Audio Decoding

You can call native APIs to perform audio decoding, which decodes media data into PCM streams.

For details about the supported decoding capabilities, see AVCodec Supported Formats.

When to Use

Audio playback

Decode audio and transmit the data to the speaker for playing.
Audio rendering

Decode audio and transmit the data to the audio processing module for audio rendering.
Audio editing

Decode audio and transmit the data for audio editing (for example, adjusting the playback speed of a channel). Audio editing is performed based on PCM streams.

NOTE

Streams generated in the MP3 audio encoding process cannot be directly decoded through the MP3 audio decoding process. The following process is recommended: PCM stream -> MP3 audio encoding -> multiplexing -> demultiplexing -> MP3 audio decoding.

Development Guidelines

Read the API reference.

Refer to the code snippet below to complete the entire audio decoding process, including creating a decoder, setting decoding parameters (such as the sample rate, bit rate, and audio channel count), and starting, refreshing, resetting, and destroying the decoder.

During application development, you must call the APIs in the defined sequence. Otherwise, the system may throw exceptions or exhibit other undefined behaviors.

The figure below shows the call relationship of audio decoding.

The dotted line indicates an optional operation.
The solid line indicates a mandatory operation.

Call relationship of audio decoding

Linking the Dynamic Libraries in the CMake Script

target_link_libraries(sample PUBLIC libnative_media_codecbase.so)
target_link_libraries(sample PUBLIC libnative_media_core.so)
target_link_libraries(sample PUBLIC libnative_media_acodec.so)

NOTE

The word sample in the preceding code snippet is only an example. Use the actual project directory name.

How to Develop

Add the required header files.

#include <multimedia/player_framework/native_avcodec_audiocodec.h>
#include <multimedia/native_audio_channel_layout.h>
#include <multimedia/player_framework/native_avcapability.h>
#include <multimedia/player_framework/native_avcodec_base.h>
#include <multimedia/player_framework/native_avformat.h>
#include <multimedia/player_framework/native_avbuffer.h>

Create a decoder instance. In the code snippet below, OH_AVCodec * is the pointer to the decoder instance created.

You can create a decoder by MIME type or codec name.

Method 1: Create a decoder by MIME type.

// Specify whether encoding is used. The value false means decoding.
bool isEncoder = false;
// Create a decoder by MIME type.
OH_AVCodec *audioDec_ = OH_AudioCodec_CreateByMime(OH_AVCODEC_MIMETYPE_AUDIO_AAC, isEncoder);

Method 2: Create a decoder by codec name.

// Create a decoder by name.
OH_AVCapability *capability = OH_AVCodec_GetCapability(OH_AVCODEC_MIMETYPE_AUDIO_AAC, false);
const char *name = OH_AVCapability_GetName(capability);
OH_AVCodec *audioDec_ = OH_AudioCodec_CreateByName(name);

Add the header files and namespace.

#include <mutex>
#include <queue>
// Namespace of the C++ standard library.
using namespace std;

The sample code is as follows:

// Initialize the queues.
class ADecBufferSignal {
public:
    std::mutex inMutex_;
    std::mutex outMutex_;
    std::mutex startMutex_;
    std::condition_variable inCond_;
    std::condition_variable outCond_;
    std::condition_variable startCond_;
    std::queue<uint32_t> inQueue_;
    std::queue<uint32_t> outQueue_;
    std::queue<OH_AVBuffer *> inBufferQueue_;
    std::queue<OH_AVBuffer *> outBufferQueue_;
};
ADecBufferSignal *signal_;

Call OH_AudioCodec_RegisterCallback() to register callback functions.

OH_AVCodecOnError, a callback used to report a codec operation error
OH_AVCodecOnStreamChanged, a callback used to report stream information changes, including changes in the sample rate, audio channel count, and audio sample format. The decoding formats that can detect these changes include AAC, FLAC, MP3, and VORBIS. (This callback is supported since API version 15.)
OH_AVCodecOnNeedInputBuffer, a callback used to report input data required, which means that the decoder is ready for receiving data
OH_AVCodecOnNewOutputBuffer, a callback used to report output data generated, which means that decoding is complete

You need to process the callback functions to ensure that the decoder runs properly.

