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@ -164,9 +164,6 @@ The contents of each file is described in detail in this document. All header
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files are provided for usage in specific C/C++ programs. The main AAC decoder
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library API functions are located in aacdecoder_lib.h header file.
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In binary releases the decoder core resides in statically linkable libraries,
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for example libAACdec.a.
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\section Calling_Sequence Calling Sequence
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@ -174,19 +171,7 @@ The following sequence is necessary for proper decoding of ISO/MPEG-2/4 AAC,
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HE-AAC v2, or MPEG-D USAC bitstreams. In the following description, input stream
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read and output write function details are left out, since they may be
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implemented in a variety of configurations depending on the user's specific
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requirements. The example implementation uses file-based input/output, and in
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such case one may call mpegFileRead_Open() to open an input file and to allocate
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memory for the required structures, and the corresponding mpegFileRead_Close()
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to close opened files and to de-allocate associated structures.
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mpegFileRead_Open() will attempt to detect the bitstream format and in case of
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MPEG-4 file format or Raw Packets file format (a proprietary Fraunhofer IIS file
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format suitable only for testing) it will read the Audio Specific Config data
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(ASC). An unsuccessful attempt to recognize the bitstream format requires the
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user to provide this information manually. For any other bitstream formats that
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are usually applicable in streaming applications, the decoder itself will try to
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synchronize and parse the given bitstream fragment using the FDK transport
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library. Hence, for streaming applications (without file access) this step is
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not necessary.
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requirements.
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-# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder
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@ -205,19 +190,17 @@ do {
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working memory (a client-supplied input buffer "inBuffer" in framework). This
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buffer will be used to load AAC bitstream data to the decoder. Only when all
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data in this buffer has been processed will the decoder signal an empty buffer.
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For file-based input, you may invoke mpegFileRead_Read() to acquire new
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bitstream data.
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-# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer
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with the client-supplied bitstream input buffer. Note, if the data loaded in to
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the internal buffer is not sufficient to decode a frame,
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aacDecoder_DecodeFrame() will return ::AAC_DEC_NOT_ENOUGH_BITS until a
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sufficient amount of data is loaded in to the internal buffer. For streaming
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formats (ADTS, LOAS), it is acceptable to load more than one frame to the
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decoder. However, for RAW file format (Fraunhofer IIS proprietary format), only
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one frame may be loaded to the decoder per aacDecoder_DecodeFrame() call. For
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least amount of communication delay, fill and decode should be performed on a
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frame by frame basis. \code ErrorStatus = aacDecoder_Fill(aacDecoderInfo,
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inBuffer, bytesRead, bytesValid); \endcode
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decoder. However, for packed based formats, only one frame may be loaded to the
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decoder per aacDecoder_DecodeFrame() call. For least amount of communication
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delay, fill and decode should be performed on a frame by frame basis. \code
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ErrorStatus = aacDecoder_Fill(aacDecoderInfo, inBuffer, bytesRead,
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bytesValid); \endcode
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-# Call aacDecoder_DecodeFrame(). This function decodes one frame and writes
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decoded PCM audio data to a client-supplied buffer. It is the client's
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responsibility to allocate a buffer which is large enough to hold the decoded
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@ -225,12 +208,9 @@ output data. \code ErrorStatus = aacDecoder_DecodeFrame(aacDecoderInfo,
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TimeData, OUT_BUF_SIZE, flags); \endcode If the bitstream configuration (number
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of channels, sample rate, frame size) is not known a priori, you may call
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aacDecoder_GetStreamInfo() to retrieve a structure that contains this
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information. You may use this data to initialize an audio output device. In the
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example program, if the number of channels or the sample rate has changed since
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program start or the previously decoded frame, the audio output device is then
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re-initialized. If WAVE file output is chosen, a new WAVE file for each new
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stream configuration is be created. \code p_si =
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aacDecoder_GetStreamInfo(aacDecoderInfo); \endcode
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information. You may use this data to initialize an audio output device. \code
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p_si = aacDecoder_GetStreamInfo(aacDecoderInfo);
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\endcode
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-# Repeat steps 5 to 7 until no data is available to decode any more, or in case
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of error. \code } while (bytesRead[0] > 0 || doFlush || doBsFlush ||
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forceContinue); \endcode
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@ -239,7 +219,7 @@ structures. \code aacDecoder_Close(aacDecoderInfo); \endcode
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\image latex decode.png "Decode calling sequence" width=11cm
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\image latex change_source.png "Change data source sequence" width 5cm
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\image latex change_source.png "Change data source sequence" width=5cm
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\image latex conceal.png "Error concealment sequence" width=14cm
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@ -296,16 +276,14 @@ input buffer, and one to hold the decoded output PCM sample data. In resource
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limited applications, the output buffer may be reused as an external input
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buffer prior to the subsequence aacDecoder_Fill() function call.
