static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
{
int samples = s->packet.frame_duration;
int redundancy = 0;
int redundancy_size, redundancy_pos;
int ret, i, consumed;
int delayed_samples = s->delayed_samples;
ret = opus_rc_init(&s->rc, data, size);
if (ret < 0)
return ret;
/* decode the silk frame */
if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
if (!avresample_is_open(s->avr)) {
ret = opus_init_resample(s);
if (ret < 0)
return ret;
}
samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
s->packet.stereo + 1,
silk_frame_duration_ms[s->packet.config]);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
return samples;
}
samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
s->packet.frame_duration,
(uint8_t**)s->silk_output,
sizeof(s->silk_buf[0]),
samples);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
return samples;
}
s->delayed_samples += s->packet.frame_duration - samples;
} else
ff_silk_flush(s->silk);
// decode redundancy information
consumed = opus_rc_tell(&s->rc);
if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
redundancy = opus_rc_p2model(&s->rc, 12);
else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
redundancy = 1;
if (redundancy) {
redundancy_pos = opus_rc_p2model(&s->rc, 1);
if (s->packet.mode == OPUS_MODE_HYBRID)
redundancy_size = opus_rc_unimodel(&s->rc, 256) + 2;
else
redundancy_size = size - (consumed + 7) / 8;
size -= redundancy_size;
if (size < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
return AVERROR_INVALIDDATA;
}
if (redundancy_pos) {
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
ff_celt_flush(s->celt);
}
}
/* decode the CELT frame */
if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
float *out_tmp[2] = { s->out[0], s->out[1] };
float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
out_tmp : s->celt_output;
int celt_output_samples = samples;
int delay_samples = av_audio_fifo_size(s->celt_delay);
if (delay_samples) {
if (s->packet.mode == OPUS_MODE_HYBRID) {
av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
delay_samples);
out_tmp[i] += delay_samples;
}
celt_output_samples -= delay_samples;
} else {
av_log(s->avctx, AV_LOG_WARNING,
"Spurious CELT delay samples present.\n");
av_audio_fifo_drain(s->celt_delay, delay_samples);
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_BUG;
}
}
opus_raw_init(&s->rc, data + size, size);
ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
s->packet.stereo + 1,
s->packet.frame_duration,
(s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
celt_band_end[s->packet.bandwidth]);
if (ret < 0)
return ret;
if (s->packet.mode == OPUS_MODE_HYBRID) {
int celt_delay = s->packet.frame_duration - celt_output_samples;
void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
s->celt_output[1] + celt_output_samples };
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i],
s->celt_output[i], 1.0,
celt_output_samples);
}
ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
if (ret < 0)
return ret;
}
} else
ff_celt_flush(s->celt);
if (s->redundancy_idx) {
for (i = 0; i < s->output_channels; i++)
opus_fade(s->out[i], s->out[i],
s->redundancy_output[i] + 120 + s->redundancy_idx,
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
s->redundancy_idx = 0;
}
if (redundancy) {
if (!redundancy_pos) {
ff_celt_flush(s->celt);
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
for (i = 0; i < s->output_channels; i++) {
opus_fade(s->out[i] + samples - 120 + delayed_samples,
s->out[i] + samples - 120 + delayed_samples,
s->redundancy_output[i] + 120,
ff_celt_window2, 120 - delayed_samples);
if (delayed_samples)
s->redundancy_idx = 120 - delayed_samples;
}
} else {
for (i = 0; i < s->output_channels; i++) {
memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
opus_fade(s->out[i] + 120 + delayed_samples,
s->redundancy_output[i] + 120,
s->out[i] + 120 + delayed_samples,
ff_celt_window2, 120);
}
}
}
return samples;
}
int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples)
{
if (!output)
return av_audio_fifo_drain(avr->out_fifo, nb_samples);
return av_audio_fifo_read(avr->out_fifo, (void**)output, nb_samples);
}
static int activate(AVFilterContext *ctx)
{
SidechainCompressContext *s = ctx->priv;
AVFrame *out = NULL, *in[2] = { NULL };
int ret, i, nb_samples;
double *dst;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
if ((ret = ff_inlink_consume_frame(ctx->inputs[0], &in[0])) > 0) {
av_audio_fifo_write(s->fifo[0], (void **)in[0]->extended_data,
in[0]->nb_samples);
av_frame_free(&in[0]);
}
if (ret < 0)
return ret;
if ((ret = ff_inlink_consume_frame(ctx->inputs[1], &in[1])) > 0) {
av_audio_fifo_write(s->fifo[1], (void **)in[1]->extended_data,
in[1]->nb_samples);
av_frame_free(&in[1]);
}
if (ret < 0)
return ret;
nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
if (nb_samples) {
out = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
if (!out)
return AVERROR(ENOMEM);
for (i = 0; i < 2; i++) {