NOTE

Do not call the decoder APIs or perform time-consuming operations in the callbacks.

// Implement the OH_AVCodecOnError callback function.
static void OnError(OH_AVCodec *codec, int32_t errorCode, void *userData)
{
    (void)codec;
    (void)errorCode;
    (void)userData;
}
// Implement the OH_AVCodecOnStreamChanged callback function.
static void OnOutputFormatChanged(OH_AVCodec *codec, OH_AVFormat *format, void *userData)
{
    (void)codec;
    (void)userData;
    // Callback processing after the decoding output parameters are changed. The application performs processing as required.
    int32_t sampleRate;
    int32_t channelCount;
    int32_t sampleFormat;
    if (OH_AVFormat_GetIntValue(format, OH_MD_KEY_AUD_SAMPLE_RATE, &sampleRate)) {
        // Check whether the sample rate changes and perform processing as required.
    }
    if (OH_AVFormat_GetIntValue(format, OH_MD_KEY_AUD_CHANNEL_COUNT, &channelCount)) {
        // Check whether the number of audio channels changes and perform processing as required.
    }
    if (OH_AVFormat_GetIntValue(format, OH_MD_KEY_AUDIO_SAMPLE_FORMAT, &sampleFormat)) {
        // Check whether the audio sample format changes and perform processing as required.
    }
}
// Implement the OH_AVCodecOnNeedInputBuffer callback function.
static void OnInputBufferAvailable(OH_AVCodec *codec, uint32_t index, OH_AVBuffer *data, void *userData)
{
    (void)codec;
    ADecBufferSignal *signal = static_cast<ADecBufferSignal *>(userData);
    unique_lock<mutex> lock(signal->inMutex_);
    signal->inQueue_.push(index);
    signal->inBufferQueue_.push(data);
    signal->inCond_.notify_all();
    // The input stream is sent to inBufferQueue_.
}
// Implement the OH_AVCodecOnNewOutputBuffer callback function.
static void OnOutputBufferAvailable(OH_AVCodec *codec, uint32_t index, OH_AVBuffer *data, void *userData)
{
    (void)codec;
    ADecBufferSignal *signal = static_cast<ADecBufferSignal *>(userData);
    unique_lock<mutex> lock(signal->outMutex_);
    signal->outQueue_.push(index);
    signal->outBufferQueue_.push(data);
    signal->outCond_.notify_all();
    // The index of the output buffer is sent to outQueue_.
    // The decoded data is sent to outBufferQueue_.
}

Configure the callback information.

signal_ = new ADecBufferSignal();
OH_AVCodecCallback cb_ = {&OnError, &OnOutputFormatChanged, &OnInputBufferAvailable, &OnOutputBufferAvailable};
// Set the asynchronous callbacks.
int32_t ret = OH_AudioCodec_RegisterCallback(audioDec_, cb_, signal_);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}

(Optional) Call OH_AudioCodec_SetDecryptionConfig to set the decryption configuration.

Call this API after the media key system information is obtained but before Prepare() is called. For details about how to obtain such information, see step 4 in Media Data Demultiplexing.

For details about DRM APIs, see DRM.

Add the header files.

#include <multimedia/drm_framework/native_mediakeysystem.h>
#include <multimedia/drm_framework/native_mediakeysession.h>
#include <multimedia/drm_framework/native_drm_err.h>
#include <multimedia/drm_framework/native_drm_common.h>

Link the dynamic libraries in the CMake script.

target_link_libraries(sample PUBLIC libnative_drm.so)

The sample code is as follows:

// Create a media key system based on the media key system information. The following uses com.clearplay.drm as an example.
MediaKeySystem *system = nullptr;
int32_t ret = OH_MediaKeySystem_Create("com.clearplay.drm", &system);
if (system == nullptr) {
    printf("create media key system failed");
    return;
}

