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2026-09-13 20:00:54

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Signed-off-by: lizzie <lizzie@eden-emu.dev>
This commit is contained in:
lizzie
2026-09-13 20:00:54 +00:00
parent 72a270fb3a
commit 0dca1bb19a
1 changed files with 16 additions and 19 deletions
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src/audio_core/adsp/apps/opus/opus_decoder.cpp
+16 -19
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@@ -94,6 +94,7 @@ public:
avc->request_sample_fmt = AV_SAMPLE_FMT_S16;
av_channel_layout_default(&avc->ch_layout, channel_count);
if (avcodec_open2(avc, codec, nullptr) >= 0) {
ASSERT(avc->request_sample_fmt == AV_SAMPLE_FMT_S16);
pkt = av_packet_alloc();
frame = av_frame_alloc();
// opus fltp -> libopus s16
@@ -140,30 +141,27 @@ public:
if ((r = av_new_packet(pkt, int(input_data_size))) >= 0) {
std::memcpy(pkt->data, reinterpret_cast<const u8*>(input_data), input_data_size);
if ((r = avcodec_send_packet(avc, pkt)) >= 0) {
int rem_output_bytes = int(output_data_size);
av_frame_unref(frame);
do {
r = avcodec_receive_frame(avc, frame);
auto const input_bsize = av_samples_get_buffer_size(frame->linesize, frame->ch_layout.nb_channels, frame->nb_samples, AV_SAMPLE_FMT_S16, 1);
ASSERT(rem_output_bytes - input_bsize >= 0);
auto const dst_data = reinterpret_cast<s16*>(output_data + (int(output_data_size) - rem_output_bytes));
if (frame->format == AV_SAMPLE_FMT_S16) {
while ((r = avcodec_receive_frame(avc, frame)) >= 0) {
u8 *dst_arr[8] = {reinterpret_cast<u8*>(output_data), 0};
if (frame->format == avc->request_sample_fmt) {
// input_bsize == output_bsize
ASSERT(size_t(int(output_data_size) - rem_output_bytes + input_bsize) <= size_t(output_data_size));
std::memcpy(dst_data, frame->data[0], size_t(input_bsize));
av_samples_copy(dst_arr, frame->data, out_sample_count, 0, frame->nb_samples, frame->ch_layout.nb_channels, (enum AVSampleFormat)frame->format);
out_sample_count += frame->nb_samples;
} else {
int out_samples = int(av_rescale_rnd(swr_get_delay(swr, frame->sample_rate) + frame->nb_samples, frame->sample_rate, frame->sample_rate, AV_ROUND_UP));
uint8_t *output = nullptr;
av_samples_alloc(&output, frame->linesize, frame->ch_layout.nb_channels, out_samples, AV_SAMPLE_FMT_S16, 0);
out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **)frame->data, frame->nb_samples);
auto const output_bsize = av_samples_get_buffer_size(frame->linesize, frame->ch_layout.nb_channels, out_samples, AV_SAMPLE_FMT_S16, 1);
std::memcpy(dst_data, output, size_t(output_bsize));
av_freep(&output);
u8 *src_arr[8] = {0};
if ((r = av_samples_alloc(src_arr, frame->linesize, frame->ch_layout.nb_channels, out_samples, avc->request_sample_fmt, 0)) >= 0) {
out_samples = swr_convert(swr, src_arr, out_samples, (const u8 **)frame->data, frame->nb_samples);
av_samples_copy(dst_arr, src_arr, out_sample_count, 0, out_samples, frame->ch_layout.nb_channels, avc->request_sample_fmt);
av_freep(src_arr);
out_sample_count += out_samples;
} else {
LOG_ERROR(Audio_DSP, "skipped");
}
}
out_sample_count += frame->nb_samples;
rem_output_bytes -= input_bsize;
av_frame_unref(frame);
} while (r >= 0 && rem_output_bytes > 0);
}
if (r == AVERROR(EAGAIN) || r == AVERROR_EOF) {
// non-errors
LOG_DEBUG(Audio_DSP, "{}", r);
@@ -171,7 +169,6 @@ public:
// errors
LOG_ERROR(Audio_DSP, "{}", r);
}
ASSERT(rem_output_bytes == 0 && "remaining bytes!");
av_packet_unref(pkt);
return ResultSuccess;
} else {