diff options
Diffstat (limited to 'audio')
| -rw-r--r-- | audio/alsaaudio.c | 27 | ||||
| -rw-r--r-- | audio/audio.c | 392 | ||||
| -rw-r--r-- | audio/audio_int.h | 20 | ||||
| -rw-r--r-- | audio/audio_template.h | 105 | ||||
| -rw-r--r-- | audio/mixeng.c | 87 | ||||
| -rw-r--r-- | audio/mixeng.h | 2 | ||||
| -rw-r--r-- | audio/rate_template.h | 21 |
7 files changed, 357 insertions, 297 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 714bfb6453..057571dd1e 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -222,11 +222,7 @@ static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask) return -1; } - pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds)); - if (!pfds) { - dolog ("Could not initialize poll mode\n"); - return -1; - } + pfds = g_new0(struct pollfd, count); err = snd_pcm_poll_descriptors (handle, pfds, count); if (err < 0) { @@ -917,28 +913,23 @@ static void *alsa_audio_init(Audiodev *dev) alsa_init_per_direction(aopts->in); alsa_init_per_direction(aopts->out); - /* - * need to define them, as otherwise alsa produces no sound - * doesn't set has_* so alsa_open can identify it wasn't set by the user - */ + /* don't set has_* so alsa_open can identify it wasn't set by the user */ if (!dev->u.alsa.out->has_period_length) { - /* 1024 frames assuming 44100Hz */ - dev->u.alsa.out->period_length = 1024 * 1000000 / 44100; + /* 256 frames assuming 44100Hz */ + dev->u.alsa.out->period_length = 5805; } if (!dev->u.alsa.out->has_buffer_length) { /* 4096 frames assuming 44100Hz */ - dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100; + dev->u.alsa.out->buffer_length = 92880; } - /* - * OptsVisitor sets unspecified optional fields to zero, but do not depend - * on it... - */ if (!dev->u.alsa.in->has_period_length) { - dev->u.alsa.in->period_length = 0; + /* 256 frames assuming 44100Hz */ + dev->u.alsa.in->period_length = 5805; } if (!dev->u.alsa.in->has_buffer_length) { - dev->u.alsa.in->buffer_length = 0; + /* 4096 frames assuming 44100Hz */ + dev->u.alsa.in->buffer_length = 92880; } return dev; diff --git a/audio/audio.c b/audio/audio.c index 4290309d18..70b096713c 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -33,6 +33,7 @@ #include "qapi/qapi-visit-audio.h" #include "qapi/qapi-commands-audio.h" #include "qemu/cutils.h" +#include "qemu/log.h" #include "qemu/module.h" #include "qemu/help_option.h" #include "sysemu/sysemu.h" @@ -148,26 +149,6 @@ static inline int audio_bits_to_index (int bits) } } -void *audio_calloc (const char *funcname, int nmemb, size_t size) -{ - int cond; - size_t len; - - len = nmemb * size; - cond = !nmemb || !size; - cond |= nmemb < 0; - cond |= len < size; - - if (audio_bug ("audio_calloc", cond)) { - AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n", - funcname); - AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len); - return NULL; - } - - return g_malloc0 (len); -} - void AUD_vlog (const char *cap, const char *fmt, va_list ap) { if (cap) { @@ -400,13 +381,6 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len) /* * Capture */ -static void noop_conv (struct st_sample *dst, const void *src, int samples) -{ - (void) src; - (void) dst; - (void) samples; -} - static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s, struct audsettings *as) { @@ -504,15 +478,8 @@ static int audio_attach_capture (HWVoiceOut *hw) sw->info = hw->info; sw->empty = 1; sw->active = hw->enabled; - sw->conv = noop_conv; - sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq; sw->vol = nominal_volume; sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq); - if (!sw->rate) { - dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw)); - g_free (sw); - return -1; - } QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries); QLIST_INSERT_HEAD (&hw->cap_head, sc, entries); #ifdef DEBUG_CAPTURE @@ -547,8 +514,8 @@ 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; @@ -557,13 +524,13 @@ 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; @@ -575,56 +542,65 @@ static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples) /* * Soft voice (capture) */ -static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) +static void audio_pcm_sw_resample_in(SWVoiceIn *sw, + size_t frames_in_max, size_t frames_out_max, + size_t *total_in, size_t *total_out) +{ + HWVoiceIn *hw = sw->hw; + struct st_sample *src, *dst; + size_t live, rpos, frames_in, frames_out; + + live = hw->total_samples_captured - sw->total_hw_samples_acquired; + rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size); + + /* resample conv_buf from rpos to end of buffer */ + src = hw->conv_buf.buffer + rpos; + frames_in = MIN(frames_in_max, hw->conv_buf.size - rpos); + dst = sw->resample_buf.buffer; + frames_out = frames_out_max; + st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out); + rpos += frames_in; + *total_in = frames_in; + *total_out = frames_out; + + /* resample conv_buf from start of buffer if there are input frames left */ + if (frames_in_max - frames_in && rpos == hw->conv_buf.size) { + src = hw->conv_buf.buffer; + frames_in = frames_in_max - frames_in; + dst += frames_out; + frames_out = frames_out_max - frames_out; + st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out); + *total_in += frames_in; + *total_out += frames_out; + } +} + +static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len) { HWVoiceIn *hw = sw->hw; - size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0; - struct st_sample *src, *dst = sw->buf; + size_t live, frames_out_max, total_in, total_out; live = hw->total_samples_captured - sw->total_hw_samples_acquired; 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); - - samples = size / sw->info.bytes_per_frame; - - swlim = (live * sw->ratio) >> 32; - swlim = MIN (swlim, samples); - - while (swlim) { - src = hw->conv_buf->samples + rpos; - if (hw->conv_buf->pos > rpos) { - isamp = hw->conv_buf->pos - rpos; - } else { - isamp = hw->conv_buf->size - rpos; - } - - if (!isamp) { - break; - } - osamp = swlim; + frames_out_max = MIN(buf_len / sw->info.bytes_per_frame, + sw->resample_buf.size); - st_rate_flow (sw->rate, src, dst, &isamp, &osamp); - swlim -= osamp; - rpos = (rpos + isamp) % hw->conv_buf->size; - dst += osamp; - ret += osamp; - total += isamp; - } + audio_pcm_sw_resample_in(sw, live, frames_out_max, &total_in, &total_out); if (!hw->pcm_ops->volume_in) { - mixeng_volume (sw->buf, ret, &sw->vol); + mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol); } + sw->clip(buf, sw->resample_buf.buffer, total_out); - sw->clip (buf, sw->buf, ret); - sw->total_hw_samples_acquired += total; - return ret * sw->info.bytes_per_frame; + sw->total_hw_samples_acquired += total_in; + return total_out * sw->info.bytes_per_frame; } /* @@ -660,8 +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; @@ -678,17 +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; } @@ -697,84 +673,113 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) /* * Soft voice (playback) */ -static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) +static void audio_pcm_sw_resample_out(SWVoiceOut *sw, + size_t frames_in_max, size_t frames_out_max, + size_t *total_in, size_t *total_out) { - size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, blck; - size_t hw_free; - size_t ret = 0, pos = 0, total = 0; + HWVoiceOut *hw = sw->hw; + struct st_sample *src, *dst; + size_t live, wpos, frames_in, frames_out; - if (!sw) { - return size; + live = sw->total_hw_samples_mixed; + wpos = (hw->mix_buf.pos + live) % hw->mix_buf.size; + + /* write to mix_buf from wpos to end