Merge branch 'v0.18.x'
This commit is contained in:
commit
c400876df1
1
NEWS
1
NEWS
@ -58,6 +58,7 @@ ver 0.18.12 (not yet released)
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- audiofile: improve responsiveness
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- audiofile: improve responsiveness
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- audiofile: fix WAV stream playback
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- audiofile: fix WAV stream playback
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- dsdiff, dsf: fix stream playback
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- dsdiff, dsf: fix stream playback
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- faad: estimate song duration for remote files
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- sndfile: improve responsiveness
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- sndfile: improve responsiveness
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* randomize next song when enabling "random" mode while not playing
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* randomize next song when enabling "random" mode while not playing
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* randomize next song when adding to single-song queue
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* randomize next song when adding to single-song queue
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@ -69,6 +69,12 @@ decoder_buffer_free(DecoderBuffer *buffer)
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DeleteVarSize(buffer);
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DeleteVarSize(buffer);
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}
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}
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const InputStream &
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decoder_buffer_get_stream(const DecoderBuffer *buffer)
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{
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return *buffer->is;
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}
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bool
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bool
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decoder_buffer_is_empty(const DecoderBuffer *buffer)
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decoder_buffer_is_empty(const DecoderBuffer *buffer)
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{
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{
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@ -123,6 +129,12 @@ decoder_buffer_fill(DecoderBuffer *buffer)
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return true;
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return true;
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}
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}
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size_t
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decoder_buffer_available(const DecoderBuffer *buffer)
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{
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return buffer->length - buffer->consumed;;
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}
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ConstBuffer<void>
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ConstBuffer<void>
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decoder_buffer_read(const DecoderBuffer *buffer)
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decoder_buffer_read(const DecoderBuffer *buffer)
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{
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{
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@ -54,6 +54,10 @@ decoder_buffer_new(Decoder *decoder, InputStream &is,
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void
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void
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decoder_buffer_free(DecoderBuffer *buffer);
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decoder_buffer_free(DecoderBuffer *buffer);
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gcc_pure
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const InputStream &
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decoder_buffer_get_stream(const DecoderBuffer *buffer);
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gcc_pure
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gcc_pure
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bool
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bool
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decoder_buffer_is_empty(const DecoderBuffer *buffer);
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decoder_buffer_is_empty(const DecoderBuffer *buffer);
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@ -75,6 +79,13 @@ decoder_buffer_clear(DecoderBuffer *buffer);
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bool
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bool
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decoder_buffer_fill(DecoderBuffer *buffer);
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decoder_buffer_fill(DecoderBuffer *buffer);
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/**
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* How many bytes are stored in the buffer?
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*/
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gcc_pure
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size_t
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decoder_buffer_available(const DecoderBuffer *buffer);
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/**
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/**
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* Reads data from the buffer. This data is not yet consumed, you
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* Reads data from the buffer. This data is not yet consumed, you
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* have to call decoder_buffer_consume() to do that. The returned
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* have to call decoder_buffer_consume() to do that. The returned
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@ -35,8 +35,6 @@
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#include <string.h>
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#include <string.h>
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#include <unistd.h>
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#include <unistd.h>
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#define AAC_MAX_CHANNELS 6
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static const unsigned adts_sample_rates[] =
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static const unsigned adts_sample_rates[] =
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{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
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{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
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16000, 12000, 11025, 8000, 7350, 0, 0, 0
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16000, 12000, 11025, 8000, 7350, 0, 0, 0
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@ -94,7 +92,7 @@ adts_find_frame(DecoderBuffer *buffer)
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}
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}
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/* is it a frame? */
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/* is it a frame? */
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size_t frame_length = adts_check_frame(data.data);
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const size_t frame_length = adts_check_frame(data.data);
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if (frame_length == 0) {
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if (frame_length == 0) {
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/* it's just some random 0xff byte; discard it
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/* it's just some random 0xff byte; discard it
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and continue searching */
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and continue searching */
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@ -124,12 +122,18 @@ adts_find_frame(DecoderBuffer *buffer)
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static float
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static float
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adts_song_duration(DecoderBuffer *buffer)
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adts_song_duration(DecoderBuffer *buffer)
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{
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{
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const InputStream &is = decoder_buffer_get_stream(buffer);
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const bool estimate = !is.CheapSeeking();
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const auto file_size = is.GetSize();
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if (estimate && file_size <= 0)
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return -1;
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unsigned sample_rate = 0;
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unsigned sample_rate = 0;
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/* Read all frames to ensure correct time and bitrate */
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/* Read all frames to ensure correct time and bitrate */
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unsigned frames = 0;
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unsigned frames = 0;
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for (;; frames++) {