in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
if (!in[i]) {
av_frame_free(&in[0]);
av_frame_free(&in[1]);
av_frame_free(&out);
return AVERROR(ENOMEM);
}
av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
}
dst = (double *)out->data[0];
out->pts = s->pts;
s->pts += nb_samples;
compressor(s, (double *)in[0]->data[0], dst,
(double *)in[1]->data[0], nb_samples,
s->level_in, s->level_sc,
ctx->inputs[0], ctx->inputs[1]);
av_frame_free(&in[0]);
av_frame_free(&in[1]);
ret = ff_filter_frame(ctx->outputs[0], out);
if (ret < 0)
return ret;
}
FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
FF_FILTER_FORWARD_STATUS(ctx->inputs[1], ctx->outputs[0]);
if (ff_outlink_frame_wanted(ctx->outputs[0])) {
if (!av_audio_fifo_size(s->fifo[0]))
ff_inlink_request_frame(ctx->inputs[0]);
if (!av_audio_fifo_size(s->fifo[1]))
ff_inlink_request_frame(ctx->inputs[1]);
}
return 0;
}
static int opus_decode_packet(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
OpusContext *c = avctx->priv_data;
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int coded_samples = 0;
int decoded_samples = INT_MAX;
int delayed_samples = 0;
int i, ret;
/* calculate the number of delayed samples */
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
s->out[0] =
s->out[1] = NULL;
delayed_samples = FFMAX(delayed_samples,
s->delayed_samples + av_audio_fifo_size(c->sync_buffers[i]));
}
/* decode the header of the first sub-packet to find out the sample count */
if (buf) {
OpusPacket *pkt = &c->streams[0].packet;
ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
return ret;
}
coded_samples += pkt->frame_count * pkt->frame_duration;
c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
}
frame->nb_samples = coded_samples + delayed_samples;
/* no input or buffered data => nothing to do */
if (!frame->nb_samples) {
*got_frame_ptr = 0;
return 0;
}
/* setup the data buffers */
ret = ff_get_buffer(avctx, frame, 0);
if (ret < 0)
return ret;
frame->nb_samples = 0;
memset(c->out, 0, c->nb_streams * 2 * sizeof(*c->out));
for (i = 0; i < avctx->channels; i++) {
ChannelMap *map = &c->channel_maps[i];
if (!map->copy)
c->out[2 * map->stream_idx + map->channel_idx] = (float*)frame->extended_data[i];
}
/* read the data from the sync buffers */
for (i = 0; i < c->nb_streams; i++) {
float **out = c->out + 2 * i;
int sync_size = av_audio_fifo_size(c->sync_buffers[i]);
float sync_dummy[32];
int out_dummy = (!out[0]) | ((!out[1]) << 1);
if (!out[0])
out[0] = sync_dummy;
if (!out[1])
out[1] = sync_dummy;
if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))
return AVERROR_BUG;
ret = av_audio_fifo_read(c->sync_buffers[i], (void**)out, sync_size);
if (ret < 0)
return ret;
if (out_dummy & 1)
out[0] = NULL;
else
out[0] += ret;
if (out_dummy & 2)
out[1] = NULL;
else
out[1] += ret;
c->out_size[i] = frame->linesize[0] - ret * sizeof(float);
}
/* decode each sub-packet */
for (i = 0; i < c->nb_streams; i++) {
OpusStreamContext *s = &c->streams[i];
if (i && buf) {
ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);
if (ret < 0) {
av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
return ret;
}
if (coded_samples != s->packet.frame_count * s->packet.frame_duration) {
av_log(avctx, AV_LOG_ERROR,
"Mismatching coded sample count in substream %d.\n", i);
return AVERROR_INVALIDDATA;
}
s->silk_samplerate = get_silk_samplerate(s->packet.config);
}
ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
c->out + 2 * i, c->out_size[i], coded_samples);
if (ret < 0)
return ret;
c->decoded_samples[i] = ret;
decoded_samples = FFMIN(decoded_samples, ret);
buf += s->packet.packet_size;
buf_size -= s->packet.packet_size;
}
/* buffer the extra samples */
for (i = 0; i < c->nb_streams; i++) {
int buffer_samples = c->decoded_samples[i] - decoded_samples;
if (buffer_samples) {
float *buf[2] = { c->out[2 * i + 0] ? c->out[2 * i + 0] : (float*)frame->extended_data[0],
c->out[2 * i + 1] ? c->out[2 * i + 1] : (float*)frame->extended_data[0] };
buf[0] += decoded_samples;
buf[1] += decoded_samples;
ret = av_audio_fifo_write(c->sync_buffers[i], (void**)buf, buffer_samples);
if (ret < 0)
return ret;
}
}
for (i = 0; i < avctx->channels; i++) {
ChannelMap *map = &c->channel_maps[i];
/* handle copied channels */
if (map->copy) {
memcpy(frame->extended_data[i],
frame->extended_data[map->copy_idx],
frame->linesize[0]);
} else if (map->silence) {
memset(frame->extended_data[i], 0, frame->linesize[0]);
}
if (c->gain_i && decoded_samples > 0) {
c->fdsp->vector_fmul_scalar((float*)frame->extended_data[i],
(float*)frame->extended_data[i],
c->gain, FFALIGN(decoded_samples, 8));
}
}
frame->nb_samples = decoded_samples;
*got_frame_ptr = !!decoded_samples;
return avpkt->size;
}
int main(int argc, char* argv[])
{
AVFormatContext *ifmt_ctx = NULL;
AVFormatContext *ifmt_ctx_a = NULL;
AVFormatContext *ofmt_ctx;
AVInputFormat* ifmt;
AVStream* video_st;
AVStream* audio_st;
AVCodecContext* pCodecCtx;
AVCodecContext* pCodecCtx_a;
