int CAEEncoderFFmpeg::Encode(uint8_t *in, int in_size, uint8_t *out, int out_size)
{
int got_output;
AVFrame *frame;
if (!m_CodecCtx)
return 0;
/* allocate the input frame
* sadly, we have to alloc/dealloc it everytime since we have no guarantee the
* data argument will be constant over iterated calls and the frame needs to
* setup pointers inside data */
frame = av_frame_alloc();
if (!frame)
return 0;
frame->nb_samples = m_CodecCtx->frame_size;
frame->format = m_CodecCtx->sample_fmt;
frame->channel_layout = m_CodecCtx->channel_layout;
avcodec_fill_audio_frame(frame, m_CodecCtx->channels, m_CodecCtx->sample_fmt,
in, in_size, 0);
/* initialize the output packet */
av_init_packet(&m_Pkt);
m_Pkt.size = out_size - IEC61937_DATA_OFFSET;
m_Pkt.data = out + IEC61937_DATA_OFFSET;
/* encode it */
int ret = avcodec_encode_audio2(m_CodecCtx, &m_Pkt, frame, &got_output);
/* free temporary data */
av_frame_free(&frame);
if (ret < 0 || !got_output)
{
CLog::Log(LOGERROR, "CAEEncoderFFmpeg::Encode - Encoding failed");
return 0;
}
/* pack it into an IEC958 frame */
m_PackFunc(NULL, m_Pkt.size, out);
/* free the packet */
av_free_packet(&m_Pkt);
/* return the number of frames used */
return m_NeededFrames;
}
int CAEEncoderFFmpeg::Encode(float *data, unsigned int frames)
{
if (!m_CodecCtx || frames < m_NeededFrames)
return 0;
/* encode it */
int size = m_dllAvCodec.avcodec_encode_audio(m_CodecCtx, m_Buffer + IEC61937_DATA_OFFSET, FF_MIN_BUFFER_SIZE, (short*)data);
/* pack it into an IEC958 frame */
m_BufferSize = m_PackFunc(NULL, size, m_Buffer);
if (m_BufferSize != m_OutputSize)
{
m_OutputSize = m_BufferSize;
m_OutputRatio = (double)m_NeededFrames / m_OutputSize;
}
/* return the number of frames used */
return m_NeededFrames;
}
int CAEEncoderFFmpeg::Encode(float *data, unsigned int frames)
{
int got_output;
AVFrame *frame;
const uint8_t *input = (const uint8_t*) data;
if (!m_CodecCtx || frames < m_NeededFrames)
return 0;
/* size of the buffer sent to the encoder: either from the input data or after
* conversion, in all cases it is in the m_CodecCtx->sample_fmt format */
int buf_size = av_samples_get_buffer_size(NULL, m_CodecCtx->channels, frames, m_CodecCtx->sample_fmt, 0);
assert(buf_size>0);
/* allocate the input frame
* sadly, we have to alloc/dealloc it everytime since we have no guarantee the
* data argument will be constant over iterated calls and the frame needs to
* setup pointers inside data */
frame = av_frame_alloc();
if (!frame)
return 0;
frame->nb_samples = m_CodecCtx->frame_size;
frame->format = m_CodecCtx->sample_fmt;
frame->channel_layout = m_CodecCtx->channel_layout;
if (m_NeedConversion)
{
if (!m_ResampBuffer || buf_size > m_ResampBufferSize)
{
m_ResampBuffer = (uint8_t*)av_realloc(m_ResampBuffer, buf_size);
if (!m_ResampBuffer)
{
CLog::Log(LOGERROR, "CAEEncoderFFmpeg::Encode - Failed to allocate %i bytes buffer for resampling", buf_size);
av_frame_free(&frame);
return 0;
}
m_ResampBufferSize = buf_size;
}
avcodec_fill_audio_frame(frame, m_CodecCtx->channels, m_CodecCtx->sample_fmt, m_ResampBuffer, buf_size, 0);
/* important note: the '&input' here works because we convert from a packed
* format (ie, interleaved). If it were to be used to convert from planar
* formats (ie, non-interleaved, which is not currently supported by AE),
* we would need to adapt it or it would segfault. */
if (swr_convert(m_SwrCtx, frame->extended_data, frames, &input, frames) < 0)
{
CLog::Log(LOGERROR, "CAEEncoderFFmpeg::Encode - Resampling failed");
av_frame_free(&frame);
return 0;
}
}
else
avcodec_fill_audio_frame(frame, m_CodecCtx->channels, m_CodecCtx->sample_fmt,
input, buf_size, 0);
/* initialize the output packet */
av_init_packet(&m_Pkt);
m_Pkt.size = sizeof(m_Buffer) - IEC61937_DATA_OFFSET;
m_Pkt.data = m_Buffer + IEC61937_DATA_OFFSET;
/* encode it */
int ret = avcodec_encode_audio2(m_CodecCtx, &m_Pkt, frame, &got_output);
/* free temporary data */
av_frame_free(&frame);
if (ret < 0 || !got_output)
{
CLog::Log(LOGERROR, "CAEEncoderFFmpeg::Encode - Encoding failed");
return 0;
}
/* pack it into an IEC958 frame */
m_BufferSize = m_PackFunc(NULL, m_Pkt.size, m_Buffer);
if (m_BufferSize != m_OutputSize)
{
m_OutputSize = m_BufferSize;
