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formatting code using astyle
Origin commit data
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Branch: master
Commit: bc17c13de4
Author: vanhofen <vanhofen@gmx.de>
Date: 2021-05-17 (Mon, 17 May 2021)
Origin message was:
------------------
- formatting code using astyle
------------------
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:
vanhofen
@@ -43,18 +43,19 @@ extern "C" {
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}
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// reference: search for TypeLpcmDVDAudio in player/frame_parser/frame_parser_audio_lpcm.cpp
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static const uint8_t clpcm_prv[14] = {
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0xA0, //sub_stream_id
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0, 0, //resvd and UPC_EAN_ISRC stuff, unused
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0x0A, //private header length
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0, 9, //first_access_unit_pointer
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0x00, //emph,rsvd,stereo,downmix
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0x0F, //quantisation word length 1,2
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0x0F, //audio sampling freqency 1,2
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0, //resvd, multi channel type
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0, //bit shift on channel GR2, assignment
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0x80, //dynamic range control
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0, 0 //resvd for copyright management
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static const uint8_t clpcm_prv[14] =
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{
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0xA0, //sub_stream_id
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0, 0, //resvd and UPC_EAN_ISRC stuff, unused
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0x0A, //private header length
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0, 9, //first_access_unit_pointer
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0x00, //emph,rsvd,stereo,downmix
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0x0F, //quantisation word length 1,2
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0x0F, //audio sampling freqency 1,2
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0, //resvd, multi channel type
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0, //bit shift on channel GR2, assignment
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0x80, //dynamic range control
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0, 0 //resvd for copyright management
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};
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class WriterPCM : public Writer
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@@ -98,7 +99,8 @@ bool WriterPCM::prepareClipPlay()
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memcpy(lpcm_prv, clpcm_prv, sizeof(lpcm_prv));
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// figure out size of subframe and set up sample rate
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switch (uSampleRate) {
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switch (uSampleRate)
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{
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case 48000:
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SubFrameLen = 40;
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break;
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@@ -136,7 +138,8 @@ bool WriterPCM::prepareClipPlay()
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//set number of channels
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lpcm_prv[10] = uNoOfChannels - 1;
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switch (uBitsPerSample) {
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switch (uBitsPerSample)
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{
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case 24:
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lpcm_prv[7] |= 0x20;
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case 16:
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@@ -154,7 +157,8 @@ bool WriterPCM::writePCM(int64_t Pts, uint8_t *data, unsigned int size)
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bool res = true;
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uint8_t PesHeader[PES_MAX_HEADER_SIZE];
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if (initialHeader) {
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if (initialHeader)
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{
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initialHeader = false;
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prepareClipPlay();
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ioctl(fd, AUDIO_CLEAR_BUFFER, NULL);
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@@ -162,7 +166,8 @@ bool WriterPCM::writePCM(int64_t Pts, uint8_t *data, unsigned int size)
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size += breakBufferFillSize;
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while (size >= SubFrameLen) {
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while (size >= SubFrameLen)
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{
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if (breakBufferFillSize)
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memcpy(injectBuffer, breakBuffer, breakBufferFillSize);
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memcpy(injectBuffer + breakBufferFillSize, data, SubFrameLen - breakBufferFillSize);
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@@ -171,17 +176,22 @@ bool WriterPCM::writePCM(int64_t Pts, uint8_t *data, unsigned int size)
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breakBufferFillSize = 0;
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//write the PCM data
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if (uBitsPerSample == 16) {
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for (unsigned int n = 0; n < SubFrameLen; n += 2) {
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if (uBitsPerSample == 16)
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{
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for (unsigned int n = 0; n < SubFrameLen; n += 2)
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{
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uint8_t tmp = injectBuffer[n];
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injectBuffer[n] = injectBuffer[n + 1];
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injectBuffer[n + 1] = tmp;
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}
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} else {
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}
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else
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{
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// 0 1 2 3 4 5 6 7 8 9 10 11
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// A1c A1b A1a B1c B1b B1a A2c A2b A2a B2c B2b B2a
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// to A1a A1b B1a B1b A2a A2b B2a B2b A1c B1c A2c B2c
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for (unsigned int n = 0; n < SubFrameLen; n += 12) {
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for (unsigned int n = 0; n < SubFrameLen; n += 12)
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{
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uint8_t t, *p = injectBuffer + n;
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t = p[0];
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p[0] = p[2];
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@@ -207,12 +217,14 @@ bool WriterPCM::writePCM(int64_t Pts, uint8_t *data, unsigned int size)
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iov[2].iov_len = SubFrameLen;
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iov[0].iov_len = InsertPesHeader(PesHeader, iov[1].iov_len + iov[2].iov_len, PCM_PES_START_CODE, Pts, 0);
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int len = writev(fd, iov, 3);
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if (len < 0) {
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if (len < 0)
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{
