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ffmpeg: Drop deprecated channel APIs usage
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@IonAgorria
IonAgorria committed Oct 27, 2024
1 parent 842d3e0 commit 60df22a
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Showing 3 changed files with 32 additions and 36 deletions.
2 changes: 0 additions & 2 deletions CMakeLists.txt
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Original file line number Diff line number Diff line change
Expand Up @@ -369,8 +369,6 @@ IF(NOT MSVC_CL_BUILD)
-Werror=format
#TODO remove these once code is more reasonable
-Wno-deprecated-copy
#TODO hide FFMPEG deprecated API usage warning for now
-Wno-deprecated-declarations
)
IF(CLANG)
set(PERIMETER_COMPILE_OPTIONS
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62 changes: 31 additions & 31 deletions Source/Util/AVWrapper.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -25,7 +25,7 @@ size_t AVWrapperFrame::getBufferSize() const {
} else if (AVWrapperType::Audio) {
AVSampleFormat format = static_cast<AVSampleFormat>(frame->format);
//return av_samples_get_buffer_size(nullptr, frame->channels, frame->nb_samples, format, 1);
int planes = av_sample_fmt_is_planar(format) ? frame->channels : 1;
int planes = av_sample_fmt_is_planar(format) ? frame->ch_layout.nb_channels : 1;
return frame->linesize[0] * planes;
} else {
xassert(0);
Expand All @@ -49,9 +49,9 @@ size_t AVWrapperFrame::copyBuffer(uint8_t** buffer) const {
return buffer_size;
} else if (AVWrapperType::Audio) {
AVSampleFormat format = static_cast<AVSampleFormat>(frame->format);
if (av_sample_fmt_is_planar(format) && 1 < frame->channels) {
if (av_sample_fmt_is_planar(format) && 1 < frame->ch_layout.nb_channels) {
//We need to pack planar audio
int planes = frame->channels;
int planes = frame->ch_layout.nb_channels;
size_t bytes_sample = av_get_bytes_per_sample(format);
xassert(buffer_size == frame->linesize[0] * planes);
//Store each plane sample as interleaved
Expand Down Expand Up @@ -120,8 +120,7 @@ int AVWrapper::close() {
if (filterGraph) {
avfilter_graph_free(&filterGraph);
}
swrChannels = 0;
swrChannelLayout = 0;
swrChannelLayout = {};
swrSampleRate = 0;
swrFormat = AV_SAMPLE_FMT_NONE;
#endif
Expand Down Expand Up @@ -163,9 +162,8 @@ void AVWrapper::pullFilterAudio() {
break;
}

int nb_channels = av_get_channel_layout_nb_channels(swrChannelLayout);
int bytes_per_sample = av_get_bytes_per_sample(swrFormat);
int data_size = converted->frame->nb_samples * nb_channels * bytes_per_sample;
int data_size = converted->frame->nb_samples * swrChannelLayout.nb_channels * bytes_per_sample;
converted->frame->linesize[0] = data_size;
converted->type = AVWrapperType::Audio;
converted->time_base = av_buffersink_get_time_base(buffersinkCtx);
Expand Down Expand Up @@ -339,8 +337,8 @@ bool AVWrapper::setupAudioConverter(AVSampleFormat dst_format, int dst_channels,
}
#ifdef PERIMETER_FFMPEG_MOVIE
swrFormat = dst_format != 0 ? dst_format : audioCodecCtx->sample_fmt;
swrChannels = dst_channels != 0 ? dst_channels : audioCodecCtx->channels;
swrChannelLayout = swrChannels == 0 ? 0 : av_get_default_channel_layout(swrChannels);
int num_channel = dst_channels != 0 ? dst_channels : audioCodecCtx->ch_layout.nb_channels;
av_channel_layout_default(&swrChannelLayout, num_channel);
swrSampleRate = dst_frequency != 0 ? dst_frequency : audioCodecCtx->sample_rate;
if (swrFormat == AV_SAMPLE_FMT_NONE) {
fprintf(stderr, "setupAudioConverter unsupported device output format %d at %s\n", dst_format, file_path.c_str());
Expand All @@ -353,23 +351,17 @@ bool AVWrapper::setupAudioConverter(AVSampleFormat dst_format, int dst_channels,
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
const int out_sample_fmts[] = { swrFormat, -1 };
const int64_t out_channel_layouts[] = { swrChannelLayout, -1 };
const int out_sample_rates[] = { swrSampleRate, -1 };
const AVFilterLink *outlink;
AVRational time_base = audioCodecCtx->time_base;

if (!audioCodecCtx->channel_layout) {
audioCodecCtx->channel_layout = av_get_default_channel_layout(audioCodecCtx->channels);
}

char channel_str[128];
snprintf(channel_str, sizeof(channel_str), "0x%" PRIx64, audioCodecCtx->channel_layout);
av_channel_layout_describe(&swrChannelLayout, channel_str, sizeof(channel_str));
std::string args = "time_base=" + std::to_string(time_base.num) + "/" + std::to_string(time_base.den) +
":sample_rate=" + std::to_string(audioCodecCtx->sample_rate) +
":sample_fmt=" + av_get_sample_fmt_name(audioCodecCtx->sample_fmt) +
":channel_layout=" + channel_str;

av_get_channel_layout_string(channel_str, sizeof(channel_str), swrChannels, swrChannelLayout);
std::string filters_descr;

