#define MAX_AUDIO_STREAMS (32)
#define MAX_VDPAU_SURFACES (10)
+#ifndef AVCODEC_MAX_AUDIO_FRAME_SIZE
+#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
+#endif
+
struct producer_avformat_s
{
mlt_producer parent;
AVCodecContext *video_codec;
AVFrame *av_frame;
AVPacket pkt;
- ReSampleContext *audio_resample[ MAX_AUDIO_STREAMS ];
mlt_position audio_expected;
mlt_position video_expected;
int audio_index;
int max_frequency;
unsigned int invalid_pts_counter;
unsigned int invalid_dts_counter;
- double resample_factor;
mlt_cache image_cache;
int colorspace;
int full_luma;
}
mlt_log_verbose( NULL, "[producer avformat] audio: total_streams %d max_stream %d total_channels %d max_channels %d\n",
self->audio_streams, self->audio_max_stream, self->total_channels, self->max_channel );
-
- // Other audio-specific initializations
- self->resample_factor = 1.0;
}
static void set_luma_transfer( struct SwsContext *context, int colorspace, int use_full_range )
// Get codec context
AVCodecContext *codec_context = self->audio_codec[ index ];
- // Obtain the resample context if it exists (not always needed)
- ReSampleContext *resample = self->audio_resample[ index ];
-
// Obtain the audio buffers
uint8_t *audio_buffer = self->audio_buffer[ index ];
uint8_t *decode_buffer = self->decode_buffer[ index ];
while ( ptr && ret >= 0 && len > 0 )
{
- int sizeof_sample = resample? sizeof( int16_t ) : sample_bytes( codec_context );
+ int sizeof_sample = sample_bytes( codec_context );
int data_size = self->audio_buffer_size[ index ];
// Decode the audio
{
// Figure out how many samples will be needed after resampling
int convert_samples = data_size / codec_context->channels / sample_bytes( codec_context );
- int samples_needed = self->resample_factor * convert_samples;
// Resize audio buffer to prevent overflow
- if ( ( audio_used + samples_needed ) * channels * sizeof_sample > self->audio_buffer_size[ index ] )
+ if ( ( audio_used + convert_samples ) * channels * sizeof_sample > self->audio_buffer_size[ index ] )
{
- self->audio_buffer_size[ index ] = ( audio_used + samples_needed * 2 ) * channels * sizeof_sample;
+ self->audio_buffer_size[ index ] = ( audio_used + convert_samples * 2 ) * channels * sizeof_sample;
audio_buffer = self->audio_buffer[ index ] = mlt_pool_realloc( audio_buffer, self->audio_buffer_size[ index ] );
}
- if ( resample )
+ uint8_t *source = decode_buffer;
+ uint8_t *dest = &audio_buffer[ audio_used * codec_context->channels * sizeof_sample ];
+ switch ( codec_context->sample_fmt )
{
- // Copy to audio buffer while resampling
- uint8_t *source = decode_buffer;
- uint8_t *dest = &audio_buffer[ audio_used * channels * sizeof_sample ];
- audio_used += audio_resample( resample, (short*) dest, (short*) source, convert_samples );
- }
- else
- {
- uint8_t *source = decode_buffer;
- uint8_t *dest = &audio_buffer[ audio_used * codec_context->channels * sizeof_sample ];
- switch ( codec_context->sample_fmt )
- {
#if LIBAVUTIL_VERSION_INT >= ((51<<16)+(17<<8)+0)
- case AV_SAMPLE_FMT_U8P:
- case AV_SAMPLE_FMT_S16P:
- case AV_SAMPLE_FMT_S32P:
- case AV_SAMPLE_FMT_FLTP:
- planar_to_interleaved( dest, source, convert_samples, codec_context->channels, sizeof_sample );
- break;
+ case AV_SAMPLE_FMT_U8P:
+ case AV_SAMPLE_FMT_S16P:
+ case AV_SAMPLE_FMT_S32P:
+ case AV_SAMPLE_FMT_FLTP:
+ planar_to_interleaved( dest, source, convert_samples, codec_context->channels, sizeof_sample );
+ break;
#endif
- default:
- // Straight copy to audio buffer
- memcpy( dest, decode_buffer, data_size );
- }
- audio_used += convert_samples;
+ default:
+ // Straight copy to audio buffer
+ memcpy( dest, decode_buffer, data_size );
}
+ audio_used += convert_samples;
// Handle ignore
while ( *ignore && audio_used )
{
*ignore -= 1;
audio_used -= audio_used > samples ? samples : audio_used;