// Create a media key session.
MediaKeySession *session = nullptr;
DRM_ContentProtectionLevel contentProtectionLevel = CONTENT_PROTECTION_LEVEL_SW_CRYPTO;
ret = OH_MediaKeySystem_CreateMediaKeySession(system, &contentProtectionLevel, &session);
if (ret != DRM_OK) {
    // If the creation fails, refer to the DRM interface document and check logs.
    printf("create media key session failed.");
    return;
}
if (session == nullptr) {
    printf("media key session is nullptr.");
    return;
}
// Generate a media key request and set the response to the media key request.
// Set the decryption configuration, that is, set the decryption session and secure channel flag to the decoder. (Currently, the secure channel is not supported for audio decryption and therefore the secure channel flag should be set to false.)
bool secureAudio = false;
ret = OH_AudioCodec_SetDecryptionConfig(audioDec_, session, secureAudio);

Call OH_AudioCodec_Configure() to configure the decoder.

Key values of configuration options are described as follows:

Audio decoder key configuration

The sample below lists the value range of each audio decoding type.

Audio decoder format range description

Starting from API version 20, you can query the sample rate range. The following audio decoding types support decoding of any sample rate within their range:

Audio Decoding Type	Sample Rate (Hz)
FLAC	8000 – 384000
Vorbis	8000 – 192000
APE	1 – 2147483647

// (Mandatory) Configure the audio sample rate.
constexpr uint32_t DEFAULT_SAMPLERATE = 44100;
// (Optional) Configure the audio bit rate.
constexpr uint32_t DEFAULT_BITRATE = 32000;
// (Mandatory) Configure the audio channel count.
constexpr uint32_t DEFAULT_CHANNEL_COUNT = 2;
// (Optional) Configure the maximum input length.
constexpr uint32_t DEFAULT_MAX_INPUT_SIZE = 1152;
// Configure whether to use ADTS decoding (optional for AAC decoding).
constexpr uint32_t DEFAULT_AAC_TYPE = 1;
// (Optional) Configure the byte count of each audio data block. This parameter is supported since API version 22 and must be configured only for WMAV1, WMAV2, and WMA PRO decoding.
constexpr int32_t DEFAULT_BLOCK_ALIGN = 1;
OH_AVFormat *format = OH_AVFormat_Create();
// Set the format.
OH_AVFormat_SetIntValue(format, OH_MD_KEY_AUD_SAMPLE_RATE, DEFAULT_SAMPLERATE);
OH_AVFormat_SetIntValue(format, OH_MD_KEY_BITRATE, DEFAULT_BITRATE);
OH_AVFormat_SetIntValue(format, OH_MD_KEY_AUD_CHANNEL_COUNT, DEFAULT_CHANNEL_COUNT);
OH_AVFormat_SetIntValue(format, OH_MD_KEY_MAX_INPUT_SIZE, DEFAULT_MAX_INPUT_SIZE);
OH_AVFormat_SetIntValue(format, OH_MD_KEY_AAC_IS_ADTS, DEFAULT_AAC_TYPE);
OH_AVFormat_SetIntValue(format, OH_MD_KEY_BLOCK_ALIGN, DEFAULT_BLOCK_ALIGN);
// Configure the decoder.
int32_t ret = OH_AudioCodec_Configure(audioDec_, format);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}

Call OH_AudioCodec_Prepare() to prepare internal resources for the decoder.

int32_t ret = OH_AudioCodec_Prepare(audioDec_);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}

Call OH_AudioCodec_Start() to start the decoder.

Add the header file.

#include <fstream>

The sample code is as follows:

ifstream inputFile_;
ofstream outFile_;

// Set the input file path based on the actual situation.
const char* inputFilePath = "/";
// Set the output file path based on the actual situation.
const char* outputFilePath = "/";
// Open the path of the binary file to be decoded.
inputFile_.open(inputFilePath, ios::in | ios::binary); 
// Set the path of the output file.
outFile_.open(outputFilePath, ios::out | ios::binary);
// Start decoding.
int32_t ret = OH_AudioCodec_Start(audioDec_);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}

(Optional) Call OH_AVCencInfo_SetAVBuffer() to set the Common Encryption Scheme (CENC) information.