AVCodec* pCodec;
AVCodec* pCodec_a;
AVPacket *dec_pkt, enc_pkt;
AVPacket *dec_pkt_a, enc_pkt_a;
AVFrame *pframe, *pFrameYUV;
struct SwsContext *img_convert_ctx;
struct SwrContext *aud_convert_ctx;
char capture_name[80] = { 0 };
char device_name[80] = { 0 };
char device_name_a[80] = { 0 };
int framecnt = 0;
int nb_samples = 0;
int videoindex;
int audioindex;
int i;
int ret;
HANDLE hThread;
const char* out_path = "rtmp://localhost/live/livestream";
int dec_got_frame, enc_got_frame;
int dec_got_frame_a, enc_got_frame_a;
int aud_next_pts = 0;
int vid_next_pts = 0;
int encode_video = 1, encode_audio = 1;
AVRational time_base_q = { 1, AV_TIME_BASE };
av_register_all();
//Register Device
avdevice_register_all();
avformat_network_init();
#if USEFILTER
//Register Filter
avfilter_register_all();
buffersrc = avfilter_get_by_name("buffer");
buffersink = avfilter_get_by_name("buffersink");
#endif
//Show Dshow Device
show_dshow_device();
printf("\nChoose video capture device: ");
if (gets(capture_name) == 0)
{
printf("Error in gets()\n");
return -1;
}
sprintf(device_name, "video=%s", capture_name);
printf("\nChoose audio capture device: ");
if (gets(capture_name) == 0)
{
printf("Error in gets()\n");
return -1;
}
sprintf(device_name_a, "audio=%s", capture_name);
//wchar_t *cam = L"video=Integrated Camera";
//wchar_t *cam = L"video=YY伴侣";
//char *device_name_utf8 = dup_wchar_to_utf8(cam);
//wchar_t *cam_a = L"audio=麦克风阵列 (Realtek High Definition Audio)";
//char *device_name_utf8_a = dup_wchar_to_utf8(cam_a);
ifmt = av_find_input_format("dshow");
// Set device params
AVDictionary *device_param = 0;
//if not setting rtbufsize, error messages will be shown in cmd, but you can still watch or record the stream correctly in most time
//setting rtbufsize will erase those error messages, however, larger rtbufsize will bring latency
//av_dict_set(&device_param, "rtbufsize", "10M", 0);
//Set own video device's name
if (avformat_open_input(&ifmt_ctx, device_name, ifmt, &device_param) != 0){
printf("Couldn't open input video stream.（无法打开输入流）\n");
return -1;
}
//Set own audio device's name
if (avformat_open_input(&ifmt_ctx_a, device_name_a, ifmt, &device_param) != 0){
printf("Couldn't open input audio stream.（无法打开输入流）\n");
return -1;
}
//input video initialize
if (avformat_find_stream_info(ifmt_ctx, NULL) < 0)
{
printf("Couldn't find video stream information.（无法获取流信息）\n");
return -1;
}
videoindex = -1;
for (i = 0; i < ifmt_ctx->nb_streams; i++)
if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
{
videoindex = i;
break;
}
if (videoindex == -1)
{
printf("Couldn't find a video stream.（没有找到视频流）\n");
return -1;
}
if (avcodec_open2(ifmt_ctx->streams[videoindex]->codec, avcodec_find_decoder(ifmt_ctx->streams[videoindex]->codec->codec_id), NULL) < 0)
{
printf("Could not open video codec.（无法打开解码器）\n");
return -1;
}
//input audio initialize
if (avformat_find_stream_info(ifmt_ctx_a, NULL) < 0)
{
printf("Couldn't find audio stream information.（无法获取流信息）\n");
return -1;
}
audioindex = -1;
for (i = 0; i < ifmt_ctx_a->nb_streams; i++)
if (ifmt_ctx_a->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
{
audioindex = i;
break;
}
if (audioindex == -1)
{
printf("Couldn't find a audio stream.（没有找到视频流）\n");
return -1;
}
if (avcodec_open2(ifmt_ctx_a->streams[audioindex]->codec, avcodec_find_decoder(ifmt_ctx_a->streams[audioindex]->codec->codec_id), NULL) < 0)
{
printf("Could not open audio codec.（无法打开解码器）\n");
return -1;
}
//output initialize
avformat_alloc_output_context2(&ofmt_ctx, NULL, "flv", out_path);
//output video encoder initialize
pCodec = avcodec_find_encoder(AV_CODEC_ID_H264);
if (!pCodec){
printf("Can not find output video encoder! (没有找到合适的编码器！)\n");
return -1;
}
pCodecCtx = avcodec_alloc_context3(pCodec);
pCodecCtx->pix_fmt = PIX_FMT_YUV420P;
pCodecCtx->width = ifmt_ctx->streams[videoindex]->codec->width;
pCodecCtx->height = ifmt_ctx->streams[videoindex]->codec->height;
pCodecCtx->time_base.num = 1;
pCodecCtx->time_base.den = 25;
pCodecCtx->bit_rate = 300000;
pCodecCtx->gop_size = 250;
/* Some formats want stream headers to be separate. */
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
pCodecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
//H264 codec param
//pCodecCtx->me_range = 16;
//pCodecCtx->max_qdiff = 4;
//pCodecCtx->qcompress = 0.6;
pCodecCtx->qmin = 10;
pCodecCtx->qmax = 51;
//Optional Param
pCodecCtx->max_b_frames = 0;
// Set H264 preset and tune
AVDictionary *param = 0;
av_dict_set(¶m, "preset", "fast", 0);
av_dict_set(¶m, "tune", "zerolatency", 0);
if (avcodec_open2(pCodecCtx, pCodec, ¶m) < 0){
printf("Failed to open output video encoder! (编码器打开失败！)\n");
return -1;
}
//Add a new stream to output,should be called by the user before avformat_write_header() for muxing
video_st = avformat_new_stream(ofmt_ctx, pCodec);
if (video_st == NULL){
return -1;
}
video_st->time_base.num = 1;
video_st->time_base.den = 25;
video_st->codec = pCodecCtx;
