diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index cfe42284a6..f37ce1ce85 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -493,13 +493,6 @@ static int alsa_open(bool in, struct alsa_params_req *req, goto err; } - if (nchannels != 1 && nchannels != 2) { - alsa_logerr2 (err, typ, - "Can not handle obtained number of channels %d\n", - nchannels); - goto err; - } - if (apdo->buffer_length) { int dir = 0; unsigned int btime = apdo->buffer_length; @@ -602,7 +595,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) { ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; size_t pos = 0; - size_t len_frames = len >> hw->info.shift; + size_t len_frames = len / hw->info.bytes_per_frame; while (len_frames) { char *src = advance(buf, pos); @@ -648,7 +641,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) } } - pos += written << hw->info.shift; + pos += written * hw->info.bytes_per_frame; if (written < len_frames) { break; } @@ -802,7 +795,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) void *dst = advance(buf, pos); snd_pcm_sframes_t nread; - nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift); + nread = snd_pcm_readi( + alsa->handle, dst, len / hw->info.bytes_per_frame); if (nread <= 0) { switch (nread) { @@ -828,8 +822,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) } } - pos += nread << hw->info.shift; - len -= nread << hw->info.shift; + pos += nread * hw->info.bytes_per_frame; + len -= nread * hw->info.bytes_per_frame; } return pos; diff --git a/audio/audio.c b/audio/audio.c index 7128ee98dc..7fc3aa9d16 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -242,7 +242,7 @@ static int audio_validate_settings (struct audsettings *as) { int invalid; - invalid = as->nchannels != 1 && as->nchannels != 2; + invalid = as->nchannels < 1; invalid |= as->endianness != 0 && as->endianness != 1; switch (as->fmt) { @@ -299,12 +299,13 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) { - int bits = 8, sign = 0, shift = 0; + int bits = 8, sign = 0, mul; switch (as->fmt) { case AUDIO_FORMAT_S8: sign = 1; case AUDIO_FORMAT_U8: + mul = 1; break; case AUDIO_FORMAT_S16: @@ -312,7 +313,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) /* fall through */ case AUDIO_FORMAT_U16: bits = 16; - shift = 1; + mul = 2; break; case AUDIO_FORMAT_S32: @@ -320,7 +321,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) /* fall through */ case AUDIO_FORMAT_U32: bits = 32; - shift = 2; + mul = 4; break; default: @@ -331,9 +332,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) info->bits = bits; info->sign = sign; info->nchannels = as->nchannels; - info->shift = (as->nchannels == 2) + shift; - info->align = (1 << info->shift) - 1; - info->bytes_per_second = info->freq << info->shift; + info->bytes_per_frame = as->nchannels * mul; + info->bytes_per_second = info->freq * info->bytes_per_frame; info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS); } @@ -344,26 +344,25 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len) } if (info->sign) { - memset (buf, 0x00, len << info->shift); + memset(buf, 0x00, len * info->bytes_per_frame); } else { switch (info->bits) { case 8: - memset (buf, 0x80, len << info->shift); + memset(buf, 0x80, len * info->bytes_per_frame); break; case 16: { int i; uint16_t *p = buf; - int shift = info->nchannels - 1; short s = INT16_MAX; if (info->swap_endianness) { s = bswap16 (s); } - for (i = 0; i < len << shift; i++) { + for (i = 0; i < len * info->nchannels; i++) { p[i] = s; } } @@ -373,14 +372,13 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len) { int i; uint32_t *p = buf; - int shift = info->nchannels - 1; int32_t s = INT32_MAX; if (info->swap_endianness) { s = bswap32 (s); } - for (i = 0; i < len << shift; i++) { + for (i = 0; i < len * info->nchannels; i++) { p[i] = s; } } @@ -558,7 +556,7 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) while (len) { st_sample *src = hw->mix_buf->samples + pos; - uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift); + uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame); size_t samples_till_end_of_buf = hw->mix_buf->size - pos; size_t samples_to_clip = MIN(len, samples_till_end_of_buf); @@ -607,7 +605,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) return 0; } - samples = size >> sw->info.shift; + samples = size / sw->info.bytes_per_frame; if (!live) { return 0; } @@ -642,7 +640,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) sw->clip (buf, sw->buf, ret); sw->total_hw_samples_acquired += total; - return ret << sw->info.shift; + return ret * sw->info.bytes_per_frame; } /* @@ -715,7 +713,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) } wpos = (sw->hw->mix_buf->pos + live) % hwsamples; - samples = size >> sw->info.shift; + samples = size / sw->info.bytes_per_frame; dead = hwsamples - live; swlim = ((int64_t) dead << 32) / sw->ratio; @@ -759,13 +757,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) dolog ( "%s: write size %zu ret %zu total sw %zu\n", SW_NAME (sw), - size >> sw->info.shift, + size / sw->info.bytes_per_frame, ret, sw->total_hw_samples_mixed ); #endif - return ret << sw->info.shift; + return ret * sw->info.bytes_per_frame; } #ifdef DEBUG_AUDIO @@ -838,37 +836,51 @@ static void audio_timer (void *opaque) */ size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size) { + HWVoiceOut *hw; + if (!sw) { /* XXX: Consider options */ return size; } + hw = sw->hw; - if (!sw->hw->enabled) { + if (!hw->enabled) { dolog ("Writing to disabled voice %s\n", SW_NAME (sw)); return 0; } - return audio_pcm_sw_write(sw, buf, size); + if (audio_get_pdo_out(hw->s->dev)->mixing_engine) { + return audio_pcm_sw_write(sw, buf, size); + } else { + return hw->pcm_ops->write(hw, buf, size); + } } size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size) { + HWVoiceIn *hw; + if (!sw) { /* XXX: Consider options */ return size; } + hw = sw->hw; - if (!sw->hw->enabled) { + if (!hw->enabled) { dolog ("Reading from disabled voice %s\n", SW_NAME (sw)); return 0; } - return audio_pcm_sw_read(sw, buf, size); + if (audio_get_pdo_in(hw->s->dev)->mixing_engine) { + return audio_pcm_sw_read(sw, buf, size); + } else { + return hw->pcm_ops->read(hw, buf, size); + } } int AUD_get_buffer_size_out (SWVoiceOut *sw) { - return sw->hw->mix_buf->size << sw->hw->info.shift; + return sw->hw->mix_buf->size * sw->hw->info.bytes_per_frame; } void AUD_set_active_out (SWVoiceOut *sw, int on) @@ -984,10 +996,10 @@ static size_t audio_get_avail (SWVoiceIn *sw) ldebug ( "%s: get_avail live %d ret %" PRId64 "\n", SW_NAME (sw), - live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift + live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame ); - return (((int64_t) live << 32) / sw->ratio) << sw->info.shift; + return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame; } static size_t audio_get_free(SWVoiceOut *sw) @@ -1011,10 +1023,11 @@ static size_t audio_get_free(SWVoiceOut *sw) #ifdef DEBUG_OUT dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n", SW_NAME (sw), - live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift); + live, dead, (((int64_t) dead << 32) / sw->ratio) * + sw->info.bytes_per_frame); #endif - return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift; + return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame; } static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, @@ -1033,7 +1046,7 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, while (n) { size_t till_end_of_hw = hw->mix_buf->size - rpos2; size_t to_write = MIN(till_end_of_hw, n); - size_t bytes = to_write << hw->info.shift; + size_t bytes = to_write * hw->info.bytes_per_frame; size_t written; sw->buf = hw->mix_buf->samples + rpos2; @@ -1068,10 +1081,11 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) return clipped + live; } - decr = MIN(size >> hw->info.shift, live); + decr = MIN(size / hw->info.bytes_per_frame, live); audio_pcm_hw_clip_out(hw, buf, decr); - proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >> - hw->info.shift; + proc = hw->pcm_ops->put_buffer_out(hw, buf, + decr * hw->info.bytes_per_frame) / + hw->info.bytes_per_frame; live -= proc; clipped += proc; @@ -1090,6 +1104,26 @@ static void audio_run_out (AudioState *s) HWVoiceOut *hw = NULL; SWVoiceOut *sw; + if (!audio_get_pdo_out(s->dev)->mixing_engine) { + while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) { + /* there is exactly 1 sw for each hw with no mixeng */ + sw = hw->sw_head.lh_first; + + if (hw->pending_disable) { + hw->enabled = 0; + hw->pending_disable = 0; + if (hw->pcm_ops->enable_out) { + hw->pcm_ops->enable_out(hw, false); + } + } + + if (sw->active) { + sw->callback.fn(sw->callback.opaque, INT_MAX); + } + } + return; + } + while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) { size_t played, live, prev_rpos, free; int nb_live, cleanup_required; @@ -1200,16 +1234,16 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples) while (samples) { size_t proc; - size_t size = samples << hw->info.shift; + size_t size = samples * hw->info.bytes_per_frame; void *buf = hw->pcm_ops->get_buffer_in(hw, &size); - assert((size & hw->info.align) == 0); + assert(size % hw->info.bytes_per_frame == 0); if (size == 0) { hw->pcm_ops->put_buffer_in(hw, buf, size); break; } - proc = MIN(size >> hw->info.shift, + proc = MIN(size / hw->info.bytes_per_frame, conv_buf->size - conv_buf->pos); hw->conv(conv_buf->samples + conv_buf->pos, buf, proc); @@ -1217,7 +1251,7 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples) samples -= proc; conv += proc; - hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift); + hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame); } return conv; @@ -1227,6 +1261,17 @@ static void audio_run_in (AudioState *s) { HWVoiceIn *hw = NULL; + if (!audio_get_pdo_in(s->dev)->mixing_engine) { + while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) { + /* there is exactly 1 sw for each hw with no mixeng */ + SWVoiceIn *sw = hw->sw_head.lh_first; + if (sw->active) { + sw->callback.fn(sw->callback.opaque, INT_MAX); + } + } + return; + } + while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) { SWVoiceIn *sw; size_t captured = 0, min; @@ -1280,7 +1325,7 @@ static void audio_run_capture (AudioState *s) for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) { cb->ops.capture (cb->opaque, cap->buf, - to_capture << hw->info.shift); + to_capture * hw->info.bytes_per_frame); } rpos = (rpos + to_capture) % hw->mix_buf->size; live -= to_capture; @@ -1333,7 +1378,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size) ssize_t start; if (unlikely(!hw->buf_emul)) { - size_t calc_size = hw->conv_buf->size << hw->info.shift; + size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame; hw->buf_emul = g_malloc(calc_size); hw->size_emul = calc_size; hw->pos_emul = hw->pending_emul = 0; @@ -1369,7 +1414,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size) void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size) { if (unlikely(!hw->buf_emul)) { - size_t calc_size = hw->mix_buf->size << hw->info.shift; + size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame; hw->buf_emul = g_malloc(calc_size); hw->size_emul = calc_size; @@ -1751,6 +1796,11 @@ CaptureVoiceOut *AUD_add_capture( s = audio_init(NULL, NULL); } + if (!audio_get_pdo_out(s->dev)->mixing_engine) { + dolog("Can't capture with mixeng disabled\n"); + return NULL; + } + if (audio_validate_settings (as)) { dolog ("Invalid settings were passed when trying to add capture\n"); audio_print_settings (as); @@ -1783,7 +1833,7 @@ CaptureVoiceOut *AUD_add_capture( audio_pcm_init_info (&hw->info, as); - cap->buf = g_malloc0_n(hw->mix_buf->size, 1 << hw->info.shift); + cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame); hw->clip = mixeng_clip [hw->info.nchannels == 2] @@ -1841,31 +1891,45 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque) } void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol) +{ + Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } }; + audio_set_volume_out(sw, &vol); +} + +void audio_set_volume_out(SWVoiceOut *sw, Volume *vol) { if (sw) { HWVoiceOut *hw = sw->hw; - sw->vol.mute = mute; - sw->vol.l = nominal_volume.l * lvol / 255; - sw->vol.r = nominal_volume.r * rvol / 255; + sw->vol.mute = vol->mute; + sw->vol.l = nominal_volume.l * vol->vol[0] / 255; + sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] / + 255; if (hw->pcm_ops->volume_out) { - hw->pcm_ops->volume_out(hw, &sw->vol); + hw->pcm_ops->volume_out(hw, vol); } } } void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol) +{ + Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } }; + audio_set_volume_in(sw, &vol); +} + +void audio_set_volume_in(SWVoiceIn *sw, Volume *vol) { if (sw) { HWVoiceIn *hw = sw->hw; - sw->vol.mute = mute; - sw->vol.l = nominal_volume.l * lvol / 255; - sw->vol.r = nominal_volume.r * rvol / 255; + sw->vol.mute = vol->mute; + sw->vol.l = nominal_volume.l * vol->vol[0] / 255; + sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] / + 255; if (hw->pcm_ops->volume_in) { - hw->pcm_ops->volume_in(hw, &sw->vol); + hw->pcm_ops->volume_in(hw, vol); } } } @@ -1905,9 +1969,13 @@ void audio_create_pdos(Audiodev *dev) static void audio_validate_per_direction_opts( AudiodevPerDirectionOptions *pdo, Error **errp) { + if (!pdo->has_mixing_engine) { + pdo->has_mixing_engine = true; + pdo->mixing_engine = true; + } if (!pdo->has_fixed_settings) { pdo->has_fixed_settings = true; - pdo->fixed_settings = true; + pdo->fixed_settings = pdo->mixing_engine; } if (!pdo->fixed_settings && (pdo->has_frequency || pdo->has_channels || pdo->has_format)) { @@ -1915,6 +1983,10 @@ static void audio_validate_per_direction_opts( "You can't use frequency, channels or format with fixed-settings=off"); return; } + if (!pdo->mixing_engine && pdo->fixed_settings) { + error_setg(errp, "You can't use fixed-settings without mixeng"); + return; + } if (!pdo->has_frequency) { pdo->has_frequency = true; @@ -1926,7 +1998,7 @@ static void audio_validate_per_direction_opts( } if (!pdo->has_voices) { pdo->has_voices = true; - pdo->voices = 1; + pdo->voices = pdo->mixing_engine ? 