If the content being played is DRM encrypted and demultiplexing is performed by the upper-layer application, call OH_AVCencInfo_SetAVBuffer() to set the CENC information to the AVBuffer so that the media data can be decrypted in the AVBuffer.

Add the header file.

#include <multimedia/player_framework/native_cencinfo.h>

Link the dynamic libraries in the CMake script.

target_link_libraries(sample PUBLIC libnative_media_avcencinfo.so)

The sample code is as follows:

auto buffer = signal_->inBufferQueue_.front();
uint32_t keyIdLen = DRM_KEY_ID_SIZE;
uint8_t keyId[] = {
    0xd4, 0xb2, 0x01, 0xe4, 0x61, 0xc8, 0x98, 0x96,
    0xcf, 0x05, 0x22, 0x39, 0x8d, 0x09, 0xe6, 0x28};
uint32_t ivLen = DRM_KEY_IV_SIZE;
uint8_t iv[] = {
    0xbf, 0x77, 0xed, 0x51, 0x81, 0xde, 0x36, 0x3e,
    0x52, 0xf7, 0x20, 0x4f, 0x72, 0x14, 0xa3, 0x95};
uint32_t encryptedBlockCount = 0;
uint32_t skippedBlockCount = 0;
uint32_t firstEncryptedOffset = 0;
uint32_t subsampleCount = 1;
DrmSubsample subsamples[1] = { {0x10, 0x16} };
// Create a CencInfo instance.
OH_AVCencInfo *cencInfo = OH_AVCencInfo_Create();
if (cencInfo == nullptr) {
    // Handle exceptions.
}
// Set the decryption algorithm.
OH_AVErrCode errNo = OH_AVCencInfo_SetAlgorithm(cencInfo, DRM_ALG_CENC_AES_CTR);
if (errNo != AV_ERR_OK) {
    // Handle exceptions.
}
// Set KeyId and Iv.
errNo = OH_AVCencInfo_SetKeyIdAndIv(cencInfo, keyId, keyIdLen, iv, ivLen);
if (errNo != AV_ERR_OK) {
    // Handle exceptions.
}
// Set the sample information.
errNo = OH_AVCencInfo_SetSubsampleInfo(cencInfo, encryptedBlockCount, skippedBlockCount, firstEncryptedOffset,
    subsampleCount, subsamples);
if (errNo != AV_ERR_OK) {
    // Handle exceptions.
}
// Set the mode. KeyId, Iv, and SubSamples have been set.
errNo = OH_AVCencInfo_SetMode(cencInfo, DRM_CENC_INFO_KEY_IV_SUBSAMPLES_SET);
if (errNo != AV_ERR_OK) {
    // Handle exceptions.
}
// Set CencInfo to the AVBuffer.
errNo = OH_AVCencInfo_SetAVBuffer(cencInfo, buffer);
if (errNo != AV_ERR_OK) {
    // Handle exceptions.
}
// Destroy the CencInfo instance.
errNo = OH_AVCencInfo_Destroy(cencInfo);
if (errNo != AV_ERR_OK) {
    // Handle exceptions.
}

Call OH_AudioCodec_PushInputBuffer() to write the data to decode.

You should fill in complete input data before calling this API.

When finished, set the flags to AVCODEC_BUFFER_FLAGS_EOS.

