qemu/audio/wavaudio.c

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/*
* QEMU WAV audio driver
*
* Copyright (c) 2004-2005 Vassili Karpov (malc)
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include "qemu/osdep.h"
#include "qemu/host-utils.h"
#include "qemu/module.h"
#include "qemu/timer.h"
#include "qapi/opts-visitor.h"
#include "audio.h"
#define AUDIO_CAP "wav"
#include "audio_int.h"
typedef struct WAVVoiceOut {
HWVoiceOut hw;
FILE *f;
RateCtl rate;
int total_samples;
} WAVVoiceOut;
static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int64_t bytes = audio_rate_get_bytes(&wav->rate, &hw->info, len);
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.bytes_per_frame;
return bytes;
}
/* VICE code: Store number as little endian. */
static void le_store (uint8_t *buf, uint32_t val, int len)
{
int i;
for (i = 0; i < len; i++) {
buf[i] = (uint8_t) (val & 0xff);
val >>= 8;
}
}
static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int bits16 = 0, stereo = 0;
uint8_t hdr[] = {
0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56,
0x45, 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00,
0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
};
Audiodev *dev = drv_opaque;
AudiodevWavOptions *wopts = &dev->u.wav;
struct audsettings wav_as = audiodev_to_audsettings(dev->u.wav.out);
const char *wav_path = wopts->path ?: "qemu.wav";
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
case AUDIO_FORMAT_S8:
case AUDIO_FORMAT_U8:
bits16 = 0;
break;
case AUDIO_FORMAT_S16:
case AUDIO_FORMAT_U16:
bits16 = 1;
break;
case AUDIO_FORMAT_S32:
case AUDIO_FORMAT_U32:
dolog ("WAVE files can not handle 32bit formats\n");
return -1;
case AUDIO_FORMAT_F32:
dolog("WAVE files can not handle float formats\n");
return -1;
default:
abort();
}
hdr[34] = bits16 ? 0x10 : 0x08;
wav_as.endianness = 0;
audio_pcm_init_info (&hw->info, &wav_as);
hw->samples = 1024;
le_store (hdr + 22, hw->info.nchannels, 2);
le_store (hdr + 24, hw->info.freq, 4);
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
le_store (hdr + 32, 1 << (bits16 + stereo), 2);
wav->f = fopen(wav_path, "wb");
if (!wav->f) {
dolog ("Failed to open wave file `%s'\nReason: %s\n",
wav_path, strerror(errno));
return -1;
}
if (fwrite (hdr, sizeof (hdr), 1, wav->f) != 1) {
dolog ("wav_init_out: failed to write header\nReason: %s\n",
strerror(errno));
return -1;
}
audio_rate_start(&wav->rate);
return 0;
}
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.bytes_per_frame;
uint32_t rifflen = datalen + 36;
if (!wav->f) {
return;
}
le_store (rlen, rifflen, 4);
le_store (dlen, datalen, 4);
if (fseek (wav->f, 4, SEEK_SET)) {
dolog ("wav_fini_out: fseek to rlen failed\nReason: %s\n",
strerror(errno));
goto doclose;
}
if (fwrite (rlen, 4, 1, wav->f) != 1) {
dolog ("wav_fini_out: failed to write rlen\nReason: %s\n",
strerror (errno));
goto doclose;
}
if (fseek (wav->f, 32, SEEK_CUR)) {
dolog ("wav_fini_out: fseek to dlen failed\nReason: %s\n",
strerror (errno));
goto doclose;
}
if (fwrite (dlen, 4, 1, wav->f) != 1) {
dolog ("wav_fini_out: failed to write dlen\nReaons: %s\n",
strerror (errno));
goto doclose;
}
doclose:
if (fclose (wav->f)) {
dolog ("wav_fini_out: fclose %p failed\nReason: %s\n",
wav->f, strerror (errno));
}
wav->f = NULL;
}
static void wav_enable_out(HWVoiceOut *hw, bool enable)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
if (enable) {
audio_rate_start(&wav->rate);
}
}
static void *wav_audio_init(Audiodev *dev, Error **errp)
{
assert(dev->driver == AUDIODEV_DRIVER_WAV);
return dev;
}
static void wav_audio_fini (void *opaque)
{
ldebug ("wav_fini");
}
static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
.write = wav_write_out,
.buffer_get_free = audio_generic_buffer_get_free,
audio: fix bug 1858488 The combined generic buffer management code and buffer run out code in function audio_generic_put_buffer_out has a problematic behaviour. A few hundred milliseconds after playback starts the mixing buffer and the generic buffer are nearly full and the following pattern can be seen. On first call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but the generic buffer will fill faster and is full when audio_pcm_hw_run_out returns. This is because emulated audio devices can produce playback data at a higher rate than the audio backend hardware consumes this data. On next call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but no audio data is transferred to the generic buffer because the buffer is already full. Then the pattern repeats. For the emulated audio device this looks like the audio timer period has doubled. This patch splits the combined generic buffer management code and buffer run out code and calls the buffer run out code after buffer management code to break this pattern. The bug report is for the wav audio backend. But the problem is not limited to this backend. All audio backends which use the audio_generic_put_buffer_out function show this problem. Buglink: https://bugs.launchpad.net/qemu/+bug/1858488 Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20200123074943.6699-5-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2020-01-23 10:49:39 +03:00
.run_buffer_out = audio_generic_run_buffer_out,
.enable_out = wav_enable_out,
};
static struct audio_driver wav_audio_driver = {
.name = "wav",
.descr = "WAV renderer http://wikipedia.org/wiki/WAV",
.init = wav_audio_init,
.fini = wav_audio_fini,
.pcm_ops = &wav_pcm_ops,
.max_voices_out = 1,
.max_voices_in = 0,
.voice_size_out = sizeof (WAVVoiceOut),
.voice_size_in = 0
};
static void register_audio_wav(void)
{
audio_driver_register(&wav_audio_driver);
}
type_init(register_audio_wav);