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93d283ba767a5ef292a20135d804cd00a6dbf5bb
ports/graphics/osg/files/patch-src_osgPlugins_ffmpeg_FFmpegDecoderAudio.cpp
T
Dima Panov 3b84553e70 graphics/osg: move to recent FFMpeg, enable COLLADA by default (+) 
Backport OSG/PR1281 for recent FFMpeg support instead of stuck with old FFMpeg4 port

COLLADA is required plugin for games/openmw

While here, adopt the port and pet portlint

Approved by:	amdmi3 (former maintainer, via PM)
2025-07-21 14:59:27 +03:00

479 lines
18 KiB
C++
Raw Blame History

--- src/osgPlugins/ffmpeg/FFmpegDecoderAudio.cpp.orig 2022-12-01 18:17:31 UTC
+++ src/osgPlugins/ffmpeg/FFmpegDecoderAudio.cpp
@@ -1,39 +1,24 @@
#include "FFmpegDecoderAudio.hpp"
-
#include <osg/Notify>
-
#include <stdexcept>
-#include <string.h>
+#include <cstring>
+#include <libavutil/channel_layout.h>
+#include <libavutil/opt.h>
-//DEBUG
-//#include <iostream>
-
-
#ifndef AVCODEC_MAX_AUDIO_FRAME_SIZE
#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000
#endif
-#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55,28,1)
-#define av_frame_alloc avcodec_alloc_frame
-#define av_frame_free avcodec_free_frame
-#endif
-
-#if LIBAVCODEC_VERSION_MAJOR < 56
- #define AV_CODEC_ID_NONE CODEC_ID_NONE
-#endif
-
namespace osgFFmpeg {
static int decode_audio(AVCodecContext *avctx, int16_t *samples,
- int *frame_size_ptr,
- const uint8_t *buf, int buf_size,
- SwrContext *swr_context,
- int out_sample_rate,
- int out_nb_channels,
- AVSampleFormat out_sample_format)
+ int *frame_size_ptr,
+ const uint8_t *buf, int buf_size,
+ SwrContext *swr_context,
+ int out_sample_rate,
+ int out_nb_channels,
+ AVSampleFormat out_sample_format)
{
-#if LIBAVCODEC_VERSION_MAJOR >= 53 || (LIBAVCODEC_VERSION_MAJOR==52 && LIBAVCODEC_VERSION_MINOR>=32)
-
AVPacket avpkt;
av_init_packet(&avpkt);
avpkt.data = const_cast<uint8_t *>(buf);
@@ -45,30 +30,39 @@ static int decode_audio(AVCodecContext *avctx, int16_t
if (!frame)
return AVERROR(ENOMEM);
- ret = avcodec_decode_audio4(avctx, frame, &got_frame, &avpkt);
+ // Send the packet to the decoder
+ ret = avcodec_send_packet(avctx, &avpkt);
+ if (ret < 0) {
+ av_frame_free(&frame);
+ return ret;
+ }
-#ifdef USE_AVRESAMPLE // libav's AVFrame structure does not contain a 'channels' field
- if (ret >= 0 && got_frame) {
-#else
- if (ret >= 0 && got_frame && av_frame_get_channels(frame)>0) {
-#endif
+ // Receive the frame from the decoder
+ ret = avcodec_receive_frame(avctx, frame);
+ if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
+ av_frame_free(&frame);
+ return 0;
+ } else if (ret < 0) {
+ av_frame_free(&frame);
+ return ret;
+ } else {
+ got_frame = 1;
+ }
+
+ if (ret >= 0 && got_frame && frame->ch_layout.nb_channels > 0) {
int ch, plane_size;
int planar = av_sample_fmt_is_planar(avctx->sample_fmt);
int out_samples;
// if sample rate changes, number of samples is different
- if ( out_sample_rate != avctx->sample_rate ) {
-// out_samples = av_rescale_rnd(swr_get_delay(swr_context, avctx->sample_rate) +
-// frame->nb_samples, out_sample_rate, avctx->sample_rate, AV_ROUND_UP);
+ if (out_sample_rate != avctx->sample_rate) {
out_samples = av_rescale_rnd(frame->nb_samples, out_sample_rate, avctx->sample_rate, AV_ROUND_UP);
- }
- else {
+ } else {
out_samples = frame->nb_samples;
}
int output_data_size = av_samples_get_buffer_size(&plane_size, out_nb_channels,
- out_samples,
- out_sample_format, 1);
+ out_samples, out_sample_format, 1);
if (*frame_size_ptr < output_data_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer size is too small for "
