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libeplayer3: change extradata handling
Origin commit data
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Branch: master
Commit: 0895d29e2a
Author: martii <m4rtii@gmx.de>
Date: 2014-04-05 (Sat, 05 Apr 2014)
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No further description and justification available within origin commit message!
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This commit was generated by Migit
This commit is contained in:
martii
@@ -42,10 +42,6 @@
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#include "common.h"
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#include "misc.h"
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#include "debug.h"
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#if 0
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#include "aac.h"
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#endif
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#include "pcm.h"
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/* ***************************** */
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/* Makros/Constants */
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@@ -381,8 +377,6 @@ static void FFMPEGThread(Context_t * context)
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avOut.data = packet_data;
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avOut.len = packet_size;
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avOut.pts = pts;
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avOut.extradata = videoTrack->extraData;
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avOut.extralen = videoTrack->extraSize;
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avOut.frameRate = videoTrack->frame_rate;
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avOut.timeScale = videoTrack->TimeScale;
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avOut.width = videoTrack->width;
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@@ -402,17 +396,14 @@ static void FFMPEGThread(Context_t * context)
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if (audioTrack->inject_raw_pcm == 1) {
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ffmpeg_printf(200, "write audio raw pcm\n");
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pcmPrivateData_t extradata;
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extradata.uNoOfChannels = ((AVStream *) audioTrack->stream)->codec->channels;
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extradata.uSampleRate = ((AVStream *) audioTrack->stream)->codec->sample_rate;
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extradata.uBitsPerSample = 16;
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extradata.bLittleEndian = 1;
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avOut.uNoOfChannels = ((AVStream *) audioTrack->stream)->codec->channels;
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avOut.uSampleRate = ((AVStream *) audioTrack->stream)->codec->sample_rate;
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avOut.uBitsPerSample = 16;
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avOut.bLittleEndian = 1;
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avOut.data = packet_data;
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avOut.len = packet_size;
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avOut.pts = pts;
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avOut.extradata = (unsigned char *) &extradata;
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avOut.extralen = sizeof(extradata);
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avOut.frameRate = 0;
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avOut.timeScale = 0;
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avOut.width = 0;
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@@ -527,19 +518,15 @@ static void FFMPEGThread(Context_t * context)
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out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **)
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&decoded_frame->data[0], in_samples);
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pcmPrivateData_t extradata;
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extradata.uSampleRate = out_sample_rate;
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extradata.uNoOfChannels = av_get_channel_layout_nb_channels(out_channel_layout);
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extradata.uBitsPerSample = 16;
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extradata.bLittleEndian = 1;
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avOut.uSampleRate = out_sample_rate;
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avOut.uNoOfChannels = av_get_channel_layout_nb_channels(out_channel_layout);
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avOut.uBitsPerSample = 16;
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avOut.bLittleEndian = 1;
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avOut.data = output;
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avOut.len = out_samples * sizeof(short) * out_channels;
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avOut.pts = videoTrack ? pts : 0;
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avOut.extradata = (unsigned char *) &extradata;
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avOut.extralen = sizeof(extradata);
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avOut.frameRate = 0;
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avOut.timeScale = 0;
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avOut.width = 0;
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@@ -552,32 +539,10 @@ static void FFMPEGThread(Context_t * context)
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ffmpeg_err("writing data to audio device failed\n");
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av_freep(&output);
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}
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#if 0
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} else if (audioTrack->have_aacheader == 1) {
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ffmpeg_printf(200, "write audio aac\n");
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avOut.data = packet_data;
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avOut.len = packet_size;
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avOut.pts = pts;
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avOut.extradata = audioTrack->aacbuf;
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avOut.extralen = audioTrack->aacbuflen;
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avOut.frameRate = 0;
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avOut.timeScale = 0;
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avOut.width = 0;
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avOut.height = 0;
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avOut.type = "audio";
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avOut.stream = audioTrack->stream;
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avOut.avfc = avContext;
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if (context->output->audio->Write(context, &avOut) < 0)
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ffmpeg_err("(aac) writing data to audio device failed\n");
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#endif
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} else {
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avOut.data = packet_data;
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avOut.len = packet_size;
