mirror of
https://github.com/tuxbox-neutrino/libstb-hal.git
synced 2025-08-26 15:02:58 +02:00
- formatting code using astyle
Conflicts: libarmbox/dmx.cpp libgeneric-pc/video_lib.h libspark/dmx.cpp Signed-off-by: Thilo Graf <dbt@novatux.de>
This commit is contained in:
svenhoefer
@@ -43,7 +43,7 @@ extern "C" {
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/* my own buf 16k */
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#define DMX_BUF_SZ 0x4000
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cAudio * audioDecoder = NULL;
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cAudio *audioDecoder = NULL;
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extern cDemux *audioDemux;
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static uint8_t *dmxbuf = NULL;
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static int bufpos;
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@@ -163,9 +163,9 @@ int cAudio::PrepareClipPlay(int ch, int srate, int bits, int le)
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adevice = ao_open_live(driver, &sformat, NULL);
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ao_info *ai = ao_driver_info(driver);
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hal_info("%s: changed params ch %d srate %d bits %d le %d adevice %p\n",
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__func__, ch, srate, bits, le, adevice);;
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__func__, ch, srate, bits, le, adevice);;
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hal_info("libao driver: %d name '%s' short '%s' author '%s'\n",
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driver, ai->name, ai->short_name, ai->author);
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driver, ai->name, ai->short_name, ai->author);
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}
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return 0;
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};
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@@ -173,7 +173,8 @@ int cAudio::PrepareClipPlay(int ch, int srate, int bits, int le)
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int cAudio::WriteClip(unsigned char *buffer, int size)
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{
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hal_debug("cAudio::%s buf 0x%p size %d\n", __func__, buffer, size);
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if (!adevice) {
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if (!adevice)
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{
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hal_info("%s: adevice not opened?\n", __func__);
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return 0;
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}
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@@ -187,7 +188,8 @@ int cAudio::StopClip()
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#if 0
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/* don't do anything - closing / reopening ao all the time makes for long delays
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* reinit on-demand (e.g. for changed parameters) instead */
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if (!adevice) {
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if (!adevice)
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{
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hal_info("%s: adevice not opened?\n", __func__);
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return 0;
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}
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@@ -200,13 +202,15 @@ int cAudio::StopClip()
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void cAudio::getAudioInfo(int &type, int &layer, int &freq, int &bitrate, int &mode)
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{
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type = 0;
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layer = 0; /* not used */
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layer = 0; /* not used */
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freq = 0;
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bitrate = 0; /* not used, but easy to get :-) */
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mode = 0; /* default: stereo */
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bitrate = 0; /* not used, but easy to get :-) */
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mode = 0; /* default: stereo */
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printf("cAudio::getAudioInfo c %p\n", c);
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if (c) {
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switch (c->codec_id) {
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if (c)
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{
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switch (c->codec_id)
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{
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case AV_CODEC_ID_MP2:
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type = AUDIO_FMT_MPEG;
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break;
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@@ -236,40 +240,42 @@ void cAudio::getAudioInfo(int &type, int &layer, int &freq, int &bitrate, int &m
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bitrate = c->bit_rate;
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if (c->channels == 1)
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mode = 3; /* for AV_CODEC_ID_MP2, only stereo / mono is detected for now */
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if (c->codec_id != AV_CODEC_ID_MP2) {
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switch (c->channel_layout) {
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if (c->codec_id != AV_CODEC_ID_MP2)
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{
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switch (c->channel_layout)
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{
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case AV_CH_LAYOUT_MONO:
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mode = 1; // "C"
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mode = 1; // "C"
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break;
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case AV_CH_LAYOUT_STEREO:
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mode = 2; // "L/R"
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mode = 2; // "L/R"
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break;
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case AV_CH_LAYOUT_2_1:
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case AV_CH_LAYOUT_SURROUND:
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mode = 3; // "L/C/R"
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mode = 3; // "L/C/R"
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break;
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case AV_CH_LAYOUT_2POINT1:
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mode = 4; // "L/R/S"
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mode = 4; // "L/R/S"
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break;
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case AV_CH_LAYOUT_3POINT1:
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mode = 5; // "L/C/R/S"
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mode = 5; // "L/C/R/S"
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break;
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case AV_CH_LAYOUT_2_2:
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case AV_CH_LAYOUT_QUAD:
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mode = 6; // "L/R/SL/SR"
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mode = 6; // "L/R/SL/SR"
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break;
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case AV_CH_LAYOUT_5POINT0:
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case AV_CH_LAYOUT_5POINT1:
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mode = 7; // "L/C/R/SL/SR"
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mode = 7; // "L/C/R/SL/SR"
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break;
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default:
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hal_info("%s: unknown ch_layout 0x%" PRIx64 "\n",
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__func__, c->channel_layout);
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__func__, c->channel_layout);
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}
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}
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}
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hal_debug("%s t: %d l: %d f: %d b: %d m: %d codec_id: %x\n",
