#include <pthread.h>
#if LIBAVUTIL_VERSION_INT < (50<<16)
+#define PIX_FMT_RGB32 PIX_FMT_RGBA32
#define PIX_FMT_YUYV422 PIX_FMT_YUV422
#endif
output.data, output.linesize);
sws_freeContext( context );
}
+ else if ( *format == mlt_image_rgb24a || *format == mlt_image_opengl )
+ {
+ struct SwsContext *context = sws_getContext( width, height, pix_fmt,
+ width, height, PIX_FMT_RGBA, SWS_FAST_BILINEAR, NULL, NULL, NULL);
+ AVPicture output;
+ avpicture_fill( &output, buffer, PIX_FMT_RGBA, width, height );
+ sws_scale( context, frame->data, frame->linesize, 0, height,
+ output.data, output.linesize);
+ sws_freeContext( context );
+ }
else
{
struct SwsContext *context = sws_getContext( width, height, pix_fmt,
avpicture_fill( &output, buffer, PIX_FMT_RGB24, width, height );
img_convert( &output, PIX_FMT_RGB24, (AVPicture *)frame, pix_fmt, width, height );
}
+ else if ( format == mlt_image_rgb24a || format == mlt_image_opengl )
+ {
+ AVPicture output;
+ avpicture_fill( &output, buffer, PIX_FMT_RGB32, width, height );
+ img_convert( &output, PIX_FMT_RGB32, (AVPicture *)frame, pix_fmt, width, height );
+ }
else
{
AVPicture output;
case mlt_image_rgb24:
size = *width * ( *height + 1 ) * 3;
break;
+ case mlt_image_rgb24a:
+ case mlt_image_opengl:
+ size = *width * ( *height + 1 ) * 4;
+ break;
default:
*format = mlt_image_yuv422;
size = *width * ( *height + 1 ) * 2;
{
int_position = req_position;
}
+ mlt_log_debug( MLT_PRODUCER_SERVICE(this), "pkt.dts %llu req_pos %d cur_pos %d pkt_pos %d",
+ pkt.dts, req_position, current_position, int_position );
+ // Make a dumb assumption on streams that contain wild timestamps
+ if ( abs( req_position - int_position ) > 999 )
+ {
+ int_position = req_position;
+ mlt_log_debug( MLT_PRODUCER_SERVICE(this), " WILD TIMESTAMP!" );
+ }
mlt_properties_set_int( properties, "_last_position", int_position );
// Decode the image
got_picture = 0;
}
}
- mlt_log_debug( MLT_PRODUCER_SERVICE(this), "pkt.dts %llu req_pos %d cur_pos %d pkt_pos %d got_pic %d key %d\n",
- pkt.dts, req_position, current_position, int_position, got_picture, pkt.flags & PKT_FLAG_KEY );
+ mlt_log_debug( MLT_PRODUCER_SERVICE(this), " got_pic %d key %d\n", got_picture, pkt.flags & PKT_FLAG_KEY );
av_free_packet( &pkt );
}
else if ( ret >= 0 )
if ( allocate_buffer( frame_properties, codec_context, buffer, format, width, height ) )
{
convert_image( av_frame, *buffer, codec_context->pix_fmt, format, *width, *height );
- mlt_properties_set_int( frame_properties, "progressive", !av_frame->interlaced_frame );
+ if ( !mlt_properties_get( properties, "force_progressive" ) )
+ mlt_properties_set_int( frame_properties, "progressive", !av_frame->interlaced_frame );
mlt_properties_set_int( properties, "top_field_first", av_frame->top_field_first );
mlt_properties_set_int( properties, "_current_position", int_position );
mlt_properties_set_int( properties, "_got_picture", 1 );
// Determine the fps
source_fps = ( double )codec_context->time_base.den / ( codec_context->time_base.num == 0 ? 1 : codec_context->time_base.num );
+ // If the muxer reports a frame rate different than the codec
+ double muxer_fps = av_q2d( context->streams[ index ]->r_frame_rate );
+ if ( source_fps != muxer_fps )
+ // Choose the lesser - the wrong tends to be off by some multiple of 10
+ source_fps = muxer_fps < source_fps ? muxer_fps : source_fps;
+
// We'll use fps if it's available
if ( source_fps > 0 )
mlt_properties_set_double( properties, "source_fps", source_fps );
mlt_properties_set_int( frame_properties, "real_width", codec_context->width );
mlt_properties_set_int( frame_properties, "real_height", codec_context->height );
mlt_properties_set_double( frame_properties, "aspect_ratio", aspect_ratio );
+ if ( mlt_properties_get( properties, "force_progressive" ) )
+ mlt_properties_set_int( frame_properties, "progressive", mlt_properties_get_int( properties, "force_progressive" ) );
mlt_frame_push_get_image( frame, producer_get_image );
mlt_properties_set_data( frame_properties, "avformat_producer", this, 0, NULL, NULL );
/** Get the audio from a frame.