//output audio encoder initialize
pCodec_a = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (!pCodec_a){
printf("Can not find output audio encoder! (没有找到合适的编码器！)\n");
return -1;
}
pCodecCtx_a = avcodec_alloc_context3(pCodec_a);
pCodecCtx_a->channels = 2;
pCodecCtx_a->channel_layout = av_get_default_channel_layout(2);
pCodecCtx_a->sample_rate = ifmt_ctx_a->streams[audioindex]->codec->sample_rate;
pCodecCtx_a->sample_fmt = pCodec_a->sample_fmts[0];
pCodecCtx_a->bit_rate = 32000;
pCodecCtx_a->time_base.num = 1;
pCodecCtx_a->time_base.den = pCodecCtx_a->sample_rate;
/** Allow the use of the experimental AAC encoder */
pCodecCtx_a->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Some formats want stream headers to be separate. */
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
pCodecCtx_a->flags |= CODEC_FLAG_GLOBAL_HEADER;
if (avcodec_open2(pCodecCtx_a, pCodec_a, NULL) < 0){
printf("Failed to open ouput audio encoder! (编码器打开失败！)\n");
return -1;
}
//Add a new stream to output,should be called by the user before avformat_write_header() for muxing
audio_st = avformat_new_stream(ofmt_ctx, pCodec_a);
if (audio_st == NULL){
return -1;
}
audio_st->time_base.num = 1;
audio_st->time_base.den = pCodecCtx_a->sample_rate;
audio_st->codec = pCodecCtx_a;
//Open output URL,set before avformat_write_header() for muxing
if (avio_open(&ofmt_ctx->pb, out_path, AVIO_FLAG_READ_WRITE) < 0){
printf("Failed to open output file! (输出文件打开失败！)\n");
return -1;
}
//Show some Information
av_dump_format(ofmt_ctx, 0, out_path, 1);
//Write File Header
avformat_write_header(ofmt_ctx, NULL);
//prepare before decode and encode
dec_pkt = (AVPacket *)av_malloc(sizeof(AVPacket));
#if USEFILTER
#else
//camera data may has a pix fmt of RGB or sth else,convert it to YUV420
img_convert_ctx = sws_getContext(ifmt_ctx->streams[videoindex]->codec->width, ifmt_ctx->streams[videoindex]->codec->height,
ifmt_ctx->streams[videoindex]->codec->pix_fmt, pCodecCtx->width, pCodecCtx->height, PIX_FMT_YUV420P, SWS_BICUBIC, NULL, NULL, NULL);
// Initialize the resampler to be able to convert audio sample formats
aud_convert_ctx = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(pCodecCtx_a->channels),
pCodecCtx_a->sample_fmt,
pCodecCtx_a->sample_rate,
av_get_default_channel_layout(ifmt_ctx_a->streams[audioindex]->codec->channels),
ifmt_ctx_a->streams[audioindex]->codec->sample_fmt,
ifmt_ctx_a->streams[audioindex]->codec->sample_rate,
0, NULL);
/**
* 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
*/
//av_assert0(pCodecCtx_a->sample_rate == ifmt_ctx_a->streams[audioindex]->codec->sample_rate);
swr_init(aud_convert_ctx);
#endif
//Initialize the buffer to store YUV frames to be encoded.
pFrameYUV = av_frame_alloc();
uint8_t *out_buffer = (uint8_t *)av_malloc(avpicture_get_size(PIX_FMT_YUV420P, pCodecCtx->width, pCodecCtx->height));
avpicture_fill((AVPicture *)pFrameYUV, out_buffer, PIX_FMT_YUV420P, pCodecCtx->width, pCodecCtx->height);
//Initialize the FIFO buffer to store audio samples to be encoded.
AVAudioFifo *fifo = NULL;
fifo = av_audio_fifo_alloc(pCodecCtx_a->sample_fmt, pCodecCtx_a->channels, 1);
//Initialize the buffer to store converted samples to be encoded.
uint8_t **converted_input_samples = NULL;
/**
* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(converted_input_samples = (uint8_t**)calloc(pCodecCtx_a->channels,
sizeof(**converted_input_samples)))) {
printf("Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
printf("\n --------call started----------\n");
#if USEFILTER
printf("\n Press differnet number for different filters:");
printf("\n 1->Mirror");
printf("\n 2->Add Watermark");
printf("\n 3->Negate");
printf("\n 4->Draw Edge");
printf("\n 5->Split Into 4");
printf("\n 6->Vintage");
printf("\n Press 0 to remove filter\n");
#endif
printf("\nPress enter to stop...\n");
hThread = CreateThread(
NULL, // default security attributes
0, // use default stack size
MyThreadFunction, // thread function name
NULL, // argument to thread function
0, // use default creation flags
NULL); // returns the thread identifier
//start decode and encode
int64_t start_time = av_gettime();
while (encode_video || encode_audio)
{
if (encode_video &&
(!encode_audio || av_compare_ts(vid_next_pts, time_base_q,
aud_next_pts, time_base_q) <= 0))
{
if ((ret=av_read_frame(ifmt_ctx, dec_pkt)) >= 0){
if (exit_thread)
break;
av_log(NULL, AV_LOG_DEBUG, "Going to reencode the frame\n");
pframe = av_frame_alloc();
if (!pframe) {
ret = AVERROR(ENOMEM);
return ret;
}
ret = avcodec_decode_video2(ifmt_ctx->streams[dec_pkt->stream_index]->codec, pframe,
&dec_got_frame, dec_pkt);
if (ret < 0) {
av_frame_free(&pframe);
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
if (dec_got_frame){
#if USEFILTER
pframe->pts = av_frame_get_best_effort_timestamp(pframe);
if (filter_change)
apply_filters(ifmt_ctx);
filter_change = 0;
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame(buffersrc_ctx, pframe) < 0) {
