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Re: [PATCH v2 01/17] audio: change type of mix_buf and conv_buf
From: |
Marc-André Lureau |
Subject: |
Re: [PATCH v2 01/17] audio: change type of mix_buf and conv_buf |
Date: |
Wed, 22 Feb 2023 14:49:28 +0400 |
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
- [PATCH v2 00/17] audio: improve callback interface for audio frontends, Volker Rümelin, 2023/02/06
- [PATCH v2 01/17] audio: change type of mix_buf and conv_buf, Volker Rümelin, 2023/02/06
- Re: [PATCH v2 01/17] audio: change type of mix_buf and conv_buf,
Marc-André Lureau <=
- [PATCH v2 02/17] audio: change type and name of the resample buffer, Volker Rümelin, 2023/02/06
- [PATCH v2 03/17] audio: make the resampling code greedy, Volker Rümelin, 2023/02/06
- [PATCH v2 06/17] audio: rename variables in audio_pcm_sw_write(), Volker Rümelin, 2023/02/06
- [PATCH v2 05/17] audio: remove sw == NULL check, Volker Rümelin, 2023/02/06
- [PATCH v2 07/17] audio: don't misuse audio_pcm_sw_write(), Volker Rümelin, 2023/02/06