/*
* ZSL
* 虽然名字叫 decode，其实并不解码，只是从 is->sampq 里拿出 af（audioframe,Frame类型）
* 把 af->frame(AVFrame类型)里的 data 经过 swr_convert() 之后，存入 is->audio_buf
* 返回存入的大小(即 resample 之后的大小)
*
*/
int audio_decode_frame(VideoState *is) {
int resampled_data_size,out_size;
Frame *af;
af = frame_queue_peek_readable(&is->sampq);
frame_queue_next(&is->sampq);
if (!is->swr_ctx) {
is->swr_ctx = swr_alloc_set_opts(NULL,
AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16, is->audio_ctx->sample_rate,
av_get_default_channel_layout(is->audio_ctx->channels), is->audio_ctx->sample_fmt, is->audio_ctx->sample_rate,
0, NULL);
swr_init(is->swr_ctx);
}
const uint8_t **in = (const uint8_t **)af->frame->extended_data;
uint8_t **out = &is->audio_buf;
//out_size = av_samples_get_buffer_size(NULL, 2, af->frame->nb_samples, AV_SAMPLE_FMT_S16, 1);
out_size = 2 * 1152 * 2;
if (out_size < 0) { /*比如 af->frame->nb_samples==0 的时候，这必须要处理一下，不然一会儿 av_fast_malloc() 就出问题了 */
av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
return -1;
}
int len2;
av_fast_malloc(&is->audio_buf, &is->audio_buf_size, out_size);
len2 = swr_convert(is->swr_ctx, out, af->frame->nb_samples, in, af->frame->nb_samples);
resampled_data_size = len2 * 2 * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
return resampled_data_size;
}
void MediaThread::Pcm::save(const AVFrame *frame, double stamp)
{
stamp_ = stamp;
#if 0
memset(buf_, 0, 4096);
data_len_ = 4096;
#else
if (!swr_) {
// 总是输出 2, s16, 32000 ...
swr_ = swr_alloc_set_opts(0, AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16, 32000,
frame->channel_layout, (AVSampleFormat)frame->format, frame->sample_rate,
0, 0);
ch_ = 2;
samplerate_ = 32000;
bitsize_ = 16;
swr_init(swr_);
}
size_t out_size = frame->nb_samples * 2 * 2; // samples * bytes per sample * channels
if (buf_len_ < out_size) {
buf_ = (unsigned char*)realloc(buf_, out_size);
buf_len_ = out_size;
}
int samples = swr_convert(swr_, &buf_, frame->nb_samples, (const uint8_t**)frame->extended_data, frame->nb_samples);
data_len_ = samples * 2 * 2; // samples * bytes per sample * channels
#endif
}
void Parser::InitResampler()
{
swr_ctx = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(cdc_ctx_out->channels),
cdc_ctx_out->sample_fmt,
cdc_ctx_out->sample_rate,
channelLayoutIn,
(AVSampleFormat)sampleFormatIn,
sampleRateIn,
0,0);
if(!swr_ctx)
throw ContextCreatorException() << errno_code(MIR_ERR_ALLOC_SWR_CONTEXT);
// set options
av_opt_set_int(swr_ctx, "in_channel_layout", channelLayoutIn, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", sampleRateIn, 0);
av_opt_set_int(swr_ctx, "in_bit_rate", bitRateIn, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", (AVSampleFormat)sampleFormatIn, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", cdc_ctx_out->channel_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", cdc_ctx_out->sample_rate, 0);
av_opt_set_int(swr_ctx, "out_bit_rate", cdc_ctx_out->bit_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", cdc_ctx_out->sample_fmt, 0);
if (swr_init(swr_ctx) < 0)
throw ContextCreatorException() << errno_code(MIR_ERR_INIT_SWR_CONTEXT);
}
// 函数实现
void* renderopen(void *surface, AVRational frate, int pixfmt, int w, int h,
int64_t ch_layout, AVSampleFormat sndfmt, int srate)
{
WAVEFORMATEX wfx = {0};
RENDER *render = (RENDER*)malloc(sizeof(RENDER));
memset(render, 0, sizeof(RENDER));
render->hRenderWnd = (HWND)surface; // save hwnd
// init for audio
wfx.cbSize = sizeof(wfx);
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.wBitsPerSample = 16; // 16bit
wfx.nSamplesPerSec = 44100; // 44.1k
wfx.nChannels = 2; // stereo
wfx.nBlockAlign = wfx.nChannels * wfx.wBitsPerSample / 8;
wfx.nAvgBytesPerSec = wfx.nBlockAlign * wfx.nSamplesPerSec;
waveOutOpen(&(render->hWaveOut), WAVE_MAPPER, &wfx, (DWORD_PTR)waveOutProc, (DWORD)render, CALLBACK_FUNCTION);
waveOutPause(render->hWaveOut);
wavqueue_create(&(render->WavQueue), render->hWaveOut, ((int64_t)44100 * 4 * frate.den / frate.num) & ~0x3);
/* allocate & init swr context */
render->pSWRContext = swr_alloc_set_opts(NULL, AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16, 44100,
ch_layout, sndfmt, srate, 0, NULL);
swr_init(render->pSWRContext);
// init for video
render->nVideoWidth = w;
render->nVideoHeight = h;
render->nRenderWidth = GetSystemMetrics(SM_CXSCREEN);
render->nRenderHeight= GetSystemMetrics(SM_CYSCREEN);
render->nRenderNewW = render->nRenderWidth;
render->nRenderNewH = render->nRenderHeight;
render->PixelFormat = (PixelFormat)pixfmt;
// create sws context
render->pSWSContext = sws_getContext(
render->nVideoWidth,
render->nVideoHeight,
render->PixelFormat,
render->nRenderWidth,
render->nRenderHeight,
PIX_FMT_RGB32,
SWS_BILINEAR,
0, 0, 0);
render->iFrameTick = 1000 * frate.den / frate.num;
render->iSleepTick = render->iFrameTick;
// create dc & bitmaps
render->hRenderDC = GetDC(render->hRenderWnd);
render->hBufferDC = CreateCompatibleDC(render->hRenderDC);
// create bmp queue
bmpqueue_create(&(render->BmpQueue), render->hBufferDC, GetSystemMetrics(SM_CXSCREEN), GetSystemMetrics(SM_CYSCREEN), 32);
render->nRenderStatus = 0;
pthread_create(&(render->hVideoThread), NULL, VideoRenderThreadProc, render);
return render;
}
EC_U32 AudioWaveScale::Init(MediaCtxInfo* pMediaInfo, AudioPCMBuffer *pFirstFrame)
{
if (EC_NULL == pMediaInfo)
return Audio_Render_Err_InitFail;
EC_S32 out_sample_rate = pMediaInfo->m_nSampleRate;
EC_S64 out_channel_layout = AV_CH_LAYOUT_STEREO;
EC_S32 out_channels = av_get_channel_layout_nb_channels(out_channel_layout);
AVSampleFormat out_sample_fmt = AV_SAMPLE_FMT_S16;
AVCodecContext *pCodecCtx = (AVCodecContext*)(pMediaInfo->m_pAudioCodecInfo);
EC_S64 in_channel_layout = av_get_default_channel_layout(pCodecCtx->channels);
EC_S32 in_sample_rate = pCodecCtx->sample_rate;
AVSampleFormat in_sample_fmt = pCodecCtx->sample_fmt;
m_nOutChannels = out_channels;
m_nOutSampleFormat = out_sample_fmt;
m_pWaveScaleContext = swr_alloc();
m_pWaveScaleContext = swr_alloc_set_opts(m_pWaveScaleContext,
out_channel_layout,
out_sample_fmt,
out_sample_rate,
in_channel_layout,
in_sample_fmt,
in_sample_rate, 0, NULL);
EC_S32 nRet = swr_init(m_pWaveScaleContext);
if (nRet < 0) return Audio_Render_Err_InitFail;
m_pScaleOutbuffer = (uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE * 2);
if (m_pScaleOutbuffer == EC_NULL) return EC_Err_Memory_Low;
return Audio_Render_Err_None;
}
static int config_output(AVFilterLink *outlink)
{
int ret;
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
AConvertContext *aconvert = ctx->priv;
char buf1[64], buf2[64];
/* if not specified in args, use the format and layout of the output */
if (aconvert->out_sample_fmt == AV_SAMPLE_FMT_NONE)
aconvert->out_sample_fmt = outlink->format;
if (aconvert->out_chlayout == 0)
aconvert->out_chlayout = outlink->channel_layout;
aconvert->swr = swr_alloc_set_opts(aconvert->swr,
aconvert->out_chlayout, aconvert->out_sample_fmt, inlink->sample_rate,
inlink->channel_layout, inlink->format, inlink->sample_rate,
0, ctx);
if (!aconvert->swr)
return AVERROR(ENOMEM);
ret = swr_init(aconvert->swr);
if (ret < 0)
return ret;
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, outlink->channel_layout);
av_log(ctx, AV_LOG_VERBOSE,
"fmt:%s cl:%s -> fmt:%s cl:%s\n",
av_get_sample_fmt_name(inlink ->format), buf1,
av_get_sample_fmt_name(outlink->format), buf2);
return 0;
}
CAudioReader::CAudioReader(CAudioSource *source, AudioTrack *info)
{
m_Source = source;
m_Info = info;
m_MediaInfo = info->m_Media;
m_DestSampleRate = source->m_SampleRate;
m_DestChannels = source->m_Channels;
m_DestPacketBytes = source->m_PacketBytes;
if (m_MediaInfo->m_IsPlanar)
{
m_SourceChannels = m_MediaInfo->m_nChannel;
m_SourcePacketBytes = av_get_bytes_per_sample((enum AVSampleFormat)m_MediaInfo->m_SampleFormat);
}
else
{
m_SourceChannels = 1;
m_SourcePacketBytes = m_MediaInfo->m_nChannel * av_get_bytes_per_sample((enum AVSampleFormat)m_MediaInfo->m_SampleFormat);
}
m_SourceSampleRate = m_MediaInfo->m_SampleRate;
for(int i = 0; i < m_SourceChannels; i++)
{