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for (;; frames++) {
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unsigned frame_length = adts_find_frame(buffer);
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const unsigned frame_length = adts_find_frame(buffer);
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if (frame_length == 0)
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if (frame_length == 0)
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break;
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break;
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@ -139,15 +143,35 @@ adts_song_duration(DecoderBuffer *buffer)
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assert(frame_length <= data.size);
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assert(frame_length <= data.size);
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sample_rate = adts_sample_rates[(data.data[2] & 0x3c) >> 2];
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sample_rate = adts_sample_rates[(data.data[2] & 0x3c) >> 2];
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if (sample_rate == 0)
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break;
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}
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}
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decoder_buffer_consume(buffer, frame_length);
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decoder_buffer_consume(buffer, frame_length);
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if (estimate && frames == 128) {
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/* if this is a remote file, don't slurp the
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whole file just for checking the song
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duration; instead, stop after some time and
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extrapolate the song duration from what we
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have until now */
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const auto offset = is.GetOffset()
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- decoder_buffer_available(buffer);
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if (offset <= 0)
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return -1;
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frames = (frames * file_size) / offset;
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break;
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}
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}
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}
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float frames_per_second = (float)sample_rate / 1024.0;
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if (sample_rate == 0)
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if (frames_per_second <= 0)
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return -1;
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return -1;
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float frames_per_second = (float)sample_rate / 1024.0;
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assert(frames_per_second > 0);
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return (float)frames / frames_per_second;
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return (float)frames / frames_per_second;
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}
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}
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@ -171,7 +195,7 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
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tagsize += 10;
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tagsize += 10;
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bool success = decoder_buffer_skip(buffer, tagsize) &&
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const bool success = decoder_buffer_skip(buffer, tagsize) &&
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decoder_buffer_fill(buffer);
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decoder_buffer_fill(buffer);
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if (!success)
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if (!success)
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return -1;
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return -1;
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@ -181,20 +205,21 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
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return -1;
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return -1;
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}
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}
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if (is.IsSeekable() && data.size >= 2 &&
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if (data.size >= 8 && adts_check_frame(data.data) > 0) {
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data.data[0] == 0xFF && ((data.data[1] & 0xF6) == 0xF0)) {
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/* obtain the duration from the ADTS header */
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/* obtain the duration from the ADTS header */
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if (!is.IsSeekable())
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return -1;
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float song_length = adts_song_duration(buffer);
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float song_length = adts_song_duration(buffer);
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is.LockSeek(tagsize, IgnoreError());
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is.LockSeek(tagsize, IgnoreError());
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decoder_buffer_clear(buffer);
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decoder_buffer_clear(buffer);
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decoder_buffer_fill(buffer);
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return song_length;
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return song_length;
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} else if (data.size >= 5 && memcmp(data.data, "ADIF", 4) == 0) {
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} else if (data.size >= 5 && memcmp(data.data, "ADIF", 4) == 0) {
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/* obtain the duration from the ADIF header */
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/* obtain the duration from the ADIF header */
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unsigned bit_rate;
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size_t skip_size = (data.data[4] & 0x80) ? 9 : 0;
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size_t skip_size = (data.data[4] & 0x80) ? 9 : 0;
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if (8 + skip_size > data.size)
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if (8 + skip_size > data.size)
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@ -202,7 +227,7 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
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header */
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header */
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return -1;
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return -1;
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bit_rate = ((data.data[4 + skip_size] & 0x0F) << 19) |
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unsigned bit_rate = ((data.data[4 + skip_size] & 0x0F) << 19) |
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(data.data[5 + skip_size] << 11) |
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(data.data[5 + skip_size] << 11) |
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(data.data[6 + skip_size] << 3) |
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(data.data[6 + skip_size] << 3) |
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(data.data[7 + skip_size] & 0xE0);
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(data.data[7 + skip_size] & 0xE0);
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@ -215,6 +240,21 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
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return -1;
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return -1;
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}
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}
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static NeAACDecHandle
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faad_decoder_new()
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{
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const NeAACDecHandle decoder = NeAACDecOpen();
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NeAACDecConfigurationPtr config =
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NeAACDecGetCurrentConfiguration(decoder);
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config->outputFormat = FAAD_FMT_16BIT;
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config->downMatrix = 1;
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config->dontUpSampleImplicitSBR = 0;
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NeAACDecSetConfiguration(decoder, config);
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return decoder;
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}
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/**
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/**
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* Wrapper for NeAACDecInit() which works around some API
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* Wrapper for NeAACDecInit() which works around some API
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* inconsistencies in libfaad.
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* inconsistencies in libfaad.