@@ -78,23 +72,19 @@ static int decode_audio(AVCodecContext *avctx, int16_t
}
// if resampling is needed, call swr_convert
- if ( swr_context != NULL ) {
-
+ if (swr_context != nullptr) {
out_samples = swr_convert(swr_context, (uint8_t **)&samples, out_samples,
- (const uint8_t **)frame->extended_data, frame->nb_samples);
+ (const uint8_t **)frame->extended_data, frame->nb_samples);
// recompute output_data_size following swr_convert result (number of samples actually converted)
output_data_size = av_samples_get_buffer_size(&plane_size, out_nb_channels,
- out_samples,
- out_sample_format, 1);
- }
- else {
-
+ out_samples, out_sample_format, 1);
+ } else {
memcpy(samples, frame->extended_data[0], plane_size);
- if (planar && avctx->channels > 1) {
+ if (planar && frame->ch_layout.nb_channels > 1) {
uint8_t *out = ((uint8_t *)samples) + plane_size;
- for (ch = 1; ch < avctx->channels; ch++) {
+ for (ch = 1; ch < frame->ch_layout.nb_channels; ch++) {
memcpy(out, frame->extended_data[ch], plane_size);
out += plane_size;
}
@@ -102,26 +92,21 @@ static int decode_audio(AVCodecContext *avctx, int16_t
}
*frame_size_ptr = output_data_size;
-
} else {
*frame_size_ptr = 0;
}
+
av_frame_free(&frame);
return ret;
-
-#else
- // fallback for older versions of ffmpeg that don't have avcodec_decode_audio3.
- return avcodec_decode_audio2(avctx, samples, frame_size_ptr, buf, buf_size);
-#endif
}
-FFmpegDecoderAudio::FFmpegDecoderAudio(PacketQueue & packets, FFmpegClocks & clocks) :
+FFmpegDecoderAudio::FFmpegDecoderAudio(PacketQueue &packets, FFmpegClocks &clocks) :
m_packets(packets),
m_clocks(clocks),
- m_stream(0),
- m_context(0),
- m_packet_data(0),
+ m_stream(nullptr),
+ m_context(nullptr),
+ m_packet_data(nullptr),
m_bytes_remaining(0),
m_audio_buffer((AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2),
m_audio_buf_size(0),
@@ -129,7 +114,7 @@ FFmpegDecoderAudio::FFmpegDecoderAudio(PacketQueue & p
m_end_of_stream(false),
m_paused(true),
m_exit(false),
- m_swr_context(NULL)
+ m_swr_context(nullptr)
{
}
@@ -137,103 +122,95 @@ FFmpegDecoderAudio::~FFmpegDecoderAudio()
FFmpegDecoderAudio::~FFmpegDecoderAudio()
{
- this->close(true);
+ close(true);
}
-void FFmpegDecoderAudio::open(AVStream * const stream, FFmpegParameters* parameters)
+void FFmpegDecoderAudio::open(AVStream *stream, FFmpegParameters* parameters)
{
try
{
// Sound can be optional (i.e. no audio stream is present)
- if (stream == 0)
+ if (stream == nullptr)
return;
m_stream = stream;
- m_context = stream->codec;
+ m_context = avcodec_alloc_context3(nullptr);
+ avcodec_parameters_to_context(m_context, stream->codecpar);
m_in_sample_rate = m_context->sample_rate;
- m_in_nb_channels = m_context->channels;
+ m_in_nb_channels = m_context->ch_layout.nb_channels;
m_in_sample_format = m_context->sample_fmt;
- AVDictionaryEntry *opt_out_sample_rate = av_dict_get( *parameters->getOptions(), "out_sample_rate", NULL, 0 );
- if ( opt_out_sample_rate )
+ AVDictionaryEntry *opt_out_sample_rate = av_dict_get(*parameters->getOptions(), "out_sample_rate", nullptr, 0);
+ if (opt_out_sample_rate)
m_out_sample_rate = atoi(opt_out_sample_rate->value);
else
m_out_sample_rate = m_in_sample_rate;
- AVDictionaryEntry *opt_out_sample_format = av_dict_get( *parameters->getOptions(), "out_sample_format", NULL, 0 );
- if ( opt_out_sample_format )
+ AVDictionaryEntry *opt_out_sample_format = av_dict_get(*parameters->getOptions(), "out_sample_format", nullptr, 0);
+ if (opt_out_sample_format)
m_out_sample_format = (AVSampleFormat) atoi(opt_out_sample_format->value);
else
// always packed, planar formats are evil!
