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2026-09-14 09:23:00

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Signed-off-by: lizzie <lizzie@eden-emu.dev>
This commit is contained in:
lizzie
2026-09-14 09:23:00 +00:00
committed by crueter
parent 9bee2c2dc1
commit 02dccd61ed
1 changed files with 8 additions and 8 deletions
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src/audio_core/adsp/apps/opus/opus_decoder.cpp
+8 -8
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@@ -58,7 +58,7 @@ public:
/// idempotency of initialize is guaranteed
Result InitializeDecoder(u32 sample_rate, u32 total_stream_count, u32 channel_count, u32 stereo_stream_count, u8 const* mappings) {
LOG_INFO(Audio_DSP, "sample_rate={}, total_stream_count={}, channel_count={}, stereo_stream_count={}, mappings={}", sample_rate, total_stream_count, channel_count, stereo_stream_count, fmt::ptr(mappings));
LOG_DEBUG(Audio_DSP, "sample_rate={}, total_stream_count={}, channel_count={}, stereo_stream_count={}, mappings={}", sample_rate, total_stream_count, channel_count, stereo_stream_count, fmt::ptr(mappings));
ASSERT(channel_count >= 1 && channel_count <= 2);
// prefer libopus, ffmpeg docs say to use libopus **if** available
// However, native opus can also work with swrescale:
@@ -116,7 +116,7 @@ public:
}
Result Shutdown() {
LOG_INFO(Audio_DSP, "called avc={}", fmt::ptr(avc));
LOG_DEBUG(Audio_DSP, "called avc={}", fmt::ptr(avc));
swr_free(&swr);
av_frame_free(&frame);
av_packet_free(&pkt);
@@ -125,7 +125,7 @@ public:
}
Result ResetDecoder() {
LOG_INFO(Audio_DSP, "called avc={}", fmt::ptr(avc));
LOG_DEBUG(Audio_DSP, "called avc={}", fmt::ptr(avc));
if (avc) {
if (avcodec_is_open(avc)) avcodec_flush_buffers(avc);
return ResultSuccess;
@@ -134,20 +134,20 @@ public:
}
Result Decode(u32& out_sample_count, u64 output_data, u64 output_data_size, u64 input_data, u64 input_data_size, u8* const* tmp_buf) {
LOG_INFO(Audio_DSP, "called out_sample_count={},output_data={:#x},output_data_size={},input_data={:#x},input_data_size={}", fmt::ptr(&out_sample_count), output_data, output_data_size, input_data, input_data_size);
LOG_DEBUG(Audio_DSP, "called out_sample_count={},output_data={:#x},output_data_size={},input_data={:#x},input_data_size={}", fmt::ptr(&out_sample_count), output_data, output_data_size, input_data, input_data_size);
ASSERT(avc && avcodec_is_open(avc));
out_sample_count = 0;
int r;
LOG_INFO(Audio_DSP, "old packet data={}", fmt::ptr(pkt->data));
LOG_DEBUG(Audio_DSP, "old packet data={}", fmt::ptr(pkt->data));
if ((r = av_new_packet(pkt, int(input_data_size))) >= 0) {
LOG_INFO(Audio_DSP, "new packet data={}", fmt::ptr(pkt->data));
LOG_DEBUG(Audio_DSP, "new packet data={}", fmt::ptr(pkt->data));
std::memcpy(pkt->data, reinterpret_cast<const u8*>(input_data), input_data_size);
r = avcodec_send_packet(avc, pkt);
av_packet_unref(pkt);
if (r >= 0) {
while ((r = avcodec_receive_frame(avc, frame)) >= 0) {
LOG_INFO(Audio_DSP, "frame data={},data[0]={},nb_samples={}", fmt::ptr(frame->data), fmt::ptr(frame->data[0]), frame->nb_samples);
LOG_DEBUG(Audio_DSP, "frame data={},data[0]={},nb_samples={}", fmt::ptr(frame->data), fmt::ptr(frame->data[0]), frame->nb_samples);
ASSERT(std::in_range<u16>(frame->nb_samples));
u8 *dst_arr[2] = {reinterpret_cast<u8*>(output_data), nullptr};
if (frame->format == avc->request_sample_fmt) {
@@ -200,7 +200,7 @@ OpusDecoder::OpusDecoder(Core::System& system) {
while (!stop_token.stop_requested()) {
auto msg = Receive(Direction::DSP, stop_token);
LOG_INFO(Audio_DSP, "msg={}, buffer={}", msg, shared_memory->host_send_data[0]);
LOG_DEBUG(Audio_DSP, "msg={}, buffer={}", msg, shared_memory->host_send_data[0]);
switch (msg) {
case Shutdown:
Send(Direction::Host, Message::ShutdownOK);