--- /dev/null
+/*
+ * audio conversion
+ * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ * Copyright (c) 2008 Peter Ross
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AUDIOCONVERT_H
+#define AVCODEC_AUDIOCONVERT_H
+
+/**
+ * @file audioconvert.h
+ * Audio format conversion routines
+ */
+
+
+#include "avcodec.h"
+
+
+/**
+ * Generate string corresponding to the sample format with
+ * number sample_fmt, or a header if sample_fmt is negative.
+ *
+ * @param[in] buf the buffer where to write the string
+ * @param[in] buf_size the size of buf
+ * @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
+ * a negative value to print the corresponding header.
+ * Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
+ */
+void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
+
+/**
+ * @return NULL on error
+ */
+const char *avcodec_get_sample_fmt_name(int sample_fmt);
+
+/**
+ * @return SAMPLE_FMT_NONE on error
+ */
+enum SampleFormat avcodec_get_sample_fmt(const char* name);
+
+/**
+ * @return NULL on error
+ */
+const char *avcodec_get_channel_name(int channel_id);
+
+/**
+ * Return description of channel layout
+ */
+void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout);
+
+/**
+ * Guess the channel layout
+ * @param nb_channels
+ * @param codec_id Codec identifier, or CODEC_ID_NONE if unknown
+ * @param fmt_name Format name, or NULL if unknown
+ * @return Channel layout mask
+ */
+int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name);
+
+
+struct AVAudioConvert;
+typedef struct AVAudioConvert AVAudioConvert;
+
+/**
+ * Create an audio sample format converter context
+ * @param out_fmt Output sample format
+ * @param out_channels Number of output channels
+ * @param in_fmt Input sample format
+ * @param in_channels Number of input channels
+ * @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
+ * @param flags See FF_MM_xx
+ * @return NULL on error
+ */
+AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
+ enum SampleFormat in_fmt, int in_channels,
+ const float *matrix, int flags);
+
+/**
+ * Free audio sample format converter context
+ */
+void av_audio_convert_free(AVAudioConvert *ctx);
+
+/**
+ * Convert between audio sample formats
+ * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
+ * @param[in] out_stride distance between consecutive input samples (measured in bytes)
+ * @param[in] in array of input buffers for each channel
+ * @param[in] in_stride distance between consecutive output samples (measured in bytes)
+ * @param len length of audio frame size (measured in samples)
+ */
+int av_audio_convert(AVAudioConvert *ctx,
+ void * const out[6], const int out_stride[6],
+ const void * const in[6], const int in_stride[6], int len);
+
+#endif /* AVCODEC_AUDIOCONVERT_H */
// ffmpeg Header files
#include <avformat.h>
#ifdef SWSCALE
-#include <swscale.h>
+# include <swscale.h>
+#endif
+#if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(71<<8)+0))
+# include "audioconvert.h"
#endif
// System header files
// Obtain the resample context if it exists (not always needed)
ReSampleContext *resample = mlt_properties_get_data( properties, "audio_resample", NULL );
- // Obtain the audio buffer
+#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" );
int paused = 0;
// Check for resample and create if necessary
- if ( resample == NULL && codec_context->channels <= 2 )
+ if ( resample == NULL && ( *frequency != codec_context->sample_rate || codec_context->channels <= 2 ) )
{
// Create the resampler
resample = audio_resample_init( *channels, codec_context->channels, *frequency, codec_context->sample_rate );
*frequency = codec_context->sample_rate;
}
+#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 );
+ }
+#endif
+
// Check for audio buffer and create if necessary
if ( audio_buffer == NULL )
{
mlt_properties_set_data( properties, "audio_buffer", audio_buffer, 0, ( mlt_destructor )mlt_pool_release, NULL );
}
+ // Check for decoder buffer and create if necessary
+ if ( decode_buffer == NULL )
+ {
+ // Allocate the audio buffer
+ decode_buffer = mlt_pool_alloc( AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t ) );
+
+ // And store it on properties for reuse
+ mlt_properties_set_data( properties, "decode_buffer", decode_buffer, 0, ( mlt_destructor )mlt_pool_release, NULL );
+ }
+
+ // 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 );
+ }
+
// Seek if necessary
if ( position != expected )
{
{
int ret = 0;
int got_audio = 0;
- int16_t *temp = av_malloc( sizeof( int16_t ) * AVCODEC_MAX_AUDIO_FRAME_SIZE );
av_init_packet( &pkt );
// Decode the audio
#if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(29<<8)+0))
- ret = avcodec_decode_audio2( codec_context, temp, &data_size, ptr, len );
+ ret = avcodec_decode_audio2( codec_context, decode_buffer, &data_size, ptr, len );
#else
- ret = avcodec_decode_audio( codec_context, temp, &data_size, ptr, len );
+ ret = avcodec_decode_audio( codec_context, decode_buffer, &data_size, ptr, len );
#endif
if ( ret < 0 )
{
if ( data_size > 0 )
{
- if ( resample != NULL )
+ 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 )
{
- audio_used += audio_resample( resample, &audio_buffer[ audio_used * *channels ], temp, data_size / ( codec_context->channels * sizeof( int16_t ) ) );
+ int16_t *source = decode_buffer;
+ int16_t *dest = &audio_buffer[ audio_used * *channels ];
+ int convert_samples = data_size / src_stride[0];
+
+#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 );
}
else
{
- memcpy( &audio_buffer[ audio_used * *channels ], temp, data_size );
- audio_used += data_size / ( codec_context->channels * sizeof( int16_t ) );
+#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];
}
// Handle ignore
// Store the number of audio samples still available
mlt_properties_set_int( properties, "_audio_used", audio_used );
-
- // Release the temporary audio
- av_free( temp );
}
else
{
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