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The external input buffer is set in the example program and its size is defined
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by ::IN_BUF_SIZE. You may freely choose different buffer sizes. To feed the data
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to the decoder-internal input buffer, use the function aacDecoder_Fill(). This
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function returns important information regarding the number of bytes in the
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external input buffer that have not yet been copied into the internal input
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buffer (variable bytesValid). Once the external buffer has been fully copied, it
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can be completely re-filled again. In case you wish to refill the buffer while
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there are unprocessed bytes (bytesValid is unequal 0), you should preserve the
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unconsumed data. However, we recommend to refill the buffer only when bytesValid
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returns 0.
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To feed the data to the decoder-internal input buffer, use the
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function aacDecoder_Fill(). This function returns important information
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regarding the number of bytes in the external input buffer that have not yet
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been copied into the internal input buffer (variable bytesValid). Once the
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external buffer has been fully copied, it can be completely re-filled again. In
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case you wish to refill the buffer while there are unprocessed bytes (bytesValid
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is unequal 0), you should preserve the unconsumed data. However, we recommend to
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refill the buffer only when bytesValid returns 0.
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The bytesValid parameter is an input and output parameter to the FDK decoder. As
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an input, it signals how many valid bytes are available in the external buffer.
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@ -340,10 +318,7 @@ explanation, please refer to ISO/IEC 13818-7:2005(E), chapter 8.5.3.2.
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In case a Program Config is included in the audio configuration, the channel
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mapping described within it will be adopted.
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In case of MPEG-D Surround the channel mapping will follow the same criteria
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described in ISO/IEC 13818-7:2005(E), but adding corresponding top channels (if
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available) to the channel types in order to avoid ambiguity. The examples below
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explain these aspects in detail.
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The examples below explain these aspects in detail.
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\section OutputFormatChange Changing the audio output format
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@ -689,9 +664,7 @@ typedef enum {
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2. If the parameter value is greater than that of
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::AAC_PCM_MAX_OUTPUT_CHANNELS both will be set to the same
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value. \n
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3. This parameter does not affect MPEG Surround processing.
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\n
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4. This parameter will be ignored if the number of encoded
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3. This parameter will be ignored if the number of encoded
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audio channels is greater than 8. */
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AAC_PCM_MAX_OUTPUT_CHANNELS =
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0x0012, /*!< Maximum number of PCM output channels. If lower than the
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@ -718,11 +691,7 @@ typedef enum {
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2. If the parameter value is greater than zero but smaller
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than ::AAC_PCM_MIN_OUTPUT_CHANNELS both will be set to same
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value. \n
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3. The operating mode of the MPEG Surround module will be
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set accordingly. \n
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4. Setting this parameter with any value will disable the
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binaural processing of the MPEG Surround module
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5. This parameter will be ignored if the number of encoded
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3. This parameter will be ignored if the number of encoded
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audio channels is greater than 8. */
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AAC_METADATA_PROFILE =
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0x0020, /*!< See ::AAC_MD_PROFILE for all available values. */
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@ -803,11 +772,11 @@ typedef enum {
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sequences for fading in and out, if provided in the
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bitstream.\n Enabled album mode makes use of dedicated album
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loudness information, if provided in the bitstream.\n */
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AAC_QMF_LOWPOWER = 0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing
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mode. \n -1: Use internal default. Implies MPEG
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Surround partially complex accordingly. \n 0:
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Use complex QMF data mode. \n 1: Use real (low
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power) QMF data mode. \n */
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AAC_QMF_LOWPOWER =
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0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing mode. \n
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-1: Use internal default. \n
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0: Use complex QMF data mode. \n
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1: Use real (low power) QMF data mode. \n */
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AAC_TPDEC_CLEAR_BUFFER =
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0x0603 /*!< Clear internal bit stream buffer of transport layers. The
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decoder will start decoding at new data passed after this event
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@ -1038,21 +1007,24 @@ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_Fill(HANDLE_AACDECODER self,
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const UINT bufferSize[],
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UINT *bytesValid);
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#define AACDEC_CONCEAL \
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1 /*!< Flag for aacDecoder_DecodeFrame(): Trigger the built-in error \
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concealment module to generate a substitute signal for one lost frame. \
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New input data will not be considered. */
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#define AACDEC_FLUSH \
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2 /*!< Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all \
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delayed audio without having new input data. Thus new input data will \
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not be considered.*/
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#define AACDEC_INTR \
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4 /*!< Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data \
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discontinuity. Resync any internals as necessary. */
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#define AACDEC_CLRHIST \
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8 /*!< Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and \
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history buffers. CAUTION: This can cause discontinuities in the output \
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signal. */
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/** Flag for aacDecoder_DecodeFrame(): Trigger the built-in error concealment
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* module to generate a substitute signal for one lost frame. New input data
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* will not be considered.
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*/
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#define AACDEC_CONCEAL 1
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/** Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all delayed
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* audio without having new input data. Thus new input data will not be
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* considered.
