forked from bartvdbraak/blender
FFMPEG: adding support for ffmpeg planar formats also to writeffmpeg.c.
This commit is contained in:
Joerg Mueller
parent
c35dec60b5
commit
8588fec935
@ -88,10 +88,11 @@ static uint8_t *video_buffer = 0;
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static int video_buffersize = 0;
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static uint8_t *audio_input_buffer = 0;
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static uint8_t *audio_deinterleave_buffer = 0;
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static int audio_input_samples = 0;
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static uint8_t *audio_output_buffer = 0;
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static int audio_outbuf_size = 0;
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static double audio_time = 0.0f;
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static bool audio_deinterleave = false;
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static int audio_sample_size = 0;
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#ifdef WITH_AUDASPACE
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static AUD_Device *audio_mixdown_device = 0;
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@ -122,24 +123,48 @@ static int write_audio_frame(void)
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{
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AVCodecContext *c = NULL;
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AVPacket pkt;
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AVFrame* frame;
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int got_output = 0;
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c = audio_stream->codec;
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av_init_packet(&pkt);
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pkt.size = 0;
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pkt.data = NULL;
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frame = avcodec_alloc_frame();
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frame->nb_samples = audio_input_samples;
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frame->format = c->sample_fmt;
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frame->channel_layout = c->channel_layout;
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AUD_readDevice(audio_mixdown_device, audio_input_buffer, audio_input_samples);
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audio_time += (double) audio_input_samples / (double) c->sample_rate;
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pkt.size = avcodec_encode_audio(c, audio_output_buffer, audio_outbuf_size, (short *) audio_input_buffer);
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if(audio_deinterleave) {
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int channel, i;
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uint8_t* temp;
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if (pkt.size < 0) {
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for(channel = 0; channel < c->channels; channel++) {
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for(i = 0; i < frame->nb_samples; i++) {
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memcpy(audio_deinterleave_buffer + (i + channel * frame->nb_samples) * audio_sample_size,
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audio_input_buffer + (c->channels * i + channel) * audio_sample_size, audio_sample_size);
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}
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}
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temp = audio_deinterleave_buffer;
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audio_deinterleave_buffer = audio_input_buffer;
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audio_input_buffer = temp;
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}
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avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, audio_input_buffer,
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audio_input_samples * c->channels * audio_sample_size, 0);
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if(avcodec_encode_audio2(c, &pkt, frame, &got_output) < 0) {
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// XXX error("Error writing audio packet");
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return -1;
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}
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pkt.data = audio_output_buffer;
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if(got_output) {
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if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE) {
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pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base, audio_stream->time_base);
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PRINT("Audio Frame PTS: %d\n", (int) pkt.pts);
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@ -153,6 +178,12 @@ static int write_audio_frame(void)
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fprintf(stderr, "Error writing audio packet!\n");
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return -1;
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}
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av_free_packet(&pkt);
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}
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avcodec_free_frame(&frame);
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return 0;
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}
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#endif // #ifdef WITH_AUDASPACE
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@ -608,8 +639,6 @@ static AVStream *alloc_video_stream(RenderData *rd, int codec_id, AVFormatContex
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return st;
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}
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/* Prepare an audio stream for the output file */
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static AVStream *alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContext *of, char *error, int error_size)
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{
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AVStream *st;
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@ -659,11 +688,6 @@ static AVStream *alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex
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}
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}
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if (c->sample_fmt == AV_SAMPLE_FMT_FLTP) {
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BLI_strncpy(error, "Requested audio codec requires planar float sample format, which is not supported yet", error_size);
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return NULL;
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}
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if (codec->supported_samplerates) {
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const int *p = codec->supported_samplerates;
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int best = 0;
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@ -692,24 +716,23 @@ static AVStream *alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex
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st->codec->time_base.num = 1;
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st->codec->time_base.den = st->codec->sample_rate;
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audio_outbuf_size = FF_MIN_BUFFER_SIZE;
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if ((c->codec_id >= CODEC_ID_PCM_S16LE) && (c->codec_id <= CODEC_ID_PCM_DVD))
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audio_input_samples = audio_outbuf_size * 8 / c->bits_per_coded_sample / c->channels;
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if (c->frame_size == 0)
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// used to be if((c->codec_id >= CODEC_ID_PCM_S16LE) && (c->codec_id <= CODEC_ID_PCM_DVD))
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// not sure if that is needed anymore, so let's try out if there are any
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// complaints regarding some ffmpeg versions users might have
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audio_input_samples = FF_MIN_BUFFER_SIZE * 8 / c->bits_per_coded_sample / c->channels;
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else {
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audio_input_samples = c->frame_size;
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if (c->frame_size * c->channels * sizeof(int16_t) * 4 > audio_outbuf_size)
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audio_outbuf_size = c->frame_size * c->channels * sizeof(int16_t) * 4;
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}
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audio_output_buffer = (uint8_t *) av_malloc(audio_outbuf_size);
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audio_deinterleave = av_sample_fmt_is_planar(c->sample_fmt);
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if (c->sample_fmt == AV_SAMPLE_FMT_FLT) {
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audio_input_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * sizeof(float));
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}
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else {
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audio_input_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * sizeof(int16_t));
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}
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audio_sample_size = av_get_bytes_per_sample(c->sample_fmt);
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audio_input_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * audio_sample_size);
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if(audio_deinterleave)
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audio_deinterleave_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * audio_sample_size);
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audio_time = 0.0f;
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@ -1010,12 +1033,28 @@ int BKE_ffmpeg_start(struct Scene *scene, RenderData *rd, int rectx, int recty,
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AVCodecContext *c = audio_stream->codec;
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AUD_DeviceSpecs specs;
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specs.channels = c->channels;
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if (c->sample_fmt == AV_SAMPLE_FMT_FLT) {
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specs.format = AUD_FORMAT_FLOAT32;
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}
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else {
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switch(av_get_packed_sample_fmt(c->sample_fmt))
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{
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case AV_SAMPLE_FMT_U8:
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specs.format = AUD_FORMAT_U8;
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break;
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case AV_SAMPLE_FMT_S16:
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specs.format = AUD_FORMAT_S16;
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break;
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case AV_SAMPLE_FMT_S32:
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specs.format = AUD_FORMAT_S32;
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break;
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case AV_SAMPLE_FMT_FLT:
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specs.format = AUD_FORMAT_FLOAT32;
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break;
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case AV_SAMPLE_FMT_DBL:
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specs.format = AUD_FORMAT_FLOAT64;
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break;
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default:
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return -31415;
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}
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specs.rate = rd->ffcodecdata.audio_mixrate;
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audio_mixdown_device = sound_mixdown(scene, specs, rd->sfra, rd->ffcodecdata.audio_volume);
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#ifdef FFMPEG_CODEC_TIME_BASE
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@ -1138,15 +1177,16 @@ static void end_ffmpeg_impl(int is_autosplit)
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MEM_freeN(video_buffer);
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video_buffer = 0;
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}
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if (audio_output_buffer) {
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av_free(audio_output_buffer);
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audio_output_buffer = 0;
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}
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if (audio_input_buffer) {
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av_free(audio_input_buffer);
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audio_input_buffer = 0;
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}
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if (audio_deinterleave_buffer) {
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av_free(audio_deinterleave_buffer);
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audio_deinterleave_buffer = 0;
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}
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if (img_convert_ctx) {
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sws_freeContext(img_convert_ctx);
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img_convert_ctx = 0;
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