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res = false;
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break;
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}
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}
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if (size && res) {
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if (size && res)
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{
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breakBufferFillSize = size;
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memcpy(breakBuffer, data, size);
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}
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@@ -231,58 +243,67 @@ void WriterPCM::Init(int _fd, AVStream *_stream, Player *_player)
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bool WriterPCM::Write(AVPacket *packet, int64_t pts)
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{
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if (!packet) {
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if (!packet)
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{
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restart_audio_resampling = true;
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return true;
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}
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AVCodecContext *c = player->input.GetCodecContext((unsigned int)stream->index);
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if (restart_audio_resampling) {
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if (restart_audio_resampling)
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{
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restart_audio_resampling = false;
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initialHeader = true;
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if (swr) {
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if (swr)
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{
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swr_free(&swr);
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swr = NULL;
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}
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if (decoded_frame) {
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if (decoded_frame)
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{
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av_frame_free(&decoded_frame);
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decoded_frame = NULL;
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}
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#if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(57,25,101)
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AVCodec *codec = avcodec_find_decoder(c->codec_id);
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if (!codec) {
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if (!codec)
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{
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fprintf(stderr, "%s %d: avcodec_find_decoder(%llx)\n", __func__, __LINE__, (unsigned long long) c->codec_id);
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return false;
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}
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avcodec_close(c);
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if (avcodec_open2(c, codec, NULL)) {
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if (avcodec_open2(c, codec, NULL))
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{
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fprintf(stderr, "%s %d: avcodec_open2 failed\n", __func__, __LINE__);
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return false;
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}
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#endif
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}
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if (!swr) {
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if (!swr)
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{
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int in_rate = c->sample_rate;
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// rates in descending order
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int rates[] = {192000, 176400, 96000, 88200, 48000, 44100, 0};
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int i = 0;
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// find the next equal or smallest rate
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while (rates[i] && in_rate < rates[i])
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i++;
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i++;
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out_sample_rate = rates[i] ? rates[i] : 44100;
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out_channels = c->channels;
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if (c->channel_layout == 0) {
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if (c->channel_layout == 0)
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{
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// FIXME -- need to guess, looks pretty much like a bug in the FFMPEG WMA decoder
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c->channel_layout = AV_CH_LAYOUT_STEREO;
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}
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out_channel_layout = c->channel_layout;
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// player2 won't play mono
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if (out_channel_layout == AV_CH_LAYOUT_MONO) {
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if (out_channel_layout == AV_CH_LAYOUT_MONO)
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{
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out_channel_layout = AV_CH_LAYOUT_STEREO;
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out_channels = 2;
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}
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@@ -292,7 +313,8 @@ bool WriterPCM::Write(AVPacket *packet, int64_t pts)
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uBitsPerSample = 16;
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swr = swr_alloc();
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if (!swr) {
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if (!swr)
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{
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fprintf(stderr, "%s %d: swr_alloc failed\n", __func__, __LINE__);
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return false;
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}
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@@ -304,31 +326,37 @@ bool WriterPCM::Write(AVPacket *packet, int64_t pts)
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av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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int e = swr_init(swr);
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if (e < 0) {
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if (e < 0)
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{
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fprintf(stderr, "swr_init: %d (icl=%d ocl=%d isr=%d osr=%d isf=%d osf=%d)\n",
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-e, (int) c->channel_layout,
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(int) out_channel_layout, c->sample_rate, out_sample_rate, c->sample_fmt, AV_SAMPLE_FMT_S16);
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-e, (int) c->channel_layout,
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(int) out_channel_layout, c->sample_rate, out_sample_rate, c->sample_fmt, AV_SAMPLE_FMT_S16);
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restart_audio_resampling = true;
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return false;
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}
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}
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unsigned int packet_size = packet->size;
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while (packet_size > 0 || (!packet_size && !packet->data)) {
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while (packet_size > 0 || (!packet_size && !packet->data))
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{