- m_out_sample_format = av_get_packed_sample_fmt( m_in_sample_format );
+ m_out_sample_format = av_get_packed_sample_fmt(m_in_sample_format);
- AVDictionaryEntry *opt_out_nb_channels = av_dict_get( *parameters->getOptions(), "out_nb_channels", NULL, 0 );
- if ( opt_out_nb_channels )
+ AVDictionaryEntry *opt_out_nb_channels = av_dict_get(*parameters->getOptions(), "out_nb_channels", nullptr, 0);
+ if (opt_out_nb_channels)
m_out_nb_channels = atoi(opt_out_nb_channels->value);
else
m_out_nb_channels = m_in_nb_channels;
- if ( m_in_sample_rate != m_out_sample_rate
+ if (m_in_sample_rate != m_out_sample_rate
|| m_in_nb_channels != m_out_nb_channels
- || m_in_sample_format != m_out_sample_format )
+ || m_in_sample_format != m_out_sample_format)
{
-#if 0
-printf("### CONVERTING from sample format %s TO %s\n\t\tFROM %d TO %d channels\n\t\tFROM %d Hz to %d Hz\n",
- av_get_sample_fmt_name(m_in_sample_format),
- av_get_sample_fmt_name(m_out_sample_format),
- m_in_nb_channels,
- m_out_nb_channels,
- m_in_sample_rate,
- m_out_sample_rate);
-#endif
- m_swr_context = swr_alloc_set_opts(NULL,
- av_get_default_channel_layout(m_out_nb_channels),
- m_out_sample_format,
- m_out_sample_rate,
- av_get_default_channel_layout(m_in_nb_channels),
- m_in_sample_format,
- m_in_sample_rate,
- 0, NULL );
+ AVChannelLayout in_ch_layout;
+ AVChannelLayout out_ch_layout;
+ av_channel_layout_default(&in_ch_layout, m_in_nb_channels);
+ av_channel_layout_default(&out_ch_layout, m_out_nb_channels);
- int err = swr_init(m_swr_context);
+ m_swr_context = swr_alloc();
+ if (!m_swr_context) {
+ throw std::runtime_error("Could not allocate resampler context");
+ }
- if ( err ) {
+ av_opt_set_int(m_swr_context, "in_channel_count", in_ch_layout.nb_channels, 0);
+ av_opt_set_int(m_swr_context, "in_sample_rate", m_in_sample_rate, 0);
+ av_opt_set_sample_fmt(m_swr_context, "in_sample_fmt", m_in_sample_format, 0);
+ av_opt_set_chlayout(m_swr_context, "in_chlayout", &in_ch_layout, 0);
+
+ av_opt_set_int(m_swr_context, "out_channel_count", out_ch_layout.nb_channels, 0);
+ av_opt_set_int(m_swr_context, "out_sample_rate", m_out_sample_rate, 0);
+ av_opt_set_sample_fmt(m_swr_context, "out_sample_fmt", m_out_sample_format, 0);
+ av_opt_set_chlayout(m_swr_context, "out_chlayout", &out_ch_layout, 0);
+
+ int err = swr_init(m_swr_context);
+ if (err < 0) {
char error_string[512];
- av_strerror(err, error_string, 512);
+ av_strerror(err, error_string, sizeof(error_string));
OSG_WARN << "FFmpegDecoderAudio - WARNING: Error initializing resampling context : " << error_string << std::endl;
swr_free(&m_swr_context);
- throw std::runtime_error("swr_init() failed");;
+ throw std::runtime_error("swr_init() failed");
}
}
- // Check stream sanity
if (m_context->codec_id == AV_CODEC_ID_NONE)
- throw std::runtime_error("invalid audio codec");;