m_hFiles[i] = CreateFile(m_MediaInfo->m_AudioTmpFile[i],
GENERIC_READ,
FILE_SHARE_READ | FILE_SHARE_WRITE | FILE_SHARE_DELETE,
NULL,
OPEN_EXISTING,
FILE_ATTRIBUTE_NORMAL,
NULL);
}
if (
(m_MediaInfo->m_SampleRate != m_Source->m_SampleRate) ||
(m_MediaInfo->m_nChannel != m_Source->m_Channels) ||
(m_MediaInfo->m_SampleFormat != m_Source->m_SampleFormat) ||
(m_MediaInfo->m_channel_layout != m_Source->m_Layout)
)
{
swr_context =
swr_alloc_set_opts(NULL,
m_Source->m_Layout, (AVSampleFormat)source->m_SampleFormat, m_Source->m_SampleRate,
m_MediaInfo->m_channel_layout, (AVSampleFormat)m_MediaInfo->m_SampleFormat, m_MediaInfo->m_SampleRate,
0, NULL);
swr_init(swr_context);
for(int i = 0; i < m_SourceChannels; i++)
{
m_ReadBuffer[i] = (uint8_t *)MemoryAlloc(m_SourceSampleRate * m_SourcePacketBytes);
}
for(int i = 0; i < m_DestChannels; i++)
{
m_ResampleBuffer[i] = (uint8_t *)MemoryAlloc(m_DestSampleRate * m_DestPacketBytes * 2);
}
}
ResetStartStop();
}
static void audio_convert(dtaudio_decoder_t *decoder, AVFrame * dst,
AVFrame * src)
{
int nb_sample;
int dst_buf_size;
int out_channels;
//for audio post processor
//struct SwsContext *m_sws_ctx = NULL;
struct SwrContext *m_swr_ctx = NULL;
//ResampleContext *m_resample_ctx=NULL;
enum AVSampleFormat src_fmt = avctxp->sample_fmt;
enum AVSampleFormat dst_fmt = AV_SAMPLE_FMT_S16;
dst->linesize[0] = src->linesize[0];
*dst = *src;
dst->data[0] = NULL;
out_channels = decoder->para.dst_channels;
nb_sample = frame->nb_samples;
dst_buf_size = nb_sample * av_get_bytes_per_sample(dst_fmt) * out_channels;
dst->data[0] = (uint8_t *) av_malloc(dst_buf_size);
avcodec_fill_audio_frame(dst, out_channels, dst_fmt, dst->data[0], dst_buf_size,
0);
dt_debug(TAG, "SRCFMT:%d dst_fmt:%d \n", src_fmt, dst_fmt);
/* resample toAV_SAMPLE_FMT_S16 */
if (src_fmt != dst_fmt || out_channels != decoder->para.channels) {
if (!m_swr_ctx) {
uint64_t in_channel_layout = av_get_default_channel_layout(avctxp->channels);
uint64_t out_channel_layout = av_get_default_channel_layout(out_channels);
m_swr_ctx = swr_alloc_set_opts(NULL, out_channel_layout, dst_fmt,
avctxp->sample_rate, in_channel_layout, src_fmt, avctxp->sample_rate, 0, NULL);
swr_init(m_swr_ctx);
}
uint8_t **out = (uint8_t **) & dst->data;
const uint8_t **in = (const uint8_t **) src->extended_data;
if (m_swr_ctx) {
int ret, out_count;
out_count = nb_sample;
ret = swr_convert(m_swr_ctx, out, out_count, in, nb_sample);
if (ret < 0) {
//set audio mute
memset(dst->data[0], 0, dst_buf_size);
printf("audio convert failed, set mute data\n");
}
}
} else { // no need to convert ,just copy
memcpy(dst->data[0], src->data[0], src->linesize[0]);
}
//free context
if (m_swr_ctx != NULL) {
swr_free(&m_swr_ctx);
}
//if(m_resample_ctx!=NULL)
// audio_resample_close(m_resample_ctx);
}
SimpleAT3::SimpleAT3()
: codec_(0),
codecCtx_(0),
swrCtx_(0) {
frame_ = av_frame_alloc();
codec_ = avcodec_find_decoder(AV_CODEC_ID_ATRAC3P);
if (!codec_) {
// Eh, we shouldn't even have managed to compile. But meh.
ERROR_LOG(ME, "This version of FFMPEG does not support AV_CODEC_ID_ATRAC3P (Atrac3+). Update your submodule.");
return;
}
codecCtx_ = avcodec_alloc_context3(codec_);
if (!codecCtx_) {
ERROR_LOG(ME, "Failed to allocate a codec context");
return;
}
codecCtx_->channels = 2;
codecCtx_->channel_layout = AV_CH_LAYOUT_STEREO;
AVDictionary *opts = 0;
av_dict_set(&opts, "channels", "2", 0);
av_dict_set(&opts, "sample_rate", "44100", 0);
if (avcodec_open2(codecCtx_, codec_, &opts) < 0) {
ERROR_LOG(ME, "Failed to open codec");
return;
}
av_dict_free(&opts);
// Initializing the sample rate convert. We only really use it to convert float output
// into int.
int wanted_channels = 2;
int64_t wanted_channel_layout = av_get_default_channel_layout(wanted_channels);
int64_t dec_channel_layout = av_get_default_channel_layout(2);
swrCtx_ = swr_alloc_set_opts(
swrCtx_,
wanted_channel_layout,
AV_SAMPLE_FMT_S16,
codecCtx_->sample_rate,
dec_channel_layout,
codecCtx_->sample_fmt,
codecCtx_->sample_rate,
0,
NULL);
if (!swrCtx_ || swr_init(swrCtx_) < 0) {
ERROR_LOG(ME, "swr_init: Failed to initialize the resampling context");
avcodec_close(codecCtx_);
codec_ = 0;
return;
}
}
/**
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
*/
int AudioDecoder::init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context)
{
int error;
/**
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(output_codec_context->channels),
output_codec_context->sample_fmt,
output_codec_context->sample_rate,
av_get_default_channel_layout(input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (!resample_context) {
ELOG_WARN( "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/**
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
ELOG_DEBUG( "audio input sample_rate = %d, out %d", input_codec_context->sample_rate, output_codec_context->sample_rate);
/** Open the resampler with the specified parameters. */
if ((error = swr_init(resample_context)) < 0) {
ELOG_WARN( "Could not open resample context");
swr_free(&resample_context);
return error;
}
/** Open the resampler with the specified parameters. */
if ((error = swr_init(resample_context)) < 0) {
ELOG_DEBUG( "Could not open resample context");
swr_free(&resample_context);
return error;
}
ELOG_DEBUG( "swr_init done");
return 0;
}
void XAudioStream::setSpeed(float speed)
{
if(m_speed == speed || speed <= 0.0f) return;
m_speed = speed;
swr_free(&m_pSwrContext);
m_pSwrContext = swr_alloc();
if(m_pSwrContext == NULL) return;
if(m_pAudioCodecCtx->channel_layout == 0)
{
swr_alloc_set_opts(m_pSwrContext,av_get_default_channel_layout(XEG.getAudioChannelSum()),getSampleFormat(),XEG.getAudioSampleRate() * m_speed,
av_get_default_channel_layout(m_pAudioCodecCtx->channels),m_pAudioCodecCtx->sample_fmt,m_pAudioCodecCtx->sample_rate,0,NULL);
}else
{
swr_alloc_set_opts(m_pSwrContext,av_get_default_channel_layout(XEG.getAudioChannelSum()),getSampleFormat(),XEG.getAudioSampleRate() * m_speed,
m_pAudioCodecCtx->channel_layout,m_pAudioCodecCtx->sample_fmt,m_pAudioCodecCtx->sample_rate,0,NULL);
}
if(swr_init(m_pSwrContext) < 0)
{
LogStr("swr_init() fail");
return;
}
}
kxMovieError openAudioStream(size_t audioStream)
{
AVCodecContext *codecCtx = _formatCtx->streams[audioStream]->codec;
SwrContext *swrContext = NULL;
AVCodec *codec = avcodec_find_decoder(codecCtx->codec_id);
if(!codec)
return kxMovieErrorCodecNotFound;
if (avcodec_open2(codecCtx, codec, NULL) < 0)
return kxMovieErrorOpenCodec;
if (!audioCodecIsSupported(codecCtx))
{
swrContext = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(codecCtx->channels),
AV_SAMPLE_FMT_S16,
codecCtx->sample_rate,
av_get_default_channel_layout(codecCtx->channels),
codecCtx->sample_fmt,
codecCtx->sample_rate,
0,
NULL);
if (!swrContext ||
swr_init(swrContext)) {
if (swrContext)
swr_free(&swrContext);
avcodec_close(codecCtx);
return kxMovieErroReSampler;
}
}
_audioFrame = av_frame_alloc();
if (!_audioFrame) {
if (swrContext)
swr_free(&swrContext);
avcodec_close(codecCtx);
return kxMovieErrorAllocateFrame;
}
_audioStream = audioStream;
_audioCodecCtx = codecCtx;
_swrContext = swrContext;
return kxMovieErrorNone;
}
int AudioResampling(AVCodecContext * audio_dec_ctx,
AVFrame * pAudioDecodeFrame,
enum AVSampleFormat out_sample_fmt,
int out_channels,
int out_sample_rate,
uint8_t* out_buf)
{
struct SwrContext * swr_ctx = 0;
swr_ctx = swr_alloc_set_opts(swr_ctx,audio_dec_ctx->channel_layout,out_sample_fmt,out_sample_rate,
audio_dec_ctx->channel_layout,audio_dec_ctx->sample_fmt,audio_dec_ctx->sample_rate,0,0);
int ret = 0;
int dst_linesize = 0;
int resampled_data_size = av_samples_get_buffer_size(&dst_linesize, out_channels,audio_dec_ctx->frame_size,audio_dec_ctx->sample_fmt, 1);
uint8_t *dst_data = (uint8_t*)av_malloc(resampled_data_size);
if ((ret = swr_init(swr_ctx)) < 0) {
printf("Failed to initialize the resampling context\n");
return -1;
}
if (swr_ctx){
ret = swr_convert(swr_ctx, &dst_data, dst_linesize,
(const uint8_t **)pAudioDecodeFrame->data, pAudioDecodeFrame->nb_samples);
