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The Microsoft Media Foundation AAC encoder is a Media Foundation Transform that encodes Advanced Audio Coding (AAC) Low Complexity (LC) profile, as defined by ISO/IEC 13818-7 (MPEG-2 Audio Part 7) .

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The AAC encoder does not support encoding to any other AAC profiles, such as Main, SSR, or LTP.

Class Identifier

The class identifier (CLSID) of the AAC encoder is CLSID_AACMFTEncoder, defined in the header file wmcodecdsp.h.

Media Types

The AAC encoder supports the following media types. You can set the types in either order input type first, or output type first.

Input Types

Set the following attributes on the input media type.

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After the input type is set, the encoder derives the following values and adds them to the media type:

Output Types

Set the following attributes on the output media type.

The following table lists the values that can be used for the MF_MT_AAC_PROFILE_LEVEL_INDICATION attribute.

After the output type is set, the AAC encoder updates the type by adding the MF_MT_USER_DATA attribute. This attribute contains the portion of the HEAACWAVEINFO structure that appears after the WAVEFORMATEX structure (that is, after the wfx member). This is followed by the AudioSpecificConfig() data, as defined by ISO/IEC 14496-3.

Each output sample contains one compressed AAC frame with no header. This format is equivalent to the raw_data_block() element defined by MPEG-2. The MF_MT_AAC_PAYLOAD_TYPE attribute, if present in the output type, must be set to zero to indicate this payload type.

Each output sample contains one compressed AAC frame corresponding to 1024 PCM samples. For example, at 48 Khz sampling rate, the duration of one compressed frame is 21.33 msec.

If MF_MT_AAC_PAYLOAD_TYPE is zero (the default value), each output sample contains one raw_data_block() element as defined by ISO/IEC 13818-7.

Example Media Types

Here is an example of the media types needed to encode from 44.1-kHz, 160-Kbps stereo audio to raw AAC

Input media type:

Output media type:

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Remarks

In the current implementation, every input sample must have a valid time and duration. To set the sample time, call IMFSample::SetSampleTime. To set the sample duration, call IMFSample::SetSampleDuration.

If the sample time is not set, the encoder's IMFTransform::ProcessInput method returns MF_E_NO_SAMPLE_TIMESTAMP. If the sample duration is not set, the ProcessInput method returns MF_E_NO_SAMPLE_DURATION.

Sample duration can be calculated as follows:

where nAudioSamplesPerChannel is the number of PCM audio samples per channel in the input buffer, and nSamplesPerSec is the sampling rate, in samples per second.

Note

Due to a bug in the current implementation, if the sample duration is set to zero, the ProcessInput call succeeds, but a subsequent call to IMFTransform::ProcessOutput will throw a divide-by-zero exception. To avoid this error, set a valid nonzero duration on each input sample.

Requirements

See also