*/
-static int producer_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
+static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
// Get the properties from the frame
mlt_properties frame_properties = MLT_FRAME_PROPERTIES( frame );
// Obtain the resample context if it exists (not always needed)
ReSampleContext *resample = mlt_properties_get_data( properties, "audio_resample", NULL );
-#if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(71<<8)+0))
- // Get the format converter context if it exists
- AVAudioConvert *convert = mlt_properties_get_data( properties, "audio_convert", NULL );
-#endif
-
// Obtain the audio buffers
int16_t *audio_buffer = mlt_properties_get_data( properties, "audio_buffer", NULL );
int16_t *decode_buffer = mlt_properties_get_data( properties, "decode_buffer", NULL );
- int16_t *convert_buffer = mlt_properties_get_data( properties, "convert_buffer", NULL );
// Get amount of audio used
int audio_used = mlt_properties_get_int( properties, "_audio_used" );
if ( resample == NULL && codec_context->channels <= 2 )
{
// Create the resampler
+#if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0))
+ resample = av_audio_resample_init( *channels, codec_context->channels, *frequency, codec_context->sample_rate,
+ SAMPLE_FMT_S16, codec_context->sample_fmt, 16, 10, 0, 0.8 );
+#else
resample = audio_resample_init( *channels, codec_context->channels, *frequency, codec_context->sample_rate );
+#endif
// And store it on properties
mlt_properties_set_data( properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL );
}
else if ( resample == NULL )
{
- *channels = codec_context->channels;
- *frequency = codec_context->sample_rate;
- }
+ // TODO: uncomment and remove following line when full multi-channel support is ready
+ // *channels = codec_context->channels;
+ codec_context->request_channels = *channels;
-#if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(71<<8)+0))
- // Check for audio format converter and create if necessary
- // TODO: support higher resolutions than 16-bit.
- if ( convert == NULL && codec_context->sample_fmt != SAMPLE_FMT_S16 )
- {
- // Create single channel converter for interleaved with no mixing matrix
- convert = av_audio_convert_alloc( SAMPLE_FMT_S16, 1, codec_context->sample_fmt, 1, NULL, 0 );
- mlt_properties_set_data( properties, "audio_convert", convert, 0, ( mlt_destructor )av_audio_convert_free, NULL );
+ *frequency = codec_context->sample_rate;
}
-#endif
// Check for audio buffer and create if necessary
if ( audio_buffer == NULL )
mlt_properties_set_data( properties, "decode_buffer", decode_buffer, 0, ( mlt_destructor )av_free, NULL );
}
-#if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(71<<8)+0))
- // Check for format converter buffer and create if necessary
- if ( resample && convert && convert_buffer == NULL )
- {
- // Allocate the audio buffer
- convert_buffer = mlt_pool_alloc( AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t ) );
-
- // And store it on properties for reuse
- mlt_properties_set_data( properties, "convert_buffer", convert_buffer, 0, ( mlt_destructor )mlt_pool_release, NULL );
- }
-#endif
-
// Seek if necessary
if ( position != expected )
{
len -= ret;
ptr += ret;
- if ( data_size > 0 )
+ if ( data_size > 0 && ( audio_used * *channels + data_size < AVCODEC_MAX_AUDIO_FRAME_SIZE ) )
{
- int src_stride[6]= { av_get_bits_per_sample_format( codec_context->sample_fmt ) / 8 };
- int dst_stride[6]= { av_get_bits_per_sample_format( SAMPLE_FMT_S16 ) / 8 };
-
if ( resample )
{
int16_t *source = decode_buffer;
int16_t *dest = &audio_buffer[ audio_used * *channels ];
- int convert_samples = data_size / src_stride[0];
+ int convert_samples = data_size / av_get_bits_per_sample_format( codec_context->sample_fmt ) * 8 / codec_context->channels;
-#if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(71<<8)+0))
- if ( convert )
- {
- const void *src_buf[6] = { decode_buffer };
- void *dst_buf[6] = { convert_buffer };
- av_audio_convert( convert, dst_buf, dst_stride, src_buf, src_stride, convert_samples );
- source = convert_buffer;
- }
-#endif
- audio_used += audio_resample( resample, dest, source, convert_samples / codec_context->channels );
+ audio_used += audio_resample( resample, dest, source, convert_samples );
}
else
{
-#if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(71<<8)+0))
- if ( convert )
- {
- const void *src_buf[6] = { decode_buffer };
- void *dst_buf[6] = { &audio_buffer[ audio_used * *channels ] };
- av_audio_convert( convert, dst_buf, dst_stride, src_buf, src_stride, data_size / src_stride[0] );
- }
- else
-#endif
- {
- memcpy( &audio_buffer[ audio_used * *channels ], decode_buffer, data_size );
- }
- audio_used += data_size / *channels / src_stride[0];
+ memcpy( &audio_buffer[ audio_used * *channels ], decode_buffer, data_size );
+ audio_used += data_size / *channels / av_get_bits_per_sample_format( codec_context->sample_fmt ) * 8;
}
// Handle ignore
av_free_packet( &pkt );
}
- *buffer = mlt_pool_alloc( *samples * *channels * sizeof( int16_t ) );
- mlt_properties_set_data( frame_properties, "audio", *buffer, 0, ( mlt_destructor )mlt_pool_release, NULL );
+ int size = *samples * *channels * sizeof( int16_t );
+ *format = mlt_audio_s16;
+ *buffer = mlt_pool_alloc( size );
+ mlt_frame_set_audio( frame, *buffer, *format, size, mlt_pool_release );
// Now handle the audio if we have enough
if ( audio_used >= *samples )
projects / mlt / blobdiff