resampled_data_size = av_samples_get_buffer_size(&dst_linesize,out_channels,ret,out_sample_fmt,1);
if (ret < 0) {
printf("swr_convert error \n");
return -1;
}
if (resampled_data_size < 0) {
printf("av_samples_get_buffer_size error \n");
return -1;
}
} else{
printf("swr_ctx null error \n");
return -1;
}
memcpy(out_buf, dst_data, resampled_data_size);
if (dst_data) {
av_free(dst_data);
}
if (swr_ctx)
{
swr_free(&swr_ctx);
}
return resampled_data_size;
}
void render_audio(void *hrender, AVFrame *audio)
{
RENDER *render = (RENDER*)hrender;
int sampnum = 0;
DWORD apts = (DWORD)audio->pts;
if (!render->adev) return;
do {
if (render->nAdevBufAvail == 0) {
adev_request(render->adev, &render->pAdevHdrCur);
apts += render->nFramePeriod * render->nRenderSpeedCur / 100;
render->nAdevBufAvail = (int )render->pAdevHdrCur->size;
render->pAdevBufCur = (BYTE*)render->pAdevHdrCur->data;
}
if (render->nRenderSpeedCur != render->nRenderSpeedNew) {
render->nRenderSpeedCur = render->nRenderSpeedNew;
// set vdev frame rate
int framerate = (render->FrameRate.num * render->nRenderSpeedCur) / (render->FrameRate.den * 100);
vdev_setfrate(render->vdev, framerate > 1 ? framerate : 1);
//++ allocate & init swr context
if (render->pSWRContext) {
swr_free(&render->pSWRContext);
}
int samprate = 44100 * 100 / render->nRenderSpeedCur;
render->pSWRContext = swr_alloc_set_opts(NULL, AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16, samprate,
render->nChanLayout, render->SampleFormat, render->nSampleRate, 0, NULL);
swr_init(render->pSWRContext);
//-- allocate & init swr context
}
//++ do resample audio data ++//
sampnum = swr_convert(render->pSWRContext, (uint8_t**)&render->pAdevBufCur,
render->nAdevBufAvail / 4, (const uint8_t**)audio->extended_data,
audio->nb_samples);
audio->extended_data = NULL;
audio->nb_samples = 0;
render->nAdevBufAvail -= sampnum * 4;
render->pAdevBufCur += sampnum * 4;
//-- do resample audio data --//
if (render->nAdevBufAvail == 0) {
adev_post(render->adev, apts);
}
} while (sampnum > 0);
}
Swr::Swr(std::int64_t out_ch_layout, AVSampleFormat out_sample_fmt,
int out_sample_rate, std::int64_t in_ch_layout,
AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset,
void* log_ctx)
{
this->context = swr_alloc_set_opts(nullptr, out_ch_layout,
out_sample_fmt, out_sample_rate,
in_ch_layout, in_sample_fmt,
in_sample_rate, log_offset, log_ctx);
if (this->context == nullptr) {
throw std::bad_alloc();
}
assert(this->context != nullptr);
swr_init(this->context);
}
void init_swr(){
uint64_t out_channel_layout=AV_CH_LAYOUT_STEREO;
//nb_samples: AAC-1024 MP3-1152
out_sample_rate=pCodecCtx->sample_rate;
out_channels=av_get_channel_layout_nb_channels(out_channel_layout);
out_buffer=(uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE*out_channels);
//FIX:Some Codec's Context Information is missing
int in_channel_layout=av_get_default_channel_layout(pCodecCtx->channels);
//Swr
au_convert_ctx = swr_alloc();
swr_alloc_set_opts(au_convert_ctx,out_channel_layout, out_sample_fmt, out_sample_rate,
in_channel_layout, pCodecCtx->sample_fmt , pCodecCtx->sample_rate,0, NULL);
if(swr_init(au_convert_ctx)<0){
au_convert_ctx=NULL;
}
createBufferQueueAudioPlayer(2,out_sample_rate);
}
void AudioDec::set_audio_spec(AVCodecContext* pCodecCtx)
{
/* 오디오 출력을 위한 구조체 설정 파트 */
/* sample rate */
wfx.nSamplesPerSec = pCodecCtx->sample_rate;
/* number of bits per sample of mono data */
switch (pCodecCtx->sample_fmt)
{
case AV_SAMPLE_FMT_U8: wfx.wBitsPerSample = 8; break;
case AV_SAMPLE_FMT_S16: wfx.wBitsPerSample = 16; break;
case AV_SAMPLE_FMT_S32: wfx.wBitsPerSample = 32; break;
case AV_SAMPLE_FMT_FLT: wfx.wBitsPerSample = sizeof(double)* 8; break;
case AV_SAMPLE_FMT_FLTP: wfx.wBitsPerSample = 16; break;
default: wfx.wBitsPerSample = 0; break;
}
wfx.wBitsPerSample = 16;
wfx.nChannels = FFMIN(2, pCodecCtx->channels);
/* the count in bytes of the size of extra information (after cbSize) */
wfx.cbSize = 0;
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nBlockAlign = (wfx.wBitsPerSample * wfx.nChannels) / 8;
wfx.nAvgBytesPerSec = wfx.nBlockAlign * wfx.nSamplesPerSec;
MMRESULT result = waveOutOpen(&hWaveOut, WAVE_MAPPER, &wfx, (DWORD)audio_callback_func_static, (DWORD)&waveFreeBlockCount, CALLBACK_FUNCTION);
if (result != MMSYSERR_NOERROR)
{
// 오디오 출력장치를 열지 못했다...
TRACE("err waveOutOpen \n");
return;
}
pSwrContext = swr_alloc_set_opts(
NULL, AV_CH_LAYOUT_STEREO,
AV_SAMPLE_FMT_S16,
pCodecCtx->sample_rate,
wfx.nChannels,
pCodecCtx->sample_fmt,
pCodecCtx->sample_rate,
0, NULL);
swr_init(pSwrContext);
}
void render_audio(void *hrender, AVFrame *audio)
{
RENDER *render = (RENDER*)hrender;
int sampnum = 0;
int64_t apts = audio->pts;
if (!render || !render->adev) return;
do {
if (render->adev_buf_avail == 0) {
adev_lock(render->adev, &render->adev_hdr_cur);
apts += 10 * render->render_speed_cur * render->frame_rate.den / render->frame_rate.num;
render->adev_buf_avail = (int )render->adev_hdr_cur->size;
render->adev_buf_cur = (uint8_t*)render->adev_hdr_cur->data;
}
if (render->render_speed_cur != render->render_speed_new) {
render->render_speed_cur = render->render_speed_new;
//++ allocate & init swr context
if (render->swr_context) {
swr_free(&render->swr_context);
}
int samprate = (int)(ADEV_SAMPLE_RATE * 100.0 / render->render_speed_cur);
render->swr_context = swr_alloc_set_opts(NULL, AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16, samprate,
render->chan_layout, render->sample_fmt, render->sample_rate, 0, NULL);
swr_init(render->swr_context);
//-- allocate & init swr context
}
//++ do resample audio data ++//
sampnum = swr_convert(render->swr_context,
(uint8_t**)&render->adev_buf_cur, render->adev_buf_avail / 4,
(const uint8_t**)audio->extended_data, audio->nb_samples);
audio->extended_data = NULL;
audio->nb_samples = 0;
render->adev_buf_avail -= sampnum * 4;
render->adev_buf_cur += sampnum * 4;
//-- do resample audio data --//
if (render->adev_buf_avail == 0) {
adev_unlock(render->adev, apts);
}
} while (sampnum > 0);
}
void AVFile::_updateSWR()
{
if (swrCtx) {
swr_free(&swrCtx);
}
if (!_channels || !_sample_rate || !codecCtx)
return;
if (codecCtx->channel_layout != (uint64_t)av_get_default_channel_layout(_channels) ||
codecCtx->sample_fmt != AV_SAMPLE_FMT_FLT ||
codecCtx->sample_rate != (int)_sample_rate)
{
swrCtx = swr_alloc_set_opts(0, av_get_default_channel_layout(_channels), AV_SAMPLE_FMT_FLT, _sample_rate,
codecCtx->channel_layout, codecCtx->sample_fmt, codecCtx->sample_rate,
0, 0);
if (!swrCtx)
throw AVException("Unable to allocate swresample context");
swr_init(swrCtx);
}
}
void CTransCoder::SwrInit( AVCodecContext* ctx )
{
//Out Audio Param
m_out_channel_layout = AV_CH_LAYOUT_STEREO;
//nb_samples:
m_out_nb_samples = 1024;
m_out_sample_fmt = AV_SAMPLE_FMT_S16;
m_out_sample_rate = 44100;
m_out_channels = av_get_channel_layout_nb_channels(m_out_channel_layout);
//Out buffer size
int out_buffer_size = av_samples_get_buffer_size( NULL, m_out_channels, m_out_nb_samples, m_out_sample_fmt, 1 );
m_audioBuffer = (uint8_t *)av_malloc( MAX_AUDIO_FRAME_SIZE );
int64_t in_channel_layout = av_get_default_channel_layout(ctx->channels);
m_au_convert_ctx = swr_alloc();
m_au_convert_ctx = swr_alloc_set_opts( m_au_convert_ctx, m_out_channel_layout, m_out_sample_fmt, m_out_sample_rate,
in_channel_layout, ctx->sample_fmt, ctx->sample_rate, 0, NULL);
swr_init(m_au_convert_ctx);
}
void AudioOutput::readAVFrame(AVFrame* frame)
{
if(!swrCtx){
swrCtx = swr_alloc_set_opts(NULL,
aw->audioCodecCtx->channel_layout, AV_SAMPLE_FMT_S16, aw->audioCodecCtx->sample_rate,
frame->channel_layout, (AVSampleFormat)frame->format, frame->sample_rate,
0, NULL);
swr_init(swrCtx);
}
const uint8_t **in = (const uint8_t **)frame->extended_data;
uint8_t **out = &data;
int bufLen = aw->audioCodecCtx->channels * frame->nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
if(capacity < bufLen){
data = (uint8_t*)realloc(data, bufLen);
capacity = bufLen;
}
int out_count = (int64_t)(frame->nb_samples) *4* aw->audioCodecCtx->sample_rate / frame->sample_rate + 256;
