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https://github.com/sle118/squeezelite-esp32.git
synced 2025-12-06 19:47:02 +03:00
more aac correction - release
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@@ -358,7 +358,7 @@ static decode_state alac_decode(void) {
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return DECODE_COMPLETE;
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}
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// enough data for coding
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// is there enough data for decoding
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if (bytes < block_size) {
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UNLOCK_S;
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return DECODE_RUNNING;
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@@ -389,19 +389,25 @@ static decode_state helixaac_decode(void) {
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}
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if (found == 1) {
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LOG_INFO("samplerate: %u channels: %u", samplerate, channels);
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bytes_total = _buf_used(streambuf);
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bytes_wrap = min(bytes_total, _buf_cont_read(streambuf));
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LOCK_O;
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LOG_INFO("setting track_start");
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output.next_sample_rate = decode_newstream(samplerate, output.supported_rates);
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IF_DSD( output.next_fmt = PCM; )
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output.track_start = outputbuf->writep;
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if (output.fade_mode) _checkfade(true);
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decode.new_stream = false;
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UNLOCK_O;
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LOG_INFO("setting track start, samplerate: %u channels: %u", samplerate, channels);
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bytes_total = _buf_used(streambuf);
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bytes_wrap = min(bytes_total, _buf_cont_read(streambuf));
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// come back later if we don' thave enough data
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if (bytes_total < WRAPBUF_LEN) {
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UNLOCK_S;
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LOG_INFO("need more audio data");
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return DECODE_RUNNING;
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}
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} else if (found == -1) {
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@@ -418,8 +424,9 @@ static decode_state helixaac_decode(void) {
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}
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}
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if (bytes_wrap < WRAPBUF_LEN && (bytes_total > WRAPBUF_LEN || stream.state <= DISCONNECT)) {
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// make a local copy of frames which may have wrapped round the end of streambuf
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// we always have at least WRAPBUF_LEN unless it's the end of a stream
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if (bytes_wrap < WRAPBUF_LEN) {
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// build a linear buffer if we are crossing the end of streambuf
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memcpy(a->wrap_buf, streambuf->readp, bytes_wrap);
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memcpy(a->wrap_buf + bytes_wrap, streambuf->buf, min(WRAPBUF_LEN, bytes_total) - bytes_wrap);
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