of buffer */ + src = sw->resample_buf.buffer; + frames_in = frames_in_max; + dst = hw->mix_buf.buffer + wpos; + frames_out = MIN(frames_out_max, hw->mix_buf.size - wpos); + st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out); + wpos += frames_out; + *total_in = frames_in; + *total_out = frames_out; + + /* write to mix_buf from start of buffer if there are input frames left */ + if (frames_in_max - frames_in > 0 && wpos == hw->mix_buf.size) { + src += frames_in; + frames_in = frames_in_max - frames_in; + dst = hw->mix_buf.buffer; + frames_out = frames_out_max - frames_out; + st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out); + *total_in += frames_in; + *total_out += frames_out; } +} - hwsamples = sw->hw->mix_buf->size; +static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) +{ + HWVoiceOut *hw = sw->hw; + size_t live, dead, hw_free, sw_max, fe_max; + size_t frames_in_max, frames_out_max, total_in, total_out; live = sw->total_hw_samples_mixed; - if (audio_bug(__func__, live > hwsamples)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples); + 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; } - if (live == hwsamples) { + if (live == hw->mix_buf.size) { #ifdef DEBUG_OUT dolog ("%s is full %zu\n", sw->name, live); #endif return 0; } - wpos = (sw->hw->mix_buf->pos + live) % hwsamples; - - dead = hwsamples - live; - hw_free = audio_pcm_hw_get_free(sw->hw); + dead = hw->mix_buf.size - live; + hw_free = audio_pcm_hw_get_free(hw); hw_free = hw_free > live ? hw_free - live : 0; - samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio; - samples = MIN(samples, size / sw->info.bytes_per_frame); - if (samples) { - sw->conv(sw->buf, buf, samples); + frames_out_max = MIN(dead, hw_free); + sw_max = st_rate_frames_in(sw->rate, frames_out_max); + fe_max = MIN(buf_len / sw->info.bytes_per_frame + sw->resample_buf.pos, + sw->resample_buf.size); + frames_in_max = MIN(sw_max, fe_max); + + if (!frames_in_max) { + return 0; + } + if (frames_in_max > sw->resample_buf.pos) { + sw->conv(sw->resample_buf.buffer + sw->resample_buf.pos, + buf, frames_in_max - sw->resample_buf.pos); if (!sw->hw->pcm_ops->volume_out) { - mixeng_volume(sw->buf, samples, &sw->vol); + mixeng_volume(sw->resample_buf.buffer + sw->resample_buf.pos, + frames_in_max - sw->resample_buf.pos, &sw->vol); } } - while (samples) { - dead = hwsamples - live; - left = hwsamples - wpos; - blck = MIN (dead, left); - if (!blck) { - break; - } - isamp = samples; - osamp = blck; - st_rate_flow_mix ( - sw->rate, - sw->buf + pos, - sw->hw->mix_buf->samples + wpos, - &isamp, - &osamp - ); - ret += isamp; - samples -= isamp; - pos += isamp; - live += osamp; - wpos = (wpos + osamp) % hwsamples; - total += osamp; - } - - sw->total_hw_samples_mixed += total; + audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max, + &total_in, &total_out); + + sw->total_hw_samples_mixed += total_out; sw->empty = sw->total_hw_samples_mixed == 0; + /* + * Upsampling may leave one audio frame in the resample buffer. Decrement + * total_in by one if there was a leftover frame from the previous resample + * pass in the resample buffer. Increment total_in by one if the current + * resample pass left one frame in the resample buffer. + */ + if (frames_in_max - total_in == 1) { + /* copy one leftover audio frame to the beginning of the buffer */ + *sw->resample_buf.buffer = *(sw->resample_buf.buffer + total_in); + total_in += 1 - sw->resample_buf.pos; + sw->resample_buf.pos = 1; + } else if (total_in >= sw->resample_buf.pos) { + total_in -= sw->resample_buf.pos; + sw->resample_buf.pos = 0; + } + #ifdef DEBUG_OUT dolog ( - "%s: write size %zu ret %zu total sw %zu\n", - SW_NAME (sw), - size / sw->info.bytes_per_frame, - ret, + "%s: write size %zu written %zu total mixed %zu\n", + SW_NAME(sw), + buf_len / sw->info.bytes_per_frame, + total_in, sw->total_hw_samples_mixed ); #endif - return ret * sw->info.bytes_per_frame; + return total_in * sw->info.bytes_per_frame; } #ifdef DEBUG_AUDIO @@ -992,18 +997,6 @@ void AUD_set_active_in (SWVoiceIn *sw, int on) } } -/** - * audio_frontend_frames_in() - returns the number of frames the resampling - * code generates from frames_in frames - * - * @sw: audio recording frontend - * @frames_in: number of frames - */ -static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in) -{ - return (int64_t)frames_in * sw->ratio >> 32; -} - static size_t audio_get_avail (SWVoiceIn *sw) { size_t live; @@ -1013,33 +1006,21 @@ 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; } ldebug ( - "%s: get_avail live %zu frontend frames %zu\n", + "%s: get_avail live %zu frontend frames %u\n", SW_NAME (sw), - live, audio_frontend_frames_in(sw, live) + live, st_rate_frames_out(sw->rate, live) ); return live; } -/** - * audio_frontend_frames_out() - returns the number of frames needed to - * get frames_out frames after resampling - * - * @sw: audio playback frontend - * @frames_out: number of frames - */ -static size_t audio_frontend_frames_out(SWVoiceOut *sw, size_t frames_out) -{ - return ((int64_t)frames_out << 32) / sw->ratio; -} - static size_t audio_get_free(SWVoiceOut *sw) { size_t live, dead; @@ -1050,17 +1031,17 @@ 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", - SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead)); + dolog("%s: get_free live %zu dead %zu frontend frames %u\n", + SW_NAME(sw), live, dead, st_rate_frames_in(sw->rate, dead)); #endif return dead; @@ -1076,32 +1057,40 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) { SWVoiceOut *sw = &sc->sw; - int rpos2 = rpos; + size_t rpos2 = rpos; n = samples; 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.bytes_per_frame; - size_t written; - - sw->buf = hw->mix_buf->samples + rpos2; - written = audio_pcm_sw_write (sw, NULL, bytes); - if (written - bytes) { - dolog("Could not mix %zu bytes into a capture " + size_t till_end_of_hw = hw->mix_buf.size - rpos2; + size_t to_read = MIN(till_end_of_hw, n); + size_t live, frames_in, frames_out; + + sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2; + sw->resample_buf.size = to_read; + live = sw->total_hw_samples_mixed; + + audio_pcm_sw_resample_out(sw, + to_read, sw->hw->mix_buf.size - live, + &frames_in, &frames_out); + + sw->total_hw_samples_mixed += frames_out; + sw->empty = sw->total_hw_samples_mixed == 0; + + if (to_read - frames_in) { + dolog("Could not mix %zu frames into a capture " "buffer, mixed %zu\n", - bytes, written); + to_read, frames_in); break; } - n -= to_write; - rpos2 = (rpos2 + to_write) % hw->mix_buf->size; + n -= to_read; + rpos2 = (rpos2 + to_read) % 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) @@ -1127,7 +1116,7 @@ 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; @@ -1181,12 +1170,14 @@ static void audio_run_out (AudioState *s) size_t free; if (hw_free > sw->total_hw_samples_mixed) { - free = audio_frontend_frames_out(sw, + free = st_rate_frames_in(sw->rate, MIN(sw_free, hw_free - sw->total_hw_samples_mixed)); } else { free = 0; } - if (free > 0) { + if (free > sw->resample_buf.pos) { + free = MIN(free, sw->resample_buf.size) + - sw->resample_buf.pos; sw->callback.fn(sw->callback.opaque, free * sw->info.bytes_per_frame); } @@ -1198,8 +1189,8 @@ 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; } @@ -1227,13 +1218,13 @@ 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 @@ -1314,10 +1305,10 @@ 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; @@ -1330,8 +1321,9 @@ static void audio_run_in (AudioState *s) size_t sw_avail = audio_get_avail(sw); size_t avail; - avail = audio_frontend_frames_in(sw, sw_avail); + avail = st_rate_frames_out(sw->rate, sw_avail); if (avail > 0) { + avail = MIN(avail, sw->resample_buf.size); sw->callback.fn(sw->callback.opaque, avail * sw->info.bytes_per_frame); } @@ -1350,14 +1342,14 @@ 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); @@ -1365,10 +1357,10 @@ 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) { @@ -1927,7 +1919,7 @@ 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]; @@ -1979,7 +1971,7 @@ 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 e87ce014a0..d51d63f08d 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; @@ -108,8 +108,7 @@ struct SWVoiceOut { AudioState *s; struct audio_pcm_info info; t_sample *conv; - int64_t ratio; - struct st_sample *buf; + STSampleBuffer resample_buf; void *rate; size_t total_hw_samples_mixed; int active; @@ -126,10 +125,9 @@ struct SWVoiceIn { AudioState *s; int active; struct audio_pcm_info info; - int64_t ratio; void *rate; size_t total_hw_samples_acquired; - struct st_sample *buf; + STSampleBuffer resample_buf; f_sample *clip; HWVoiceIn *hw; char *name; @@ -151,8 +149,8 @@ struct audio_driver { int can_be_default; int max_voices_out; int max_voices_in; - int voice_size_out; - int voice_size_in; + size_t voice_size_out; + size_t voice_size_in; QLIST_ENTRY(audio_driver) next; }; @@ -251,7 +249,6 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as); void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len); int audio_bug (const char *funcname, int cond); -void *audio_calloc (const char *funcname, int nmemb, size_t size); void audio_run(AudioState *s, const char *msg); @@ -294,9 +291,6 @@ static inline size_t audio_ring_posb(size_t pos, size_t dist, size_t len) #define ldebug(fmt, ...) (void)0 #endif -#define AUDIO_STRINGIFY_(n) #n -#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n) - typedef struct AudiodevListEntry { Audiodev *dev; QSIMPLEQ_ENTRY(AudiodevListEntry) next; diff --git a/audio/audio_template.h b/audio/audio_template.h index 42b4712acb..e42326c20d 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -40,7 +40,7 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s, struct audio_driver *drv) { int max_voices = glue (drv->max_voices_, TYPE); - int voice_size = glue (drv->voice_size_, TYPE); + size_t voice_size = glue(drv->voice_size_, TYPE); if (glue (s->nb_hw_voices_, TYPE) > max_voices) { if (!max_voices) { @@ -63,16 +63,17 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s, } if (audio_bug(__func__, voice_size && !max_voices)) { - dolog ("drv=`%s' voice_size=%d max_voices=0\n", - drv->name, voice_size); + dolog("drv=`%s' voice_size=%zu max_voices=0\n", + drv->name, voice_size); } } 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) @@ -83,56 +84,67 @@ 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; } } static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw) { - g_free (sw->buf); + g_free(sw->resample_buf.buffer); + sw->resample_buf.buffer = NULL; + sw->resample_buf.size = 0; if (sw->rate) { st_rate_stop (sw->rate); } - - sw->buf = NULL; sw->rate = NULL; } static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) { - int samples; + HW *hw = sw->hw; + uint64_t samples; if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) { return 0; } -#ifdef DAC - samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio; -#else - samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32; -#endif + samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq); + if (samples == 0) { + uint64_t f_fe_min; + uint64_t f_be = (uint32_t)hw->info.freq; - sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample)); - if (!sw->buf) { - dolog ("Could not allocate buffer for `%s' (%d samples)\n", - SW_NAME (sw), samples); + /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */ + f_fe_min = (f_be + 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 >= %" PRIu64 " Hz" + " are supported.