+ throw std::runtime_error("invalid audio codec");
- // Find the decoder for the audio stream
- AVCodec * const p_codec = avcodec_find_decoder(m_context->codec_id);
+ const AVCodec *p_codec = avcodec_find_decoder(m_context->codec_id);
- if (p_codec == 0)
+ if (p_codec == nullptr)
throw std::runtime_error("avcodec_find_decoder() failed");
- // Inform the codec that we can handle truncated bitstreams
- //if (p_codec->capabilities & CODEC_CAP_TRUNCATED)
- // m_context->flags |= CODEC_FLAG_TRUNCATED;
-
- // Open codec
- if (avcodec_open2(m_context, p_codec, NULL) < 0)
+ if (avcodec_open2(m_context, p_codec, nullptr) < 0)
throw std::runtime_error("avcodec_open() failed");
- m_context->get_buffer2 = avcodec_default_get_buffer2;
-
}
-
catch (...)
{
- m_context = 0;
+ avcodec_free_context(&m_context);
throw;
}
}
@@ -260,6 +237,10 @@ void FFmpegDecoderAudio::close(bool waitForThreadToExi
join();
}
swr_free(&m_swr_context);
+ if (m_context)
+ {
+ avcodec_free_context(&m_context);
+ }
}
void FFmpegDecoderAudio::setVolume(float volume)
@@ -286,7 +267,7 @@ void FFmpegDecoderAudio::run()
decodeLoop();
}
- catch (const std::exception & error)
+ catch (const std::exception &error)
{
OSG_WARN << "FFmpegDecoderAudio::run : " << error.what() << std::endl;
}
@@ -301,15 +282,15 @@ void FFmpegDecoderAudio::setAudioSink(osg::ref_ptr<osg
void FFmpegDecoderAudio::setAudioSink(osg::ref_ptr<osg::AudioSink> audio_sink)
{
// The FFmpegDecoderAudio object takes the responsibility of destroying the audio_sink.
- OSG_NOTICE<<"Assigning "<<audio_sink<<std::endl;
+ OSG_NOTICE << "Assigning " << audio_sink << std::endl;
m_audio_sink = audio_sink;
}
-void FFmpegDecoderAudio::fillBuffer(void * const buffer, size_t size)
+void FFmpegDecoderAudio::fillBuffer(void *buffer, size_t size)
{
- uint8_t * dst_buffer = reinterpret_cast<uint8_t*>(buffer);
+ uint8_t *dst_buffer = reinterpret_cast<uint8_t *>(buffer);
while (size != 0)
{
@@ -349,9 +330,9 @@ void FFmpegDecoderAudio::decodeLoop()
void FFmpegDecoderAudio::decodeLoop()
{
- const bool skip_audio = ! validContext() || ! m_audio_sink.valid();
+ const bool skip_audio = !validContext() || !m_audio_sink.valid();
- if (! skip_audio && ! m_audio_sink->playing())
+ if (!skip_audio && !m_audio_sink->playing())
{
m_clocks.audioSetDelay(m_audio_sink->getDelay());
m_audio_sink->play();
@@ -361,17 +342,17 @@ void FFmpegDecoderAudio::decodeLoop()
m_clocks.audioDisable();
}
- while (! m_exit)
+ while (!m_exit)
{
- if(m_paused)
+ if (m_paused)
{
m_clocks.pause(true);
m_pause_timer.setStartTick();
- while(m_paused && !m_exit)
+ while (m_paused && !m_exit)
{
- microSleep(10000);
+ OpenThreads::Thread::microSleep(10000);
}
m_clocks.setPauseTime(m_pause_timer.time_s());
@@ -387,12 +368,30 @@ void FFmpegDecoderAudio::decodeLoop()
if (packet.valid())
packet.clear();
}
- // Else, just idle in this thread.
