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@ -20,28 +20,22 @@ void AudioStream::setAudioCallback(AudioCallback audioCallback) { |
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} |
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void AudioStream::setAudioEncInfo(int samplesInHZ, int channels) { |
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//打开编码器
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mChannels = channels; |
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//一次最大能输入编码器的样本数量 (一个样本是16位 2字节)
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//编码后的最大字节数
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//open faac encoder
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audioCodec = faacEncOpen(static_cast<unsigned long>(samplesInHZ), |
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static_cast<unsigned int>(channels), |
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&inputSamples, |
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&maxOutputBytes); |
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//设置编码器参数
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//set encoder params
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faacEncConfigurationPtr config = faacEncGetCurrentConfiguration(audioCodec); |
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//指定为 mpeg4 标准
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config->mpegVersion = MPEG4; |
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//lc 标准
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config->aacObjectType = LOW; |
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//16位
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config->inputFormat = FAAC_INPUT_16BIT; |
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// 编码出原始数据
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config->outputFormat = 0; |
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faacEncSetConfiguration(audioCodec, config); |
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//输出缓冲区 编码后的数据 用这个缓冲区来保存
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//output buffer
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buffer = new u_char[maxOutputBytes]; |
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} |
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@ -56,7 +50,7 @@ RTMPPacket *AudioStream::getAudioTag() { |
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int bodySize = static_cast<int>(2 + len); |
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RTMPPacket *packet = new RTMPPacket; |
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RTMPPacket_Alloc(packet, bodySize); |
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//双声道
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//channel layout: stereo
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packet->m_body[0] = 0xAF; |
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if (mChannels == 1) { |
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packet->m_body[0] = 0xAE; |
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@ -74,7 +68,7 @@ RTMPPacket *AudioStream::getAudioTag() { |
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} |
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void AudioStream::encodeData(int8_t *data) { |
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//返回编码后数据字节的长度
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//encode a frame, and return encoded len
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int byteLen = faacEncEncode(audioCodec, reinterpret_cast<int32_t *>(data), |
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static_cast<unsigned int>(inputSamples), |
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buffer, |
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@ -83,7 +77,7 @@ void AudioStream::encodeData(int8_t *data) { |
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int bodySize = 2 + byteLen; |
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RTMPPacket *packet = new RTMPPacket; |
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RTMPPacket_Alloc(packet, bodySize); |
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//双声道
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//stereo
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packet->m_body[0] = 0xAF; |
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if (mChannels == 1) { |
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packet->m_body[0] = 0xAE; |
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