m_OutputRatio = (double)m_NeededFrames / m_OutputSize;
}
/* free the packet */
av_free_packet(&m_Pkt);
/* return the number of frames used */
return m_NeededFrames;
}
bool CAEEncoderFFmpeg::Initialize(AEAudioFormat &format, bool allow_planar_input)
{
Reset();
bool ac3 = CSettings::Get().GetBool("audiooutput.ac3passthrough");
AVCodec *codec = NULL;
#if 0
/* the DCA encoder is currently useless for transcode, it creates a 196 kHz DTS-HD like mongrel which is useless for SPDIF */
bool dts = CSettings::Get().GetBool("audiooutput.dtspassthrough");
if (dts && (!ac3 || g_advancedSettings.m_audioTranscodeTo.Equals("dts")))
{
m_CodecName = "DTS";
m_CodecID = AV_CODEC_ID_DTS;
m_PackFunc = &CAEPackIEC61937::PackDTS_1024;
m_BitRate = DTS_ENCODE_BITRATE;
codec = avcodec_find_encoder(m_CodecID);
}
#endif
/* fallback to ac3 if we support it, we might not have DTS support */
if (!codec && ac3)
{
m_CodecName = "AC3";
m_CodecID = AV_CODEC_ID_AC3;
m_PackFunc = &CAEPackIEC61937::PackAC3;
m_BitRate = AC3_ENCODE_BITRATE;
codec = avcodec_find_encoder(m_CodecID);
}
/* check we got the codec */
if (!codec)
return false;
m_CodecCtx = avcodec_alloc_context3(codec);
m_CodecCtx->bit_rate = m_BitRate;
m_CodecCtx->sample_rate = format.m_sampleRate;
m_CodecCtx->channel_layout = AV_CH_LAYOUT_5POINT1_BACK;
/* select a suitable data format */
if (codec->sample_fmts)
{
bool hasFloat = false;
bool hasDouble = false;
bool hasS32 = false;
bool hasS16 = false;
bool hasU8 = false;
bool hasFloatP = false;
bool hasUnknownFormat = false;
for(int i = 0; codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; ++i)
{
switch (codec->sample_fmts[i])
{
case AV_SAMPLE_FMT_FLT: hasFloat = true; break;
case AV_SAMPLE_FMT_DBL: hasDouble = true; break;
case AV_SAMPLE_FMT_S32: hasS32 = true; break;
case AV_SAMPLE_FMT_S16: hasS16 = true; break;
case AV_SAMPLE_FMT_U8 : hasU8 = true; break;
case AV_SAMPLE_FMT_FLTP:
if (allow_planar_input)
hasFloatP = true;
else
hasUnknownFormat = true;
break;
case AV_SAMPLE_FMT_NONE: return false;
default: hasUnknownFormat = true; break;
}
}
if (hasFloat)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_FLT;
format.m_dataFormat = AE_FMT_FLOAT;
}
else if (hasFloatP)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP;
format.m_dataFormat = AE_FMT_FLOATP;
}
else if (hasDouble)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_DBL;
format.m_dataFormat = AE_FMT_DOUBLE;
}
else if (hasS32)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_S32;
format.m_dataFormat = AE_FMT_S32NE;
}
else if (hasS16)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
format.m_dataFormat = AE_FMT_S16NE;
}
else if (hasU8)
{
m_CodecCtx->sample_fmt = AV_SAMPLE_FMT_U8;
format.m_dataFormat = AE_FMT_U8;
}
else if (hasUnknownFormat)
{
m_CodecCtx->sample_fmt = codec->sample_fmts[0];
format.m_dataFormat = AE_FMT_FLOAT;
m_NeedConversion = true;
CLog::Log(LOGNOTICE, "CAEEncoderFFmpeg::Initialize - Unknown audio format, it will be resampled.");
}
else
{
CLog::Log(LOGERROR, "CAEEncoderFFmpeg::Initialize - Unable to find a suitable data format for the codec (%s)", m_CodecName.c_str());
return false;
}
}
m_CodecCtx->channels = BuildChannelLayout(m_CodecCtx->channel_layout, m_Layout);
/* open the codec */
if (avcodec_open2(m_CodecCtx, codec, NULL))
{
av_freep(&m_CodecCtx);
return false;
}
format.m_frames = m_CodecCtx->frame_size;
format.m_frameSamples = m_CodecCtx->frame_size * m_CodecCtx->channels;
format.m_frameSize = m_CodecCtx->channels * (CAEUtil::DataFormatToBits(format.m_dataFormat) >> 3);
format.m_channelLayout = m_Layout;
m_CurrentFormat = format;
m_NeededFrames = format.m_frames;
m_OutputSize = m_PackFunc(NULL, 0, m_Buffer);
m_OutputRatio = (double)m_NeededFrames / m_OutputSize;
m_SampleRateMul = 1.0 / (double)m_CodecCtx->sample_rate;
if (m_NeedConversion)
{
m_SwrCtx = swr_alloc_set_opts(NULL,
m_CodecCtx->channel_layout, m_CodecCtx->sample_fmt, m_CodecCtx->sample_rate,
m_CodecCtx->channel_layout, AV_SAMPLE_FMT_FLT, m_CodecCtx->sample_rate,
0, NULL);
if (!m_SwrCtx || swr_init(m_SwrCtx) < 0)
{
CLog::Log(LOGERROR, "CAEEncoderFFmpeg::Initialize - Failed to initialise resampler.");
return false;
}
}
CLog::Log(LOGNOTICE, "CAEEncoderFFmpeg::Initialize - %s encoder ready", m_CodecName.c_str());
return true;
}