printf("Error while feeding the filtergraph\n");
break;
}
picref = av_frame_alloc();
/* pull filtered pictures from the filtergraph */
while (1) {
ret = av_buffersink_get_frame_flags(buffersink_ctx, picref, 0);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
return ret;
if (picref) {
img_convert_ctx = sws_getContext(picref->width, picref->height, (AVPixelFormat)picref->format, pCodecCtx->width, pCodecCtx->height, AV_PIX_FMT_YUV420P, SWS_BICUBIC, NULL, NULL, NULL);
sws_scale(img_convert_ctx, (const uint8_t* const*)picref->data, picref->linesize, 0, pCodecCtx->height, pFrameYUV->data, pFrameYUV->linesize);
sws_freeContext(img_convert_ctx);
pFrameYUV->width = picref->width;
pFrameYUV->height = picref->height;
pFrameYUV->format = PIX_FMT_YUV420P;
#else
sws_scale(img_convert_ctx, (const uint8_t* const*)pframe->data, pframe->linesize, 0, pCodecCtx->height, pFrameYUV->data, pFrameYUV->linesize);
pFrameYUV->width = pframe->width;
pFrameYUV->height = pframe->height;
pFrameYUV->format = PIX_FMT_YUV420P;
#endif
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = avcodec_encode_video2(pCodecCtx, &enc_pkt, pFrameYUV, &enc_got_frame);
av_frame_free(&pframe);
if (enc_got_frame == 1){
//printf("Succeed to encode frame: %5d\tsize:%5d\n", framecnt, enc_pkt.size);
framecnt++;
enc_pkt.stream_index = video_st->index;
//Write PTS
AVRational time_base = ofmt_ctx->streams[0]->time_base;//{ 1, 1000 };
AVRational r_framerate1 = ifmt_ctx->streams[videoindex]->r_frame_rate;//{ 50, 2 };
//Duration between 2 frames (us)
int64_t calc_duration = (double)(AV_TIME_BASE)*(1 / av_q2d(r_framerate1)); //内部时间戳
//Parameters
//enc_pkt.pts = (double)(framecnt*calc_duration)*(double)(av_q2d(time_base_q)) / (double)(av_q2d(time_base));
enc_pkt.pts = av_rescale_q(framecnt*calc_duration, time_base_q, time_base);
enc_pkt.dts = enc_pkt.pts;
enc_pkt.duration = av_rescale_q(calc_duration, time_base_q, time_base); //(double)(calc_duration)*(double)(av_q2d(time_base_q)) / (double)(av_q2d(time_base));
enc_pkt.pos = -1;
//printf("video pts : %d\n", enc_pkt.pts);
vid_next_pts=framecnt*calc_duration; //general timebase
//Delay
int64_t pts_time = av_rescale_q(enc_pkt.pts, time_base, time_base_q);
int64_t now_time = av_gettime() - start_time;
if ((pts_time > now_time) && ((vid_next_pts + pts_time - now_time)<aud_next_pts))
av_usleep(pts_time - now_time);
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
av_free_packet(&enc_pkt);
}
#if USEFILTER
av_frame_unref(picref);
}
}
#endif
}
else {
av_frame_free(&pframe);
}
av_free_packet(dec_pkt);
}
else
if (ret == AVERROR_EOF)
encode_video = 0;
else
{
printf("Could not read video frame\n");
return ret;
}
}
else
{
//audio trancoding here
const int output_frame_size = pCodecCtx_a->frame_size;
if (exit_thread)
break;
/**
* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples.
*/
while (av_audio_fifo_size(fifo) < output_frame_size) {
/**
* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer.
*/
AVFrame *input_frame = av_frame_alloc();
if (!input_frame)
{
ret = AVERROR(ENOMEM);
return ret;
}
/** Decode one frame worth of audio samples. */
/** Packet used for temporary storage. */
AVPacket input_packet;
av_init_packet(&input_packet);
input_packet.data = NULL;
input_packet.size = 0;
/** Read one audio frame from the input file into a temporary packet. */
if ((ret = av_read_frame(ifmt_ctx_a, &input_packet)) < 0) {
/** If we are at the end of the file, flush the decoder below. */
if (ret == AVERROR_EOF)
{
encode_audio = 0;
}
else
{
printf("Could not read audio frame\n");
return ret;
}
}
/**
* Decode the audio frame stored in the temporary packet.
* The input audio stream decoder is used to do this.
* If we are at the end of the file, pass an empty packet to the decoder
* to flush it.
*/
if ((ret = avcodec_decode_audio4(ifmt_ctx_a->streams[audioindex]->codec, input_frame,
&dec_got_frame_a, &input_packet)) < 0) {
printf("Could not decode audio frame\n");
return ret;
}
av_packet_unref(&input_packet);
/** If there is decoded data, convert and store it */
if (dec_got_frame_a) {
/**
* Allocate memory for the samples of all channels in one consecutive
* block for convenience.
*/
if ((ret = av_samples_alloc(converted_input_samples, NULL,
pCodecCtx_a->channels,
input_frame->nb_samples,
pCodecCtx_a->sample_fmt, 0)) < 0) {
printf("Could not allocate converted input samples\n");
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return ret;
}
/**
* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples.
*/
/** Convert the samples using the resampler. */
if ((ret = swr_convert(aud_convert_ctx,
converted_input_samples, input_frame->nb_samples,
(const uint8_t**)input_frame->extended_data, input_frame->nb_samples)) < 0) {
printf("Could not convert input samples\n");
return ret;
}
/** Add the converted input samples to the FIFO buffer for later processing. */
/**
* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples.