1 : INT_MAX; } if (!pdo->has_format) { pdo->has_format = true; @@ -2081,14 +2153,14 @@ size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate, now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); ticks = now - rate->start_ticks; bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND); - samples = (bytes - rate->bytes_sent) >> info->shift; + samples = (bytes - rate->bytes_sent) / info->bytes_per_frame; if (samples < 0 || samples > 65536) { AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples); audio_rate_start(rate); samples = 0; } - ret = MIN(samples << info->shift, bytes_avail); + ret = MIN(samples * info->bytes_per_frame, bytes_avail); rate->bytes_sent += ret; return ret; } diff --git a/audio/audio.h b/audio/audio.h index c74abb8c47..0db3c7dd5e 100644 --- a/audio/audio.h +++ b/audio/audio.h @@ -124,6 +124,16 @@ uint64_t AUD_get_elapsed_usec_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts); void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol); void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol); +#define AUDIO_MAX_CHANNELS 16 +typedef struct Volume { + bool mute; + int channels; + uint8_t vol[AUDIO_MAX_CHANNELS]; +} Volume; + +void audio_set_volume_out(SWVoiceOut *sw, Volume *vol); +void audio_set_volume_in(SWVoiceIn *sw, Volume *vol); + SWVoiceIn *AUD_open_in ( QEMUSoundCard *card, SWVoiceIn *sw, diff --git a/audio/audio_int.h b/audio/audio_int.h index 22a703c13e..5ba2078346 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -43,8 +43,7 @@ struct audio_pcm_info { int sign; int freq; int nchannels; - int align; - int shift; + int bytes_per_frame; int bytes_per_second; int swap_endianness; }; @@ -166,7 +165,7 @@ struct audio_pcm_ops { */ size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size); void (*enable_out)(HWVoiceOut *hw, bool enable); - void (*volume_out)(HWVoiceOut *hw, struct mixeng_volume *vol); + void (*volume_out)(HWVoiceOut *hw, Volume *vol); int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque); void (*fini_in) (HWVoiceIn *hw); @@ -174,7 +173,7 @@ struct audio_pcm_ops { void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size); void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size); void (*enable_in)(HWVoiceIn *hw, bool enable); - void (*volume_in)(HWVoiceIn *hw, struct mixeng_volume *vol); + void (*volume_in)(HWVoiceIn *hw, Volume *vol); }; void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size); diff --git a/audio/audio_template.h b/audio/audio_template.h index 235d1acbbe..3287d7075e 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -78,13 +78,17 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw) static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) { - size_t samples = hw->samples; - if (audio_bug(__func__, samples == 0)) { - dolog("Attempted to allocate empty buffer\n"); - } + if (glue(audio_get_pdo_, TYPE)(hw->s->dev)->mixing_engine) { + size_t samples = hw->samples; + if (audio_bug(__func__, samples == 0)) { + dolog("Attempted to allocate empty buffer\n"); + } - HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples); - HWBUF->size = samples; + HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples); + HWBUF->size = samples; + } else { + HWBUF = NULL; + } } static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw) @@ -103,6 +107,10 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) { int samples; + if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) { + return 0; + } + samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio; sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample)); @@ -328,9 +336,9 @@ static HW *glue(audio_pcm_hw_add_, TYPE)(AudioState *s, struct audsettings *as) HW *hw; AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev); - if (pdo->fixed_settings) { + if (!pdo->mixing_engine || pdo->fixed_settings) { hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as); - if (hw) { + if (!pdo->mixing_engine || hw) { return hw; } } @@ -425,8 +433,8 @@ SW *glue (AUD_open_, TYPE) ( struct audsettings *as ) { - AudioState *s = card->state; - AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev); + AudioState *s; + AudiodevPerDirectionOptions *pdo; if (audio_bug(__func__, !card || !name || !callback_fn || !as)) { dolog ("card=%p name=%p callback_fn=%p as=%p\n", @@ -434,6 +442,9 @@ SW *glue (AUD_open_, TYPE) ( goto fail; } + s = card->state; + pdo = glue(audio_get_pdo_, TYPE)(s->dev); + ldebug ("open %s, freq %d, nchannels %d, fmt %d\n", name, as->freq, as->nchannels, as->fmt); diff --git a/audio/coreaudio.c b/audio/coreaudio.c index 1427c9f622..66f0f459cf 100644 --- a/audio/coreaudio.c +++ b/audio/coreaudio.c @@ -440,7 +440,7 @@ static OSStatus audioDeviceIOProc( } frameCount = core->audioDevicePropertyBufferFrameSize; - pending_frames = hw->pending_emul >> hw->info.shift; + pending_frames = hw->pending_emul / hw->info.bytes_per_frame; /* if there are not enough samples, set signal and return */ if (pending_frames < frameCount) { @@ -449,7 +449,7 @@ static OSStatus audioDeviceIOProc( return 0; } - len = frameCount << hw->info.shift; + len = frameCount * hw->info.bytes_per_frame; while (len) { size_t write_len; ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul; diff --git a/audio/dsound_template.h b/audio/dsound_template.h index 9f10b688df..7a15f91ce5 100644 --- a/audio/dsound_template.h +++ b/audio/dsound_template.h @@ -98,8 +98,8 @@ static int glue (dsound_lock_, TYPE) ( goto fail; } - if ((p1p && *p1p && (*blen1p & info->align)) || - (p2p && *p2p && (*blen2p & info->align))) { + if ((p1p && *p1p && (*blen1p % info->bytes_per_frame)) || + (p2p && *p2p && (*blen2p % info->bytes_per_frame))) { dolog("DirectSound returned misaligned buffer %ld %ld\n", *blen1p, *blen2p); glue(dsound_unlock_, TYPE)(buf, *p1p, p2p ? *p2p : NULL, *blen1p, @@ -247,14 +247,14 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as, obt_as.endianness = 0; audio_pcm_init_info (&hw->info, &obt_as); - if (bc.dwBufferBytes & hw->info.align) { + if (bc.dwBufferBytes % hw->info.bytes_per_frame) { dolog ( "GetCaps returned misaligned buffer size %ld, alignment %d\n", - bc.dwBufferBytes, hw->info.align + 1 + bc.dwBufferBytes, hw->info.bytes_per_frame ); } hw->size_emul = bc.dwBufferBytes; - hw->samples = bc.dwBufferBytes >> hw->info.shift; + hw->samples = bc.dwBufferBytes / hw->info.bytes_per_frame; ds->s = s; #ifdef DEBUG_DSOUND diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c index d4a4757445..c265c0094b 100644 --- a/audio/dsoundaudio.c +++ b/audio/dsoundaudio.c @@ -320,8 +320,8 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb, return; } - len1 = blen1 >> hw->info.shift; - len2 = blen2 >> hw->info.shift; + len1 = blen1 / hw->info.bytes_per_frame; + len2 = blen2 / hw->info.bytes_per_frame; #ifdef DEBUG_DSOUND dolog ("clear %p,%ld,%ld %p,%ld,%ld\n", diff --git a/audio/noaudio.c b/audio/noaudio.c index ec8a287f36..ff99b253ff 100644 --- a/audio/noaudio.c +++ b/audio/noaudio.c @@ -91,7 +91,7 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size) NoVoiceIn *no = (NoVoiceIn *) hw; int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size); - audio_pcm_info_clear_buf(&hw->info, buf, bytes >> hw->info.shift); + audio_pcm_info_clear_buf(&hw->info, buf, bytes / hw->info.bytes_per_frame); return bytes; } diff --git a/audio/ossaudio.c b/audio/ossaudio.c index 0c4451e972..c43faeeea4 100644 --- a/audio/ossaudio.c +++ b/audio/ossaudio.c @@ -506,16 +506,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as, oss->nfrags = obt.nfrags; oss->fragsize = obt.fragsize; - if (obt.nfrags * obt.fragsize & hw->info.align) { + if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) { dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n", - obt.nfrags * obt.fragsize, hw->info.align + 1); + obt.nfrags * obt.fragsize, hw->info.bytes_per_frame); } - hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift; + hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame; oss->mmapped = 0; if (oopts->has_try_mmap && oopts->try_mmap) { - hw->size_emul = hw->samples << hw->info.shift; + hw->size_emul = hw->samples * hw->info.bytes_per_frame; hw->buf_emul = mmap( NULL, hw->size_emul, @@ -644,12 +644,12 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) oss->nfrags = obt.nfrags; oss->fragsize = obt.fragsize; - if (obt.nfrags * obt.fragsize & hw->info.align) { + if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) { dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n", - obt.nfrags * obt.fragsize, hw->info.align + 1); + obt.nfrags * obt.fragsize, hw->info.bytes_per_frame); } - hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift; + hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame; oss->fd = fd; oss->dev = dev; diff --git a/audio/paaudio.c b/audio/paaudio.c index ed31f863f7..df541a72d3 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -2,6 +2,7 @@ #include "qemu/osdep.h" #include "qemu/module.h" +#include "qemu-common.h" #include "audio.h" #include "qapi/opts-visitor.h" @@ -98,6 +99,59 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x) } \ } while (0) +static void *qpa_get_buffer_in(HWVoiceIn *hw, size_t *size) +{ + PAVoiceIn *p = (PAVoiceIn *) hw; + PAConnection *c = p->g->conn; + int r; + + pa_threaded_mainloop_lock(c->mainloop); + + CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail, + "pa_threaded_mainloop_lock failed\n"); + + if (!p->read_length) { + r = pa_stream_peek(p->stream, &p->read_data, &p->read_length); + CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail, + "pa_stream_peek failed\n"); + } + + *size = MIN(p->read_length, *size); + + pa_threaded_mainloop_unlock(c->mainloop); + return (void *) p->read_data; + +unlock_and_fail: + pa_threaded_mainloop_unlock(c->mainloop); + *size = 0; + return NULL; +} + +static void qpa_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size) +{ + PAVoiceIn *p = (PAVoiceIn *) hw; + PAConnection *c = p->g->conn; + int r; + + pa_threaded_mainloop_lock(c->mainloop); + + CHECK_DEAD_GOTO(c, p->stream, unlock, + "pa_threaded_mainloop_lock failed\n"); + + assert(buf == p->read_data && size <= p->read_length); + + p->read_data += size; + p->read_length -= size; + + if (size && !p->read_length) { + r = pa_stream_drop(p->stream); + CHECK_SUCCESS_GOTO(c, r == 0, unlock, "pa_stream_drop failed\n"); + } + +unlock: + pa_threaded_mainloop_unlock(c->mainloop); +} + static size_t qpa_read(HWVoiceIn *hw, void *data, size_t length) { PAVoiceIn *p = (PAVoiceIn *) hw; @@ -136,6 +190,32 @@ unlock_and_fail: return 0; } +static void *qpa_get_buffer_out(HWVoiceOut *hw, size_t *size) +{ + PAVoiceOut *p = (PAVoiceOut *) hw; + PAConnection *c = p->g->conn; + void *ret; + int r; + + pa_threaded_mainloop_lock(c->mainloop); + + CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail, + "pa_threaded_mainloop_lock failed\n"); + + *size = -1; + r = pa_stream_begin_write(p->stream, &ret, size); + CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail, + "pa_stream_begin_write failed\n"); + + pa_threaded_mainloop_unlock(c->mainloop); + return ret; + +unlock_and_fail: + pa_threaded_mainloop_unlock(c->mainloop); + *size = 0; + return NULL; +} + static size_t qpa_write(HWVoiceOut *hw, void *data, size_t length) { PAVoiceOut *p = (PAVoiceOut *) hw; @@ -259,17 +339,59 @@ static pa_stream *qpa_simple_new ( pa_stream_direction_t dir, const char *dev, const pa_sample_spec *ss, - const pa_channel_map *map, const pa_buffer_attr *attr, int *rerror) { int r; - pa_stream *stream; + pa_stream *stream = NULL; pa_stream_flags_t flags; + pa_channel_map map; pa_threaded_mainloop_lock(c->mainloop); - stream = pa_stream_new(c->context, name, ss, map); + pa_channel_map_init(&map); + map.channels = ss->channels; + + /* + * TODO: This currently expects the only frontend supporting more than 2 + * channels is the usb-audio. We will need some means to set channel + * order when a new frontend gains multi-channel support. + */ + switch (ss->channels) { + case 1: + map.map[0] = PA_CHANNEL_POSITION_MONO; + break; + + case 2: + map.map[0] = PA_CHANNEL_POSITION_LEFT; + map.map[1] = PA_CHANNEL_POSITION_RIGHT; + break; + + case 6: + map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT; + map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT; + map.map[2] = PA_CHANNEL_POSITION_CENTER; + map.map[3] = PA_CHANNEL_POSITION_LFE; + map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT; + map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT; + break; + + case 8: + map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT; + map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT; + map.map[2] = PA_CHANNEL_POSITION_CENTER; + map.map[3] = PA_CHANNEL_POSITION_LFE; + map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT; + map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT; + map.map[6] = PA_CHANNEL_POSITION_SIDE_LEFT; + map.map[7] = PA_CHANNEL_POSITION_SIDE_RIGHT; + + default: + dolog("Internal error: unsupported channel count %d\n", ss->channels); + goto fail; + } + + stream = pa_stream_new(c->context, name, ss, &map); if (!stream) { goto fail; } @@ -338,11 +460,10 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as, pa->stream = qpa_simple_new ( c, - "qemu", + ppdo->has_stream_name ? ppdo->stream_name : g->dev->id, PA_STREAM_PLAYBACK, ppdo->has_name ? ppdo->name : NULL, &ss, - NULL, /* channel map */ &ba, /* buffering attributes */ &error ); @@ -387,11 +508,10 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) pa->stream = qpa_simple_new ( c, - "qemu", + ppdo->has_stream_name ? ppdo->stream_name : g->dev->id, PA_STREAM_RECORD, ppdo->has_name ? ppdo->name : NULL, &ss, - NULL, /* channel map */ &ba, /* buffering attributes */ &error ); @@ -452,20 +572,22 @@ static void qpa_fini_in (HWVoiceIn *hw) } } -static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol) +static void qpa_volume_out(HWVoiceOut *hw, Volume *vol) { PAVoiceOut *pa = (PAVoiceOut *) hw; pa_operation *op; pa_cvolume v; PAConnection *c = pa->g->conn; + int i; #ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */ pa_cvolume_init (&v); /* function is present in 0.9.13+ */ #endif - v.channels = 2; - v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX; - v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX; + v.channels = vol->channels; + for (i = 0; i < vol->channels; ++i) { + v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255; + } pa_threaded_mainloop_lock(c->mainloop); @@ -492,20 +614,22 @@ static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol) pa_threaded_mainloop_unlock(c->mainloop); } -static void qpa_volume_in(HWVoiceIn *hw, struct mixeng_volume *vol) +static void qpa_volume_in(HWVoiceIn *hw, Volume *vol) { PAVoiceIn *pa = (PAVoiceIn *) hw; pa_operation *op; pa_cvolume v; PAConnection *c = pa->g->conn; + int i; #ifdef PA_CHECK_VERSION pa_cvolume_init (&v); #endif - v.channels = 2; - v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX; - v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX; + v.channels = vol->channels; + for (i = 0; i < vol->channels; ++i) { + v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255; + } pa_threaded_mainloop_lock(c->mainloop); @@ -549,6 +673,7 @@ static int qpa_validate_per_direction_opts(Audiodev *dev, /* common */ static void *qpa_conn_init(const char *server) { + const char *vm_name; PAConnection *c = g_malloc0(sizeof(PAConnection)); QTAILQ_INSERT_TAIL(&pa_conns, c, list); @@ -557,8 +682,9 @@ static void *qpa_conn_init(const char *server) goto fail; } + vm_name = qemu_get_vm_name(); c->context = pa_context_new(pa_threaded_mainloop_get_api(c->mainloop), - server); + vm_name ? vm_name : "qemu"); if (!c->context) { goto fail; } @@ -698,11 +824,15 @@ static struct audio_pcm_ops qpa_pcm_ops = { .init_out = qpa_init_out, .fini_out = qpa_fini_out, .write = qpa_write, + .get_buffer_out = qpa_get_buffer_out, + .put_buffer_out = qpa_write, /* pa handles it */ .volume_out = qpa_volume_out, .init_in = qpa_init_in, .fini_in = qpa_fini_in, .read = qpa_read, + .get_buffer_in = qpa_get_buffer_in, + .put_buffer_in = qpa_put_buffer_in, .volume_in = qpa_volume_in }; diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c index 9860f9c5e1..b6b5da4812 100644 --- a/audio/spiceaudio.c +++ b/audio/spiceaudio.c @@ -131,7 +131,8 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size) if (out->frame) { *size = audio_rate_get_bytes( - &hw->info, &out->rate, (out->fsize - out->fpos) << hw->info.shift); + &hw->info, &out->rate, + (out->fsize - out->fpos) * hw->info.bytes_per_frame); } else { audio_rate_start(&out->rate); } @@ -179,13 +180,14 @@ static void line_out_enable(HWVoiceOut *hw, bool enable) } #if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2)) -static void line_out_volume(HWVoiceOut *hw, struct mixeng_volume *vol) +static void line_out_volume(HWVoiceOut *hw, Volume *vol) { SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw); uint16_t svol[2]; - svol[0] = vol->l / ((1ULL << 16) + 1); - svol[1] = vol->r / ((1ULL << 16) + 1); + assert(vol->channels == 2); + svol[0] = vol->vol[0] * 257; + svol[1] = vol->vol[1] * 257; spice_server_playback_set_volume(&out->sin, 2, svol); spice_server_playback_set_mute(&out->sin, vol->mute); } @@ -262,13 +264,14 @@ static void line_in_enable(HWVoiceIn *hw, bool enable) } #if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2)) -static void line_in_volume(HWVoiceIn *hw, struct mixeng_volume *vol) +static void line_in_volume(HWVoiceIn *hw, Volume *vol) { SpiceVoiceIn *in = container_of(hw, SpiceVoiceIn, hw); uint16_t svol[2]; - svol[0] = vol->l / ((1ULL << 16) + 1); - svol[1] = vol->r / ((1ULL << 16) + 1); + assert(vol->channels == 2); + svol[0] = vol->vol[0] * 257; + svol[1] = vol->vol[1] * 257; spice_server_record_set_volume(&in->sin, 2, svol); spice_server_record_set_mute(&in->sin, vol->mute); } diff --git a/audio/wavaudio.c b/audio/wavaudio.c index 47efdc1b1e..e46d834bd3 100644 --- a/audio/wavaudio.c +++ b/audio/wavaudio.c @@ -43,14 +43,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len) { WAVVoiceOut *wav = (WAVVoiceOut *) hw; int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len); - assert(bytes >> hw->info.shift << hw->info.shift == bytes); + assert(bytes % hw->info.bytes_per_frame == 0); if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) { dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n", bytes, strerror(errno)); } - wav->total_samples += bytes >> hw->info.shift; + wav->total_samples += bytes / hw->info.bytes_per_frame; return bytes; } @@ -134,7 +134,7 @@ static void wav_fini_out (HWVoiceOut *hw) WAVVoiceOut *wav = (WAVVoiceOut *) hw; uint8_t rlen[4]; uint8_t dlen[4]; - uint32_t datalen = wav->total_samples << hw->info.shift; + uint32_t datalen = wav->total_samples * hw->info.bytes_per_frame; uint32_t rifflen = datalen + 36; if (!wav->f) { diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c index ae42e5a2f1..ea604bbb8e 100644 --- a/hw/usb/dev-audio.c +++ b/hw/usb/dev-audio.c @@ -37,11 +37,15 @@ #include "desc.h" #include "audio/audio.h" +static void usb_audio_reinit(USBDevice *dev, unsigned channels); + #define USBAUDIO_VENDOR_NUM 0x46f4 /* CRC16() of "QEMU" */ #define USBAUDIO_PRODUCT_NUM 0x0002 #define DEV_CONFIG_VALUE 1 /* The one and only */ +#define USBAUDIO_MAX_CHANNELS(s) (s->multi ? 8 : 2) + /* Descriptor subtypes for AC interfaces */ #define DST_AC_HEADER 1 #define DST_AC_INPUT_TERMINAL 2 @@ -80,6 +84,27 @@ static const USBDescStrings usb_audio_stringtable = { [STRING_REAL_STREAM] = "Audio Output - 48 kHz Stereo", }; +/* + * A USB audio device supports an arbitrary number of alternate + * interface settings for each interface. Each corresponds to a block + * diagram of parameterized blocks. This can thus refer to things like + * number of channels, data rates, or in fact completely different + * block diagrams. Alternative setting 0 is always the null block diagram, + * which is used by a disabled device. + */ +enum usb_audio_altset { + ALTSET_OFF = 0x00, /* No endpoint */ + ALTSET_STEREO = 0x01, /* Single endpoint */ + ALTSET_51 = 0x02, + ALTSET_71 = 0x03, +}; + +static unsigned altset_channels[] = { + [ALTSET_STEREO] = 2, + [ALTSET_51] = 6, + [ALTSET_71] = 8, +}; + #define U16(x) ((x) & 0xff), (((x) >> 8) & 0xff) #define U24(x) U16(x), (((x) >> 16) & 0xff) #define U32(x) U24(x), (((x) >> 24) & 0xff) @@ -87,7 +112,8 @@ static const USBDescStrings usb_audio_stringtable = { /* * A Basic Audio Device uses these