/*
int out_size = av_samples_get_buffer_size(NULL, frame->channels, out_count, AV_SAMPLE_FMT_S16, 0);
int len2;
if (out_size < 0) {
std::cout<<"av_samples_get_buffer_size() failed\n";
return;
}
*/
int len2 = swr_convert(swrCtx, out, out_count, in, frame->nb_samples);
//std::cout<<len2<<"\n";
if (len2 < 0) {
std::cout<<"swr_convert() failed\n";
return;
}
index = 0;
size = aw->audioCodecCtx->channels * len2 * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
}
int CAudioDecoder::open(void)
{
int ret = 0;
bOpen = true;
pFormatCtx = avformat_alloc_context();
const int BUF_LEN = (1024 * 200);
iobuffer = (unsigned char *)av_malloc(BUF_LEN);
avio = avio_alloc_context(iobuffer, BUF_LEN, 0, this, fill_iobuffer, NULL, NULL);
pFormatCtx->pb = avio;
pFormatCtx->flags = AVFMT_FLAG_CUSTOM_IO;
if (avformat_open_input(&pFormatCtx, NULL, NULL, NULL) != 0){
bOpen = false;
return -1; // Couldn't open file
}
// Retrieve stream information
if (avformat_find_stream_info(pFormatCtx, NULL)<0){
bOpen = false;
return -1; // Couldn't find stream information
}
// Find the first video stream
audioStream = -1;
unsigned int i;
for (i = 0; i<pFormatCtx->nb_streams; i++)
if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
audioStream = i;
break;
}
if (audioStream == -1){
bOpen = false;
return -1; // Didn't find a video stream
}
/*AV_SAMPLE_FMT_S16;*/
// Get a pointer to the codec context for the video stream
pCodecCtxOrig = pFormatCtx->streams[audioStream]->codec;
if (!pCodecCtxOrig){
fprintf(stderr, "Unsupported codec!\n");
bOpen = false;
return -1; // Codec not found
}
// Find the decoder for the video stream
pCodec = avcodec_find_decoder(pCodecCtxOrig->codec_id);
if (pCodec == NULL) {
fprintf(stderr, "Unsupported codec!\n");
bOpen = false;
return -1; // Codec not found
}
// Copy context
pCodecCtx = avcodec_alloc_context3(pCodec);
if (avcodec_copy_context(pCodecCtx, pCodecCtxOrig) != 0) {
fprintf(stderr, "Couldn't copy codec context");
bOpen = false;
return -1; // Error copying codec context
}
// Open codec
if (avcodec_open2(pCodecCtx, pCodec, NULL)<0){
bOpen = false;
return -1; // Could not open codec
}
// Allocate audio frame
pFrame = av_frame_alloc();
pFrameYUV = av_frame_alloc();
av_init_packet(&packet);
out_buffer = (uint8_t *)av_malloc(MAX_AUDIO_FRAME_SIZE * 2);
//Out Audio Param
out_channel_layout = AV_CH_LAYOUT_STEREO;
out_nb_samples = 1024;
out_sample_fmt = AV_SAMPLE_FMT_S16;
out_sample_rate = 44100;
out_channels = av_get_channel_layout_nb_channels(out_channel_layout);
//Out Buffer Size
out_buffer_size = av_samples_get_buffer_size(NULL, out_channels, out_nb_samples, out_sample_fmt, 1);
//SDL_AudioSpec
wanted_spec.freq = out_sample_rate;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.channels = out_channels;
wanted_spec.silence = 0;
wanted_spec.samples = 1024/*out_nb_samples*/;
wanted_spec.callback = fill_audio;
wanted_spec.userdata = pCodecCtx;
if (SDL_OpenAudio(&wanted_spec, NULL)<0){
printf("can't open audio.\n");
return -1;
}
int64_t in_channel_layout = av_get_default_channel_layout(pCodecCtx->channels);
au_convert_ctx = swr_alloc();
au_convert_ctx = swr_alloc_set_opts(au_convert_ctx, out_channel_layout, out_sample_fmt, out_sample_rate,
in_channel_layout, pCodecCtx->sample_fmt, pCodecCtx->sample_rate, 0, NULL);
swr_init(au_convert_ctx);
bOpen = true;
//bStop = false;
//start();
return ret;
}
bool FFmpegEncoderOpen(struct FFmpegEncoder* encoder, const char* outfile) {
AVCodec* acodec = avcodec_find_encoder_by_name(encoder->audioCodec);
AVCodec* vcodec = avcodec_find_encoder_by_name(encoder->videoCodec);
if ((encoder->audioCodec && !acodec) || !vcodec || !FFmpegEncoderVerifyContainer(encoder)) {
return false;
}
encoder->currentAudioSample = 0;
encoder->currentAudioFrame = 0;
encoder->currentVideoFrame = 0;
encoder->nextAudioPts = 0;
AVOutputFormat* oformat = av_guess_format(encoder->containerFormat, 0, 0);
#ifndef USE_LIBAV
avformat_alloc_output_context2(&encoder->context, oformat, 0, outfile);
#else
encoder->context = avformat_alloc_context();
strncpy(encoder->context->filename, outfile, sizeof(encoder->context->filename) - 1);
encoder->context->filename[sizeof(encoder->context->filename) - 1] = '\0';
encoder->context->oformat = oformat;
#endif
if (acodec) {
#ifdef FFMPEG_USE_CODECPAR
encoder->audioStream = avformat_new_stream(encoder->context, NULL);
encoder->audio = avcodec_alloc_context3(acodec);
#else
encoder->audioStream = avformat_new_stream(encoder->context, acodec);
encoder->audio = encoder->audioStream->codec;
#endif
encoder->audio->bit_rate = encoder->audioBitrate;
encoder->audio->channels = 2;
encoder->audio->channel_layout = AV_CH_LAYOUT_STEREO;
encoder->audio->sample_rate = encoder->sampleRate;
encoder->audio->sample_fmt = encoder->sampleFormat;
AVDictionary* opts = 0;
av_dict_set(&opts, "strict", "-2", 0);
if (encoder->context->oformat->flags & AVFMT_GLOBALHEADER) {
#ifdef AV_CODEC_FLAG_GLOBAL_HEADER
encoder->audio->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
#else
encoder->audio->flags |= CODEC_FLAG_GLOBAL_HEADER;
#endif
}
avcodec_open2(encoder->audio, acodec, &opts);
av_dict_free(&opts);
#if LIBAVCODEC_VERSION_MAJOR >= 55
encoder->audioFrame = av_frame_alloc();
#else
encoder->audioFrame = avcodec_alloc_frame();
#endif
if (!encoder->audio->frame_size) {
encoder->audio->frame_size = 1;
}
encoder->audioFrame->nb_samples = encoder->audio->frame_size;
encoder->audioFrame->format = encoder->audio->sample_fmt;
encoder->audioFrame->pts = 0;
#ifdef USE_LIBAVRESAMPLE
encoder->resampleContext = avresample_alloc_context();
av_opt_set_int(encoder->resampleContext, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_int(encoder->resampleContext, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
av_opt_set_int(encoder->resampleContext, "in_sample_rate", PREFERRED_SAMPLE_RATE, 0);
av_opt_set_int(encoder->resampleContext, "out_sample_rate", encoder->sampleRate, 0);
av_opt_set_int(encoder->resampleContext, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int(encoder->resampleContext, "out_sample_fmt", encoder->sampleFormat, 0);
avresample_open(encoder->resampleContext);
#else
encoder->resampleContext = swr_alloc_set_opts(NULL, AV_CH_LAYOUT_STEREO, encoder->sampleFormat, encoder->sampleRate,
AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16, PREFERRED_SAMPLE_RATE, 0, NULL);
swr_init(encoder->resampleContext);
#endif
encoder->audioBufferSize = (encoder->audioFrame->nb_samples * PREFERRED_SAMPLE_RATE / encoder->sampleRate) * 4;
encoder->audioBuffer = av_malloc(encoder->audioBufferSize);
encoder->postaudioBufferSize = av_samples_get_buffer_size(0, encoder->audio->channels, encoder->audio->frame_size, encoder->audio->sample_fmt, 0);
encoder->postaudioBuffer = av_malloc(encoder->postaudioBufferSize);
avcodec_fill_audio_frame(encoder->audioFrame, encoder->audio->channels, encoder->audio->sample_fmt, (const uint8_t*) encoder->postaudioBuffer, encoder->postaudioBufferSize, 0);
if (encoder->audio->codec->id == AV_CODEC_ID_AAC &&
(strcasecmp(encoder->containerFormat, "mp4") ||
strcasecmp(encoder->containerFormat, "m4v") ||
strcasecmp(encoder->containerFormat, "mov"))) {
// MP4 container doesn't support the raw ADTS AAC format that the encoder spits out
#ifdef FFMPEG_USE_NEW_BSF
av_bsf_alloc(av_bsf_get_by_name("aac_adtstoasc"), &encoder->absf);
avcodec_parameters_from_context(encoder->absf->par_in, encoder->audio);
av_bsf_init(encoder->absf);
#else
encoder->absf = av_bitstream_filter_init("aac_adtstoasc");
#endif
}
#ifdef FFMPEG_USE_CODECPAR
avcodec_parameters_from_context(encoder->audioStream->codecpar, encoder->audio);
#endif
}
#ifdef FFMPEG_USE_CODECPAR
encoder->videoStream = avformat_new_stream(encoder->context, NULL);
encoder->video = avcodec_alloc_context3(vcodec);
#else
encoder->videoStream = avformat_new_stream(encoder->context, vcodec);
encoder->video = encoder->videoStream->codec;
#endif
encoder->video->bit_rate = encoder->videoBitrate;
encoder->video->width = encoder->width;