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avOut.pts = pts;
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avOut.extradata = NULL;
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avOut.extralen = 0;
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avOut.frameRate = 0;
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avOut.timeScale = 0;
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avOut.width = 0;
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@@ -914,16 +879,8 @@ int container_ffmpeg_update_tracks(Context_t * context, char *filename)
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track.width = stream->codec->width;
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track.height = stream->codec->height;
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track.extraData = stream->codec->extradata;
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track.extraSize = stream->codec->extradata_size;
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track.frame_rate = stream->r_frame_rate.num;
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#if 0
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track.aacbuf = 0;
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track.have_aacheader = -1;
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#endif
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double frame_rate = av_q2d(stream->r_frame_rate); /* rational to double */
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ffmpeg_printf(10, "frame_rate = %f\n", frame_rate);
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@@ -986,10 +943,6 @@ int container_ffmpeg_update_tracks(Context_t * context, char *filename)
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track.Encoding = encoding;
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track.Id = stream->id;
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track.duration = (double) stream->duration * av_q2d(stream->time_base) * 1000.0;
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#if 0
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track.aacbuf = 0;
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track.have_aacheader = -1;
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#endif
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if (stream->duration == AV_NOPTS_VALUE) {
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ffmpeg_printf(10, "Stream has no duration so we take the duration from context\n");
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@@ -1002,151 +955,6 @@ int container_ffmpeg_update_tracks(Context_t * context, char *filename)
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track.inject_as_pcm = 1;
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ffmpeg_printf(10, " Handle inject_as_pcm = %d\n", track.inject_as_pcm);
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}
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#if 0
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else if (stream->codec->codec_id == AV_CODEC_ID_AAC) {
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ffmpeg_printf(10, "Create AAC ExtraData\n");
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ffmpeg_printf(10, "stream->codec->extradata_size %d\n", stream->codec->extradata_size);
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Hexdump(stream->codec->extradata, stream->codec->extradata_size);
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/* extradata
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13 10 56 e5 9d 48 00 (anderen cops)
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object_type: 00010 2 = LC
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sample_rate: 011 0 6 = 24000
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chan_config: 0010 2 = Stereo
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000 0
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1010110 111 = 0x2b7
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00101 = SBR
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1
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0011 = 48000
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101 01001000 = 0x548
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ps = 0
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0000000
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*/
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unsigned int object_type = 2; // LC
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unsigned int sample_index = aac_get_sample_rate_index(stream->codec->sample_rate);
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unsigned int chan_config = stream->codec->channels;
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if (stream->codec->extradata_size >= 2) {
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object_type = stream->codec->extradata[0] >> 3;
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sample_index = ((stream->codec->extradata[0] & 0x7) << 1)
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+ (stream->codec->extradata[1] >> 7);
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chan_config = (stream->codec->extradata[1] >> 3)
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&& 0xf;
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}
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ffmpeg_printf(10, "aac object_type %d\n", object_type);
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ffmpeg_printf(10, "aac sample_index %d\n", sample_index);
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ffmpeg_printf(10, "aac chan_config %d\n", chan_config);
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object_type -= 1; // Cause of ADTS
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track.aacbuflen = AAC_HEADER_LENGTH;
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track.aacbuf = malloc(8);
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track.aacbuf[0] = 0xFF;
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track.aacbuf[1] = 0xF1;
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track.aacbuf[2] = ((object_type & 0x03) << 6) | (sample_index << 2) | ((chan_config >> 2) & 0x01);
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track.aacbuf[3] = (chan_config & 0x03) << 6;
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track.aacbuf[4] = 0x00;
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track.aacbuf[5] = 0x1F;
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track.aacbuf[6] = 0xFC;
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printf("AAC_HEADER -> ");
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Hexdump(track.aacbuf, 7);
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track.have_aacheader = 1;
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} else if (stream->codec->codec_id == AV_CODEC_ID_WMAV1 || stream->codec->codec_id == AV_CODEC_ID_WMAV2 || stream->codec->codec_id == AV_CODEC_ID_WMAPRO) //if (stream->codec->extradata_size > 0)
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{
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ffmpeg_printf(10, "Create WMA ExtraData\n");
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track.aacbuflen = 104 + stream->codec->extradata_size;
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track.aacbuf = malloc(track.aacbuflen);
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memset(track.aacbuf, 0, track.aacbuflen);
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unsigned char ASF_Stream_Properties_Object[16] = { 0x91, 0x07, 0xDC, 0xB7, 0xB7, 0xA9, 0xCF, 0x11,
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0x8E, 0xE6, 0x00, 0xC0, 0x0C, 0x20, 0x53, 0x65
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};