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*/
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#define AACDEC_FLUSH 2
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/** Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data
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* discontinuity. Resync any internals as necessary.
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*/
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#define AACDEC_INTR 4
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/** Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and history
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* buffers. CAUTION: This can cause discontinuities in the output signal.
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*/
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#define AACDEC_CLRHIST 8
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/**
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* \brief Decode one audio frame
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@ -1,7 +1,7 @@
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/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
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© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
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Forschung e.V. All rights reserved.
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1. INTRODUCTION
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@ -149,12 +149,6 @@ All API header files are located in the folder /include of the release package.
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All header files are provided for usage in C/C++ programs. The AAC encoder
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library API functions are located in aacenc_lib.h.
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In binary releases the encoder core resides in statically linkable libraries
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called for example libAACenc.a/libFDK.a (LINUX) or FDK_fastaaclib.lib (MS Visual
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C++) for the plain AAC-LC core encoder and libSBRenc.a (LINUX) or
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FDK_sbrEncLib.lib (MS Visual C++) for the SBR (Spectral Band Replication) and PS
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(Parametric Stereo) modules.
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\section CallingSequence Calling Sequence
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For encoding of ISO/MPEG-2/4 AAC bitstreams the following sequence is mandatory.
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@ -326,18 +320,12 @@ input buffer, simulating a modulo buffer: \code if (outargs.numInSamples>0) {
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\endcode
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\section writeOutData Output Bitstream Data
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If any AAC bitstream data is available, write it to output file or device. This
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can be done once the following condition is true: \code if
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(outargs.numOutBytes>0) {
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If any AAC bitstream data is available, write it to output file or device as
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follows. \code if (outargs.numOutBytes>0) { FDKfwrite(outputBuffer,
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outargs.numOutBytes, 1, pOutFile);
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}
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\endcode
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If you use file I/O then for example call mpegFileWrite_Write() from the library
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libMpegFileWrite \code mpegFileWrite_Write(hMpegFile, outputBuffer,
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outargs.numOutBytes, aacEncoder_GetParam(hAacEncoder, AACENC_GRANULE_LENGTH));
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\endcode
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\section cfgMetaData Meta Data Configuration
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If the present library is configured with Metadata support, it is possible to
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@ -427,7 +415,7 @@ switch (nChannels) {
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return chMode;
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\endcode
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\subsection bitreservoir Bitreservoir Configuration
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\subsection peakbitrate Peak Bitrate Configuration
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In AAC, the default bitreservoir configuration depends on the chosen bitrate per
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frame and the number of effective channels. The size can be determined as below.
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\f[
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@ -436,17 +424,10 @@ bitreservoir = nEffChannels*6144 - (bitrate*framelength/samplerate)
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Due to audio quality concerns it is not recommended to change the bitreservoir
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size to a lower value than the default setting! However, for minimizing the
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delay for streaming applications or for achieving a constant size of the
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bitstream packages in each frame, it may be necessaray to change the
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bitreservoir size. This can be done with the ::AACENC_PEAK_BITRATE parameter.
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\code
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bitstream packages in each frame, it may be necessaray to limit the maximum bits
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per frame size. This can be done with the ::AACENC_PEAK_BITRATE parameter. \code
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aacEncoder_SetParam(hAacEncoder, AACENC_PEAK_BITRATE, value);
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\endcode
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By setting ::AACENC_BITRATEMODE to fixed framing, the bitreservoir is disabled.
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A disabled bitreservoir results in a constant size for each bitstream package.
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Please note that especially at lower bitrates a disabled bitreservoir can
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downgrade the audio quality considerably! The default bitreservoir configuration
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can be achieved as follows. \code aacEncoder_SetParam(hAacEncoder,
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AACENC_BITRESERVOIR, -1); \endcode
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To achieve acceptable audio quality with a reduced bitreservoir size setting at
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least 1000 bits per audio channel is recommended. For a multichannel audio file
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@ -967,9 +948,7 @@ in this Fraunhofer IIS AAC encoder. AAC has been designed in that way.
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\subsection BEHAVIOUR_ESTIM_AVG_FRAMESIZES Estimating Average Frame Sizes
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A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel (there is
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also one mode with 1920 samples per channel but this is only for special
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purposes such as DAB+ digital radio).
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A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel.
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The number of HE-AAC frames \f$N\_FRAMES\f$ per second at 44.1 kHz is:
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@ -1082,9 +1061,7 @@ typedef struct AACENCODER *HANDLE_AACENCODER;
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typedef struct {
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UINT maxOutBufBytes; /*!< Maximum number of encoder bitstream bytes within one
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frame. Size depends on maximum number of supported
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channels in encoder instance. For superframing (as
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used for example in DAB+), size has to be a multiple
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accordingly. */
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channels in encoder instance. */
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UINT maxAncBytes; /*!< Maximum number of ancillary data bytes which can be
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inserted into bitstream within one frame. */
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