encoder->video->height = encoder->height;
encoder->video->time_base = (AVRational) { VIDEO_TOTAL_LENGTH, GBA_ARM7TDMI_FREQUENCY };
encoder->video->pix_fmt = encoder->pixFormat;
encoder->video->gop_size = 60;
encoder->video->max_b_frames = 3;
if (encoder->context->oformat->flags & AVFMT_GLOBALHEADER) {
#ifdef AV_CODEC_FLAG_GLOBAL_HEADER
encoder->video->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
#else
encoder->video->flags |= CODEC_FLAG_GLOBAL_HEADER;
#endif
}
if (encoder->video->codec->id == AV_CODEC_ID_H264 &&
(strcasecmp(encoder->containerFormat, "mp4") ||
strcasecmp(encoder->containerFormat, "m4v") ||
strcasecmp(encoder->containerFormat, "mov"))) {
// QuickTime and a few other things require YUV420
encoder->video->pix_fmt = AV_PIX_FMT_YUV420P;
}
if (strcmp(vcodec->name, "libx264") == 0) {
// Try to adaptively figure out when you can use a slower encoder
if (encoder->width * encoder->height > 1000000) {
av_opt_set(encoder->video->priv_data, "preset", "superfast", 0);
} else if (encoder->width * encoder->height > 500000) {
av_opt_set(encoder->video->priv_data, "preset", "veryfast", 0);
} else {
av_opt_set(encoder->video->priv_data, "preset", "faster", 0);
}
if (encoder->videoBitrate == 0) {
av_opt_set(encoder->video->priv_data, "crf", "0", 0);
encoder->video->pix_fmt = AV_PIX_FMT_YUV444P;
}
}
avcodec_open2(encoder->video, vcodec, 0);
#if LIBAVCODEC_VERSION_MAJOR >= 55
encoder->videoFrame = av_frame_alloc();
#else
encoder->videoFrame = avcodec_alloc_frame();
#endif
encoder->videoFrame->format = encoder->video->pix_fmt;
encoder->videoFrame->width = encoder->video->width;
encoder->videoFrame->height = encoder->video->height;
encoder->videoFrame->pts = 0;
_ffmpegSetVideoDimensions(&encoder->d, encoder->iwidth, encoder->iheight);
av_image_alloc(encoder->videoFrame->data, encoder->videoFrame->linesize, encoder->video->width, encoder->video->height, encoder->video->pix_fmt, 32);
#ifdef FFMPEG_USE_CODECPAR
avcodec_parameters_from_context(encoder->videoStream->codecpar, encoder->video);
#endif
if (avio_open(&encoder->context->pb, outfile, AVIO_FLAG_WRITE) < 0) {
return false;
}
return avformat_write_header(encoder->context, 0) >= 0;
}
int audio_decode_frame(VideoState *is, double *pts_ptr) {
int len1, data_size = 0, n;
AVPacket *pkt = &is->audio_pkt;
double pts;
uint8_t *out[] = {is->audio_buf};
int64_t wanted_channel_layout = 0;
wanted_channel_layout = is->audio_st->codec->channel_layout;
SwrContext* t_audio_conv = swr_alloc_set_opts(NULL,
wanted_channel_layout,AV_SAMPLE_FMT_S16,is->audio_st->codec->sample_rate,
wanted_channel_layout,is->audio_st->codec->sample_fmt, is->audio_st->codec->sample_rate,
0,NULL);
swr_init(t_audio_conv);
for(;;) {
while(is->audio_pkt_size > 0)
{
int got_frame = 0;
int t_audio_size= 0;
len1 = avcodec_decode_audio4(is->audio_st->codec, &is->audio_frame, &got_frame, pkt);
if(len1 < 0) {
/* if error, skip frame */
is->audio_pkt_size = 0;
break;
}
if (got_frame)
{
int size1 = av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
int len = swr_convert(t_audio_conv,
out, (MAX_AUDIO_FRAME_SIZE * 3) / 2/is->audio_st->codec->channels/size1,
(const uint8_t **)is->audio_frame.extended_data, is->audio_frame.nb_samples);
t_audio_size = len * is->audio_st->codec->channels * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
//av_packet_unref(pkt);
swr_free(&t_audio_conv);
//return len;
}
is->audio_pkt_data += len1;
is->audio_pkt_size -= len1;
//if(data_size <= 0) {
if(t_audio_size <= 0) {
/* No data yet, get more frames */
continue;
}
pts = is->audio_clock;
*pts_ptr = pts;
n = 2*is->audio_st->codec->channels;
is->audio_clock += (double)data_size /
(double)(n * is->audio_st->codec->sample_rate);
/* We have data, return it and come back for more later */
//return data_size;
return t_audio_size;
}
if(pkt->data)
av_free_packet(pkt);
if(is->quit) {
return -1;
}
/* next packet */
if(packet_queue_get(&is->audioq, pkt, 1) < 0) {
return -1;
}
is->audio_pkt_data = pkt->data;
is->audio_pkt_size = pkt->size;
/* if update, update the audio clock w/pts */
if(pkt->pts != AV_NOPTS_VALUE) {
is->audio_clock = av_q2d(is->audio_st->time_base)*pkt->pts;
}
}
//
// for(;;) {
// while(is->audio_pkt_size > 0) {
// int got_frame;
// len1 = avcodec_decode_audio4(is->audio_st->codec, &is->audio_frame, &got_frame, pkt);
// if(len1 < 0) {
// /* if error, skip frame */
// is->audio_pkt_size = 0;
// break;
// }
// if (got_frame)
// {
// data_size =
// av_samples_get_buffer_size
// (
// NULL,
// is->audio_st->codec->channels,
// is->audio_frame.nb_samples,
// is->audio_st->codec->sample_fmt,
// 1
// );
// memcpy(is->audio_buf, is->audio_frame.data[0], data_size);
// }
// is->audio_pkt_data += len1;
// is->audio_pkt_size -= len1;
// if(data_size <= 0) {
// /* No data yet, get more frames */
// continue;
// }
// pts = is->audio_clock;
// *pts_ptr = pts;
// n = 2 * is->audio_st->codec->channels;
// is->audio_clock += (double)data_size /
// (double)(n * is->audio_st->codec->sample_rate);
//
// /* We have data, return it and come back for more later */
// return data_size;
// }
// if(pkt->data)
// av_free_packet(pkt);
//
// if(is->quit) {
// return -1;
// }
// /* next packet */
// if(packet_queue_get(&is->audioq, pkt, 1) < 0) {
// return -1;
// }
// is->audio_pkt_data = pkt->data;
// is->audio_pkt_size = pkt->size;
// /* if update, update the audio clock w/pts */
// if(pkt->pts != AV_NOPTS_VALUE) {
// is->audio_clock = av_q2d(is->audio_st->time_base)*pkt->pts;
// }
//
// }
return 0;
}
/*
*TODO: broken sample rate(AAC), see mplayer
*/
bool AudioResamplerFF::prepare()
{
DPTR_D(AudioResamplerFF);
if (!d.in_format.isValid()) {
qWarning("src audio parameters 'channel layout(or channels), sample rate and sample format must be set before initialize resampler");
return false;
}
//TODO: also in do this statistics
if (!d.in_format.channels()) {
if (!d.in_format.channelLayoutFFmpeg()) { //FIXME: already return
d.in_format.setChannels(2);
d.in_format.setChannelLayoutFFmpeg(av_get_default_channel_layout(d.in_format.channels())); //from mplayer2
qWarning("both channels and channel layout are not available, assume channels=%d, channel layout=%lld", d.in_format.channels(), d.in_format.channelLayoutFFmpeg());
} else {
d.in_format.setChannels(av_get_channel_layout_nb_channels(d.in_format.channelLayoutFFmpeg()));
}
}
if (!d.in_format.channels())
d.in_format.setChannels(2); //TODO: why av_get_channel_layout_nb_channels() may return 0?
if (!d.in_format.channelLayoutFFmpeg()) {
qWarning("channel layout not available, use default layout");
d.in_format.setChannelLayoutFFmpeg(av_get_default_channel_layout(d.in_format.channels()));
}
if (!d.out_format.channels()) {
if (d.out_format.channelLayoutFFmpeg()) {
d.out_format.setChannels(av_get_channel_layout_nb_channels(d.out_format.channelLayoutFFmpeg()));
} else {
d.out_format.setChannels(d.in_format.channels());
d.out_format.setChannelLayoutFFmpeg(d.in_format.channelLayoutFFmpeg());
}
}
if (d.out_format.channelLayout() == AudioFormat::ChannelLayout_Unsupported) {
d.out_format.setChannels(d.in_format.channels());
d.out_format.setChannelLayoutFFmpeg(d.in_format.channelLayoutFFmpeg());
}
//now we have out channels
if (!d.out_format.channelLayoutFFmpeg())
d.out_format.setChannelLayoutFFmpeg(av_get_default_channel_layout(d.out_format.channels()));
if (!d.out_format.sampleRate())
d.out_format.setSampleRate(inAudioFormat().sampleRate());
if (d.speed <= 0)
d.speed = 1.0;
//DO NOT set sample rate here, we should keep the original and multiply 1/speed when needed
//if (d.speed != 1.0)
// d.out_format.setSampleRate(int(qreal(d.out_format.sampleFormat())/d.speed));
qDebug("swr speed=%.2f", d.speed);
//d.in_planes = av_sample_fmt_is_planar((enum AVSampleFormat)d.in_sample_format) ? d.in_channels : 1;
//d.out_planes = av_sample_fmt_is_planar((enum AVSampleFormat)d.out_sample_format) ? d.out_channels : 1;
if (d.context)
swr_free(&d.context); //TODO: if no free(of cause free is required), why channel mapping and layout not work if change from left to stero?