*/
if ((ret = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + input_frame->nb_samples)) < 0) {
printf("Could not reallocate FIFO\n");
return ret;
}
/** Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
input_frame->nb_samples) < input_frame->nb_samples) {
printf("Could not write data to FIFO\n");
return AVERROR_EXIT;
}
}
}
/**
* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder.
*/
if (av_audio_fifo_size(fifo) >= output_frame_size)
/**
* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file.
*/
{
/** Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame=av_frame_alloc();
if (!output_frame)
{
ret = AVERROR(ENOMEM);
return ret;
}
/**
* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size
*/
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
pCodecCtx_a->frame_size);
/** Initialize temporary storage for one output frame. */
/**
* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity.
*/
output_frame->nb_samples = frame_size;
output_frame->channel_layout = pCodecCtx_a->channel_layout;
output_frame->format = pCodecCtx_a->sample_fmt;
output_frame->sample_rate = pCodecCtx_a->sample_rate;
/**
* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified.
*/
if ((ret = av_frame_get_buffer(output_frame, 0)) < 0) {
printf("Could not allocate output frame samples\n");
av_frame_free(&output_frame);
return ret;
}
/**
* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily.
*/
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
printf("Could not read data from FIFO\n");
return AVERROR_EXIT;
}
/** Encode one frame worth of audio samples. */
/** Packet used for temporary storage. */
AVPacket output_packet;
av_init_packet(&output_packet);
output_packet.data = NULL;
output_packet.size = 0;
/** Set a timestamp based on the sample rate for the container. */
if (output_frame) {
nb_samples += output_frame->nb_samples;
}
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
*/
if ((ret = avcodec_encode_audio2(pCodecCtx_a, &output_packet,
output_frame, &enc_got_frame_a)) < 0) {
printf("Could not encode frame\n");
av_packet_unref(&output_packet);
return ret;
}
/** Write one audio frame from the temporary packet to the output file. */
if (enc_got_frame_a) {
output_packet.stream_index = 1;
AVRational time_base = ofmt_ctx->streams[1]->time_base;
AVRational r_framerate1 = { ifmt_ctx_a->streams[audioindex]->codec->sample_rate, 1 };// { 44100, 1};
int64_t calc_duration = (double)(AV_TIME_BASE)*(1 / av_q2d(r_framerate1)); //内部时间戳
output_packet.pts = av_rescale_q(nb_samples*calc_duration, time_base_q, time_base);
output_packet.dts = output_packet.pts;
output_packet.duration = output_frame->nb_samples;
//printf("audio pts : %d\n", output_packet.pts);
aud_next_pts = nb_samples*calc_duration;
int64_t pts_time = av_rescale_q(output_packet.pts, time_base, time_base_q);
int64_t now_time = av_gettime() - start_time;
if ((pts_time > now_time) && ((aud_next_pts + pts_time - now_time)<vid_next_pts))
av_usleep(pts_time - now_time);
if ((ret = av_interleaved_write_frame(ofmt_ctx, &output_packet)) < 0) {
printf("Could not write frame\n");
av_packet_unref(&output_packet);
return ret;
}
av_packet_unref(&output_packet);
}
av_frame_free(&output_frame);
}
}
}
//Flush Encoder
ret = flush_encoder(ifmt_ctx, ofmt_ctx, 0, framecnt);
if (ret < 0) {
printf("Flushing encoder failed\n");
return -1;
}
ret = flush_encoder_a(ifmt_ctx_a, ofmt_ctx, 1, nb_samples);
if (ret < 0) {
printf("Flushing encoder failed\n");
return -1;
}
//Write file trailer
av_write_trailer(ofmt_ctx);
cleanup:
//Clean
#if USEFILTER
if (filter_graph)
avfilter_graph_free(&filter_graph);
#endif
if (video_st)
avcodec_close(video_st->codec);
if (audio_st)
avcodec_close(audio_st->codec);
av_free(out_buffer);
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
//free(converted_input_samples);
}
if (fifo)
av_audio_fifo_free(fifo);
avio_close(ofmt_ctx->pb);
avformat_free_context(ifmt_ctx);
avformat_free_context(ifmt_ctx_a);
avformat_free_context(ofmt_ctx);
CloseHandle(hThread);
return 0;
}
static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
{
int samples = s->packet.frame_duration;
int redundancy = 0;
int redundancy_size, redundancy_pos;
int ret, i, consumed;
int delayed_samples = s->delayed_samples;
ret = ff_opus_rc_dec_init(&s->rc, data, size);
if (ret < 0)
return ret;
/* decode the silk frame */
if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
#if CONFIG_SWRESAMPLE
if (!swr_is_initialized(s->swr)) {
#elif CONFIG_AVRESAMPLE
if (!avresample_is_open(s->avr)) {
#endif
ret = opus_init_resample(s);
if (ret < 0)
return ret;
}
samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
s->packet.stereo + 1,
silk_frame_duration_ms[s->packet.config]);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
return samples;
}
#if CONFIG_SWRESAMPLE
samples = swr_convert(s->swr,
(uint8_t**)s->out, s->packet.frame_duration,
(const uint8_t**)s->silk_output, samples);
#elif CONFIG_AVRESAMPLE
samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
s->packet.frame_duration,
(uint8_t**)s->silk_output,
sizeof(s->silk_buf[0]),
samples);
#endif
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
return samples;
}
av_assert2((samples & 7) == 0);
s->delayed_samples += s->packet.frame_duration - samples;