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__func__, type, layer, freq, bitrate, mode, c?c->codec_id:-1);
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__func__, type, layer, freq, bitrate, mode, c ? c->codec_id : -1);
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};
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void cAudio::SetSRS(int /*iq_enable*/, int /*nmgr_enable*/, int /*iq_mode*/, int /*iq_level*/)
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@@ -310,8 +316,10 @@ static int _my_read(void *, uint8_t *buf, int buf_size)
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int cAudio::my_read(uint8_t *buf, int buf_size)
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{
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int tmp = 0;
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if (audioDecoder && bufpos < DMX_BUF_SZ - 4096) {
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while (bufpos < buf_size && ++tmp < 20) { /* retry max 20 times */
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if (audioDecoder && bufpos < DMX_BUF_SZ - 4096)
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{
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while (bufpos < buf_size && ++tmp < 20) /* retry max 20 times */
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{
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int ret = audioDemux->Read(dmxbuf + bufpos, DMX_BUF_SZ - bufpos, 10);
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if (ret > 0)
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bufpos += ret;
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@@ -322,7 +330,8 @@ int cAudio::my_read(uint8_t *buf, int buf_size)
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if (bufpos == 0)
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return 0;
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//hal_info("%s buf_size %d bufpos %d th %d tmp %d\n", __func__, buf_size, bufpos, thread_started, tmp);
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if (bufpos > buf_size) {
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if (bufpos > buf_size)
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{
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memcpy(buf, dmxbuf, buf_size);
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memmove(dmxbuf, dmxbuf + buf_size, bufpos - buf_size);
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bufpos -= buf_size;
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@@ -367,18 +376,19 @@ void cAudio::run()
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av_init_packet(&avpkt);
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inp = av_find_input_format("mpegts");
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AVIOContext *pIOCtx = avio_alloc_context(inbuf, INBUF_SIZE, // internal Buffer and its size
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0, // bWriteable (1=true,0=false)
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NULL, // user data; will be passed to our callback functions
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_my_read, // read callback
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NULL, // write callback
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NULL); // seek callback
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0, // bWriteable (1=true,0=false)
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NULL, // user data; will be passed to our callback functions
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_my_read, // read callback
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NULL, // write callback
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NULL); // seek callback
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avfc = avformat_alloc_context();
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avfc->pb = pIOCtx;
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avfc->iformat = inp;
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avfc->probesize = 188*5;
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avfc->probesize = 188 * 5;
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thread_started = true;
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if (avformat_open_input(&avfc, NULL, inp, NULL) < 0) {
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if (avformat_open_input(&avfc, NULL, inp, NULL) < 0)
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{
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hal_info("%s: avformat_open_input() failed.\n", __func__);
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goto out;
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}
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@@ -394,31 +404,35 @@ void cAudio::run()
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hal_info("%s: stream 0 no audio codec? 0x%x\n", __func__, p->codec_type);
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codec = avcodec_find_decoder(p->codec_id);
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if (!codec) {
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if (!codec)
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{
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hal_info("%s: Codec for %s not found\n", __func__, avcodec_get_name(p->codec_id));
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goto out;
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}
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if (c)
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av_free(c);
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c = avcodec_alloc_context3(codec);
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if (avcodec_open2(c, codec, NULL) < 0) {
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if (avcodec_open2(c, codec, NULL) < 0)
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{
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hal_info("%s: avcodec_open2() failed\n", __func__);
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goto out;
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}
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if(p->sample_rate == 0 || p->channels == 0){
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if (p->sample_rate == 0 || p->channels == 0)
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{
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av_get_sample_fmt_string(tmp, sizeof(tmp), c->sample_fmt);
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hal_info("Header missing %s, sample_fmt %d (%s) sample_rate %d channels %d\n",avcodec_get_name(p->codec_id), c->sample_fmt, tmp, p->sample_rate, p->channels);
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hal_info("Header missing %s, sample_fmt %d (%s) sample_rate %d channels %d\n", avcodec_get_name(p->codec_id), c->sample_fmt, tmp, p->sample_rate, p->channels);
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goto out2;
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}
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frame = av_frame_alloc();
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if (!frame) {
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if (!frame)
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{
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hal_info("%s: av_frame_alloc failed\n", __func__);
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goto out2;
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}
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/* output sample rate, channels, layout could be set here if necessary */
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o_ch = p->channels; /* 2 */
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o_sr = p->sample_rate; /* 48000 */
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o_layout = p->channel_layout; /* AV_CH_LAYOUT_STEREO */
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o_ch = p->channels; /* 2 */
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o_sr = p->sample_rate; /* 48000 */
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o_layout = p->channel_layout; /* AV_CH_LAYOUT_STEREO */
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if (sformat.channels != o_ch || sformat.rate != o_sr ||
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sformat.byte_format != AO_FMT_NATIVE || sformat.bits != 16 || adevice == NULL)
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{
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@@ -433,10 +447,10 @@ void cAudio::run()
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adevice = ao_open_live(driver, &sformat, NULL);
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ai = ao_driver_info(driver);
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hal_info("%s: changed params ch %d srate %d bits %d adevice %p\n",
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__func__, o_ch, o_sr, 16, adevice);
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if(ai)
