mirrors/libstb-hal
1
0
Fork 0
mirror of https://github.com/tuxbox-neutrino/libstb-hal.git synced 2025-08-26 23:13:16 +02:00
Code Issues Packages Projects Releases Wiki Activity

libeplayer3: switch back to software decoding for AAC; crude fix for PCM injection (newer ffmpeg versions return floats instead of shorts), injected PCM streams (OGG, for example), will now work again. I'd be grateful if anybody with a deeper understanding of the FFMPEG API would supply a cleaner fix ...

Browse Source
This commit is contained in:
martii
2013-03-03 13:38:39 +01:00
parent 3085e72c01
commit fbb9dcbb03
2 changed files with 108 additions and 1 deletions
Download Patch File Download Diff File Expand all files Collapse all files

106
libeplayer3/container/container_ffmpeg.c
View File

@@ -190,7 +190,11 @@ static char* Codec2Encoding(enum CodecID id, int* version)
case CODEC_ID_MP3:
return "A_MP3";
case CODEC_ID_AAC:
#ifdef MARTII
return "A_IPCM";
#else
return "A_AAC";
#endif
case CODEC_ID_AC3:
return "A_AC3";
case CODEC_ID_DTS:
@@ -339,8 +343,10 @@ static void FFMPEGThread(Context_t *context) {
int err = 0, gotlastPts = 0, audioMute = 0;
AudioVideoOut_t avOut;
#ifndef MARTII
/* Softdecoding buffer*/
unsigned char *samples = NULL;
#endif
ffmpeg_printf(10, "\n");
@@ -630,6 +636,93 @@ static void FFMPEGThread(Context_t *context) {
}
else if (audioTrack->inject_as_pcm == 1)
{
#ifdef MARTII
int bytesDone = 0;
unsigned int samples_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
AVPacket avpkt;
avpkt = packet;
int is_planar = av_sample_fmt_is_planar(((AVStream*) audioTrack->stream)->codec->sample_fmt);
int nc = ((AVStream*) audioTrack->stream)->codec->channels;
if (is_planar)
samples_size *= nc;
uint8_t *samples = (unsigned char *)malloc(samples_size);
while(avpkt.size > 0)
{
int decoded_data_size = samples_size;
bytesDone = avcodec_decode_audio3(( (AVStream*) audioTrack->stream)->codec,
(short *)(samples), &decoded_data_size, &avpkt);
if(bytesDone < 0)
break;
avpkt.data += bytesDone;
avpkt.size -= bytesDone;
if(decoded_data_size <= 0)
continue;
int i;
if (is_planar && nc > 1) {
int ds = decoded_data_size/sizeof(float)/nc;
short *shortSamples = (short *)malloc(nc * ds * sizeof(short));
float *floatSamples = (float *)samples;
int j;
for (i = 0; i < ds; i++)
for (j = 0; j < nc; j++)
shortSamples[i * nc + j] = floatSamples[j * ds + i] * 32767.0;
free(samples);
samples = (unsigned char *) shortSamples;
} else if (nc == 1) {
// mono doesn't seem to work ... convert to stereo
int ds = decoded_data_size/sizeof(float);
short *shortSamples = (short *)samples;
float *floatSamples = (float *)samples;
for (i = 0; i < ds; i++)
shortSamples[2 * i] = shortSamples[2 * i + 1] = floatSamples[i] * 32767.0;
decoded_data_size *= 2;
nc++;
} else {
int ds = decoded_data_size/sizeof(float);
short *shortSamples = (short *)samples;
float *floatSamples = (float *)samples;
for (i = 0; i < ds; i++)
shortSamples[i] = floatSamples[i] * 32767.0;
}
pcmPrivateData_t extradata;
extradata.uNoOfChannels = nc;
extradata.uSampleRate = ((AVStream*) audioTrack->stream)->codec->sample_rate;
extradata.uBitsPerSample = 16;
extradata.bLittleEndian = 1;
avOut.data = samples;
avOut.len = (decoded_data_size * sizeof(short))/sizeof(float);
avOut.pts = pts;
avOut.extradata = (unsigned char *) &extradata;
avOut.extralen = sizeof(extradata);
avOut.frameRate = 0;
avOut.timeScale = 0;
avOut.width = 0;
avOut.height = 0;
avOut.type = "audio";
#ifdef reverse_playback_3
if (!context->playback->BackWard)
#endif
if (context->output->audio->Write(context, &avOut) < 0)
ffmpeg_err("writing data to audio device failed\n");
}
free(samples);
#else
int bytesDone = 0;
unsigned int samples_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
AVPacket avpkt;
@@ -646,7 +739,6 @@ static void FFMPEGThread(Context_t *context) {
bytesDone = avcodec_decode_audio3(( (AVStream*) audioTrack->stream)->codec,
(short *)(samples), &decoded_data_size, &avpkt);
if(bytesDone < 0) // Error Happend
break;
@@ -684,6 +776,7 @@ static void FFMPEGThread(Context_t *context) {
if (context->output->audio->Write(context, &avOut) < 0)
ffmpeg_err("writing data to audio device failed\n");
}
#endif
}
else if (audioTrack->have_aacheader == 1)
{
@@ -915,11 +1008,13 @@ static void FFMPEGThread(Context_t *context) {
} /* while */
#ifndef MARTII
// Freeing the allocated buffer for softdecoding
if (samples != NULL) {
free(samples);
samples = NULL;
}
#endif
hasPlayThreadStarted = 0;
@@ -1138,6 +1233,9 @@ int container_ffmpeg_init(Context_t *context, char * filename)
track.inject_raw_pcm = 1;
ffmpeg_printf(10, " Handle inject_raw_pcm = %d\n", track.inject_as_pcm);
}
#ifdef MARTII
else
#endif
if(!strncmp(encoding, "A_IPCM", 6))
{
track.inject_as_pcm = 1;
@@ -1150,6 +1248,7 @@ int container_ffmpeg_init(Context_t *context, char * filename)
printf("AVCODEC__INIT__SUCCESS\n");
else
printf("AVCODEC__INIT__FAILED\n");
#ifndef MARTII
}
else if(stream->codec->codec_id == CODEC_ID_AAC) {
ffmpeg_printf(10,"Create AAC ExtraData\n");
@@ -1201,9 +1300,14 @@ int container_ffmpeg_init(Context_t *context, char * filename)
Hexdump(track.aacbuf,7);
track.have_aacheader = 1;
#endif
} else if(stream->codec->codec_id == CODEC_ID_WMAV1
|| stream->codec->codec_id == CODEC_ID_WMAV2
#ifdef MARTII
|| stream->codec->codec_id == 86056 ) //CODEC_ID_WMAPRO) //if (stream->codec->extradata_size > 0)
#else
|| 86056 ) //CODEC_ID_WMAPRO) //if (stream->codec->extradata_size > 0)
#endif
{
ffmpeg_printf(10,"Create WMA ExtraData\n");
track.aacbuflen = 104 + stream->codec->extradata_size;

3
libeplayer3/output/writer/pcm.c
View File

@@ -88,6 +88,9 @@ static const unsigned char clpcm_pes[18] = { 0x00, 0x00, 0x01, 0xBD, //start c
0x1E, 0x60, 0x0A, //first pes only, 0xFF after
0xFF
};
#ifdef MARTII
// reference: search for TypeLpcmDVDAudio in player/frame_parser/frame_parser_audio_lpcm.cpp
#endif
static const unsigned char clpcm_prv[14] = { 0xA0, //sub_stream_id
0, 0, //resvd and UPC_EAN_ISRC stuff, unused
0x0A, //private header length