Fix audio buffer overflow on large resampling (2902193).

[mlt] / src / modules / avformat / producer_avformat.c

diff --git a/src/modules/avformat/producer_avformat.c b/src/modules/avformat/producer_avformat.c
index 6fd930d958eb268944863f630e0ca1f9a495d19c..2d0be647bc3c5fe446c602d8cf597cd945eb70ee 100644 (file)
--- a/src/modules/avformat/producer_avformat.c
+++ b/src/modules/avformat/producer_avformat.c
@@ -85,6 +85,7 @@ struct producer_avformat_s
        int max_channel;
        int max_frequency;
        unsigned int invalid_pts_counter;
+       double resample_factor;
 };
 typedef struct producer_avformat_s *producer_avformat;
 
@@ -154,6 +155,8 @@ mlt_producer producer_avformat_init( mlt_profile profile, char *file )
 
                        // Register our get_frame implementation
                        producer->get_frame = producer_get_frame;
+                       
+                       this->resample_factor = 1.0;
 
                        // Open the file
                        if ( producer_open( this, profile, file ) != 0 )
@@ -1417,26 +1420,28 @@ static int decode_audio( producer_avformat this, int *ignore, AVPacket *pkt, int
                // If decoded successfully
                if ( data_size > 0 )
                {
+                       // Figure out how many samples will be needed after resampling
+                       int convert_samples = data_size / codec_context->channels / ( av_get_bits_per_sample_format( codec_context->sample_fmt ) / 8 );
+                       int samples_needed = lrint( this->resample_factor * convert_samples );
+                       
                        // Resize audio buffer to prevent overflow
-                       if ( audio_used * channels + data_size > this->audio_buffer_size[ index ] )
+                       if ( audio_used * channels + samples_needed > this->audio_buffer_size[ index ] )
                        {
-                               mlt_pool_release( this->audio_buffer[ index ] );
-                               this->audio_buffer_size[ index ] = audio_used * channels * sizeof(int16_t) + data_size * 2;
-                               audio_buffer = this->audio_buffer[ index ] = mlt_pool_alloc( this->audio_buffer_size[ index ] );
+                               this->audio_buffer_size[ index ] *= 2;
+                               audio_buffer = this->audio_buffer[ index ] = mlt_pool_realloc( audio_buffer, this->audio_buffer_size[ index ] * sizeof(int16_t) );
                        }
                        if ( resample )
                        {
                                // Copy to audio buffer while resampling
                                int16_t *source = decode_buffer;
                                int16_t *dest = &audio_buffer[ audio_used * channels ];
-                               int convert_samples = data_size / codec_context->channels / ( av_get_bits_per_sample_format( codec_context->sample_fmt ) / 8 );
                                audio_used += audio_resample( resample, dest, source, convert_samples );
                        }
                        else
                        {
                                // Straight copy to audio buffer
                                memcpy( &audio_buffer[ audio_used * codec_context->channels ], decode_buffer, data_size );
-                               audio_used += data_size / codec_context->channels / ( av_get_bits_per_sample_format( codec_context->sample_fmt ) / 8 );
+                               audio_used += convert_samples;
                        }
 
                        // Handle ignore
@@ -1493,7 +1498,7 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format
 
        // Fetch the audio_format
        AVFormatContext *context = this->audio_format;
-
+       
        // Determine the tracks to use
        int index = this->audio_index;
        int index_max = this->audio_index + 1;
@@ -1516,6 +1521,12 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format
                        // 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 = this->audio_index == INT_MAX ? 1 : (double) *channels / codec_context->channels;
+                               resample_factor *= (double) *frequency / codec_context->sample_rate;
+                               if ( resample_factor > this->resample_factor )
+                                       this->resample_factor = resample_factor;
+                               
                                // Create the resampler
 #if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0))
                                this->audio_resample[ index ] = av_audio_resample_init(
@@ -1534,8 +1545,8 @@ static int producer_get_audio( mlt_frame frame, void **buffer, mlt_audio_format
                        }
 
                        // Check for audio buffer and create if necessary
-                       this->audio_buffer[ index ] = mlt_pool_alloc( AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t ) );
                        this->audio_buffer_size[ index ] = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+                       this->audio_buffer[ index ] = mlt_pool_alloc( this->audio_buffer_size[ index ] * sizeof( int16_t ) );
 
                        // Check for decoder buffer and create if necessary
                        this->decode_buffer[ index ] = av_malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t ) );