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https://github.com/mpv-player/mpv
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ao_dsound: switch to new AO API
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wm4
parent
cee56e8623
commit
8afcb84ee5
@ -39,22 +39,10 @@
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#include "audio/format.h"
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#include "ao.h"
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#include "audio/reorder_ch.h"
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#include "audio_out_internal.h"
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#include "core/mp_msg.h"
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#include "osdep/timer.h"
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#include "core/subopt-helper.h"
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static const ao_info_t info =
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{
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"Windows DirectSound audio output",
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"dsound",
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"Gabor Szecsi <deje@miki.hu>",
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""
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};
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LIBAO_EXTERN(dsound)
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/**
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\todo use the definitions from the win32 api headers when they define these
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*/
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@ -104,6 +92,8 @@ static int device_num = 0; ///wanted device number
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static GUID device; ///guid of the device
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static int audio_volume;
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static float get_delay(struct ao *ao);
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/***************************************************************************************/
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/**
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@ -296,7 +286,7 @@ static void DestroyBuffer(void)
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\param len length of the data to copy in bytes
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\return number of copyed bytes
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*/
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static int write_buffer(unsigned char *data, int len)
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static int write_buffer(struct ao *ao, unsigned char *data, int len)
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{
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HRESULT res;
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LPVOID lpvPtr1;
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@ -318,7 +308,7 @@ static int write_buffer(unsigned char *data, int len)
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if (SUCCEEDED(res)) {
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if (!AF_FORMAT_IS_AC3(ao_data.format)) {
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if (!AF_FORMAT_IS_AC3(ao->format)) {
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memcpy(lpvPtr1, data, dwBytes1);
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if (lpvPtr2 != NULL)
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memcpy(lpvPtr2, (char *)data + dwBytes1, dwBytes2);
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@ -358,9 +348,9 @@ static int write_buffer(unsigned char *data, int len)
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\brief handle control commands
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\param cmd command
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\param arg argument
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\return CONTROL_OK or -1 in case the command can't be handled
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\return CONTROL_OK or CONTROL_UNKNOWN in case the command is not supported
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*/
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static int control(int cmd, void *arg)
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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DWORD volume;
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switch (cmd) {
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@ -379,7 +369,7 @@ static int control(int cmd, void *arg)
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return CONTROL_OK;
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}
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}
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return -1;
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return CONTROL_UNKNOWN;
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}
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/**
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@ -388,31 +378,31 @@ static int control(int cmd, void *arg)
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\param channels number of channels
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\param format format
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\param flags unused
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\return 1=success 0=fail
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\return 0=success -1=fail
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*/
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static int init(int rate, const struct mp_chmap *channels, int format, int flags)
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static int init(struct ao *ao, char *params)
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{
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int res;
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if (!InitDirectSound())
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return 0;
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return -1;
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global_ao->no_persistent_volume = true;
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ao->no_persistent_volume = true;
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audio_volume = 100;
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// ok, now create the buffers
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WAVEFORMATEXTENSIBLE wformat;
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DSBUFFERDESC dsbpridesc;
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DSBUFFERDESC dsbdesc;
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int format = ao->format;
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int rate = ao->samplerate;
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if (AF_FORMAT_IS_AC3(format))
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format = AF_FORMAT_AC3_NE;
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else {
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struct mp_chmap_sel sel = {
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0
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};
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struct mp_chmap_sel sel = {0};
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mp_chmap_sel_add_waveext(&sel);
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if (!ao_chmap_sel_adjust(&ao_data, &sel, &ao_data.channels))
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return 0;
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if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
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return -1;
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}
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switch (format) {
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case AF_FORMAT_AC3_NE:
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@ -426,30 +416,30 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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af_fmt2str_short(format));
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format = AF_FORMAT_S16_LE;
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}
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//fill global ao_data
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ao_data.samplerate = rate;
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ao_data.format = format;
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ao_data.bps = ao_data.channels.num * rate * (af_fmt2bits(format) >> 3);
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if (ao_data.buffersize == -1)
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ao_data.buffersize = ao_data.bps; // space for 1 sec
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//set our audio parameters
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ao->samplerate = rate;
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ao->format = format;
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ao->bps = ao->channels.num * rate * (af_fmt2bits(format) >> 3);
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if (ao->buffersize == -1)
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ao->buffersize = ao->bps; // space for 1 sec
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mp_msg(MSGT_AO, MSGL_V,
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"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate,
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ao_data.channels.num, af_fmt2str_short(format));
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ao->channels.num, af_fmt2str_short(format));
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: Buffersize:%d bytes (%d msec)\n",
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ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);
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ao->buffersize, ao->buffersize / ao->bps * 1000);
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//fill waveformatex
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ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
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wformat.Format.cbSize = (ao_data.channels.num > 2)
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wformat.Format.cbSize = (ao->channels.num > 2)
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? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
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wformat.Format.nChannels = ao_data.channels.num;
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wformat.Format.nChannels = ao->channels.num;
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wformat.Format.nSamplesPerSec = rate;
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if (AF_FORMAT_IS_AC3(format)) {
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wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
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wformat.Format.wBitsPerSample = 16;
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wformat.Format.nBlockAlign = 4;
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} else {
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wformat.Format.wFormatTag = (ao_data.channels.num > 2)
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wformat.Format.wFormatTag = (ao->channels.num > 2)
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? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
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wformat.Format.wBitsPerSample = af_fmt2bits(format);
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wformat.Format.nBlockAlign = wformat.Format.nChannels *
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@ -470,8 +460,8 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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| DSBCAPS_GLOBALFOCUS /** Allows background playing */
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| DSBCAPS_CTRLVOLUME; /** volume control enabled */
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if (ao_data.channels.num > 2) {
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wformat.dwChannelMask = mp_chmap_to_waveext(&ao_data.channels);
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if (ao->channels.num > 2) {
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wformat.dwChannelMask = mp_chmap_to_waveext(&ao->channels);