\n", + sw->info.freq, sw->name, f_fe_min); return -1; } + /* + * Allocate one additional audio frame that is needed for upsampling + * if the resample buffer size is small. For large buffer sizes take + * care of overflows and truncation. + */ + samples = samples < SIZE_MAX ? samples + 1 : SIZE_MAX; + sw->resample_buf.buffer = g_new0(st_sample, samples); + sw->resample_buf.size = samples; + sw->resample_buf.pos = 0; + #ifdef DAC - sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq); + sw->rate = st_rate_start(sw->info.freq, hw->info.freq); #else - sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq); + sw->rate = st_rate_start(hw->info.freq, sw->info.freq); #endif - if (!sw->rate) { - g_free (sw->buf); - sw->buf = NULL; - return -1; - } + return 0; } @@ -149,11 +161,8 @@ static int glue (audio_pcm_sw_init_, TYPE) ( sw->hw = hw; sw->active = 0; #ifdef DAC - sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq; sw->total_hw_samples_mixed = 0; sw->empty = 1; -#else - sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq; #endif if (sw->info.is_float) { @@ -264,13 +273,11 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s, return NULL; } - hw = audio_calloc(__func__, 1, glue(drv->voice_size_, TYPE)); - if (!hw) { - dolog ("Can not allocate voice `%s' size %d\n", - drv->name, glue (drv->voice_size_, TYPE)); - return NULL; - } - + /* + * Since glue(s->nb_hw_voices_, TYPE) is != 0, glue(drv->voice_size_, TYPE) + * is guaranteed to be != 0. See the audio_init_nb_voices_* functions. + */ + hw = g_malloc0(glue(drv->voice_size_, TYPE)); hw->s = s; hw->pcm_ops = drv->pcm_ops; @@ -418,33 +425,28 @@ static SW *glue(audio_pcm_create_voice_pair_, TYPE)( hw_as = *as; } - sw = audio_calloc(__func__, 1, sizeof(*sw)); - if (!sw) { - dolog ("Could not allocate soft voice `%s' (%zu bytes)\n", - sw_name ? sw_name : "unknown", sizeof (*sw)); - goto err1; - } + sw = g_new0(SW, 1); sw->s = s; hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as); if (!hw) { - goto err2; + dolog("Could not create a backend for voice `%s'\n", sw_name); + goto err1; } glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw); if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) { - goto err3; + goto err2; } return sw; -err3: +err2: glue (audio_pcm_hw_del_sw_, TYPE) (sw); glue (audio_pcm_hw_gc_, TYPE) (&hw); -err2: - g_free (sw); err1: + g_free(sw); return NULL; } @@ -515,8 +517,8 @@ SW *glue (AUD_open_, TYPE) ( HW *hw = sw->hw; if (!hw) { - dolog ("Internal logic error voice `%s' has no hardware store\n", - SW_NAME (sw)); + dolog("Internal logic error: voice `%s' has no backend\n", + SW_NAME(sw)); goto fail; } @@ -527,7 +529,6 @@ SW *glue (AUD_open_, TYPE) ( } else { sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as); if (!sw) { - dolog ("Failed to create voice `%s'\n", name); return NULL; } } diff --git a/audio/mixeng.c b/audio/mixeng.c index 100a306d6f..69f6549224 100644 --- a/audio/mixeng.c +++ b/audio/mixeng.c @@ -414,12 +414,7 @@ struct rate { */ void *st_rate_start (int inrate, int outrate) { - struct rate *rate = audio_calloc(__func__, 1, sizeof(*rate)); - - if (!rate) { - dolog ("Could not allocate resampler (%zu bytes)\n", sizeof (*rate)); - return NULL; - } + struct rate *rate = g_new0(struct rate, 1); rate->opos = 0; @@ -445,6 +440,86 @@ void st_rate_stop (void *opaque) g_free (opaque); } +/** + * st_rate_frames_out() - returns the number of frames the resampling code + * generates from frames_in frames + * + * @opaque: pointer to struct rate + * @frames_in: number of frames + * + * When upsampling, there may be more than one correct result. In this case, + * the function