//If use swr_alloc() need to set the parameters (av_opt_set_xxx() manually or with swr_alloc_set_opts()) before calling swr_init()
d.context = swr_alloc_set_opts(d.context
, d.out_format.channelLayoutFFmpeg()
, (enum AVSampleFormat)outAudioFormat().sampleFormatFFmpeg()
, qreal(outAudioFormat().sampleRate())/d.speed
, d.in_format.channelLayoutFFmpeg()
, (enum AVSampleFormat)inAudioFormat().sampleFormatFFmpeg()
, inAudioFormat().sampleRate()
, 0 /*log_offset*/, 0 /*log_ctx*/);
/*
av_opt_set_int(d.context, "in_channel_layout", d.in_channel_layout, 0);
av_opt_set_int(d.context, "in_sample_rate", d.in_format.sampleRate(), 0);
av_opt_set_sample_fmt(d.context, "in_sample_fmt", (enum AVSampleFormat)in_format.sampleFormatFFmpeg(), 0);
av_opt_set_int(d.context, "out_channel_layout", d.out_channel_layout, 0);
av_opt_set_int(d.context, "out_sample_rate", d.out_format.sampleRate(), 0);
av_opt_set_sample_fmt(d.context, "out_sample_fmt", (enum AVSampleFormat)out_format.sampleFormatFFmpeg(), 0);
*/
qDebug("out: {cl: %lld, fmt: %s, freq: %d}"
, d.out_format.channelLayoutFFmpeg()
, qPrintable(d.out_format.sampleFormatName())
, d.out_format.sampleRate());
qDebug("in {cl: %lld, fmt: %s, freq: %d}"
, d.in_format.channelLayoutFFmpeg()
, qPrintable(d.in_format.sampleFormatName())
, d.in_format.sampleRate());
if (!d.context) {
qWarning("Allocat swr context failed!");
return false;
}
//avresample 0.0.2(FFmpeg 0.11)~1.0.1(FFmpeg 1.1) has no channel mapping. but has remix matrix, so does swresample
//TODO: why crash if use channel mapping for L or R?
#if QTAV_HAVE(SWR_AVR_MAP) //LIBAVRESAMPLE_VERSION_INT < AV_VERSION_INT(1, 1, 0)
bool remix = false;
int in_c = d.in_format.channels();
int out_c = d.out_format.channels();
/*
* matrix[i + stride * o] is the weight of input channel i in output channel o.
*/
double *matrix = 0;
if (d.out_format.channelLayout() == AudioFormat::ChannelLayout_Left) {
remix = true;
matrix = (double*)calloc(in_c*out_c, sizeof(double));
for (int o = 0; o < out_c; ++o) {
matrix[0 + in_c * o] = 1;
}
}
if (d.out_format.channelLayout() == AudioFormat::ChannelLayout_Right) {
remix = true;
matrix = (double*)calloc(in_c*out_c, sizeof(double));
for (int o = 0; o < out_c; ++o) {
matrix[1 + in_c * o] = 1;
}
}
if (!remix && in_c < out_c) {
remix = true;
//double matrix[in_c*out_c]; //C99, VLA
matrix = (double*)calloc(in_c*out_c, sizeof(double));
for (int i = 0, o = 0; o < out_c; ++o) {
matrix[i + in_c * o] = 1;
i = (i + i)%in_c;
}
}
if (remix && matrix) {
avresample_set_matrix(d.context, matrix, in_c);
free(matrix);
}
#else
bool use_channel_map = false;
if (d.out_format.channelLayout() == AudioFormat::ChannelLayout_Left) {
use_channel_map = true;
memset(d.channel_map, 0, sizeof(d.channel_map));
for (int i = 0; i < d.out_format.channels(); ++i) {
d.channel_map[i] = 0;
}
}
if (d.out_format.channelLayout() == AudioFormat::ChannelLayout_Right) {
use_channel_map = true;
memset(d.channel_map, 0, sizeof(d.channel_map));
for (int i = 0; i < d.out_format.channels(); ++i) {
d.channel_map[i] = 1;
}
}
if (!use_channel_map && d.in_format.channels() < d.out_format.channels()) {
use_channel_map = true;
memset(d.channel_map, 0, sizeof(d.channel_map));
for (int i = 0; i < d.out_format.channels(); ++i) {
d.channel_map[i] = i % d.in_format.channels();
}
}
if (use_channel_map) {
av_opt_set_int(d.context, "icl", d.out_format.channelLayoutFFmpeg(), 0);
//TODO: why crash if layout is mono and set uch(i.e. always the next line)
av_opt_set_int(d.context, "uch", d.out_format.channels(), 0);
swr_set_channel_mapping(d.context, d.channel_map);
}
#endif //QTAV_HAVE(SWR_AVR_MAP)
int ret = swr_init(d.context);
if (ret < 0) {
qWarning("swr_init failed: %s", av_err2str(ret));
swr_free(&d.context);
return false;
}
return true;
}
int decode_audio(AVCodecContext *ctx, queue_t *qa)
{
static struct SwrContext *swr_ctx;
static int64_t src_layout;
static int src_freq;
static int src_channels;
static enum AVSampleFormat src_fmt = -1;
static AVFrame *aFrame;
AVPacket pkt;
AVPacket pkt_tmp;
int64_t dec_channel_layout;
int len, len2;
int got_frame;
int data_size;
if( astream.count > 192000*2)
return -1;
if( get_packet(qa, &pkt) == 0 )
return 0;
// __asm__("int3");
if (!aFrame)
{
if (!(aFrame = avcodec_alloc_frame()))
return -1;
} else
avcodec_get_frame_defaults(aFrame);
pkt_tmp = pkt;
while(pkt_tmp.size > 0)
{
data_size = 192000;
// len = avcodec_decode_audio3(ctx,(int16_t*)decoder_buffer,
// &data_size, &pkt_tmp);
got_frame = 0;
len = avcodec_decode_audio4(ctx, aFrame, &got_frame, &pkt_tmp);
if(len >= 0 && got_frame)
{
char *samples;
int ch, plane_size;
int planar = av_sample_fmt_is_planar(ctx->sample_fmt);
int data_size = av_samples_get_buffer_size(&plane_size, ctx->channels,
aFrame->nb_samples,
ctx->sample_fmt, 1);
// if(audio_base == -1.0)
// {
// if (pkt.pts != AV_NOPTS_VALUE)
// audio_base = get_audio_base() * pkt.pts;
// printf("audio base %f\n", audio_base);
// };
pkt_tmp.data += len;
pkt_tmp.size -= len;
dec_channel_layout =
(aFrame->channel_layout && aFrame->channels == av_get_channel_layout_nb_channels(aFrame->channel_layout)) ?
aFrame->channel_layout : av_get_default_channel_layout(aFrame->channels);
if (aFrame->format != src_fmt ||
dec_channel_layout != src_layout ||
aFrame->sample_rate != src_freq ||
!swr_ctx)
{
swr_free(&swr_ctx);
swr_ctx = swr_alloc_set_opts(NULL, AV_CH_LAYOUT_STEREO, AV_SAMPLE_FMT_S16,
aFrame->sample_rate, dec_channel_layout,aFrame->format,
aFrame->sample_rate, 0, NULL);
if (!swr_ctx || swr_init(swr_ctx) < 0)
{
printf("Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
aFrame->sample_rate, av_get_sample_fmt_name(aFrame->format), (int)aFrame->channels,
aFrame->sample_rate, av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 2);
break;
}
src_layout = dec_channel_layout;
src_channels = aFrame->channels;
src_freq = aFrame->sample_rate;
src_fmt = aFrame->format;
};
if (swr_ctx)
{
const uint8_t **in = (const uint8_t **)aFrame->extended_data;
uint8_t *out[] = {decoder_buffer};
int out_count = 192000 * 3 / 2 / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
len2 = swr_convert(swr_ctx, out, out_count, in, aFrame->nb_samples);
if (len2 < 0) {
printf("swr_convert() failed\n");
break;
}
if (len2 == out_count) {
printf("warning: audio buffer is probably too small\n");
swr_init(swr_ctx);
}
data_size = len2 * 2 * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
mutex_lock(&astream.lock);
samples = astream.buffer+astream.count;
memcpy(samples, decoder_buffer, data_size);
/*
memcpy(samples, aFrame->extended_data[0], plane_size);
if (planar && ctx->channels > 1)
{
uint8_t *out = ((uint8_t *)samples) + plane_size;
for (ch = 1; ch < ctx->channels; ch++)
{
memcpy(out, aFrame->extended_data[ch], plane_size);
out += plane_size;
}
}
*/
astream.count += data_size;
mutex_unlock(&astream.lock);
};
}
else pkt_tmp.size = 0;
}
av_free_packet(&pkt);
return 1;
};
static int
ca_create_swrctx(WAVEFORMATEX *w) {
struct RTSPConf *rtspconf = rtspconf_global();
int bufreq, samples;
//
if(swrctx != NULL)
swr_free(&swrctx);
if(audio_buf != NULL)
free(audio_buf);
//
ga_error("CoreAudio: create swr context - format[%x] freq[%d] channels[%d]\n",
w->wFormatTag, w->nSamplesPerSec, w->nChannels);
//
swrctx = swr_alloc_set_opts(NULL,
rtspconf->audio_device_channel_layout,
rtspconf->audio_device_format,
rtspconf->audio_samplerate,
CA2SWR_chlayout(w->nChannels),
CA2SWR_format(w),
w->nSamplesPerSec, 0, NULL);
if(swrctx == NULL) {
ga_error("CoreAudio: cannot create resample context.\n");
return -1;
} else {
ga_error("CoreAudio: resample context (%x,%d,%d) -> (%x,%d,%d)\n",
(int) CA2SWR_chlayout(w->nChannels),
(int) CA2SWR_format(w),
(int) w->nSamplesPerSec,
(int) rtspconf->audio_device_channel_layout,
(int) rtspconf->audio_device_format,
(int) rtspconf->audio_samplerate);
}
if(swr_init(swrctx) < 0) {
swr_free(&swrctx);
swrctx = NULL;
ga_error("CoreAudio: resample context init failed.\n");
return -1;
}
// allocate buffer?
