#if LIBAVUTIL_VERSION_INT >= ((50<<16)+(8<<8)+0)
#include <libavutil/pixdesc.h>
#endif
+#if LIBAVUTIL_VERSION_INT >= ((50<<16)+(38<<8)+0)
+#include <libavutil/samplefmt.h>
+#endif
#include <libavutil/mathematics.h>
#if LIBAVUTIL_VERSION_INT < (50<<16)
typedef struct
{
- int16_t *buffer;
+ uint8_t *buffer;
int size;
int used;
double time;
return fifo;
}
-// sample_fifo_clear and check are temporarily aborted (not working as intended)
-
-void sample_fifo_clear( sample_fifo fifo, double time )
-{
- int words = ( float )( time - fifo->time ) * fifo->frequency * fifo->channels;
- if ( ( int )( ( float )time * 100 ) < ( int )( ( float )fifo->time * 100 ) && fifo->used > words && words > 0 )
- {
- memmove( fifo->buffer, &fifo->buffer[ words ], ( fifo->used - words ) * sizeof( int16_t ) );
- fifo->used -= words;
- fifo->time = time;
- }
- else if ( ( int )( ( float )time * 100 ) != ( int )( ( float )fifo->time * 100 ) )
- {
- fifo->used = 0;
- fifo->time = time;
- }
-}
-
-void sample_fifo_check( sample_fifo fifo, double time )
-{
- if ( fifo->used == 0 )
- {
- if ( ( int )( ( float )time * 100 ) < ( int )( ( float )fifo->time * 100 ) )
- fifo->time = time;
- }
-}
-
-void sample_fifo_append( sample_fifo fifo, int16_t *samples, int count )
+// count is the number of samples multiplied by the number of bytes per sample
+void sample_fifo_append( sample_fifo fifo, uint8_t *samples, int count )
{
if ( ( fifo->size - fifo->used ) < count )
{
fifo->size += count * 5;
- fifo->buffer = realloc( fifo->buffer, fifo->size * sizeof( int16_t ) );
+ fifo->buffer = realloc( fifo->buffer, fifo->size );
}
- memcpy( &fifo->buffer[ fifo->used ], samples, count * sizeof( int16_t ) );
+ memcpy( &fifo->buffer[ fifo->used ], samples, count );
fifo->used += count;
}
return fifo->used;
}
-int sample_fifo_fetch( sample_fifo fifo, int16_t *samples, int count )
+int sample_fifo_fetch( sample_fifo fifo, uint8_t *samples, int count )
{
if ( count > fifo->used )
count = fifo->used;
- memcpy( samples, fifo->buffer, count * sizeof( int16_t ) );
+ memcpy( samples, fifo->buffer, count );
fifo->used -= count;
- memmove( fifo->buffer, &fifo->buffer[ count ], fifo->used * sizeof( int16_t ) );
+ memmove( fifo->buffer, &fifo->buffer[ count ], fifo->used );
fifo->time += ( double )count / fifo->channels / fifo->frequency;
}
}
+static int get_mlt_audio_format( int av_sample_fmt )
+{
+ switch ( av_sample_fmt )
+ {
+ case AV_SAMPLE_FMT_S32:
+ return mlt_audio_s32le;
+ case AV_SAMPLE_FMT_FLT:
+ return mlt_audio_f32le;
+#if LIBAVUTIL_VERSION_INT >= ((51<<16)+(17<<8)+0)
+ case AV_SAMPLE_FMT_S32P:
+ return mlt_audio_s32;
+ case AV_SAMPLE_FMT_FLTP:
+ return mlt_audio_float;
+#endif
+ default:
+ return mlt_audio_s16;
+ }
+}
+
+static int pick_sample_fmt( mlt_properties properties, AVCodec *codec )
+{
+ int sample_fmt = AV_SAMPLE_FMT_S16;
+ const char *format = mlt_properties_get( properties, "mlt_audio_format" );
+ const int *p = codec->sample_fmts;
+
+ // get default av_sample_fmt from mlt_audio_format
+ if ( format )
+ {
+ if ( !strcmp( format, "s32le" ) )
+ sample_fmt = AV_SAMPLE_FMT_S32;
+ else if ( !strcmp( format, "f32le" ) )
+ sample_fmt = AV_SAMPLE_FMT_FLT;
+#if LIBAVUTIL_VERSION_INT >= ((51<<16)+(17<<8)+0)
+ else if ( !strcmp( format, "s32" ) )
+ sample_fmt = AV_SAMPLE_FMT_S32P;
+ else if ( !strcmp( format, "float" ) )
+ sample_fmt = AV_SAMPLE_FMT_FLTP;
+#endif
+ }
+ // check if codec supports our mlt_audio_format
+ for ( ; *p != -1; p++ )
+ {
+ if ( *p == sample_fmt )
+ return sample_fmt;
+ }
+ // no match - pick first one we support
+ for ( p = codec->sample_fmts; *p != -1; p++ )
+ {
+ switch (*p)
+ {
+ case AV_SAMPLE_FMT_S16:
+ case AV_SAMPLE_FMT_S32:
+ case AV_SAMPLE_FMT_FLT:
+#if LIBAVUTIL_VERSION_INT >= ((51<<16)+(17<<8)+0)
+ case AV_SAMPLE_FMT_S32P:
+ case AV_SAMPLE_FMT_FLTP:
+#endif
+ return *p;
+ default:
+ break;
+ }
+ }
+ mlt_log_error( properties, "audio codec sample_fmt not compatible" );
+
+ return AV_SAMPLE_FMT_NONE;
+}
+
/** Add an audio output stream
*/
c->codec_id = codec->id;
c->codec_type = CODEC_TYPE_AUDIO;
- c->sample_fmt = SAMPLE_FMT_S16;
+ c->sample_fmt = pick_sample_fmt( properties, codec );
#if 0 // disabled until some audio codecs are multi-threaded
// Setup multi-threading
int img_height = height;
// Get default audio properties
- mlt_audio_format aud_fmt = mlt_audio_s16;
int channels = mlt_properties_get_int( properties, "channels" );
int total_channels = channels;
int frequency = mlt_properties_get_int( properties, "frequency" );
- int16_t *pcm = NULL;
+ void *pcm = NULL;
int samples = 0;
// AVFormat audio buffer and frame size
mlt_image_format img_fmt = mlt_image_yuv422;
// For receiving audio samples back from the fifo
- int16_t *audio_buf_1 = av_malloc( AUDIO_ENCODE_BUFFER_SIZE );
- int16_t *audio_buf_2 = NULL;
+ uint8_t *audio_buf_1 = av_malloc( AUDIO_ENCODE_BUFFER_SIZE );
+ uint8_t *audio_buf_2 = NULL;
int count = 0;
// Allocate the context
}
mlt_properties_set_int( properties, "channels", total_channels );
+ // Audio format is determined when adding the audio stream
+ mlt_audio_format aud_fmt = mlt_audio_none;
+ if ( audio_st[0] )
+ aud_fmt = get_mlt_audio_format( audio_st[0]->codec->sample_fmt );
+ int sample_bytes = mlt_audio_format_size( aud_fmt, 1, 1 );
+ sample_bytes = sample_bytes ? sample_bytes : 1; // prevent divide by zero
+
// Set the parameters (even though we have none...)