specific values */ -#define USBAUDIO_PACKET_SIZE 192 +#define USBAUDIO_PACKET_SIZE_BASE 96 +#define USBAUDIO_PACKET_SIZE(channels) (USBAUDIO_PACKET_SIZE_BASE * channels) #define USBAUDIO_SAMPLE_RATE 48000 #define USBAUDIO_PACKET_INTERVAL 1 @@ -121,7 +147,7 @@ static const USBDescIface desc_iface[] = { 0x01, /* u8 bTerminalID */ U16(0x0101), /* u16 wTerminalType */ 0x00, /* u8 bAssocTerminal */ - 0x02, /* u16 bNrChannels */ + 0x02, /* u8 bNrChannels */ U16(0x0003), /* u16 wChannelConfig */ 0x00, /* u8 iChannelNames */ STRING_INPUT_TERMINAL, /* u8 iTerminal */ @@ -156,14 +182,14 @@ static const USBDescIface desc_iface[] = { }, },{ .bInterfaceNumber = 1, - .bAlternateSetting = 0, + .bAlternateSetting = ALTSET_OFF, .bNumEndpoints = 0, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING, .iInterface = STRING_NULL_STREAM, },{ .bInterfaceNumber = 1, - .bAlternateSetting = 1, + .bAlternateSetting = ALTSET_STEREO, .bNumEndpoints = 1, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING, @@ -199,7 +225,7 @@ static const USBDescIface desc_iface[] = { { .bEndpointAddress = USB_DIR_OUT | 0x01, .bmAttributes = 0x0d, - .wMaxPacketSize = USBAUDIO_PACKET_SIZE, + .wMaxPacketSize = USBAUDIO_PACKET_SIZE(2), .bInterval = 1, .is_audio = 1, /* Stereo Headphone Class-specific @@ -247,17 +273,274 @@ static const USBDesc desc_audio = { .str = usb_audio_stringtable, }; -/* - * A USB audio device supports an arbitrary number of alternate - * interface settings for each interface. Each corresponds to a block - * diagram of parameterized blocks. This can thus refer to things like - * number of channels, data rates, or in fact completely different - * block diagrams. Alternative setting 0 is always the null block diagram, - * which is used by a disabled device. - */ -enum usb_audio_altset { - ALTSET_OFF = 0x00, /* No endpoint */ - ALTSET_ON = 0x01, /* Single endpoint */ +/* multi channel compatible desc */ + +static const USBDescIface desc_iface_multi[] = { + { + .bInterfaceNumber = 0, + .bNumEndpoints = 0, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceProtocol = 0x04, + .iInterface = STRING_USBAUDIO_CONTROL, + .ndesc = 4, + .descs = (USBDescOther[]) { + { + /* Headphone Class-Specific AC Interface Header Descriptor */ + .data = (uint8_t[]) { + 0x09, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AC_HEADER, /* u8 bDescriptorSubtype */ + U16(0x0100), /* u16 bcdADC */ + U16(0x38), /* u16 wTotalLength */ + 0x01, /* u8 bInCollection */ + 0x01, /* u8 baInterfaceNr */ + } + },{ + /* Generic Stereo Input Terminal ID1 Descriptor */ + .data = (uint8_t[]) { + 0x0c, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AC_INPUT_TERMINAL, /* u8 bDescriptorSubtype */ + 0x01, /* u8 bTerminalID */ + U16(0x0101), /* u16 wTerminalType */ + 0x00, /* u8 bAssocTerminal */ + 0x08, /* u8 bNrChannels */ + U16(0x063f), /* u16 wChannelConfig */ + 0x00, /* u8 iChannelNames */ + STRING_INPUT_TERMINAL, /* u8 iTerminal */ + } + },{ + /* Generic Stereo Feature Unit ID2 Descriptor */ + .data = (uint8_t[]) { + 0x19, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AC_FEATURE_UNIT, /* u8 bDescriptorSubtype */ + 0x02, /* u8 bUnitID */ + 0x01, /* u8 bSourceID */ + 0x02, /* u8 bControlSize */ + U16(0x0001), /* u16 bmaControls(0) */ + U16(0x0002), /* u16 bmaControls(1) */ + U16(0x0002), /* u16 bmaControls(2) */ + U16(0x0002), /* u16 bmaControls(3) */ + U16(0x0002), /* u16 bmaControls(4) */ + U16(0x0002), /* u16 bmaControls(5) */ + U16(0x0002), /* u16 bmaControls(6) */ + U16(0x0002), /* u16 bmaControls(7) */ + U16(0x0002), /* u16 bmaControls(8) */ + STRING_FEATURE_UNIT, /* u8 iFeature */ + } + },{ + /* Headphone Ouptut Terminal ID3 Descriptor */ + .data = (uint8_t[]) { + 0x09, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AC_OUTPUT_TERMINAL, /* u8 bDescriptorSubtype */ + 0x03, /* u8 bUnitID */ + U16(0x0301), /* u16 wTerminalType (SPK) */ + 0x00, /* u8 bAssocTerminal */ + 0x02, /* u8 bSourceID */ + STRING_OUTPUT_TERMINAL, /* u8 iTerminal */ + } + } + }, + },{ + .bInterfaceNumber = 1, + .bAlternateSetting = ALTSET_OFF, + .bNumEndpoints = 0, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING, + .iInterface = STRING_NULL_STREAM, + },{ + .bInterfaceNumber = 1, + .bAlternateSetting = ALTSET_STEREO, + .bNumEndpoints = 1, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING, + .iInterface = STRING_REAL_STREAM, + .ndesc = 2, + .descs = (USBDescOther[]) { + { + /* Headphone Class-specific AS General Interface Descriptor */ + .data = (uint8_t[]) { + 0x07, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AS_GENERAL, /* u8 bDescriptorSubtype */ + 0x01, /* u8 bTerminalLink */ + 0x00, /* u8 bDelay */ + 0x01, 0x00, /* u16 wFormatTag */ + } + },{ + /* Headphone Type I Format Type Descriptor */ + .data = (uint8_t[]) { + 0x0b, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AS_FORMAT_TYPE, /* u8 bDescriptorSubtype */ + 0x01, /* u8 bFormatType */ + 0x02, /* u8 bNrChannels */ + 0x02, /* u8 bSubFrameSize */ + 0x10, /* u8 bBitResolution */ + 0x01, /* u8 bSamFreqType */ + U24(USBAUDIO_SAMPLE_RATE), /* u24 tSamFreq */ + } + } + }, + .eps = (USBDescEndpoint[]) { + { + .bEndpointAddress = USB_DIR_OUT | 0x01, + .bmAttributes = 0x0d, + .wMaxPacketSize = USBAUDIO_PACKET_SIZE(2), + .bInterval = 1, + .is_audio = 1, + /* Stereo Headphone Class-specific + AS Audio Data Endpoint Descriptor */ + .extra = (uint8_t[]) { + 0x07, /* u8 bLength */ + USB_DT_CS_ENDPOINT, /* u8 bDescriptorType */ + DST_EP_GENERAL, /* u8 bDescriptorSubtype */ + 0x00, /* u8 bmAttributes */ + 0x00, /* u8 bLockDelayUnits */ + U16(0x0000), /* u16 wLockDelay */ + }, + }, + } + },{ + .bInterfaceNumber = 1, + .bAlternateSetting = ALTSET_51, + .bNumEndpoints = 1, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING, + .iInterface = STRING_REAL_STREAM, + .ndesc = 2, + .descs = (USBDescOther[]) { + { + /* Headphone Class-specific AS General Interface Descriptor */ + .data = (uint8_t[]) { + 0x07, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AS_GENERAL, /* u8 bDescriptorSubtype */ + 0x01, /* u8 bTerminalLink */ + 0x00, /* u8 bDelay */ + 0x01, 0x00, /* u16 wFormatTag */ + } + },{ + /* Headphone Type I Format Type Descriptor */ + .data = (uint8_t[]) { + 0x0b, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AS_FORMAT_TYPE, /* u8 bDescriptorSubtype */ + 0x01, /* u8 bFormatType */ + 0x06, /* u8 bNrChannels */ + 0x02, /* u8 bSubFrameSize */ + 0x10, /* u8 bBitResolution */ + 0x01, /* u8 bSamFreqType */ + U24(USBAUDIO_SAMPLE_RATE), /* u24 tSamFreq */ + } + } + }, + .eps = (USBDescEndpoint[]) { + { + .bEndpointAddress = USB_DIR_OUT | 0x01, + .bmAttributes = 0x0d, + .wMaxPacketSize = USBAUDIO_PACKET_SIZE(6), + .bInterval = 1, + .is_audio = 1, + /* Stereo Headphone Class-specific + AS Audio Data Endpoint Descriptor */ + .extra = (uint8_t[]) { + 0x07, /* u8 bLength */ + USB_DT_CS_ENDPOINT, /* u8 bDescriptorType */ + DST_EP_GENERAL, /* u8 bDescriptorSubtype */ + 0x00, /* u8 