ga_samplerate = rtspconf->audio_samplerate;
ga_channels = av_get_channel_layout_nb_channels(rtspconf->audio_device_channel_layout);
ca_samplerate = w->nSamplesPerSec;
ca_bytes_per_sample = w->wBitsPerSample/8;
samples = av_rescale_rnd(CA_MAX_SAMPLES,
rtspconf->audio_samplerate, w->nSamplesPerSec, AV_ROUND_UP);
bufreq = av_samples_get_buffer_size(NULL,
rtspconf->audio_channels, samples*2,
rtspconf->audio_device_format,
1/*no-alignment*/);
if((audio_buf = (unsigned char *) malloc(bufreq)) == NULL) {
ga_error("CoreAudio: cannot allocate resample memory.\n");
return -1;
}
if(audio_source_setup(bufreq, rtspconf->audio_samplerate,
16/* depends on format */,
rtspconf->audio_channels) < 0) {
ga_error("CoreAudio: audio source setup failed.\n");
return -1;
}
ga_error("CoreAudio: max %d samples with %d byte(s) resample buffer allocated.\n",
samples, bufreq);
//
return 0;
}
bool CAEEncoderFFmpeg::Initialize(AEAudioFormat &format, bool allow_planar_input)
{
Reset();
bool ac3 = CSettings::Get().GetBool("audiooutput.ac3passthrough");
AVCodec *codec = NULL;
#if 0
/* the DCA encoder is currently useless for transcode, it creates a 196 kHz DTS-HD like mongrel which is useless for SPDIF */
bool dts = CSettings::Get().GetBool("audiooutput.dtspassthrough");
if (dts && (!ac3 || g_advancedSettings.m_audioTranscodeTo.Equals("dts")))
{
m_CodecName = "DTS";
m_CodecID = AV_CODEC_ID_DTS;
m_PackFunc = &CAEPackIEC61937::PackDTS_1024;
m_BitRate = DTS_ENCODE_BITRATE;
codec = avcodec_find_encoder(m_CodecID);
}
#endif
/* fallback to ac3 if we support it, we might not have DTS support */
if (!codec && ac3)
{
m_CodecName = "AC3";
m_CodecID = AV_CODEC_ID_AC3;
m_PackFunc = &CAEPackIEC61937::PackAC3;
m_BitRate = AC3_ENCODE_BITRATE;
codec = avcodec_find_encoder(m_CodecID);
}
/* check we got the codec */
if (!codec)
return false;
m_CodecCtx = avcodec_alloc_context3(codec);
m_CodecCtx->bit_rate = m_BitRate;
m_CodecCtx->sample_rate = format.m_sampleRate;
m_CodecCtx->channel_layout = AV_CH_LAYOUT_5POINT1_BACK;
/* select a suitable data format */
if (codec->sample_fmts)
{
bool hasFloat = false;
bool hasDouble = false;
bool hasS32 = false;
bool hasS16 = false;
bool hasU8 = false;
bool hasFloatP = false;
bool hasUnknownFormat = false;
for(int i = 0; codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; ++i)
{
switch (codec->sample_fmts[i])
{
case AV_SAMPLE_FMT_FLT: hasFloat = true; break;
case AV_SAMPLE_FMT_DBL: hasDouble = true; break;
case AV_SAMPLE_FMT_S32: hasS32 = true; break;
case AV_SAMPLE_FMT_S16: hasS16 = true; break;
case AV_SAMPLE_FMT_U8 : hasU8 = true; break;
case AV_SAMPLE_FMT_FLTP:
if (allow_planar_input)
hasFloatP = true;
else
hasUnknownFormat = true;
break;
case AV_SAMPLE_FMT_NONE: return false;
default: hasUnknownFormat = true; break;
}
}
if (hasFloat)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_FLT;
format.m_dataFormat = AE_FMT_FLOAT;
}
else if (hasFloatP)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP;
format.m_dataFormat = AE_FMT_FLOATP;
}
else if (hasDouble)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_DBL;
format.m_dataFormat = AE_FMT_DOUBLE;
}
else if (hasS32)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_S32;
format.m_dataFormat = AE_FMT_S32NE;
}
else if (hasS16)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
format.m_dataFormat = AE_FMT_S16NE;
}
else if (hasU8)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_U8;
format.m_dataFormat = AE_FMT_U8;
}
else if (hasUnknownFormat)
{
m_CodecCtx->sample_fmt = codec->sample_fmts[0];
format.m_dataFormat = AE_FMT_FLOAT;
m_NeedConversion = true;
CLog::Log(LOGNOTICE, "CAEEncoderFFmpeg::Initialize - Unknown audio format, it will be resampled.");
}
else
{
CLog::Log(LOGERROR, "CAEEncoderFFmpeg::Initialize - Unable to find a suitable data format for the codec (%s)", m_CodecName.c_str());
return false;
}
}
m_CodecCtx->channels = BuildChannelLayout(m_CodecCtx->channel_layout, m_Layout);
/* open the codec */
if (avcodec_open2(m_CodecCtx, codec, NULL))
{
av_freep(&m_CodecCtx);
return false;
}
format.m_frames = m_CodecCtx->frame_size;
format.m_frameSamples = m_CodecCtx->frame_size * m_CodecCtx->channels;
format.m_frameSize = m_CodecCtx->channels * (CAEUtil::DataFormatToBits(format.m_dataFormat) >> 3);
format.m_channelLayout = m_Layout;
m_CurrentFormat = format;
m_NeededFrames = format.m_frames;
m_OutputSize = m_PackFunc(NULL, 0, m_Buffer);
m_OutputRatio = (double)m_NeededFrames / m_OutputSize;
m_SampleRateMul = 1.0 / (double)m_CodecCtx->sample_rate;
if (m_NeedConversion)
{
m_SwrCtx = swr_alloc_set_opts(NULL,
m_CodecCtx->channel_layout, m_CodecCtx->sample_fmt, m_CodecCtx->sample_rate,
m_CodecCtx->channel_layout, AV_SAMPLE_FMT_FLT, m_CodecCtx->sample_rate,
0, NULL);
if (!m_SwrCtx || swr_init(m_SwrCtx) < 0)
{
CLog::Log(LOGERROR, "CAEEncoderFFmpeg::Initialize - Failed to initialise resampler.");
return false;
}
}
CLog::Log(LOGNOTICE, "CAEEncoderFFmpeg::Initialize - %s encoder ready", m_CodecName.c_str());
return true;
}
void BE_ST_InitAudio(void)
{
g_sdlAudioSubsystemUp = false;
g_sdlEmulatedOPLChipReady = false;
int inSampleRate = BE_Cross_GetSelectedGameVerSampleRate();
bool doDigitized = (inSampleRate != 0);
if (!doDigitized)
inSampleRate = OPL_SAMPLE_RATE;
if (g_refKeenCfg.sndSubSystem)
{
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0)
{
BE_Cross_LogMessage(BE_LOG_MSG_WARNING, "SDL audio system initialization failed,\n%s\n", SDL_GetError());
}
else
{
g_sdlAudioSpec.freq = g_refKeenCfg.sndSampleRate;
#ifdef MIXER_SAMPLE_FORMAT_FLOAT
g_sdlAudioSpec.format = AUDIO_F32SYS;
#elif (defined MIXER_SAMPLE_FORMAT_SINT16)
g_sdlAudioSpec.format = AUDIO_S16SYS;
#endif
g_sdlAudioSpec.channels = 1;
// Should be some power-of-two roughly proportional to the sample rate; Using 1024 for 48000Hz.