uint32_t index = signal_->inQueue_.front();
auto buffer = signal_->inBufferQueue_.front();
int32_t size;
int64_t pts;
// size is the length of each frame of the data to decode. pts is the timestamp of each frame and is used to indicate when the audio should be played.
// The values of size and pts are obtained from an audio and video resource file or data stream to decode.
// In the case of an audio and video resource file, the values are obtained from the buffer in the demultiplexed OH_AVDemuxer_ReadSampleBuffer.
// In the case of a data stream, the values are obtained from the data stream provider.
// In this example, the values of size and pts are obtained from the test file.
inputFile_.read(reinterpret_cast<char *>(&size), sizeof(size));
inputFile_.read(reinterpret_cast<char *>(&pts), sizeof(pts));
inputFile_.read((char *)OH_AVBuffer_GetAddr(buffer), size);
OH_AVCodecBufferAttr attr = {0};
if (inputFile_.eof()) {
    attr.size = 0;
    attr.flags = AVCODEC_BUFFER_FLAGS_EOS;
} else {
    attr.size = size;
    attr.flags = AVCODEC_BUFFER_FLAGS_NONE;
}
attr.pts = pts;
OH_AVBuffer_SetBufferAttr(buffer, &attr);
int32_t ret = OH_AudioCodec_PushInputBuffer(audioDec_, index);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}

Call OH_AudioCodec_FreeOutputBuffer() to release the decoded data.

Once you have retrieved the decoded PCM stream, call OH_AudioCodec_FreeOutputBuffer() to free up the data.

uint32_t index = signal_->outQueue_.front();
OH_AVBuffer *data = signal_->outBufferQueue_.front();
if (data == nullptr) {
    // Handle exceptions.
}
// Obtain the buffer attributes.
OH_AVCodecBufferAttr attr = {0};
int32_t ret = OH_AVBuffer_GetBufferAttr(data, &attr);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}
// Write the decoded data (specified by data) to the output file.
outFile_.write(reinterpret_cast<char *>(OH_AVBuffer_GetAddr(data)), attr.size);
ret = OH_AudioCodec_FreeOutputBuffer(audioDec_, index);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}
if (attr.flags == AVCODEC_BUFFER_FLAGS_EOS) {
    // End.
}

(Optional) Call OH_AudioCodec_Flush() to refresh the decoder.

After OH_AudioCodec_Flush() is called, the decoder remains in the running state, but the current queue is cleared and the buffer storing the decoded data is freed.

To continue decoding, you must call OH_AudioCodec_Start() again.

You need to call OH_AudioCodec_Start() in the following cases:
- To use the same decoder configuration after AVCODEC_BUFFER_FLAGS_EOS of the output buffer is set, call OH_AudioCodec_Flush() to refresh the decoder.
- If a recoverable error occurs during the execution (OH_AudioCodec_IsValid() returns true), you can call OH_AudioCodec_Flush() to refresh the decoder and then call OH_AudioCodec_Start() to start decoding again.
```
// Refresh the decoder.
int32_t ret = OH_AudioCodec_Flush(audioDec_);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}
// Start decoding again.
ret = OH_AudioCodec_Start(audioDec_);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}
```
(Optional) Call OH_AudioCodec_Reset() to reset the decoder.

After OH_AudioCodec_Reset() is called, the decoder returns to the initialized state. The input and output buffers obtained before the reset cannot be used. You must call OH_AudioCodec_Configure() to reconfigure the decoder and then call OH_AudioCodec_Start() to start decoding again. Obtain the input and output buffers again after the decoder is started.
```
// Reset the decoder.
int32_t ret = OH_AudioCodec_Reset(audioDec_);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}
// Reconfigure the decoder.
ret = OH_AudioCodec_Configure(audioDec_, format);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}
```
Call OH_AudioCodec_Stop() to stop the decoder.

After the decoder is stopped, you can call OH_AudioCodec_Start() to start it again. The input and output buffers obtained before the decoder is stopped cannot be reused. You must obtain them again after the decoder is started.
```
// Stop the decoder.
int32_t ret = OH_AudioCodec_Stop(audioDec_);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
}
```

Call OH_AudioCodec_Destroy() to destroy the decoder instance and release resources.

NOTE

You only need to call this API once.

// Call OH_AudioCodec_Destroy to destroy the decoder.
int32_t ret = OH_AudioCodec_Destroy(audioDec_);
if (ret != AV_ERR_OK) {
    // Handle exceptions.
} else {
    audioDec_ = NULL; // The decoder cannot be destroyed repeatedly.
}

Samples

The following sample is provided to help you better understand how to develop audio decoding:

Audio Decoding