| Message ID | 20230206185237.8358-1-vr_qemu@t-online.de |
|---|---|
| State | New |
| Headers | show |
| Series | audio: improve callback interface for audio frontends | expand |
On Mon, Feb 6, 2023 at 10:52 PM Volker Rümelin <vr_qemu@t-online.de> wrote: > > Change the type of mix_buf in struct HWVoiceOut and conv_buf > in struct HWVoiceIn from STSampleBuffer * to STSampleBuffer. > However, a buffer pointer is still needed. For this reason in > struct STSampleBuffer samples[] is changed to *buffer. > > This is a preparation for the next patch. The next patch will > add this line, which is not possible with the current struct > STSampleBuffer definition. > > + sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2; > > There are no functional changes. > > Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk> > Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com> > --- > audio/audio.c | 106 ++++++++++++++++++++--------------------- > audio/audio_int.h | 6 +-- > audio/audio_template.h | 19 ++++---- > 3 files changed, 67 insertions(+), 64 deletions(-) > > diff --git a/audio/audio.c b/audio/audio.c > index 772c3cc320..a0b54e4a2e 100644 > --- a/audio/audio.c > +++ b/audio/audio.c > @@ -523,8 +523,8 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw) > static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) > { > size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw); > - if (audio_bug(__func__, live > hw->conv_buf->size)) { > - dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); > + if (audio_bug(__func__, live > hw->conv_buf.size)) { > + dolog("live=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size); > return 0; > } > return live; > @@ -533,13 +533,13 @@ static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) > static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples) > { > size_t conv = 0; > - STSampleBuffer *conv_buf = hw->conv_buf; > + STSampleBuffer *conv_buf = &hw->conv_buf; > > while (samples) { > uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame); > size_t proc = MIN(samples, conv_buf->size - conv_buf->pos); > > - hw->conv(conv_buf->samples + conv_buf->pos, src, proc); > + hw->conv(conv_buf->buffer + conv_buf->pos, src, proc); > conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size; > samples -= proc; > conv += proc; > @@ -561,12 +561,12 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) > if (!live) { > return 0; > } > - if (audio_bug(__func__, live > hw->conv_buf->size)) { > - dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); > + if (audio_bug(__func__, live > hw->conv_buf.size)) { > + dolog("live_in=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size); > return 0; > } > > - rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size); > + rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size); > > samples = size / sw->info.bytes_per_frame; > > @@ -574,11 +574,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) > swlim = MIN (swlim, samples); > > while (swlim) { > - src = hw->conv_buf->samples + rpos; > - if (hw->conv_buf->pos > rpos) { > - isamp = hw->conv_buf->pos - rpos; > + src = hw->conv_buf.buffer + rpos; > + if (hw->conv_buf.pos > rpos) { > + isamp = hw->conv_buf.pos - rpos; > } else { > - isamp = hw->conv_buf->size - rpos; > + isamp = hw->conv_buf.size - rpos; > } > > if (!isamp) { > @@ -588,7 +588,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) > > st_rate_flow (sw->rate, src, dst, &isamp, &osamp); > swlim -= osamp; > - rpos = (rpos + isamp) % hw->conv_buf->size; > + rpos = (rpos + isamp) % hw->conv_buf.size; > dst += osamp; > ret += osamp; > total += isamp; > @@ -636,8 +636,8 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) > if (nb_live1) { > size_t live = smin; > > - if (audio_bug(__func__, live > hw->mix_buf->size)) { > - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); > + if (audio_bug(__func__, live > hw->mix_buf.size)) { > + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size); > return 0; > } > return live; > @@ -654,17 +654,17 @@ static size_t audio_pcm_hw_get_free(HWVoiceOut *hw) > static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) > { > size_t clipped = 0; > - size_t pos = hw->mix_buf->pos; > + size_t pos = hw->mix_buf.pos; > > while (len) { > - st_sample *src = hw->mix_buf->samples + pos; > + st_sample *src = hw->mix_buf.buffer + pos; > 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_till_end_of_buf = hw->mix_buf.size - pos; > size_t samples_to_clip = MIN(len, samples_till_end_of_buf); > > hw->clip(dst, src, samples_to_clip); > > - pos = (pos + samples_to_clip) % hw->mix_buf->size; > + pos = (pos + samples_to_clip) % hw->mix_buf.size; > len -= samples_to_clip; > clipped += samples_to_clip; > } > @@ -683,11 +683,11 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) > return size; > } > > - hwsamples = sw->hw->mix_buf->size; > + hwsamples = sw->hw->mix_buf.size; > > live = sw->total_hw_samples_mixed; > if (audio_bug(__func__, live > hwsamples)) { > - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples); > + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hwsamples); > return 0; > } > > @@ -698,7 +698,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) > return 0; > } > > - wpos = (sw->hw->mix_buf->pos + live) % hwsamples; > + wpos = (sw->hw->mix_buf.pos + live) % hwsamples; > > dead = hwsamples - live; > hw_free = audio_pcm_hw_get_free(sw->hw); > @@ -725,7 +725,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) > st_rate_flow_mix ( > sw->rate, > sw->buf + pos, > - sw->hw->mix_buf->samples + wpos, > + sw->hw->mix_buf.buffer + wpos, > &isamp, > &osamp > ); > @@ -989,9 +989,9 @@ static size_t audio_get_avail (SWVoiceIn *sw) > } > > live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired; > - if (audio_bug(__func__, live > sw->hw->conv_buf->size)) { > - dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live, > - sw->hw->conv_buf->size); > + if (audio_bug(__func__, live > sw->hw->conv_buf.size)) { > + dolog("live=%zu sw->hw->conv_buf.size=%zu\n", live, > + sw->hw->conv_buf.size); > return 0; > } > > @@ -1026,13 +1026,13 @@ static size_t audio_get_free(SWVoiceOut *sw) > > live = sw->total_hw_samples_mixed; > > - if (audio_bug(__func__, live > sw->hw->mix_buf->size)) { > - dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live, > - sw->hw->mix_buf->size); > + if (audio_bug(__func__, live > sw->hw->mix_buf.size)) { > + dolog("live=%zu sw->hw->mix_buf.size=%zu\n", live, > + sw->hw->mix_buf.size); > return 0; > } > > - dead = sw->hw->mix_buf->size - live; > + dead = sw->hw->mix_buf.size - live; > > #ifdef DEBUG_OUT > dolog("%s: get_free live %zu dead %zu frontend frames %zu\n", > @@ -1056,12 +1056,12 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, > > n = samples; > while (n) { > - size_t till_end_of_hw = hw->mix_buf->size - rpos2; > + 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.bytes_per_frame; > size_t written; > > - sw->buf = hw->mix_buf->samples + rpos2; > + sw->buf = hw->mix_buf.buffer + rpos2; > written = audio_pcm_sw_write (sw, NULL, bytes); > if (written - bytes) { > dolog("Could not mix %zu bytes into a capture " > @@ -1070,14 +1070,14 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, > break; > } > n -= to_write; > - rpos2 = (rpos2 + to_write) % hw->mix_buf->size; > + rpos2 = (rpos2 + to_write) % hw->mix_buf.size; > } > } > } > > - n = MIN(samples, hw->mix_buf->size - rpos); > - mixeng_clear(hw->mix_buf->samples + rpos, n); > - mixeng_clear(hw->mix_buf->samples, samples - n); > + n = MIN(samples, hw->mix_buf.size - rpos); > + mixeng_clear(hw->mix_buf.buffer + rpos, n); > + mixeng_clear(hw->mix_buf.buffer, samples - n); > } > > static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) > @@ -1103,7 +1103,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) > > live -= proc; > clipped += proc; > - hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size; > + hw->mix_buf.pos = (hw->mix_buf.pos + proc) % hw->mix_buf.size; > > if (proc == 0 || proc < decr) { > break; > @@ -1174,8 +1174,8 @@ static void audio_run_out (AudioState *s) > live = 0; > } > > - if (audio_bug(__func__, live > hw->mix_buf->size)) { > - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); > + if (audio_bug(__func__, live > hw->mix_buf.size)) { > + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size); > continue; > } > > @@ -1203,13 +1203,13 @@ static void audio_run_out (AudioState *s) > continue; > } > > - prev_rpos = hw->mix_buf->pos; > + prev_rpos = hw->mix_buf.pos; > played = audio_pcm_hw_run_out(hw, live); > replay_audio_out(&played); > - if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) { > - dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n", > - hw->mix_buf->pos, hw->mix_buf->size, played); > - hw->mix_buf->pos = 0; > + if (audio_bug(__func__, hw->mix_buf.pos >= hw->mix_buf.size)) { > + dolog("hw->mix_buf.pos=%zu hw->mix_buf.size=%zu played=%zu\n", > + hw->mix_buf.pos, hw->mix_buf.size, played); > + hw->mix_buf.pos = 0; > } > > #ifdef DEBUG_OUT > @@ -1290,10 +1290,10 @@ static void audio_run_in (AudioState *s) > > if (replay_mode != REPLAY_MODE_PLAY) { > captured = audio_pcm_hw_run_in( > - hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw)); > + hw, hw->conv_buf.size - audio_pcm_hw_get_live_in(hw)); > } > - replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos, > - hw->conv_buf->size); > + replay_audio_in(&captured, hw->conv_buf.buffer, &hw->conv_buf.pos, > + hw->conv_buf.size); > > min = audio_pcm_hw_find_min_in (hw); > hw->total_samples_captured += captured - min; > @@ -1326,14 +1326,14 @@ static void audio_run_capture (AudioState *s) > SWVoiceOut *sw; > > captured = live = audio_pcm_hw_get_live_out (hw, NULL); > - rpos = hw->mix_buf->pos; > + rpos = hw->mix_buf.pos; > while (live) { > - size_t left = hw->mix_buf->size - rpos; > + size_t left = hw->mix_buf.size - rpos; > size_t to_capture = MIN(live, left); > struct st_sample *src; > struct capture_callback *cb; > > - src = hw->mix_buf->samples + rpos; > + src = hw->mix_buf.buffer + rpos; > hw->clip (cap->buf, src, to_capture); > mixeng_clear (src, to_capture); > > @@ -1341,10 +1341,10 @@ static void audio_run_capture (AudioState *s) > cb->ops.capture (cb->opaque, cap->buf, > to_capture * hw->info.bytes_per_frame); > } > - rpos = (rpos + to_capture) % hw->mix_buf->size; > + rpos = (rpos + to_capture) % hw->mix_buf.size; > live -= to_capture; > } > - hw->mix_buf->pos = rpos; > + hw->mix_buf.pos = rpos; > > for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) { > if (!sw->active && sw->empty) { > @@ -1903,7 +1903,7 @@ CaptureVoiceOut *AUD_add_capture( > > audio_pcm_init_info (&hw->info, as); > > - cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame); > + cap->buf = g_malloc0_n(hw->mix_buf.size, hw->info.bytes_per_frame); > > if (hw->info.is_float) { > hw->clip = mixeng_clip_float[hw->info.nchannels == 2]; > @@ -1955,7 +1955,7 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque) > sw = sw1; > } > QLIST_REMOVE (cap, entries); > - g_free (cap->hw.mix_buf); > + g_free(cap->hw.mix_buf.buffer); > g_free (cap->buf); > g_free (cap); > } > diff --git a/audio/audio_int.h b/audio/audio_int.h > index 5028f2354a..061845dcc2 100644 > --- a/audio/audio_int.h > +++ b/audio/audio_int.h > @@ -58,7 +58,7 @@ typedef struct SWVoiceCap SWVoiceCap; > > typedef struct STSampleBuffer { > size_t pos, size; > - st_sample samples[]; > + st_sample *buffer; > } STSampleBuffer; > > typedef struct HWVoiceOut { > @@ -71,7 +71,7 @@ typedef struct HWVoiceOut { > f_sample *clip; > uint64_t ts_helper; > > - STSampleBuffer *mix_buf; > + STSampleBuffer mix_buf; > void *buf_emul; > size_t pos_emul, pending_emul, size_emul; > > @@ -93,7 +93,7 @@ typedef struct HWVoiceIn { > size_t total_samples_captured; > uint64_t ts_helper; > > - STSampleBuffer *conv_buf; > + STSampleBuffer conv_buf; > void *buf_emul; > size_t pos_emul, pending_emul, size_emul; > > diff --git a/audio/audio_template.h b/audio/audio_template.h > index 980e1f4bd0..dd87170cbd 100644 > --- a/audio/audio_template.h > +++ b/audio/audio_template.h > @@ -71,8 +71,9 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s, > static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw) > { > g_free(hw->buf_emul); > - g_free (HWBUF); > - HWBUF = NULL; > + g_free(HWBUF.buffer); > + HWBUF.buffer = NULL; > + HWBUF.size = 0; > } > > static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) > @@ -83,10 +84,12 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) > dolog("Attempted to allocate empty buffer\n"); > } > > - HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples); > - HWBUF->size = samples; > + HWBUF.buffer = g_new0(st_sample, samples); > + HWBUF.size = samples; > + HWBUF.pos = 0; > } else { > - HWBUF = NULL; > + HWBUF.buffer = NULL; > + HWBUF.size = 0; > } > } > > @@ -111,9 +114,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) > } > > #ifdef DAC > - samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio; > + samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; > #else > - samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32; > + samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; > #endif > if (audio_bug(__func__, samples < 0)) { > dolog("Can not allocate buffer for `%s' (%d samples)\n", > @@ -126,7 +129,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) > size_t f_fe_min; > > /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */ > - f_fe_min = (hw->info.freq + HWBUF->size - 1) / HWBUF->size; > + f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size; > qemu_log_mask(LOG_UNIMP, > AUDIO_CAP ": The guest selected a " NAME " sample rate" > " of %d Hz for %s. Only sample rates >= %zu Hz are" > -- > 2.35.3 > -- Marc-André Lureau
diff --git a/audio/audio.c b/audio/audio.c index 772c3cc320..a0b54e4a2e 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -523,8 +523,8 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw) static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) { size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw); - if (audio_bug(__func__, live > hw->conv_buf->size)) { - dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); + if (audio_bug(__func__, live > hw->conv_buf.size)) { + dolog("live=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size); return 0; } return live; @@ -533,13 +533,13 @@ static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples) { size_t conv = 0; - STSampleBuffer *conv_buf = hw->conv_buf; + STSampleBuffer *conv_buf = &hw->conv_buf; while (samples) { uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame); size_t proc = MIN(samples, conv_buf->size - conv_buf->pos); - hw->conv(conv_buf->samples + conv_buf->pos, src, proc); + hw->conv(conv_buf->buffer + conv_buf->pos, src, proc); conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size; samples -= proc; conv += proc; @@ -561,12 +561,12 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) if (!live) { return 0; } - if (audio_bug(__func__, live > hw->conv_buf->size)) { - dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); + if (audio_bug(__func__, live > hw->conv_buf.size)) { + dolog("live_in=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size); return 0; } - rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size); + rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size); samples = size / sw->info.bytes_per_frame; @@ -574,11 +574,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) swlim = MIN (swlim, samples); while (swlim) { - src = hw->conv_buf->samples + rpos; - if (hw->conv_buf->pos > rpos) { - isamp = hw->conv_buf->pos - rpos; + src = hw->conv_buf.buffer + rpos; + if (hw->conv_buf.pos > rpos) { + isamp = hw->conv_buf.pos - rpos; } else { - isamp = hw->conv_buf->size - rpos; + isamp = hw->conv_buf.size - rpos; } if (!isamp) { @@ -588,7 +588,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) st_rate_flow (sw->rate, src, dst, &isamp, &osamp); swlim -= osamp; - rpos = (rpos + isamp) % hw->conv_buf->size; + rpos = (rpos + isamp) % hw->conv_buf.size; dst += osamp; ret += osamp; total += isamp; @@ -636,8 +636,8 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) if (nb_live1) { size_t live = smin; - if (audio_bug(__func__, live > hw->mix_buf->size)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); + if (audio_bug(__func__, live > hw->mix_buf.size)) { + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size); return 0; } return live; @@ -654,17 +654,17 @@ static size_t audio_pcm_hw_get_free(HWVoiceOut *hw) static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) { size_t clipped = 0; - size_t pos = hw->mix_buf->pos; + size_t pos = hw->mix_buf.pos; while (len) { - st_sample *src = hw->mix_buf->samples + pos; + st_sample *src = hw->mix_buf.buffer + pos; 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_till_end_of_buf = hw->mix_buf.size - pos; size_t samples_to_clip = MIN(len, samples_till_end_of_buf); hw->clip(dst, src, samples_to_clip); - pos = (pos + samples_to_clip) % hw->mix_buf->size; + pos = (pos + samples_to_clip) % hw->mix_buf.size; len -= samples_to_clip; clipped += samples_to_clip; } @@ -683,11 +683,11 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) return size; } - hwsamples = sw->hw->mix_buf->size; + hwsamples = sw->hw->mix_buf.size; live = sw->total_hw_samples_mixed; if (audio_bug(__func__, live > hwsamples)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples); + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hwsamples); return 0; } @@ -698,7 +698,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) return 0; } - wpos = (sw->hw->mix_buf->pos + live) % hwsamples; + wpos = (sw->hw->mix_buf.pos + live) % hwsamples; dead = hwsamples - live; hw_free = audio_pcm_hw_get_free(sw->hw); @@ -725,7 +725,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) st_rate_flow_mix ( sw->rate, sw->buf + pos, - sw->hw->mix_buf->samples + wpos, + sw->hw->mix_buf.buffer + wpos, &isamp, &osamp ); @@ -989,9 +989,9 @@ static size_t audio_get_avail (SWVoiceIn *sw) } live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired; - if (audio_bug(__func__, live > sw->hw->conv_buf->size)) { - dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live, - sw->hw->conv_buf->size); + if (audio_bug(__func__, live > sw->hw->conv_buf.size)) { + dolog("live=%zu sw->hw->conv_buf.size=%zu\n", live, + sw->hw->conv_buf.size); return 0; } @@ -1026,13 +1026,13 @@ static size_t audio_get_free(SWVoiceOut *sw) live = sw->total_hw_samples_mixed; - if (audio_bug(__func__, live > sw->hw->mix_buf->size)) { - dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live, - sw->hw->mix_buf->size); + if (audio_bug(__func__, live > sw->hw->mix_buf.size)) { + dolog("live=%zu sw->hw->mix_buf.size=%zu\n", live, + sw->hw->mix_buf.size); return 0; } - dead = sw->hw->mix_buf->size - live; + dead = sw->hw->mix_buf.size - live; #ifdef DEBUG_OUT dolog("%s: get_free live %zu dead %zu frontend frames %zu\n", @@ -1056,12 +1056,12 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, n = samples; while (n) { - size_t till_end_of_hw = hw->mix_buf->size - rpos2; + 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.bytes_per_frame; size_t written; - sw->buf = hw->mix_buf->samples + rpos2; + sw->buf = hw->mix_buf.buffer + rpos2; written = audio_pcm_sw_write (sw, NULL, bytes); if (written - bytes) { dolog("Could not mix %zu bytes into a capture " @@ -1070,14 +1070,14 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, break; } n -= to_write; - rpos2 = (rpos2 + to_write) % hw->mix_buf->size; + rpos2 = (rpos2 + to_write) % hw->mix_buf.size; } } } - n = MIN(samples, hw->mix_buf->size - rpos); - mixeng_clear(hw->mix_buf->samples + rpos, n); - mixeng_clear(hw->mix_buf->samples, samples - n); + n = MIN(samples, hw->mix_buf.size - rpos); + mixeng_clear(hw->mix_buf.buffer + rpos, n); + mixeng_clear(hw->mix_buf.buffer, samples - n); } static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) @@ -1103,7 +1103,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) live -= proc; clipped += proc; - hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size; + hw->mix_buf.pos = (hw->mix_buf.pos + proc) % hw->mix_buf.size; if (proc == 0 || proc < decr) { break; @@ -1174,8 +1174,8 @@ static void audio_run_out (AudioState *s) live = 0; } - if (audio_bug(__func__, live > hw->mix_buf->size)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); + if (audio_bug(__func__, live > hw->mix_buf.size)) { + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size); continue; } @@ -1203,13 +1203,13 @@ static void audio_run_out (AudioState *s) continue; } - prev_rpos = hw->mix_buf->pos; + prev_rpos = hw->mix_buf.pos; played = audio_pcm_hw_run_out(hw, live); replay_audio_out(&played); - if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) { - dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n", - hw->mix_buf->pos, hw->mix_buf->size, played); - hw->mix_buf->pos = 0; + if (audio_bug(__func__, hw->mix_buf.pos >= hw->mix_buf.size)) { + dolog("hw->mix_buf.pos=%zu hw->mix_buf.size=%zu played=%zu\n", + hw->mix_buf.pos, hw->mix_buf.size, played); + hw->mix_buf.pos = 0; } #ifdef DEBUG_OUT @@ -1290,10 +1290,10 @@ static void audio_run_in (AudioState *s) if (replay_mode != REPLAY_MODE_PLAY) { captured = audio_pcm_hw_run_in( - hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw)); + hw, hw->conv_buf.size - audio_pcm_hw_get_live_in(hw)); } - replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos, - hw->conv_buf->size); + replay_audio_in(&captured, hw->conv_buf.buffer, &hw->conv_buf.pos, + hw->conv_buf.size); min = audio_pcm_hw_find_min_in (hw); hw->total_samples_captured += captured - min; @@ -1326,14 +1326,14 @@ static void audio_run_capture (AudioState *s) SWVoiceOut *sw; captured = live = audio_pcm_hw_get_live_out (hw, NULL); - rpos = hw->mix_buf->pos; + rpos = hw->mix_buf.pos; while (live) { - size_t left = hw->mix_buf->size - rpos; + size_t left = hw->mix_buf.size - rpos; size_t to_capture = MIN(live, left); struct st_sample *src; struct capture_callback *cb; - src = hw->mix_buf->samples + rpos; + src = hw->mix_buf.buffer + rpos; hw->clip (cap->buf, src, to_capture); mixeng_clear (src, to_capture); @@ -1341,10 +1341,10 @@ static void audio_run_capture (AudioState *s) cb->ops.capture (cb->opaque, cap->buf, to_capture * hw->info.bytes_per_frame); } - rpos = (rpos + to_capture) % hw->mix_buf->size; + rpos = (rpos + to_capture) % hw->mix_buf.size; live -= to_capture; } - hw->mix_buf->pos = rpos; + hw->mix_buf.pos = rpos; for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) { if (!sw->active && sw->empty) { @@ -1903,7 +1903,7 @@ CaptureVoiceOut *AUD_add_capture( audio_pcm_init_info (&hw->info, as); - cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame); + cap->buf = g_malloc0_n(hw->mix_buf.size, hw->info.bytes_per_frame); if (hw->info.is_float) { hw->clip = mixeng_clip_float[hw->info.nchannels == 2]; @@ -1955,7 +1955,7 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque) sw = sw1; } QLIST_REMOVE (cap, entries); - g_free (cap->hw.mix_buf); + g_free(cap->hw.mix_buf.buffer); g_free (cap->buf); g_free (cap); } diff --git a/audio/audio_int.h b/audio/audio_int.h index 5028f2354a..061845dcc2 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -58,7 +58,7 @@ typedef struct SWVoiceCap SWVoiceCap; typedef struct STSampleBuffer { size_t pos, size; - st_sample samples[]; + st_sample *buffer; } STSampleBuffer; typedef struct HWVoiceOut { @@ -71,7 +71,7 @@ typedef struct HWVoiceOut { f_sample *clip; uint64_t ts_helper; - STSampleBuffer *mix_buf; + STSampleBuffer mix_buf; void *buf_emul; size_t pos_emul, pending_emul, size_emul; @@ -93,7 +93,7 @@ typedef struct HWVoiceIn { size_t total_samples_captured; uint64_t ts_helper; - STSampleBuffer *conv_buf; + STSampleBuffer conv_buf; void *buf_emul; size_t pos_emul, pending_emul, size_emul; diff --git a/audio/audio_template.h b/audio/audio_template.h index 980e1f4bd0..dd87170cbd 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -71,8 +71,9 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s, static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw) { g_free(hw->buf_emul); - g_free (HWBUF); - HWBUF = NULL; + g_free(HWBUF.buffer); + HWBUF.buffer = NULL; + HWBUF.size = 0; } static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) @@ -83,10 +84,12 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) dolog("Attempted to allocate empty buffer\n"); } - HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples); - HWBUF->size = samples; + HWBUF.buffer = g_new0(st_sample, samples); + HWBUF.size = samples; + HWBUF.pos = 0; } else { - HWBUF = NULL; + HWBUF.buffer = NULL; + HWBUF.size = 0; } } @@ -111,9 +114,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) } #ifdef DAC - samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio; + samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; #else - samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32; + samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; #endif if (audio_bug(__func__, samples < 0)) { dolog("Can not allocate buffer for `%s' (%d samples)\n", @@ -126,7 +129,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) size_t f_fe_min; /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */ - f_fe_min = (hw->info.freq + HWBUF->size - 1) / HWBUF->size; + f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size; qemu_log_mask(LOG_UNIMP, AUDIO_CAP ": The guest selected a " NAME " sample rate" " of %d Hz for %s. Only sample rates >= %zu Hz are"