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__func__, o_ch, o_sr, 16, adevice);
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if (ai)
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hal_info("libao driver: %d name '%s' short '%s' author '%s'\n",
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driver, ai->name, ai->short_name, ai->author);
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driver, ai->name, ai->short_name, ai->author);
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}
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#if 0
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hal_info(" driver options:");
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@@ -446,17 +460,19 @@ void cAudio::run()
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#endif
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av_get_sample_fmt_string(tmp, sizeof(tmp), c->sample_fmt);
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hal_info("decoding %s, sample_fmt %d (%s) sample_rate %d channels %d\n",
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avcodec_get_name(p->codec_id), c->sample_fmt, tmp, p->sample_rate, p->channels);
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avcodec_get_name(p->codec_id), c->sample_fmt, tmp, p->sample_rate, p->channels);
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swr = swr_alloc_set_opts(swr,
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o_layout, AV_SAMPLE_FMT_S16, o_sr, /* output */
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p->channel_layout, c->sample_fmt, p->sample_rate, /* input */
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0, NULL);
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if (! swr) {
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o_layout, AV_SAMPLE_FMT_S16, o_sr, /* output */
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p->channel_layout, c->sample_fmt, p->sample_rate, /* input */
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0, NULL);
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if (! swr)
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{
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hal_info("could not alloc resample context\n");
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goto out3;
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}
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swr_init(swr);
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while (thread_started) {
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while (thread_started)
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{
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int gotframe = 0;
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if (av_read_frame(avfc, &avpkt) < 0)
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break;
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@@ -464,26 +480,35 @@ void cAudio::run()
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avcodec_decode_audio4(c, frame, &gotframe, &avpkt);
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#else
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av_ret = avcodec_send_packet(c, &avpkt);
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if (av_ret != 0 && av_ret != AVERROR(EAGAIN)) {
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if (av_ret != 0 && av_ret != AVERROR(EAGAIN))
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{
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hal_info("%s: avcodec_send_packet %d\n", __func__, av_ret);
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}else {
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}
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else
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{
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av_ret = avcodec_receive_frame(c, frame);
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if (av_ret != 0 && av_ret != AVERROR(EAGAIN)) {
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if (av_ret != 0 && av_ret != AVERROR(EAGAIN))
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{
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hal_info("%s: avcodec_send_packet %d\n", __func__, av_ret);
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}else {
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}
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else
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{
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gotframe = 1;
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}
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}
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#endif
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if (gotframe && thread_started) {
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if (gotframe && thread_started)
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{
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int out_linesize;
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obuf_sz = av_rescale_rnd(swr_get_delay(swr, p->sample_rate) + frame->nb_samples, o_sr, p->sample_rate, AV_ROUND_UP);
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if (obuf_sz > obuf_sz_max) {
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if (obuf_sz > obuf_sz_max)
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{
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hal_info("obuf_sz: %d old: %d\n", obuf_sz, obuf_sz_max);
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av_free(obuf);
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if (av_samples_alloc(&obuf, &out_linesize, o_ch,
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frame->nb_samples, AV_SAMPLE_FMT_S16, 1) < 0) {
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frame->nb_samples, AV_SAMPLE_FMT_S16, 1) < 0)
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{
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hal_info("av_samples_alloc failed\n");
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av_packet_unref(&avpkt);
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break; /* while (thread_started) */
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@@ -491,13 +516,13 @@ void cAudio::run()
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obuf_sz_max = obuf_sz;
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}
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obuf_sz = swr_convert(swr, &obuf, obuf_sz,
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(const uint8_t **)frame->extended_data, frame->nb_samples);
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(const uint8_t **)frame->extended_data, frame->nb_samples);
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#if (LIBAVUTIL_VERSION_MAJOR < 54)
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curr_pts = av_frame_get_best_effort_timestamp(frame);
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#else
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curr_pts = frame->best_effort_timestamp;
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#endif
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hal_debug("%s: pts 0x%" PRIx64 " %3f\n", __func__, curr_pts, curr_pts/90000.0);
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hal_debug("%s: pts 0x%" PRIx64 " %3f\n", __func__, curr_pts, curr_pts / 90000.0);
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int o_buf_sz = av_samples_get_buffer_size(&out_linesize, o_ch, obuf_sz, AV_SAMPLE_FMT_S16, 1);
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if (o_buf_sz > 0)
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ao_play(adevice, (char *)obuf, o_buf_sz);
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@@ -507,13 +532,13 @@ void cAudio::run()
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// ao_close(adevice); /* can take long :-( */
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av_free(obuf);
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swr_free(&swr);
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out3:
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out3:
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av_frame_free(&frame);
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out2:
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out2:
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avcodec_close(c);
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av_free(c);
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c = NULL;
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out:
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out:
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avformat_close_input(&avfc);
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av_free(pIOCtx->buffer);
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av_free(pIOCtx);
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Reference in New Issue
Block a user