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memcpy(track.aacbuf + 0, ASF_Stream_Properties_Object, 16); // ASF_Stream_Properties_Object
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memcpy(track.aacbuf + 16, &track.aacbuflen, 4); //FrameDateLength
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unsigned int sizehi = 0;
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memcpy(track.aacbuf + 20, &sizehi, 4); // sizehi (not used)
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unsigned char ASF_Audio_Media[16] = { 0x40, 0x9E, 0x69, 0xF8, 0x4D, 0x5B, 0xCF, 0x11,
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0xA8, 0xFD, 0x00, 0x80, 0x5F, 0x5C, 0x44, 0x2B
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};
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memcpy(track.aacbuf + 24, ASF_Audio_Media, 16); //ASF_Audio_Media
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unsigned char ASF_Audio_Spread[16] = { 0x50, 0xCD, 0xC3, 0xBF, 0x8F, 0x61, 0xCF, 0x11,
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0x8B, 0xB2, 0x00, 0xAA, 0x00, 0xB4, 0xE2, 0x20
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};
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memcpy(track.aacbuf + 40, ASF_Audio_Spread, 16); //ASF_Audio_Spread
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memset(track.aacbuf + 56, 0, 4); // time_offset (not used)
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memset(track.aacbuf + 60, 0, 4); // time_offset_hi (not used)
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unsigned int type_specific_data_length = 18 + stream->codec->extradata_size;
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memcpy(track.aacbuf + 64, &type_specific_data_length, 4); //type_specific_data_length
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unsigned int error_correction_data_length = 8;
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memcpy(track.aacbuf + 68, &error_correction_data_length, 4); //error_correction_data_length
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unsigned short flags = 1; // stream_number
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memcpy(track.aacbuf + 72, &flags, 2); //flags
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unsigned int reserved = 0;
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memcpy(track.aacbuf + 74, &reserved, 4); // reserved
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// type_specific_data
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#define WMA_VERSION_1 0x160
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#define WMA_VERSION_2_9 0x161
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#define WMA_VERSION_9_PRO 0x162
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#define WMA_LOSSLESS 0x163
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unsigned short codec_id = 0;
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switch (stream->codec->codec_id) {
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//TODO: What code for lossless ?
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case AV_CODEC_ID_WMAPRO:
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codec_id = WMA_VERSION_9_PRO;
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break;
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case AV_CODEC_ID_WMAV2:
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codec_id = WMA_VERSION_2_9;
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break;
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case AV_CODEC_ID_WMAV1:
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default:
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codec_id = WMA_VERSION_1;
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break;
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}
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memcpy(track.aacbuf + 78, &codec_id, 2); //codec_id
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unsigned short number_of_channels = stream->codec->channels;
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memcpy(track.aacbuf + 80, &number_of_channels, 2); //number_of_channels
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unsigned int samples_per_second = stream->codec->sample_rate;
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ffmpeg_printf(1, "samples_per_second = %d\n", samples_per_second);
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memcpy(track.aacbuf + 82, &samples_per_second, 4); //samples_per_second
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unsigned int average_number_of_bytes_per_second = stream->codec->bit_rate / 8;
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ffmpeg_printf(1, "average_number_of_bytes_per_second = %d\n", average_number_of_bytes_per_second);
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memcpy(track.aacbuf + 86, &average_number_of_bytes_per_second, 4); //average_number_of_bytes_per_second
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unsigned short block_alignment = stream->codec->block_align;
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ffmpeg_printf(1, "block_alignment = %d\n", block_alignment);
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memcpy(track.aacbuf + 90, &block_alignment, 2); //block_alignment
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unsigned short bits_per_sample = stream->codec->sample_fmt >= 0 ? (stream->codec->sample_fmt + 1) * 8 : 8;
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ffmpeg_printf(1, "bits_per_sample = %d (%d)\n", bits_per_sample, stream->codec->sample_fmt);
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memcpy(track.aacbuf + 92, &bits_per_sample, 2); //bits_per_sample
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memcpy(track.aacbuf + 94, &stream->codec->extradata_size, 2); //bits_per_sample
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memcpy(track.aacbuf + 96, stream->codec->extradata, stream->codec->extradata_size);
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ffmpeg_printf(1, "aacbuf:\n");
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Hexdump(track.aacbuf, track.aacbuflen);
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//ffmpeg_printf(1, "priv_data:\n");
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//Hexdump(stream->codec->priv_data, track.aacbuflen);
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track.have_aacheader = 1;
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}
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#endif
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if (context->manager->audio) {
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if (context->manager->audio->Command(context, MANAGER_ADD, &track) < 0) {
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@@ -1183,9 +991,6 @@ int container_ffmpeg_update_tracks(Context_t * context, char *filename)
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track.width = -1; /* will be filled online from videotrack */
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track.height = -1; /* will be filled online from videotrack */
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track.extraData = stream->codec->extradata;
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track.extraSize = stream->codec->extradata_size;
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ffmpeg_printf(10, "subtitle codec %d\n", stream->codec->codec_id);
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ffmpeg_printf(10, "subtitle width %d\n", stream->codec->width);
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ffmpeg_printf(10, "subtitle height %d\n", stream->codec->height);
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