- memmove( audio_buffer, &audio_buffer[ samples * (resample? channels : codec_context->channels) * sizeof_sample ],
+ memmove( audio_buffer, &audio_buffer[ samples * codec_context->channels * sizeof_sample ],
audio_used * sizeof_sample );
}
}
*frequency = FFMAX( self->max_frequency, *frequency );
}
- // Initialize the resamplers and buffers
+ // Initialize the buffers
for ( ; index < index_max && index < MAX_AUDIO_STREAMS; index++ )
{
// Get codec context
if ( codec_context && !self->audio_buffer[ index ] )
{
-#if LIBAVCODEC_VERSION_INT < ((54<<16)+(26<<8)+0)
- // Check for resample and create if necessary
- if ( codec_context->channels <= 2 )
- {
- // Determine by how much resampling will increase number of samples
- double resample_factor = self->audio_index == INT_MAX ? 1 : (double) *channels / codec_context->channels;
- resample_factor *= (double) *frequency / codec_context->sample_rate;
- if ( resample_factor > self->resample_factor )
- self->resample_factor = resample_factor;
-
- // Create the resampler
-#if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0))
- self->audio_resample[ index ] = av_audio_resample_init(
- self->audio_index == INT_MAX ? codec_context->channels : *channels,
- codec_context->channels, *frequency, codec_context->sample_rate,
- AV_SAMPLE_FMT_S16, codec_context->sample_fmt, 16, 10, 0, 0.8 );
-#else
- self->audio_resample[ index ] = audio_resample_init(
- self->audio_index == INT_MAX ? codec_context->channels : *channels,
- codec_context->channels, *frequency, codec_context->sample_rate );
-#endif
- }
- else
-#endif
- {
- codec_context->request_channels = self->audio_index == INT_MAX ? codec_context->channels : *channels;
- sizeof_sample = sample_bytes( codec_context );
- }
+ codec_context->request_channels = self->audio_index == INT_MAX ? codec_context->channels : *channels;
+ sizeof_sample = sample_bytes( codec_context );
// Check for audio buffer and create if necessary
self->audio_buffer_size[ index ] = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof_sample;
av_init_packet( &pkt );
- // If not resampling, give consumer more than requested.
// It requested number samples based on requested frame rate.
// Do not clean this up with a samples *= ...!
- if ( self->audio_index != INT_MAX && ! self->audio_resample[ self->audio_index ] )
+ if ( self->audio_index != INT_MAX )
*samples = *samples * self->audio_codec[ self->audio_index ]->sample_rate / *frequency;
while ( ret >= 0 && !got_audio )
if ( index < MAX_AUDIO_STREAMS && ret >= 0 && pkt.data && pkt.size > 0 && ( index == self->audio_index ||
( self->audio_index == INT_MAX && context->streams[ index ]->codec->codec_type == CODEC_TYPE_AUDIO ) ) )
{
- int channels2 = ( self->audio_index == INT_MAX || !self->audio_resample[index] ) ?
- self->audio_codec[index]->channels : *channels;
+ int channels2 = self->audio_codec[index]->channels;
ret = decode_audio( self, &ignore[index], pkt, channels2, *samples, real_timecode, fps );
}
// Set some additional return values
*format = mlt_audio_s16;
- if ( self->audio_index != INT_MAX && !self->audio_resample[ self->audio_index ] )
+ if ( self->audio_index != INT_MAX )
{
index = self->audio_index;
*channels = self->audio_codec[ index ]->channels;
else if ( self->audio_index == INT_MAX )
{
for ( index = 0; index < index_max; index++ )
- if ( self->audio_codec[ index ] && !self->audio_resample[ index ] )
+ if ( self->audio_codec[ index ] )
{
// XXX: This only works if all audio tracks have the same sample format.
*format = pick_audio_format( self->audio_codec[ index ]->sample_fmt );
int i;
for ( i = 0; i < MAX_AUDIO_STREAMS; i++ )
{
- if ( self->audio_resample[i] )
- audio_resample_close( self->audio_resample[i] );
mlt_pool_release( self->audio_buffer[i] );
av_free( self->decode_buffer[i] );
if ( self->audio_codec[i] )
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