} else
ff_silk_flush(s->silk);
// decode redundancy information
consumed = opus_rc_tell(&s->rc);
if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
redundancy = ff_opus_rc_dec_log(&s->rc, 12);
else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
redundancy = 1;
if (redundancy) {
redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1);
if (s->packet.mode == OPUS_MODE_HYBRID)
redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2;
else
redundancy_size = size - (consumed + 7) / 8;
size -= redundancy_size;
if (size < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
return AVERROR_INVALIDDATA;
}
if (redundancy_pos) {
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
ff_celt_flush(s->celt);
}
}
/* decode the CELT frame */
if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
float *out_tmp[2] = { s->out[0], s->out[1] };
float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
out_tmp : s->celt_output;
int celt_output_samples = samples;
int delay_samples = av_audio_fifo_size(s->celt_delay);
if (delay_samples) {
if (s->packet.mode == OPUS_MODE_HYBRID) {
av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
delay_samples);
out_tmp[i] += delay_samples;
}
celt_output_samples -= delay_samples;
} else {
av_log(s->avctx, AV_LOG_WARNING,
"Spurious CELT delay samples present.\n");
av_audio_fifo_drain(s->celt_delay, delay_samples);
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_BUG;
}
}
ff_opus_rc_dec_raw_init(&s->rc, data + size, size);
ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
s->packet.stereo + 1,
s->packet.frame_duration,
(s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
ff_celt_band_end[s->packet.bandwidth]);
if (ret < 0)
return ret;
if (s->packet.mode == OPUS_MODE_HYBRID) {
int celt_delay = s->packet.frame_duration - celt_output_samples;
void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
s->celt_output[1] + celt_output_samples };
for (i = 0; i < s->output_channels; i++) {
s->fdsp->vector_fmac_scalar(out_tmp[i],
s->celt_output[i], 1.0,
celt_output_samples);
}
ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
if (ret < 0)
return ret;
}
} else
ff_celt_flush(s->celt);
if (s->redundancy_idx) {
for (i = 0; i < s->output_channels; i++)
opus_fade(s->out[i], s->out[i],
s->redundancy_output[i] + 120 + s->redundancy_idx,
ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
s->redundancy_idx = 0;
}
if (redundancy) {
if (!redundancy_pos) {
ff_celt_flush(s->celt);
ret = opus_decode_redundancy(s, data + size, redundancy_size);
if (ret < 0)
return ret;
for (i = 0; i < s->output_channels; i++) {
opus_fade(s->out[i] + samples - 120 + delayed_samples,
s->out[i] + samples - 120 + delayed_samples,
s->redundancy_output[i] + 120,
ff_celt_window2, 120 - delayed_samples);
if (delayed_samples)
s->redundancy_idx = 120 - delayed_samples;
}
} else {
for (i = 0; i < s->output_channels; i++) {
memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
opus_fade(s->out[i] + 120 + delayed_samples,
s->redundancy_output[i] + 120,
s->out[i] + 120 + delayed_samples,
ff_celt_window2, 120);
}
}
}
return samples;
}
static int opus_decode_subpacket(OpusStreamContext *s,
const uint8_t *buf, int buf_size,
float **out, int out_size,
int nb_samples)
{
int output_samples = 0;
int flush_needed = 0;
int i, j, ret;
s->out[0] = out[0];
s->out[1] = out[1];
s->out_size = out_size;
/* check if we need to flush the resampler */
#if CONFIG_SWRESAMPLE
if (swr_is_initialized(s->swr)) {
if (buf) {
int64_t cur_samplerate;
av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
} else {
flush_needed = !!s->delayed_samples;
}
}
#elif CONFIG_AVRESAMPLE
if (avresample_is_open(s->avr)) {
if (buf) {
int64_t cur_samplerate;
av_opt_get_int(s->avr, "in_sample_rate", 0, &cur_samplerate);
flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
} else {
flush_needed = !!s->delayed_samples;
}
}
#endif
if (!buf && !flush_needed)
return 0;
/* use dummy output buffers if the channel is not mapped to anything */
if (!s->out[0] ||
(s->output_channels == 2 && !s->out[1])) {
av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
if (!s->out_dummy)
return AVERROR(ENOMEM);
if (!s->out[0])
s->out[0] = s->out_dummy;
if (!s->out[1])
s->out[1] = s->out_dummy;
}
/* flush the resampler if necessary */
if (flush_needed) {
ret = opus_flush_resample(s, s->delayed_samples);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
return ret;
}
#if CONFIG_SWRESAMPLE
swr_close(s->swr);
#elif CONFIG_AVRESAMPLE
avresample_close(s->avr);
#endif
output_samples += s->delayed_samples;
s->delayed_samples = 0;
if (!buf)
goto finish;
}
/* decode all the frames in the packet */
for (i = 0; i < s->packet.frame_count; i++) {
int size = s->packet.frame_size[i];
int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return samples;
for (j = 0; j < s->output_channels; j++)
memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
samples = s->packet.frame_duration;
}
output_samples += samples;
for (j = 0; j < s->output_channels; j++)
s->out[j] += samples;
s->out_size -= samples * sizeof(float);
}
finish:
s->out[0] = s->out[1] = NULL;
s->out_size = 0;
return output_samples;
}
/**
* Read samples from the input FIFOs, mix, and write to the output link.