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@ -223,6 +263,13 @@ static bool
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faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
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faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
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AudioFormat &audio_format, Error &error)
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AudioFormat &audio_format, Error &error)
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{
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{
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auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer));
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if (data.IsEmpty()) {
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error.Set(faad_decoder_domain, "Empty file");
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return false;
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}
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uint8_t channels;
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uint32_t sample_rate;
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uint32_t sample_rate;
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#ifdef HAVE_FAAD_LONG
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#ifdef HAVE_FAAD_LONG
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/* neaacdec.h declares all arguments as "unsigned long", but
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/* neaacdec.h declares all arguments as "unsigned long", but
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@ -232,19 +279,11 @@ faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
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#else
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#else
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uint32_t *sample_rate_p = &sample_rate;
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uint32_t *sample_rate_p = &sample_rate;
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#endif
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#endif
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long nbytes = NeAACDecInit(decoder,
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auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer));
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/* deconst hack, libfaad requires this */
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if (data.IsEmpty()) {
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const_cast<unsigned char *>(data.data),
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error.Set(faad_decoder_domain, "Empty file");
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data.size,
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return false;
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sample_rate_p, &channels);
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}
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uint8_t channels;
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int32_t nbytes = NeAACDecInit(decoder,
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/* deconst hack, libfaad requires this */
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const_cast<uint8_t *>(data.data),
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data.size,
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sample_rate_p, &channels);
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if (nbytes < 0) {
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if (nbytes < 0) {
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error.Set(faad_decoder_domain, "Not an AAC stream");
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error.Set(faad_decoder_domain, "Not an AAC stream");
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return false;
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return false;
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@ -284,16 +323,11 @@ faad_get_file_time_float(InputStream &is)
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{
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{
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DecoderBuffer *buffer =
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DecoderBuffer *buffer =
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decoder_buffer_new(nullptr, is,
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decoder_buffer_new(nullptr, is,
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FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
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FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
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float length = faad_song_duration(buffer, is);
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float length = faad_song_duration(buffer, is);
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if (length < 0) {
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if (length < 0) {
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NeAACDecHandle decoder = NeAACDecOpen();
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NeAACDecHandle decoder = faad_decoder_new();
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NeAACDecConfigurationPtr config =
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NeAACDecGetCurrentConfiguration(decoder);
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config->outputFormat = FAAD_FMT_16BIT;
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NeAACDecSetConfiguration(decoder, config);
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decoder_buffer_fill(buffer);
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decoder_buffer_fill(buffer);
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@ -318,13 +352,11 @@ faad_get_file_time_float(InputStream &is)
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static int
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static int
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faad_get_file_time(InputStream &is)
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faad_get_file_time(InputStream &is)
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{
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{
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int file_time = -1;
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float length = faad_get_file_time_float(is);
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float length;
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if (length < 0)
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return -1;
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if ((length = faad_get_file_time_float(is)) >= 0)
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return int(length + 0.5);
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file_time = length + 0.5;
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return file_time;
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}
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}
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static void
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static void
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@ -332,19 +364,12 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
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{
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{
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DecoderBuffer *buffer =
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DecoderBuffer *buffer =
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decoder_buffer_new(&mpd_decoder, is,
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decoder_buffer_new(&mpd_decoder, is,
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FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
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FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
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const float total_time = faad_song_duration(buffer, is);
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const float total_time = faad_song_duration(buffer, is);
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/* create the libfaad decoder */
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/* create the libfaad decoder */
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NeAACDecHandle decoder = NeAACDecOpen();
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const NeAACDecHandle decoder = faad_decoder_new();
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NeAACDecConfigurationPtr config =
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NeAACDecGetCurrentConfiguration(decoder);
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config->outputFormat = FAAD_FMT_16BIT;
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config->downMatrix = 1;
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config->dontUpSampleImplicitSBR = 0;
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NeAACDecSetConfiguration(decoder, config);
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while (!decoder_buffer_is_full(buffer) && !is.LockIsEOF() &&
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while (!decoder_buffer_is_full(buffer) && !is.LockIsEOF() &&
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decoder_get_command(mpd_decoder) == DecoderCommand::NONE) {
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decoder_get_command(mpd_decoder) == DecoderCommand::NONE) {
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@ -372,20 +397,18 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
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DecoderCommand cmd;
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DecoderCommand cmd;
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unsigned bit_rate = 0;
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unsigned bit_rate = 0;
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do {
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do {
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size_t frame_size;
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const void *decoded;
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NeAACDecFrameInfo frame_info;
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/* find the next frame */
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/* find the next frame */
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frame_size = adts_find_frame(buffer);
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const size_t frame_size = adts_find_frame(buffer);
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if (frame_size == 0)
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if (frame_size == 0)
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/* end of file */
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/* end of file */
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break;
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break;
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/* decode it */
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/* decode it */
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decoded = faad_decoder_decode(decoder, buffer, &frame_info);
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NeAACDecFrameInfo frame_info;
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const void *const decoded =
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faad_decoder_decode(decoder, buffer, &frame_info);
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if (frame_info.error > 0) {
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if (frame_info.error > 0) {
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FormatWarning(faad_decoder_domain,
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FormatWarning(faad_decoder_domain,
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@ -437,7 +460,6 @@ faad_scan_stream(InputStream &is,
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const struct tag_handler *handler, void *handler_ctx)
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const struct tag_handler *handler, void *handler_ctx)
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{
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{
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int file_time = faad_get_file_time(is);
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int file_time = faad_get_file_time(is);
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if (file_time < 0)
|
if (file_time < 0)
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return false;
|
return false;
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||||||
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|
||||||
|
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