for (g_sdlAudioSpec.samples = 1; g_sdlAudioSpec.samples < g_refKeenCfg.sndSampleRate/64; g_sdlAudioSpec.samples *= 2)
{
}
if (doDigitized)
g_sdlAudioSpec.callback = (g_refKeenCfg.sndSampleRate == inSampleRate) ? BEL_ST_Simple_DigiCallBack : BEL_ST_Resampling_DigiCallBack;
else
g_sdlAudioSpec.callback = ((g_refKeenCfg.sndSampleRate == inSampleRate) || !g_refKeenCfg.oplEmulation) ? BEL_ST_Simple_EmuCallBack : BEL_ST_Resampling_EmuCallBack;
g_sdlAudioSpec.userdata = NULL;
if (SDL_OpenAudio(&g_sdlAudioSpec, NULL))
{
BE_Cross_LogMessage(BE_LOG_MSG_WARNING, "Cannot open SDL audio device,\n%s\n", SDL_GetError());
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
else
{
#ifdef REFKEEN_CONFIG_THREADS
g_sdlCallbackMutex = SDL_CreateMutex();
if (!g_sdlCallbackMutex)
{
BE_Cross_LogMessage(BE_LOG_MSG_ERROR, "Cannot create recursive mutex for SDL audio callback,\n%s\nClosing SDL audio subsystem\n", SDL_GetError());
SDL_CloseAudio();
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
else
#endif
{
BE_Cross_LogMessage(BE_LOG_MSG_NORMAL, "Audio subsystem initialized, requested spec: freq %d, format %u, channels %d, samples %u\n", (int)g_sdlAudioSpec.freq, (unsigned int)g_sdlAudioSpec.format, (int)g_sdlAudioSpec.channels, (unsigned int)g_sdlAudioSpec.samples);
g_sdlAudioSubsystemUp = true;
}
}
}
}
// If the audio subsystem is off, let us simulate a byte rate
// of 1000Hz (same as SDL_GetTicks() time units)
if (!g_sdlAudioSubsystemUp)
{
g_sdlAudioSpec.freq = doDigitized ? inSampleRate : (NUM_OF_BYTES_FOR_SOUND_CALLBACK_WITH_DISABLED_SUBSYSTEM / sizeof(BE_ST_SndSample_T));
g_sdlAudioSpec.callback = doDigitized ? BEL_ST_Resampling_DigiCallBack : BEL_ST_Resampling_EmuCallBack;
return;
}
if (g_refKeenCfg.oplEmulation)
{
YM3812Init(1, 3579545, OPL_SAMPLE_RATE);
g_sdlEmulatedOPLChipReady = true;
}
if ((doDigitized || g_sdlEmulatedOPLChipReady) && (g_sdlAudioSpec.freq != inSampleRate))
{
// Should allocate this first, for g_sdlSrcData.data_in
g_sdlMiscOutNumOfSamples = 2*g_sdlAudioSpec.samples;
g_sdlMiscOutSamples = (BE_ST_SndSample_T *)malloc(sizeof(BE_ST_SndSample_T) * g_sdlMiscOutNumOfSamples);
if (g_sdlMiscOutSamples == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: Out of memory! (Failed to allocate g_sdlMiscOutSamples.)");
#ifndef REFKEEN_RESAMPLER_NONE
if (g_refKeenCfg.useResampler)
{
#if (!defined REFKEEN_RESAMPLER_LIBRESAMPLE) && (!defined REFKEEN_RESAMPLER_LIBAVCODEC)
char errMsg[160];
#endif
#if (defined REFKEEN_RESAMPLER_LIBSWRESAMPLE)
g_sdlSwrContext = swr_alloc_set_opts(
NULL, // allocating a new context
AV_CH_LAYOUT_MONO, // out channels layout
AV_SAMPLE_FMT_S16, // out format
g_sdlAudioSpec.freq, // out rate
AV_CH_LAYOUT_MONO, // in channels layout
AV_SAMPLE_FMT_S16, // in format
inSampleRate, // in rate
0,
NULL
);
if (g_sdlSwrContext == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: swr_alloc_set_opts failed!");
int error = swr_init(g_sdlSwrContext);
if (error != 0)
{
// av_err2str requires libavutil/libavutil-ffmpeg, so don't convert code to string
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: swr_init failed! Error code: %d", error);
BE_ST_ExitWithErrorMsg(errMsg);
}
#elif (defined REFKEEN_RESAMPLER_LIBAVRESAMPLE)
g_sdlAvAudioResampleContext = avresample_alloc_context();
if (g_sdlAvAudioResampleContext == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: avresample_alloc_context failed!");
av_opt_set_int(g_sdlAvAudioResampleContext, "in_channel_layout", AV_CH_LAYOUT_MONO, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "out_channel_layout", AV_CH_LAYOUT_MONO, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "in_sample_rate", inSampleRate, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "out_sample_rate", g_sdlAudioSpec.freq, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int(g_sdlAvAudioResampleContext, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
int error = avresample_open(g_sdlAvAudioResampleContext);
if (error != 0)
{
// av_err2str requires libavutil/libavutil-ffmpeg, so don't convert code to string
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: swr_init failed! Error code: %d", error);
BE_ST_ExitWithErrorMsg(errMsg);
}
#elif (defined REFKEEN_RESAMPLER_LIBAVCODEC)
avcodec_register_all();
g_sdlAvResampleContext = av_resample_init(
g_sdlAudioSpec.freq, // out rate
inSampleRate, // in rate
16, // filter length
10, // phase count
0, // linear FIR filter
1.0 // cutoff frequency
);
if (g_sdlAvResampleContext == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: av_resample_init failed!");
#elif (defined REFKEEN_RESAMPLER_LIBRESAMPLE)
g_sdlResampleFactor = (double)g_sdlAudioSpec.freq/inSampleRate;
g_sdlResampleHandle = resample_open(0, g_sdlResampleFactor, g_sdlResampleFactor);
if (g_sdlResampleHandle == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: resample_open failed!");
#elif (defined REFKEEN_RESAMPLER_LIBSOXR)
soxr_io_spec_t io_spec = soxr_io_spec(SOXR_INT16, SOXR_INT16);
soxr_quality_spec_t q_spec = soxr_quality_spec(SOXR_LQ, 0); // Default quality spec adds an audible latency for resampling to 8000Hz
soxr_error_t error;
g_sdlSoxr = soxr_create(
inSampleRate, // in rate
g_sdlAudioSpec.freq, // out rate
1, // channels
&error,
&io_spec,
&q_spec,
NULL // runtime spec
);
if (g_sdlSoxr == NULL)
{
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: soxr_create failed!\nError: %s", soxr_strerror(error));
BE_ST_ExitWithErrorMsg(errMsg);
}
#elif (defined REFKEEN_RESAMPLER_LIBSPEEXDSP)
int error;
g_sdlSpeexResamplerState = speex_resampler_init(
1, // channels
inSampleRate, // in rate
g_sdlAudioSpec.freq, // out rate
0, // quality in the range 0-10 (10 is higher)
&error
);
if (g_sdlSpeexResamplerState == NULL)
{
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: speex_resampler_init failed! Error code: %d\nError: %s", error, speex_resampler_strerror(error));
BE_ST_ExitWithErrorMsg(errMsg);
}
#elif (defined REFKEEN_RESAMPLER_LIBSAMPLERATE)
int error;
g_sdlSrcResampler = src_new(SRC_SINC_FASTEST, 1, &error);
if (g_sdlSrcResampler == NULL)
{
snprintf(errMsg, sizeof(errMsg), "BE_ST_InitAudio: src_new failed!\nError code: %d", error);
BE_ST_ExitWithErrorMsg(errMsg);
}
g_sdlSrcData.data_in = doDigitized ? g_sdlMiscOutSamples : g_sdlALOutSamples;
g_sdlSrcData.src_ratio = (double)g_sdlAudioSpec.freq / inSampleRate;
#endif
}
else
#endif // REFKEEN_RESAMPLER_NONE
{
// The sum of all entries should be g_sdlAudioSpec.freq,
// "uniformly" distributed over g_sdlALSampleRateConvTable
g_sdlSampleRateConvTable = (int *)malloc(sizeof(int) * inSampleRate);
if (g_sdlSampleRateConvTable == NULL)
BE_ST_ExitWithErrorMsg("BE_ST_InitAudio: Failed to allocate memory for sample rate conversion!");
g_sdlSampleRateConvTableSize = inSampleRate;
for (int i = 0; i < inSampleRate; ++i)
{
// Using uint64_t cause an overflow is possible
g_sdlSampleRateConvTable[i] = ((uint64_t)(i+1)*(uint64_t)g_sdlAudioSpec.freq/inSampleRate)-(uint64_t)i*(uint64_t)g_sdlAudioSpec.freq/inSampleRate;
}
g_sdlSampleRateConvCurrIndex = 0;
g_sdlSampleRateConvCounter = 0;
}
}
}
void THMovie::play(const SDL_Rect &destination_rect, int iChannel)
{
m_destination_rect = SDL_Rect{ destination_rect.x, destination_rect.y, destination_rect.w, destination_rect.h };
if(!m_pRenderer)
{
m_sLastError = std::string("Cannot play before setting the renderer");
return;
}
m_pVideoQueue = new THAVPacketQueue();
m_pMoviePictureBuffer->reset();
m_pMoviePictureBuffer->allocate(m_pRenderer, m_pVideoCodecContext->width, m_pVideoCodecContext->height);
m_pAudioPacket = nullptr;
m_iAudioPacketSize = 0;
m_pbAudioPacketData = nullptr;
m_iAudioBufferSize = 0;
m_iAudioBufferIndex = 0;
m_iAudioBufferMaxSize = 0;
m_pAudioQueue = new THAVPacketQueue();
m_iCurSyncPts = 0;
m_iCurSyncPtsSystemTime = SDL_GetTicks();
if(m_iAudioStream >= 0)
{
Mix_QuerySpec(&m_iMixerFrequency, nullptr, &m_iMixerChannels);
#ifdef CORSIX_TH_USE_FFMPEG
m_pAudioResampleContext = swr_alloc_set_opts(
m_pAudioResampleContext,
m_iMixerChannels==1?AV_CH_LAYOUT_MONO:AV_CH_LAYOUT_STEREO,
AV_SAMPLE_FMT_S16,
m_iMixerFrequency,
m_pAudioCodecContext->channel_layout,
m_pAudioCodecContext->sample_fmt,
m_pAudioCodecContext->sample_rate,
0,
nullptr);
swr_init(m_pAudioResampleContext);
#elif defined(CORSIX_TH_USE_LIBAV)
m_pAudioResampleContext = avresample_alloc_context();
av_opt_set_int(m_pAudioResampleContext, "in_channel_layout", m_pAudioCodecContext->channel_layout, 0);
av_opt_set_int(m_pAudioResampleContext, "out_channel_layout", m_iMixerChannels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO, 0);
av_opt_set_int(m_pAudioResampleContext, "in_sample_rate", m_pAudioCodecContext->sample_rate, 0);
av_opt_set_int(m_pAudioResampleContext, "out_sample_rate", m_iMixerFrequency, 0);
av_opt_set_int(m_pAudioResampleContext, "in_sample_fmt", m_pAudioCodecContext->sample_fmt, 0);
av_opt_set_int(m_pAudioResampleContext, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
avresample_open(m_pAudioResampleContext);
#endif
m_pChunk = Mix_QuickLoad_RAW(m_pbChunkBuffer, ms_audioBufferSize);
m_iChannel = Mix_PlayChannel(iChannel, m_pChunk, -1);
if(m_iChannel < 0)
{
m_iChannel = -1;
m_sLastError = std::string(Mix_GetError());
}
else
{
Mix_RegisterEffect(m_iChannel, th_movie_audio_callback, nullptr, this);
}
}
m_pStreamThread = SDL_CreateThread(th_movie_stream_reader_thread, "Stream", this);
m_pVideoThread = SDL_CreateThread(th_movie_video_thread, "Video", this);
}