bmAttributes */ + 0x00, /* u8 bLockDelayUnits */ + U16(0x0000), /* u16 wLockDelay */ + }, + }, + } + },{ + .bInterfaceNumber = 1, + .bAlternateSetting = ALTSET_71, + .bNumEndpoints = 1, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING, + .iInterface = STRING_REAL_STREAM, + .ndesc = 2, + .descs = (USBDescOther[]) { + { + /* Headphone Class-specific AS General Interface Descriptor */ + .data = (uint8_t[]) { + 0x07, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AS_GENERAL, /* u8 bDescriptorSubtype */ + 0x01, /* u8 bTerminalLink */ + 0x00, /* u8 bDelay */ + 0x01, 0x00, /* u16 wFormatTag */ + } + },{ + /* Headphone Type I Format Type Descriptor */ + .data = (uint8_t[]) { + 0x0b, /* u8 bLength */ + USB_DT_CS_INTERFACE, /* u8 bDescriptorType */ + DST_AS_FORMAT_TYPE, /* u8 bDescriptorSubtype */ + 0x01, /* u8 bFormatType */ + 0x08, /* u8 bNrChannels */ + 0x02, /* u8 bSubFrameSize */ + 0x10, /* u8 bBitResolution */ + 0x01, /* u8 bSamFreqType */ + U24(USBAUDIO_SAMPLE_RATE), /* u24 tSamFreq */ + } + } + }, + .eps = (USBDescEndpoint[]) { + { + .bEndpointAddress = USB_DIR_OUT | 0x01, + .bmAttributes = 0x0d, + .wMaxPacketSize = USBAUDIO_PACKET_SIZE(8), + .bInterval = 1, + .is_audio = 1, + /* Stereo Headphone Class-specific + AS Audio Data Endpoint Descriptor */ + .extra = (uint8_t[]) { + 0x07, /* u8 bLength */ + USB_DT_CS_ENDPOINT, /* u8 bDescriptorType */ + DST_EP_GENERAL, /* u8 bDescriptorSubtype */ + 0x00, /* u8 bmAttributes */ + 0x00, /* u8 bLockDelayUnits */ + U16(0x0000), /* u16 wLockDelay */ + }, + }, + } + } +}; + +static const USBDescDevice desc_device_multi = { + .bcdUSB = 0x0100, + .bMaxPacketSize0 = 64, + .bNumConfigurations = 1, + .confs = (USBDescConfig[]) { + { + .bNumInterfaces = 2, + .bConfigurationValue = DEV_CONFIG_VALUE, + .iConfiguration = STRING_CONFIG, + .bmAttributes = USB_CFG_ATT_ONE | USB_CFG_ATT_SELFPOWER, + .bMaxPower = 0x32, + .nif = ARRAY_SIZE(desc_iface_multi), + .ifs = desc_iface_multi, + } + }, +}; + +static const USBDesc desc_audio_multi = { + .id = { + .idVendor = USBAUDIO_VENDOR_NUM, + .idProduct = USBAUDIO_PRODUCT_NUM, + .bcdDevice = 0, + .iManufacturer = STRING_MANUFACTURER, + .iProduct = STRING_PRODUCT, + .iSerialNumber = STRING_SERIALNUMBER, + }, + .full = &desc_device_multi, + .str = usb_audio_stringtable, }; /* @@ -295,15 +578,16 @@ enum usb_audio_altset { struct streambuf { uint8_t *data; - uint32_t size; - uint32_t prod; - uint32_t cons; + size_t size; + uint64_t prod; + uint64_t cons; }; -static void streambuf_init(struct streambuf *buf, uint32_t size) +static void streambuf_init(struct streambuf *buf, uint32_t size, + uint32_t channels) { g_free(buf->data); - buf->size = size - (size % USBAUDIO_PACKET_SIZE); + buf->size = size - (size % USBAUDIO_PACKET_SIZE(channels)); buf->data = g_malloc(buf->size); buf->prod = 0; buf->cons = 0; @@ -315,34 +599,37 @@ static void streambuf_fini(struct streambuf *buf) buf->data = NULL; } -static int streambuf_put(struct streambuf *buf, USBPacket *p) +static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels) { - uint32_t free = buf->size - (buf->prod - buf->cons); + int64_t free = buf->size - (buf->prod - buf->cons); - if (!free) { + if (free < USBAUDIO_PACKET_SIZE(channels)) { return 0; } - if (p->iov.size != USBAUDIO_PACKET_SIZE) { + if (p->iov.size != USBAUDIO_PACKET_SIZE(channels)) { return 0; } - assert(free >= USBAUDIO_PACKET_SIZE); + + /* can happen if prod overflows */ + assert(buf->prod % USBAUDIO_PACKET_SIZE(channels) == 0); usb_packet_copy(p, buf->data + (buf->prod % buf->size), - USBAUDIO_PACKET_SIZE); - buf->prod += USBAUDIO_PACKET_SIZE; - return USBAUDIO_PACKET_SIZE; + USBAUDIO_PACKET_SIZE(channels)); + buf->prod += USBAUDIO_PACKET_SIZE(channels); + return USBAUDIO_PACKET_SIZE(channels); } -static uint8_t *streambuf_get(struct streambuf *buf) +static uint8_t *streambuf_get(struct streambuf *buf, size_t *len) { - uint32_t used = buf->prod - buf->cons; + int64_t used = buf->prod - buf->cons; uint8_t *data; - if (!used) { + if (used <= 0) { + *len = 0; return NULL; } - assert(used >= USBAUDIO_PACKET_SIZE); data = buf->data + (buf->cons % buf->size); - buf->cons += USBAUDIO_PACKET_SIZE; + *len = MIN(buf->prod - buf->cons, + buf->size - (buf->cons % buf->size)); return data; } @@ -356,14 +643,15 @@ typedef struct USBAudioState { enum usb_audio_altset altset; struct audsettings as; SWVoiceOut *voice; - bool mute; - uint8_t vol[2]; + Volume vol; struct streambuf buf; + uint32_t channels; } out; /* properties */ uint32_t debug; - uint32_t buffer; + uint32_t buffer_user, buffer; + bool multi; } USBAudioState; #define TYPE_USB_AUDIO "usb-audio" @@ -374,16 +662,21 @@ static void output_callback(void *opaque, int avail) USBAudioState *s = opaque; uint8_t *data; - for (;;) { - if (avail < USBAUDIO_PACKET_SIZE) { - return; - } - data = streambuf_get(&s->out.buf); + while (avail) { + size_t written, len; + + data = streambuf_get(&s->out.buf, &len); if (!data) { return; } - AUD_write(s->out.voice, data, USBAUDIO_PACKET_SIZE); - avail -= USBAUDIO_PACKET_SIZE; + + written = AUD_write(s->out.voice, data, len); + avail -= written; + s->out.buf.cons += written; + + if (written < len) { + return; + } } } @@ -391,10 +684,15 @@ static int usb_audio_set_output_altset(USBAudioState *s, int altset) { switch (altset) { case ALTSET_OFF: - streambuf_init(&s->out.buf, s->buffer); AUD_set_active_out(s->out.voice, false); break; - case ALTSET_ON: + case ALTSET_STEREO: + case ALTSET_51: + case ALTSET_71: + if (s->out.channels != altset_channels[altset]) { + usb_audio_reinit(USB_DEVICE(s), altset_channels[altset]); + } + streambuf_init(&s->out.buf, s->buffer, s->out.channels); AUD_set_active_out(s->out.voice, true); break; default: @@ -425,33 +723,33 @@ static int usb_audio_get_control(USBAudioState *s, uint8_t attrib, switch (aid) { case ATTRIB_ID(MUTE_CONTROL, CR_GET_CUR, 0x0200): - data[0] = s->out.mute; + data[0] = s->out.vol.mute; ret = 1; break; case ATTRIB_ID(VOLUME_CONTROL, CR_GET_CUR, 0x0200): - if (cn < 2) { - uint16_t vol = (s->out.vol[cn] * 0x8800 + 127) / 255 + 0x8000; + if (cn < USBAUDIO_MAX_CHANNELS(s)) { + uint16_t vol = (s->out.vol.vol[cn] * 0x8800 + 127) / 255 + 0x8000; data[0] = vol; data[1] = vol >> 8; ret = 2; } break; case ATTRIB_ID(VOLUME_CONTROL, CR_GET_MIN, 0x0200): - if (cn < 2) { + if (cn < USBAUDIO_MAX_CHANNELS(s)) { data[0] = 0x01; data[1] = 0x80; ret = 2; } break; case ATTRIB_ID(VOLUME_CONTROL, CR_GET_MAX, 0x0200): - if (cn < 2) { + if (cn < USBAUDIO_MAX_CHANNELS(s)) { data[0] = 0x00; data[1] = 0x08; ret = 2; } break; case ATTRIB_ID(VOLUME_CONTROL, CR_GET_RES, 0x0200): - if (cn < 2) { + if (cn < USBAUDIO_MAX_CHANNELS(s)) { data[0] = 0x88; data[1] = 0x00; ret = 2; @@ -473,16 +771,17 @@ static int usb_audio_set_control(USBAudioState *s, uint8_t attrib, switch (aid) { case ATTRIB_ID(MUTE_CONTROL, CR_SET_CUR, 0x0200): - s->out.mute = data[0] & 1; + s->out.vol.mute = data[0] & 