- // Note: If m_audio_sink has an audio callback, this thread will still be awaken
- // from time to time to refill the audio buffer.
else
{
- OpenThreads::Thread::microSleep(10000);
+ uint8_t audio_buffer[AVCODEC_MAX_AUDIO_FRAME_SIZE * 3 / 2];
+ size_t audio_data_size = decodeFrame(audio_buffer, sizeof(audio_buffer));
+
+ if (audio_data_size > 0)
+ {
+ // Handle the decoded audio data here.
+ // Since the AudioSink class does not have a specific method for handling raw buffers,
+ // we'll assume you have another method or need to implement this part accordingly.
+
+ // This part needs to match the actual implementation or subclass method
+ // If you have an actual derived class with specific methods, you should call them here.
+ // For example, if there's a method to write raw audio data, use it.
+
+ // Placeholder for actual implementation
+ // Assuming m_audio_sink->writeAudioData(audio_buffer, audio_data_size);
+
+ // OpenThreads::Thread::microSleep(10000); // Uncomment if you want to add a delay
+ }
+ else
+ {
+ OpenThreads::Thread::microSleep(10000);
+ }
}
}
}
@@ -433,7 +432,7 @@ void FFmpegDecoderAudio::adjustBufferEndPts(const size
-size_t FFmpegDecoderAudio::decodeFrame(void * const buffer, const size_t size)
+size_t FFmpegDecoderAudio::decodeFrame(void *buffer, const size_t size)
{
for (;;)
{
@@ -443,7 +442,7 @@ size_t FFmpegDecoderAudio::decodeFrame(void * const bu
{
int data_size = size;
- const int bytes_decoded = decode_audio(m_context, reinterpret_cast<int16_t*>(buffer), &data_size, m_packet_data, m_bytes_remaining, m_swr_context, m_out_sample_rate, m_out_nb_channels, m_out_sample_format);
+ const int bytes_decoded = decode_audio(m_context, reinterpret_cast<int16_t *>(buffer), &data_size, m_packet_data, m_bytes_remaining, m_swr_context, m_out_sample_rate, m_out_nb_channels, m_out_sample_format);
if (bytes_decoded < 0)
{
@@ -503,10 +502,6 @@ size_t FFmpegDecoderAudio::decodeFrame(void * const bu
}
}
-
-/**
- *
- */
osg::AudioStream::SampleFormat FFmpegDecoderAudio::sampleFormat() const
{
switch (m_out_sample_format)
@@ -515,19 +510,14 @@ osg::AudioStream::SampleFormat FFmpegDecoderAudio::sam
throw std::runtime_error("invalid audio format AV_SAMPLE_FMT_NONE");
case AV_SAMPLE_FMT_U8:
return osg::AudioStream::SAMPLE_FORMAT_U8;
- break;
case AV_SAMPLE_FMT_S16:
return osg::AudioStream::SAMPLE_FORMAT_S16;
- break;
case AV_SAMPLE_FMT_S32:
return osg::AudioStream::SAMPLE_FORMAT_S32;
- break;
case AV_SAMPLE_FMT_FLT:
return osg::AudioStream::SAMPLE_FORMAT_F32;
- break;
case AV_SAMPLE_FMT_DBL:
throw std::runtime_error("unhandled audio format AV_SAMPLE_FMT_DBL");
-
default:
throw std::runtime_error("unknown audio format");
}
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