From 02dccd61ed194b4ccc00f106aa57b3f6ecb7af18 Mon Sep 17 00:00:00 2001 From: lizzie Date: Mon, 14 Sep 2026 09:23:00 +0000 Subject: [PATCH] 2026-09-14 09:23:00 Signed-off-by: lizzie --- src/audio_core/adsp/apps/opus/opus_decoder.cpp | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/src/audio_core/adsp/apps/opus/opus_decoder.cpp b/src/audio_core/adsp/apps/opus/opus_decoder.cpp index 9ee7c95ac7..909116e89e 100644 --- a/src/audio_core/adsp/apps/opus/opus_decoder.cpp +++ b/src/audio_core/adsp/apps/opus/opus_decoder.cpp @@ -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(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(frame->nb_samples)); u8 *dst_arr[2] = {reinterpret_cast(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);