//Add aresample filter if sample rates mismatch
Expand All @@ -381,12 +373,18 @@ bool AVWrapper::setupAudioConverter(AVSampleFormat dst_format, int dst_channels,
filters_descr += ":min_hard_comp=0.5";
filters_descr += ":async=" + std::to_string(swrSampleRate / 2);
}
//Convert into SDL audio format
if (!filters_descr.empty()) filters_descr += ",";
filters_descr += std::string("aformat=sample_fmts=") + av_get_sample_fmt_name(swrFormat) + ":channel_layouts=" + channel_str;

int ret = 0;

//Convert into SDL audio format
if (!filters_descr.empty()) filters_descr += ",";
const char* swrFormatName = av_get_sample_fmt_name(swrFormat);
if (!swrFormatName) {
fprintf(stderr, "setupAudioConverter not enough memory\n");
goto end;
}
filters_descr += std::string("aformat=sample_fmts=") + swrFormatName + ":channel_layouts=" + channel_str;

filterGraph = avfilter_graph_alloc();
if (!outputs || !inputs || !filterGraph) {
fprintf(stderr, "setupAudioConverter not enough memory\n");
Expand All @@ -396,32 +394,32 @@ bool AVWrapper::setupAudioConverter(AVSampleFormat dst_format, int dst_channels,
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
ret = avfilter_graph_create_filter(&buffersrcCtx, abuffersrc, "in", args.c_str(), nullptr, filterGraph);
if (ret < 0) {
fprintf(stderr, "Cannot create audio buffer source\n");
fprintf(stderr, "setupAudioConverter Cannot create audio buffer source\n");
goto end;
}

/* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersinkCtx, abuffersink, "out", nullptr, nullptr, filterGraph);
if (ret < 0) {
fprintf(stderr, "Cannot create audio buffer sink\n");
fprintf(stderr, "setupAudioConverter Cannot create audio buffer sink\n");
goto end;
}

ret = av_opt_set_int_list(buffersinkCtx, "sample_fmts", out_sample_fmts, -1, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
fprintf(stderr, "Cannot set output sample format\n");
fprintf(stderr, "setupAudioConverter Cannot set output sample format\n");
goto end;
}

ret = av_opt_set_int_list(buffersinkCtx, "channel_layouts", out_channel_layouts, -1, AV_OPT_SEARCH_CHILDREN);
ret = av_opt_set(buffersinkCtx, "ch_layouts", channel_str, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
fprintf(stderr, "Cannot set output channel layout\n");
fprintf(stderr, "setupAudioConverter Cannot set output channel layout\n");
goto end;
}

ret = av_opt_set_int_list(buffersinkCtx, "sample_rates", out_sample_rates, -1, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
fprintf(stderr, "Cannot set output sample rate\n");
fprintf(stderr, "setupAudioConverter Cannot set output sample rate\n");
goto end;
}

Expand Down Expand Up @@ -452,24 +450,26 @@ bool AVWrapper::setupAudioConverter(AVSampleFormat dst_format, int dst_channels,
inputs->pad_idx = 0;
inputs->next = nullptr;

if ((ret = avfilter_graph_parse_ptr(filterGraph, filters_descr.c_str(), &inputs, &outputs, nullptr)) < 0)
if ((ret = avfilter_graph_parse_ptr(filterGraph, filters_descr.c_str(), &inputs, &outputs, nullptr)) < 0) {
fprintf(stderr, "setupAudioConverter Error occurred at avfilter_graph_parse_ptr\n");
goto end;
}

if ((ret = avfilter_graph_config(filterGraph, nullptr)) < 0)
if ((ret = avfilter_graph_config(filterGraph, nullptr)) < 0) {
fprintf(stderr, "setupAudioConverter Error occurred at avfilter_graph_config\n");
goto end;
}

#if 1
/* Print summary of the sink buffer */
outlink = buffersinkCtx->inputs[0];
av_get_channel_layout_string(channel_str, sizeof(channel_str), -1, outlink->channel_layout);
av_channel_layout_describe(&outlink->ch_layout, channel_str, sizeof(channel_str));
fprintf(
stdout, "AVWrapper audio conv: %s\n -> %s\n -> %dHz %s %s\n",
args.c_str(), filters_descr.c_str(),
outlink->sample_rate,
static_cast<char*>(av_x_if_null(av_get_sample_fmt_name(static_cast<AVSampleFormat>(outlink->format)), "?")),
channel_str
);
#endif

end:
avfilter_inout_free(&inputs);
Expand All @@ -478,7 +478,7 @@ bool AVWrapper::setupAudioConverter(AVSampleFormat dst_format, int dst_channels,
if (filterGraph) {
avfilter_graph_free(&filterGraph);
}
fprintf(stderr, "Error occurred when setting up audio conv\n");
fprintf(stderr, "setupAudioConverter Error occurred when setting up audio conv\n");
xassert(0);
return false;
} else {
Expand Down
4 changes: 1 addition & 3 deletions Source/Util/AVWrapper.h
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Original file line number Diff line number Diff line change
Expand Up @@ -111,10 +111,8 @@ class AVWrapper
int swsHeight = 0;
///Software scaler output frame format
AVPixelFormat swsFormat = AV_PIX_FMT_NONE;
///Audio converter output channels
int swrChannels = 0;
///Audio converter output channel layout
int64_t swrChannelLayout = 0;
AVChannelLayout swrChannelLayout = {};
///Audio converter output rate
int swrSampleRate = 0;
///Audio converter output format
Expand Down

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