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wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
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wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
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// Needed for 5.1 on emu101k - shit soundblaster
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@ -480,12 +470,12 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec *
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wformat.Format.nBlockAlign;
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dsbdesc.dwBufferBytes = ao_data.buffersize;
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dsbdesc.dwBufferBytes = ao->buffersize;
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dsbdesc.lpwfxFormat = (WAVEFORMATEX *)&wformat;
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buffer_size = dsbdesc.dwBufferBytes;
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write_offset = 0;
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min_free_space = wformat.Format.nBlockAlign;
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ao_data.outburst = wformat.Format.nBlockAlign * 512;
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ao->outburst = wformat.Format.nBlockAlign * 512;
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// create primary buffer and set its format
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@ -495,7 +485,7 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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mp_msg(MSGT_AO, MSGL_ERR,
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"ao_dsound: cannot create primary buffer (%s)\n",
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dserr2str(res));
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return 0;
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return -1;
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}
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res = IDirectSoundBuffer_SetFormat(hdspribuf, (WAVEFORMATEX *)&wformat);
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if (res != DS_OK) {
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@ -520,11 +510,11 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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mp_msg(MSGT_AO, MSGL_ERR,
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"ao_dsound: cannot create secondary (stream)buffer (%s)\n",
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dserr2str(res));
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return 0;
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return -1;
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}
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}
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mp_msg(MSGT_AO, MSGL_V, "ao_dsound: secondary (stream)buffer created\n");
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return 1;
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return 0;
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}
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@ -532,7 +522,7 @@ static int init(int rate, const struct mp_chmap *channels, int format, int flags
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/**
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\brief stop playing and empty buffers (for seeking/pause)
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*/
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static void reset(void)
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static void reset(struct ao *ao)
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{
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IDirectSoundBuffer_Stop(hdsbuf);
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// reset directsound buffer
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@ -544,7 +534,7 @@ static void reset(void)
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/**
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\brief stop playing, keep buffers (for pause)
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*/
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static void audio_pause(void)
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static void audio_pause(struct ao *ao)
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{
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IDirectSoundBuffer_Stop(hdsbuf);
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}
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@ -552,7 +542,7 @@ static void audio_pause(void)
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/**
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\brief resume playing, after audio_pause()
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*/
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static void audio_resume(void)
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static void audio_resume(struct ao *ao)
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{
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IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
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}
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@ -561,18 +551,18 @@ static void audio_resume(void)
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\brief close audio device
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\param immed stop playback immediately
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*/
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static void uninit(int immed)
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static void uninit(struct ao *ao, bool immed)
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{
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if (!immed)
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mp_sleep_us(get_delay() * 1000000);
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reset();
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mp_sleep_us(get_delay(ao) * 1000000);
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reset(ao);
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DestroyBuffer();
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UninitDirectSound();
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}
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// return exact number of free (safe to write) bytes
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static int check_free_buffer_size(void)
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static int check_free_buffer_size(struct ao *ao)
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{
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int space;
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DWORD play_offset;
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@ -594,7 +584,7 @@ static int check_free_buffer_size(void)
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if (space < underrun_check) {
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// there's no useful data in the buffers
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space = buffer_size;
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reset();
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reset(ao);
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}
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underrun_check = space;
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return space;
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@ -604,9 +594,9 @@ static int check_free_buffer_size(void)
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\brief find out how many bytes can be written into the audio buffer without
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\return free space in bytes, has to return 0 if the buffer is almost full
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*/
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static int get_space(void)
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static int get_space(struct ao *ao)
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{
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int space = check_free_buffer_size();
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int space = check_free_buffer_size(ao);
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if (space < min_free_space)
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return 0;
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return space - min_free_space;
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@ -616,26 +606,45 @@ static int get_space(void)
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\brief play 'len' bytes of 'data'
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\param data pointer to the data to play
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\param len size in bytes of the data buffer, gets rounded down to outburst*n
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NOTE: outburst stuff might be outdated/deprecated
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\param flags currently unused
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\return number of played bytes
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*/
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static int play(void *data, int len, int flags)
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static int play(struct ao *ao, void *data, int len, int flags)
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{
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int space = check_free_buffer_size();
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int space = check_free_buffer_size(ao);
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if (space < len)
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len = space;
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if (!(flags & AOPLAY_FINAL_CHUNK))
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len = (len / ao_data.outburst) * ao_data.outburst;
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return write_buffer(data, len);
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len = (len / ao->outburst) * ao->outburst;
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return write_buffer(ao, data, len);
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}
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/**
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\brief get the delay between the first and last sample in the buffer
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\return delay in seconds
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*/
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static float get_delay(void)
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static float get_delay(struct ao *ao)
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{
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int space = check_free_buffer_size();
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return (float)(buffer_size - space) / (float)ao_data.bps;
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int space = check_free_buffer_size(ao);
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return (float)(buffer_size - space) / (float)ao->bps;
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}
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const struct ao_driver audio_out_dsound = {
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.info = &(const struct ao_info) {
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"Windows DirectSound audio output",
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"dsound",
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"Gabor Szecsi <deje@miki.hu>",
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""
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},
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.init = init,
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.uninit = uninit,
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.control = control,
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.get_space = get_space,
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.play = play,
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.get_delay = get_delay,
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.pause = audio_pause,
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.resume = audio_resume,
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.reset = reset,
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};
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