我有一个视频提要,用ADPCM编解码器发送给我音频。然而,android只支持PCM格式。如何将ADPCM音频提要转换为PCM音频提要?
对此的答案可能类似于this question的答案。
我已经用以下代码成功地解码了该帧:
int len = avcodec_decode_audio4(pAudioCodecCtx, pAudioFrame, &frameFinished, &packet);这里的秘密是使用反向编码函数吗?
以下是我到目前为止在音频解码功能中所拥有的内容:
<!-- language: c -->
if(packet_queue_get(env, javaThread, pAudioPacketQueue, &packet, 1) < 0) {
LOGE("audio - after get packet failed");
return;
}
LOGD("Dequeued audio packet");
// calculate frame size
int frameSize;
if (pPcmAudioCodecCtx->frame_size) {
frameSize = pPcmAudioCodecCtx->frame_size;
} else {
/* if frame_size is not set, the number of samples must be
* calculated from the buffer size */
int64_t nb_samples = (int64_t)AUDIO_PCM_OUTBUFF_SIZE * 8 /
(av_get_bits_per_sample(pPcmAudioCodecCtx->codec_id) *
pPcmAudioCodecCtx->channels);
frameSize = nb_samples;
}
int pcmBytesPerSample = av_get_bytes_per_sample(pPcmAudioCodecCtx->sample_fmt);
int pcmFrameBytes = frameSize * pcmBytesPerSample * pPcmAudioCodecCtx->channels;
uint8_t *pDataStart = packet.data;
while(packet.size > 0) {
int len = avcodec_decode_audio4(pAudioCodecCtx, pAudioFrame, &frameFinished, &packet);
LOGD("Decoded ADPCM frame");
if (len < 0) {
LOGE("Error while decoding audio");
return;
}
if (frameFinished) {
// store frame data in FIFO buffer
uint8_t *inputBuffer = pAudioFrame->data[0];
int inputBufferSize = pAudioFrame->linesize[0];
av_fifo_generic_write(fifoBuffer, inputBuffer, inputBufferSize, NULL);
LOGD("Added ADPCM frame to FIFO buffer");
// check if fifo buffer has enough data for a PCM frame
while (av_fifo_size(fifoBuffer) >= pcmFrameBytes) {
LOGI("PCM frame data in FIFO buffer");
// read frame's worth of data from FIFO buffer
av_fifo_generic_read(fifoBuffer, pAudioPcmOutBuffer, pcmFrameBytes, NULL);
LOGD("Read data from FIFO buffer into pcm frame");
avcodec_get_frame_defaults(pPcmAudioFrame);
LOGD("Got frame defaults");
pPcmAudioFrame->nb_samples = pcmFrameBytes / (pPcmAudioCodecCtx->channels *
pcmBytesPerSample);
avcodec_fill_audio_frame(pPcmAudioFrame, pPcmAudioCodecCtx->channels,
pPcmAudioCodecCtx->sample_fmt,
pAudioPcmOutBuffer, pcmFrameBytes, 1);
LOGD("Filled frame audio with data");
// fill audio play buffer
int dataSize = pPcmAudioFrame->linesize[0];
LOGD("Data to output: %d", dataSize);
jbyteArray audioPlayBuffer = (jbyteArray) env->GetObjectField(ffmpegCtx, env->GetFieldID(cls, "audioPlayBuffer", "[B"));
jbyte *bytes = env->GetByteArrayElements(audioPlayBuffer, NULL);
memcpy(bytes, pPcmAudioFrame->data[0], dataSize);
env->ReleaseByteArrayElements(audioPlayBuffer, bytes, 0);
LOGD("Copied data into Java array");
env->CallVoidMethod(player, env->GetMethodID(playerCls, "updateAudio", "(I)V"), dataSize);
}发布于 2013-01-26 18:07:00
事实证明,audio_decode_函数返回16位PCM格式,而我只是不知道如何正确地访问它。
下面是基于avcodec_decode_audio4播放音频的数据包循环中修改的代码。
int len = avcodec_decode_audio4(pAudioCodecCtx, pAudioFrame, &frameFinished, &packet);
if (len < 0) {
LOGE("Error while decoding audio");
return;
}
if (frameFinished) {
int planeSize;
uint8_t *pcmBuffer = pAudioFrame->extended_data[0];
int dataSize = av_samples_get_buffer_size(&planeSize, pAudioCodecCtx->channels,
pAudioFrame->nb_samples,
pAudioCodecCtx->sample_fmt, 1);
// fill audio play buffer
jbyteArray audioPlayBuffer = (jbyteArray) env->GetObjectField(ffmpegCtx, env->GetFieldID(cls, "audioPlayBuffer", "[B"));
jbyte *bytes = env->GetByteArrayElements(audioPlayBuffer, NULL);
memcpy(bytes, pcmBuffer, dataSize);
env->ReleaseByteArrayElements(audioPlayBuffer, bytes, 0);
env->CallVoidMethod(player, env->GetMethodID(playerCls, "updateAudio", "(I)V"), dataSize);
}发布于 2013-01-24 08:40:02
您可以在8c-example.html上看到示例代码
参见audio_encode_example函数。
https://stackoverflow.com/questions/14492678
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