7 @@ static void open_voice(AC97LinkState *s, int index, int freq) as.freq = freq; as.nchannels = 2; as.fmt = AUDIO_FORMAT_S16; - as.endianness = 0; + as.big_endian = false; if (freq > 0) { s->invalid_freq[index] = 0; diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c index ce17e21d5fd..52ee5cb6256 100644 --- a/hw/audio/adlib.c +++ b/hw/audio/adlib.c @@ -254,7 +254,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp) as.freq = s->freq; as.nchannels = SHIFT; as.fmt = AUDIO_FORMAT_S16; - as.endianness = HOST_BIG_ENDIAN; + as.big_endian = HOST_BIG_ENDIAN; s->voice = audio_be_open_out( s->audio_be, diff --git a/hw/audio/asc.c b/hw/audio/asc.c index 35c7b5750d6..ea59bdde7b8 100644 --- a/hw/audio/asc.c +++ b/hw/audio/asc.c @@ -648,7 +648,7 @@ static void asc_realize(DeviceState *dev, Error **errp) as.freq = ASC_FREQ; as.nchannels = 2; as.fmt = AUDIO_FORMAT_U8; - as.endianness = HOST_BIG_ENDIAN; + as.big_endian = HOST_BIG_ENDIAN; s->voice = audio_be_open_out(s->audio_be, s->voice, "asc.out", s, asc_out_cb, &as); diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c index e6cae9c988e..c589670e855 100644 --- a/hw/audio/cs4231a.c +++ b/hw/audio/cs4231a.c @@ -289,7 +289,7 @@ static void cs_reset_voices (CSState *s, uint32_t val) } as.nchannels = (val & (1 << 4)) ? 2 : 1; - as.endianness = 0; + as.big_endian = false; s->tab = NULL; switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) { @@ -305,12 +305,12 @@ static void cs_reset_voices (CSState *s, uint32_t val) s->tab = ALawDecompressTable; x_law: as.fmt = AUDIO_FORMAT_S16; - as.endianness = HOST_BIG_ENDIAN; + as.big_endian = HOST_BIG_ENDIAN; s->shift = as.nchannels == 2; break; case 6: - as.endianness = 1; + as.big_endian = true; /* fall through */ case 2: as.fmt = AUDIO_FORMAT_S16; diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c index e1658393c6a..ca7ad16df49 100644 --- a/hw/audio/es1370.c +++ b/hw/audio/es1370.c @@ -407,7 +407,7 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl) as.freq = new_freq; as.nchannels = 1 << (new_fmt & 1); as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8; - as.endianness = 0; + as.big_endian = false; if (i == ADC_CHANNEL) { s->adc_voice = diff --git a/hw/audio/gus.c b/hw/audio/gus.c index 5c2a34c09d2..196c4f72205 100644 --- a/hw/audio/gus.c +++ b/hw/audio/gus.c @@ -256,7 +256,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp) as.freq = s->freq; as.nchannels = 2; as.fmt = AUDIO_FORMAT_S16; - as.endianness = HOST_BIG_ENDIAN; + as.big_endian = HOST_BIG_ENDIAN; s->voice = audio_be_open_out( s->audio_be, diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c index 14e15a844ba..a891e975106 100644 --- a/hw/audio/lm4549.c +++ b/hw/audio/lm4549.c @@ -202,7 +202,7 @@ void lm4549_write(lm4549_state *s, as.freq = value; as.nchannels = 2; as.fmt = AUDIO_FORMAT_S16; - as.endianness = 0; + as.big_endian = false; s->voice = audio_be_open_out( s->audio_be, @@ -272,7 +272,7 @@ static int lm4549_post_load(void *opaque, int version_id) as.freq = freq; as.nchannels = 2; as.fmt = AUDIO_FORMAT_S16; - as.endianness = 0; + as.big_endian = false; s->voice = audio_be_open_out( s->audio_be, @@ -312,7 +312,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback data_req_cb, void* opaque, as.freq = 48000; as.nchannels = 2; as.fmt = AUDIO_FORMAT_S16; - as.endianness = 0; + as.big_endian = false; s->voice = audio_be_open_out( s->audio_be, diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c index c8fc7df8b44..1b5e452a29b 100644 --- a/hw/audio/sb16.c +++ b/hw/audio/sb16.c @@ -213,7 +213,7 @@ static void continue_dma8 (SB16State *s) as.freq = s->freq; as.nchannels = 1 << s->fmt_stereo; as.fmt = s->fmt; - as.endianness = 0; + as.big_endian = false; s->voice = audio_be_open_out( s->audio_be, @@ -376,7 +376,7 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len) as.freq = s->freq; as.nchannels = 1 << s->fmt_stereo; as.fmt = s->fmt; - as.endianness = 0; + as.big_endian = false; s->voice = audio_be_open_out( s->audio_be, @@ -877,7 +877,7 @@ static void legacy_reset (SB16State *s) as.freq = s->freq; as.nchannels = 1; as.fmt = AUDIO_FORMAT_U8; - as.endianness = 0; + as.big_endian = false; s->voice = audio_be_open_out( s->audio_be, @@ -1300,7 +1300,7 @@ static int sb16_post_load (void *opaque, int version_id) as.freq = s->freq; as.nchannels = 1 << s->fmt_stereo; as.fmt = s->fmt; - as.endianness = 0; + as.big_endian = false; s->voice = audio_be_open_out( s->audio_be, diff --git a/hw/audio/via-ac97.c b/hw/audio/via-ac97.c index 84d137b41a3..9d61283542a 100644 --- a/hw/audio/via-ac97.c +++ b/hw/audio/via-ac97.c @@ -237,7 +237,7 @@ static void open_voice_out(ViaAC97State *s) .freq = CODEC_REG(s, AC97_PCM_Front_DAC_Rate), .nchannels = s->aur.type & BIT(4) ? 2 : 1, .fmt = s->aur.type & BIT(5) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_S8, - .endianness = 0, + .big_endian = false, }; s->vo = audio_be_open_out(s->audio_be, s->vo, "via-ac97.out", s, out_cb, &as); } diff --git a/hw/audio/virtio-snd.c b/hw/audio/virtio-snd.c index 89e24c0a8e0..8b949146468 100644 --- a/hw/audio/virtio-snd.c +++ b/hw/audio/virtio-snd.c @@ -378,7 +378,7 @@ static void virtio_snd_get_qemu_audsettings(audsettings *as, as->nchannels = MIN(AUDIO_MAX_CHANNELS, params->channels); as->fmt = virtio_snd_get_qemu_format(params->format); as->freq = virtio_snd_get_qemu_freq(params->rate); - as->endianness = 0; /* Conforming to VIRTIO 1.0: always little endian. */ + as->big_endian = false; /* Conforming to VIRTIO 1.0: always little endian. */ } /* diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c index 2a286515b14..e2507b0269a 100644 --- a/hw/audio/wm8750.c +++ b/hw/audio/wm8750.c @@ -202,7 +202,7 @@ static void wm8750_set_format(WM8750State *s) return; /* Setup input */ - in_fmt.endianness = 0; + in_fmt.big_endian = false; in_fmt.nchannels = 2; in_fmt.freq = s->adc_hz; in_fmt.fmt = AUDIO_FORMAT_S16; @@ -215,7 +215,7 @@ static void wm8750_set_format(WM8750State *s) CODEC ".input3", s, wm8750_audio_in_cb, &in_fmt); /* Setup output */ - out_fmt.endianness = 0; + out_fmt.big_endian = false; out_fmt.nchannels = 2; out_fmt.freq = s->dac_hz; out_fmt.fmt = AUDIO_FORMAT_S16; diff --git a/hw/display/xlnx_dp.c b/hw/display/xlnx_dp.c index 9aa4709b411..7d037b46a35 100644 --- a/hw/display/xlnx_dp.c +++ b/hw/display/xlnx_dp.c @@ -1393,7 +1393,7 @@ static void xlnx_dp_realize(DeviceState *dev, Error **errp) as.freq = 44100; as.nchannels = 2; as.fmt = AUDIO_FORMAT_S16; - as.endianness = 0; + as.big_endian = false; s->amixer_output_stream = audio_be_open_out(s->audio_be, s->amixer_output_stream, diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c index 7b758718c12..e18e0a1dfd6 100644 --- a/hw/usb/dev-audio.c +++ b/hw/usb/dev-audio.c @@ -975,7 +975,7 @@ static void usb_audio_reinit(USBDevice *dev, unsigned channels) s->out.as.freq = USBAUDIO_SAMPLE_RATE; s->out.as.nchannels = s->out.channels; s->out.as.fmt = AUDIO_FORMAT_S16; - s->out.as.endianness = 0; + s->out.as.big_endian = false; streambuf_init(&s->out.buf, s->buffer, s->out.channels); s->out.voice = audio_be_open_out(s->audio_be, s->out.voice, TYPE_USB_AUDIO, diff --git a/tests/audio/test-audio.c b/tests/audio/test-audio.c index 90d96ae0575..b87a12eb55b 100644 --- a/tests/audio/test-audio.c +++ b/tests/audio/test-audio.c @@ -57,7 +57,7 @@ static const struct audsettings default_test_settings = { .freq = SAMPLE_RATE, .nchannels = CHANNELS, .fmt = AUDIO_FORMAT_S16, - .endianness = 0, + .big_endian = false, }; static void dummy_audio_callback(void *opaque, int avail) diff --git a/ui/vnc.c b/ui/vnc.c index d56fe2c180e..daf5b01d342 100644 --- a/ui/vnc.c +++ b/ui/vnc.c @@ -3372,7 +3372,7 @@ static void vnc_connect(VncDisplay *vd, QIOChannelSocket *sioc, vs->as.freq = 44100; vs->as.nchannels = 2; vs->as.fmt = AUDIO_FORMAT_S16; - vs->as.endianness = 0; + vs->as.big_endian = false; qemu_mutex_init(&vs->output_mutex); vs->bh = qemu_bh_new(vnc_jobs_bh, vs); -- 2.53.0