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if (!decoded_frame) {
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if (!(decoded_frame = av_frame_alloc())) {
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if (!decoded_frame)
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{
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if (!(decoded_frame = av_frame_alloc()))
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{
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fprintf(stderr, "out of memory\n");
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exit(1);
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}
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} else
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}
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else
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av_frame_unref(decoded_frame);
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#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(57,37,100)
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int got_frame = 0;
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int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, packet);
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if (len < 0) {
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if (len < 0)
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{
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restart_audio_resampling = true;
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break;
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}
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@@ -336,35 +364,43 @@ bool WriterPCM::Write(AVPacket *packet, int64_t pts)
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if (packet->data)
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packet_size -= len;
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if (!got_frame) {
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if (!got_frame)
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{
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if (!packet->data || !packet_size)
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break;
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continue;
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}
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#else
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int ret = avcodec_send_packet(c, packet);
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if (ret < 0) {
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if (ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) {
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restart_audio_resampling = true;
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break;
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}
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int ret = avcodec_send_packet(c, packet);
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if (ret < 0)
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{
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if (ret != AVERROR(EAGAIN) && ret != AVERROR_EOF)
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{
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restart_audio_resampling = true;
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break;
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}
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if (ret >= 0) {
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packet_size = 0;
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}
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if (ret >= 0)
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{
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packet_size = 0;
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}
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ret = avcodec_receive_frame(c, decoded_frame);
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if (ret < 0)
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{
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if (ret == AVERROR_EOF)
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{
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restart_audio_resampling = true;
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break;
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}
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ret = avcodec_receive_frame(c, decoded_frame);
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if (ret < 0) {
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if (ret == AVERROR_EOF) {
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restart_audio_resampling = true;
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break;
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}
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if (ret != AVERROR(EAGAIN)) {
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break;
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}
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else {
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continue;
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}
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if (ret != AVERROR(EAGAIN))
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{
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break;
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}
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else
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{
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continue;
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}
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}
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#endif
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#if (LIBAVUTIL_VERSION_MAJOR < 54)
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@@ -375,11 +411,13 @@ bool WriterPCM::Write(AVPacket *packet, int64_t pts)
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int in_samples = decoded_frame->nb_samples;
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int out_samples = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) + in_samples, out_sample_rate, c->sample_rate, AV_ROUND_UP);
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if (out_samples > out_samples_max) {
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if (out_samples > out_samples_max)
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{
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if (output)
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av_freep(&output);
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int e = av_samples_alloc(&output, NULL, out_channels, out_samples, AV_SAMPLE_FMT_S16, 1);
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if (e < 0) {
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if (e < 0)
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{
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fprintf(stderr, "av_samples_alloc: %d\n", -e);
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break;
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}
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@@ -388,7 +426,8 @@ bool WriterPCM::Write(AVPacket *packet, int64_t pts)
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out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **) &decoded_frame->data[0], in_samples);
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if (!writePCM(pts, output, out_samples * sizeof(short) * out_channels)) {
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if (!writePCM(pts, output, out_samples * sizeof(short) * out_channels))
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{
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restart_audio_resampling = true;
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break;
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}
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@@ -409,4 +448,4 @@ WriterPCM::WriterPCM()
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Register(this, AV_CODEC_ID_INJECTPCM, AUDIO_ENCODING_LPCMA);
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}
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static WriterPCM writer_pcm __attribute__ ((init_priority (300)));
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static WriterPCM writer_pcm __attribute__((init_priority(300)));
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