#if LIBAVFORMAT_VERSION_INT < ((53<<16)+(2<<8)+0)
if ( av_set_parameters(oc, NULL) >= 0 )
{
samples = mlt_sample_calculator( fps, frequency, count ++ );
channels = total_channels;
- mlt_frame_get_audio( frame, (void**) &pcm, &aud_fmt, &frequency, &channels, &samples );
+ mlt_frame_get_audio( frame, &pcm, &aud_fmt, &frequency, &channels, &samples );
// Save the audio channel remap properties for later
mlt_properties_pass( frame_meta_properties, frame_properties, "meta.map.audio." );
// Silence if not normal forward speed
if ( mlt_properties_get_double( frame_properties, "_speed" ) != 1.0 )
- memset( pcm, 0, samples * channels * 2 );
+ memset( pcm, 0, samples * channels * sample_bytes );
// Append the samples
- sample_fifo_append( fifo, pcm, samples * channels );
+ sample_fifo_append( fifo, pcm, samples * channels * sample_bytes );
total_time += ( samples * 1000000 ) / frequency;
if ( !video_st )
if ( !video_st || ( video_st && audio_st[0] && audio_pts < video_pts ) )
{
// Write audio
- if ( ( video_st && terminated ) || ( channels * audio_input_frame_size ) < sample_fifo_used( fifo ) )
+ if ( ( video_st && terminated ) || ( channels * audio_input_frame_size ) < sample_fifo_used( fifo ) / sample_bytes )
{
int j = 0; // channel offset into interleaved source buffer
- int n = FFMIN( FFMIN( channels * audio_input_frame_size, sample_fifo_used( fifo ) ), AUDIO_ENCODE_BUFFER_SIZE );
+ int n = FFMIN( FFMIN( channels * audio_input_frame_size, sample_fifo_used( fifo ) / sample_bytes ), AUDIO_ENCODE_BUFFER_SIZE );
// Get the audio samples
if ( n > 0 )
{
- sample_fifo_fetch( fifo, audio_buf_1, n );
+ sample_fifo_fetch( fifo, audio_buf_1, n * sample_bytes );
}
else if ( audio_codec_id == CODEC_ID_VORBIS && terminated )
{
// Optimized for single track and no channel remap
if ( !audio_st[1] && !mlt_properties_count( frame_meta_properties ) )
{
- pkt.size = avcodec_encode_audio( codec, audio_outbuf, audio_outbuf_size, audio_buf_1 );
+ pkt.size = avcodec_encode_audio( codec, audio_outbuf, audio_outbuf_size, (short*) audio_buf_1 );
}
else
{
// Interleave the audio buffer with the # channels for this stream/mapping.
for ( k = 0; k < map_channels; k++, j++, source_offset++, dest_offset++ )
{
- int16_t *src = audio_buf_1 + source_offset;
- int16_t *dest = audio_buf_2 + dest_offset;
+ void *src = audio_buf_1 + source_offset * sample_bytes;
+ void *dest = audio_buf_2 + dest_offset * sample_bytes;
int s = samples + 1;
while ( --s ) {
- *dest = *src;
- dest += current_channels;
- src += channels;
+ memcpy( dest, src, sample_bytes );
+ dest += current_channels * sample_bytes;
+ src += channels * sample_bytes;
}
}
}
dest_offset += current_channels;
}
}
- pkt.size = avcodec_encode_audio( codec, audio_outbuf, audio_outbuf_size, audio_buf_2 );
+ pkt.size = avcodec_encode_audio( codec, audio_outbuf, audio_outbuf_size, (short*) audio_buf_2 );
}
// Write the compressed frame in the media file
long passed = time_difference( &ante );
if ( fifo != NULL )
{
- long pending = ( ( ( long )sample_fifo_used( fifo ) * 1000 ) / frequency ) * 1000;
+ long pending = ( ( ( long )sample_fifo_used( fifo ) / sample_bytes * 1000 ) / frequency ) * 1000;
passed -= pending;
}
if ( passed < total_time )
pkt.size = 0;
if ( /*( c->capabilities & CODEC_CAP_SMALL_LAST_FRAME ) &&*/
- ( channels * audio_input_frame_size < sample_fifo_used( fifo ) ) )
+ ( channels * audio_input_frame_size < sample_fifo_used( fifo ) / sample_bytes ) )
{
- sample_fifo_fetch( fifo, audio_buf_1, channels * audio_input_frame_size );
- pkt.size = avcodec_encode_audio( c, audio_outbuf, audio_outbuf_size, audio_buf_1 );
+ sample_fifo_fetch( fifo, audio_buf_1, channels * audio_input_frame_size * sample_bytes );
+ pkt.size = avcodec_encode_audio( c, audio_outbuf, audio_outbuf_size, (short*) audio_buf_1 );
}
if ( pkt.size <= 0 )
pkt.size = avcodec_encode_audio( c, audio_outbuf, audio_outbuf_size, NULL );
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