*/
static int output_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
MixContext *s = ctx->priv;
AVFrame *out_buf, *in_buf;
int nb_samples, ns, i;
if (s->input_state[0] & INPUT_ON) {
/* first input live: use the corresponding frame size */
nb_samples = frame_list_next_frame_size(s->frame_list);
for (i = 1; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
ns = av_audio_fifo_size(s->fifos[i]);
if (ns < nb_samples) {
if (!(s->input_state[i] & INPUT_EOF))
/* unclosed input with not enough samples */
return 0;
/* closed input to drain */
nb_samples = ns;
}
}
}
} else {
/* first input closed: use the available samples */
nb_samples = INT_MAX;
for (i = 1; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
ns = av_audio_fifo_size(s->fifos[i]);
nb_samples = FFMIN(nb_samples, ns);
}
}
if (nb_samples == INT_MAX) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
return 0;
}
}
s->next_pts = frame_list_next_pts(s->frame_list);
frame_list_remove_samples(s->frame_list, nb_samples);
calculate_scales(s, nb_samples);
if (nb_samples == 0)
return 0;
out_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!out_buf)
return AVERROR(ENOMEM);
in_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!in_buf) {
av_frame_free(&out_buf);
return AVERROR(ENOMEM);
}
for (i = 0; i < s->nb_inputs; i++) {
if (s->input_state[i] & INPUT_ON) {
int planes, plane_size, p;
av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
nb_samples);
planes = s->planar ? s->nb_channels : 1;
plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
plane_size = FFALIGN(plane_size, 16);
if (out_buf->format == AV_SAMPLE_FMT_FLT ||
out_buf->format == AV_SAMPLE_FMT_FLTP) {
for (p = 0; p < planes; p++) {
s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
(float *) in_buf->extended_data[p],
s->input_scale[i], plane_size);
}
} else {
for (p = 0; p < planes; p++) {
s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
(double *) in_buf->extended_data[p],
s->input_scale[i], plane_size);
}
}
}
}
av_frame_free(&in_buf);
out_buf->pts = s->next_pts;
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += nb_samples;
return ff_filter_frame(outlink, out_buf);
}
/**
*
* buffer is a float array like:
*
* - float buffer[nsamples * nchannels];
*
*
*/
bool AV::addAudioFrame(unsigned char* buffer, int nsamples, int nchannels) {
if(!use_audio) {
printf("Cannot add audio stream, we're not using audio.\n");
return false;
}
AVCodecContext* c = ct.as->codec;
// BUFFER HANDLING
int samples_stored = av_audio_fifo_write(ct.afifo, (void**)&buffer, nsamples);
if(samples_stored != nsamples) {
return false;
}
int nstored = av_audio_fifo_size(ct.afifo);
if(nstored < c->frame_size) {
return false;
}
AVPacket packet = {0}; // data and size must be '0' (allocation is done for you :> )
AVFrame* frame = avcodec_alloc_frame();
int got_packet = 0;
av_init_packet(&packet);
packet.data = NULL;
packet.size = 0;
int use_nsamples = c->frame_size;
frame->nb_samples = use_nsamples; // <-- important, must be set before avcodec_fill_audio_frame
// GET DATA FROM BUFFER
int num_bytes = av_samples_get_buffer_size(NULL, c->channels, use_nsamples, c->sample_fmt, 0);
uint8_t* my_buffer = (uint8_t*)av_malloc(num_bytes);
uint8_t** my_ptr = &my_buffer;
int nread = av_audio_fifo_read(ct.afifo, (void**)my_ptr, use_nsamples);
if(nread != use_nsamples) {
printf("We only read: %d but we wanted to read %d samples.\n", nread, use_nsamples);
av_free(my_buffer);
return false;
}
// FILL
int fill_result = avcodec_fill_audio_frame(
frame
,c->channels
,c->sample_fmt
,(uint8_t*)my_buffer
,num_bytes
,1
);
if(fill_result != 0) {
char buf[1024];
av_strerror(fill_result, buf, 1024);
printf("av error: %s\n",buf);
av_free(my_buffer);
return false;
}
// ENCODE
int64_t now = av_gettime();
AVRational my_time_base = (AVRational){1,1e6};
AVRational stream_time_base = ct.as->time_base; // stream time base
AVRational codec_time_base = ct.as->codec->time_base; // codec time base
int64_t now_frame_pts = av_rescale_q(now, my_time_base, codec_time_base);
if(frame->pts == AV_NOPTS_VALUE) {
frame->pts = ct.acounter;
}
ct.acounter = frame->pts + use_nsamples;
printf("frame->nb_samples: %d, counter: %d\n", frame->nb_samples, ct.acounter);
int enc_result = avcodec_encode_audio2(c, &packet, frame, &got_packet);
packet.stream_index = ct.as->index;
if(!got_packet) {
av_free(my_buffer);
return false;
}
if(enc_result < 0) {
char buf[1024];
av_strerror(enc_result, buf, 1024);
printf("av error: %s\n",buf);
}
// CORRECT THE PTS, FROM VIDEO_CODEC.time_base TO STREAM.time_base
packet.pts = av_rescale_q(packet.pts, codec_time_base, stream_time_base);
packet.dts = av_rescale_q(packet.dts, codec_time_base, stream_time_base);
//packet.duration = av_rescale_q(packet.duration, codec_time_base, stream_time_base);
//packet.dts = packet.pts; // just a wild guess
packet.duration = 0;
/*
printf("Audio: stream: %d\n", packet.stream_index);
printf("Audio: ct.acounter: %d\n", ct.acounter);
printf("Audio: packet.duration: %d\n", packet.duration);
printf("Audio: stream.time_base, num=%d, den=%d\n", stream_time_base.num, stream_time_base.den);
printf("Audio: codec.time_base, num=%d, den=%d\n", codec_time_base.num, codec_time_base.den);
printf("Audio: coded_frame.pts: %lld\n", ct.as->codec->coded_frame->pts);
printf("Audio: packet.pts: %lld\n" ,packet.pts);
printf("-------------------\n");
*/
// WRITE
if(av_interleaved_write_frame(ct.c, &packet) != 0) {
printf("Cannot write audio frame.\n");
av_free(my_buffer);
return false;
}
av_free(my_buffer);
return true;
}