returns the maximum number of output frames the resampling + * code can generate. + */ +uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in) +{ + struct rate *rate = opaque; + uint64_t opos_end, opos_delta; + uint32_t ipos_end; + uint32_t frames_out; + + if (rate->opos_inc == 1ULL << 32) { + return frames_in; + } + + /* no output frame without at least one input frame */ + if (!frames_in) { + return 0; + } + + /* last frame read was at rate->ipos - 1 */ + ipos_end = rate->ipos - 1 + frames_in; + opos_end = (uint64_t)ipos_end << 32; + + /* last frame written was at rate->opos - rate->opos_inc */ + if (opos_end + rate->opos_inc <= rate->opos) { + return 0; + } + opos_delta = opos_end - rate->opos + rate->opos_inc; + frames_out = opos_delta / rate->opos_inc; + + return opos_delta % rate->opos_inc ? frames_out : frames_out - 1; +} + +/** + * st_rate_frames_in() - returns the number of frames needed to + * get frames_out frames after resampling + * + * @opaque: pointer to struct rate + * @frames_out: number of frames + * + * When downsampling, there may be more than one correct result. In this + * case, the function returns the maximum number of input frames needed. + */ +uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out) +{ + struct rate *rate = opaque; + uint64_t opos_start, opos_end; + uint32_t ipos_start, ipos_end; + + if (rate->opos_inc == 1ULL << 32) { + return frames_out; + } + + if (frames_out) { + opos_start = rate->opos; + ipos_start = rate->ipos; + } else { + uint64_t offset; + + /* add offset = ceil(opos_inc) to opos and ipos to avoid an underflow */ + offset = (rate->opos_inc + (1ULL << 32) - 1) & ~((1ULL << 32) - 1); + opos_start = rate->opos + offset; + ipos_start = rate->ipos + (offset >> 32); + } + /* last frame written was at opos_start - rate->opos_inc */ + opos_end = opos_start - rate->opos_inc + rate->opos_inc * frames_out; + ipos_end = (opos_end >> 32) + 1; + + /* last frame read was at ipos_start - 1 */ + return ipos_end + 1 > ipos_start ? ipos_end + 1 - ipos_start : 0; +} + void mixeng_clear (struct st_sample *buf, int len) { memset (buf, 0, len * sizeof (struct st_sample)); diff --git a/audio/mixeng.h b/audio/mixeng.h index 2dcd6df245..f9de7cffeb 100644 --- a/audio/mixeng.h +++ b/audio/mixeng.h @@ -52,6 +52,8 @@ void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf, void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf, size_t *isamp, size_t *osamp); void st_rate_stop (void *opaque); +uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in); +uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out); void mixeng_clear (struct st_sample *buf, int len); void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol); diff --git a/audio/rate_template.h b/audio/rate_template.h index b432719ebb..6648f0d2e5 100644 --- a/audio/rate_template.h +++ b/audio/rate_template.h @@ -40,8 +40,6 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf, int64_t t; #endif - ilast = rate->ilast; - istart = ibuf; iend = ibuf + *isamp; @@ -59,15 +57,17 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf, return; } - while (obuf < oend) { + /* without input samples, there's nothing to do */ + if (ibuf >= iend) { + *osamp = 0; + return; + } - /* Safety catch to make sure we have input samples. */ - if (ibuf >= iend) { - break; - } + ilast = rate->ilast; - /* read as many input samples so that ipos > opos */ + while (true) { + /* read as many input samples so that ipos > opos */ while (rate->ipos <= (rate->opos >> 32)) { ilast = *ibuf++; rate->ipos++; @@ -78,6 +78,11 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf, } } + /* make sure that the next output sample can be written */ + if (obuf >= oend) { + break; + } + icur = *ibuf; /* wrap ipos and opos around long before they overflow */ |