1; set_vol = true; ret = 0; break; case ATTRIB_ID(VOLUME_CONTROL, CR_SET_CUR, 0x0200): - if (cn < 2) { + if (cn < USBAUDIO_MAX_CHANNELS(s)) { uint16_t vol = data[0] + (data[1] << 8); if (s->debug) { - fprintf(stderr, "usb-audio: vol %04x\n", (uint16_t)vol); + fprintf(stderr, "usb-audio: cn %d vol %04x\n", cn, + (uint16_t)vol); } vol -= 0x8000; @@ -491,7 +790,7 @@ static int usb_audio_set_control(USBAudioState *s, uint8_t attrib, vol = 255; } - s->out.vol[cn] = vol; + s->out.vol.vol[cn] = vol; set_vol = true; ret = 0; } @@ -500,11 +799,14 @@ static int usb_audio_set_control(USBAudioState *s, uint8_t attrib, if (set_vol) { if (s->debug) { - fprintf(stderr, "usb-audio: mute %d, lvol %3d, rvol %3d\n", - s->out.mute, s->out.vol[0], s->out.vol[1]); + int i; + fprintf(stderr, "usb-audio: mute %d", s->out.vol.mute); + for (i = 0; i < USBAUDIO_MAX_CHANNELS(s); ++i) { + fprintf(stderr, ", vol[%d] %3d", i, s->out.vol.vol[i]); + } + fprintf(stderr, "\n"); } - AUD_set_volume_out(s->out.voice, s->out.mute, - s->out.vol[0], s->out.vol[1]); + audio_set_volume_out(s->out.voice, &s->out.vol); } return ret; @@ -597,7 +899,7 @@ static void usb_audio_handle_dataout(USBAudioState *s, USBPacket *p) return; } - streambuf_put(&s->out.buf, p); + streambuf_put(&s->out.buf, p, s->out.channels); if (p->actual_length < p->iov.size && s->debug > 1) { fprintf(stderr, "usb-audio: output overrun (%zd bytes)\n", p->iov.size - p->actual_length); @@ -639,6 +941,9 @@ static void usb_audio_unrealize(USBDevice *dev, Error **errp) static void usb_audio_realize(USBDevice *dev, Error **errp) { USBAudioState *s = USB_AUDIO(dev); + int i; + + dev->usb_desc = s->multi ? &desc_audio_multi : &desc_audio; usb_desc_create_serial(dev); usb_desc_init(dev); @@ -646,18 +951,35 @@ static void usb_audio_realize(USBDevice *dev, Error **errp) AUD_register_card(TYPE_USB_AUDIO, &s->card); s->out.altset = ALTSET_OFF; - s->out.mute = false; - s->out.vol[0] = 240; /* 0 dB */ - s->out.vol[1] = 240; /* 0 dB */ + s->out.vol.mute = false; + for (i = 0; i < USBAUDIO_MAX_CHANNELS(s); ++i) { + s->out.vol.vol[i] = 240; /* 0 dB */ + } + + usb_audio_reinit(dev, 2); +} + +static void usb_audio_reinit(USBDevice *dev, unsigned channels) +{ + USBAudioState *s = USB_AUDIO(dev); + + s->out.channels = channels; + if (!s->buffer_user) { + s->buffer = 32 * USBAUDIO_PACKET_SIZE(s->out.channels); + } else { + s->buffer = s->buffer_user; + } + + s->out.vol.channels = s->out.channels; s->out.as.freq = USBAUDIO_SAMPLE_RATE; - s->out.as.nchannels = 2; + s->out.as.nchannels = s->out.channels; s->out.as.fmt = AUDIO_FORMAT_S16; s->out.as.endianness = 0; - streambuf_init(&s->out.buf, s->buffer); + streambuf_init(&s->out.buf, s->buffer, s->out.channels); s->out.voice = AUD_open_out(&s->card, s->out.voice, TYPE_USB_AUDIO, s, output_callback, &s->out.as); - AUD_set_volume_out(s->out.voice, s->out.mute, s->out.vol[0], s->out.vol[1]); + audio_set_volume_out(s->out.voice, &s->out.vol); AUD_set_active_out(s->out.voice, 0); } @@ -669,8 +991,8 @@ static const VMStateDescription vmstate_usb_audio = { static Property usb_audio_properties[] = { DEFINE_AUDIO_PROPERTIES(USBAudioState, card), DEFINE_PROP_UINT32("debug", USBAudioState, debug, 0), - DEFINE_PROP_UINT32("buffer", USBAudioState, buffer, - 32 * USBAUDIO_PACKET_SIZE), + DEFINE_PROP_UINT32("buffer", USBAudioState, buffer_user, 0), + DEFINE_PROP_BOOL("multi", USBAudioState, multi, false), DEFINE_PROP_END_OF_LIST(), }; @@ -683,7 +1005,6 @@ static void usb_audio_class_init(ObjectClass *klass, void *data) dc->props = usb_audio_properties; set_bit(DEVICE_CATEGORY_SOUND, dc->categories); k->product_desc = "QEMU USB Audio Interface"; - k->usb_desc = &desc_audio; k->realize = usb_audio_realize; k->handle_reset = usb_audio_handle_reset; k->handle_control = usb_audio_handle_control; diff --git a/qapi/audio.json b/qapi/audio.json index 9fefdf5186..83312b2339 100644 --- a/qapi/audio.json +++ b/qapi/audio.json @@ -11,6 +11,11 @@ # General audio backend options that are used for both playback and # recording. # +# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU and +# convert audio formats when not supported by the backend. When +# set to off, fixed-settings must be also off (default on, +# since 4.2) +# # @fixed-settings: use fixed settings for host input/output. When off, # frequency, channels and format must not be # specified (default true) @@ -31,6 +36,7 @@ ## { 'struct': 'AudiodevPerDirectionOptions', 'data': { + '*mixing-engine': 'bool', '*fixed-settings': 'bool', '*frequency': 'uint32', '*channels': 'uint32', @@ -206,6 +212,11 @@ # # @name: name of the sink/source to use # +# @stream-name: name of the PulseAudio stream created by qemu. Can be +# used to identify the stream in PulseAudio when you +# create multiple PulseAudio devices or run multiple qemu +# instances (default: audiodev's id, since 4.2) +# # @latency: latency you want PulseAudio to achieve in microseconds # (default 15000) # @@ -215,6 +226,7 @@ 'base': 'AudiodevPerDirectionOptions', 'data': { '*name': 'str', + '*stream-name': 'str', '*latency': 'uint32' } } ## diff --git a/qemu-options.hx b/qemu-options.hx index 793d70ff93..996b6fba74 100644 --- a/qemu-options.hx +++ b/qemu-options.hx @@ -433,6 +433,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev, " specifies the audio backend to use\n" " id= identifier of the backend\n" " timer-period= timer period in microseconds\n" + " in|out.mixing-engine= use mixing engine to mix streams inside QEMU\n" " in|out.fixed-settings= use fixed settings for host audio\n" " in|out.frequency= frequency to use with fixed settings\n" " in|out.channels= number of channels to use with fixed settings\n" @@ -493,6 +494,10 @@ output's property with @code{out.@var{prop}}. For example: -audiodev alsa,id=example,out.channels=1 # leaves in.channels unspecified @end example +NOTE: parameter validation is known to be incomplete, in many cases +specifying an invalid option causes QEMU to print an error message and +continue emulation without sound. + Valid global options are: @table @option @@ -503,6 +508,16 @@ Identifies the audio backend. Sets the timer @var{period} used by the audio subsystem in microseconds. Default is 10000 (10 ms). +@item in|out.mixing-engine=on|off +Use QEMU's mixing engine to mix all streams inside QEMU and convert +audio formats when not supported by the backend. When off, +@var{fixed-settings} must be off too. Note that disabling this option +means that the selected backend must support multiple streams and the +audio formats used by the virtual cards, otherwise you'll get no sound. +It's not recommended to disable this option unless you want to use 5.1 +or 7.1 audio, as mixing engine only supports mono and stereo audio. +Default is on. + @item in|out.fixed-settings=on|off Use fixed settings for host audio. When off, it will change based on how the guest opens the sound card. In this case you must not specify