diff options
| -rw-r--r-- | NEWS | 1 | ||||
| -rw-r--r-- | src/decoder/DecoderBuffer.cxx | 12 | ||||
| -rw-r--r-- | src/decoder/DecoderBuffer.hxx | 11 | ||||
| -rw-r--r-- | src/decoder/plugins/FaadDecoderPlugin.cxx | 130 |
4 files changed, 100 insertions, 54 deletions
@@ -58,6 +58,7 @@ ver 0.18.12 (not yet released) - audiofile: improve responsiveness - audiofile: fix WAV stream playback - dsdiff, dsf: fix stream playback + - faad: estimate song duration for remote files - sndfile: improve responsiveness * randomize next song when enabling "random" mode while not playing * randomize next song when adding to single-song queue diff --git a/src/decoder/DecoderBuffer.cxx b/src/decoder/DecoderBuffer.cxx index 47671513e..258edfab8 100644 --- a/src/decoder/DecoderBuffer.cxx +++ b/src/decoder/DecoderBuffer.cxx @@ -69,6 +69,12 @@ decoder_buffer_free(DecoderBuffer *buffer) DeleteVarSize(buffer); } +const InputStream & +decoder_buffer_get_stream(const DecoderBuffer *buffer) +{ + return *buffer->is; +} + bool decoder_buffer_is_empty(const DecoderBuffer *buffer) { @@ -123,6 +129,12 @@ decoder_buffer_fill(DecoderBuffer *buffer) return true; } +size_t +decoder_buffer_available(const DecoderBuffer *buffer) +{ + return buffer->length - buffer->consumed;; +} + ConstBuffer<void> decoder_buffer_read(const DecoderBuffer *buffer) { diff --git a/src/decoder/DecoderBuffer.hxx b/src/decoder/DecoderBuffer.hxx index d6f303c36..f295eb0b5 100644 --- a/src/decoder/DecoderBuffer.hxx +++ b/src/decoder/DecoderBuffer.hxx @@ -55,6 +55,10 @@ void decoder_buffer_free(DecoderBuffer *buffer); gcc_pure +const InputStream & +decoder_buffer_get_stream(const DecoderBuffer *buffer); + +gcc_pure bool decoder_buffer_is_empty(const DecoderBuffer *buffer); @@ -76,6 +80,13 @@ bool decoder_buffer_fill(DecoderBuffer *buffer); /** + * How many bytes are stored in the buffer? + */ +gcc_pure +size_t +decoder_buffer_available(const DecoderBuffer *buffer); + +/** * Reads data from the buffer. This data is not yet consumed, you * have to call decoder_buffer_consume() to do that. The returned * buffer becomes invalid after a decoder_buffer_fill() or a diff --git a/src/decoder/plugins/FaadDecoderPlugin.cxx b/src/decoder/plugins/FaadDecoderPlugin.cxx index c7f72da15..7025536d6 100644 --- a/src/decoder/plugins/FaadDecoderPlugin.cxx +++ b/src/decoder/plugins/FaadDecoderPlugin.cxx @@ -35,8 +35,6 @@ #include <string.h> #include <unistd.h> -#define AAC_MAX_CHANNELS 6 - static const unsigned adts_sample_rates[] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 0 @@ -94,7 +92,7 @@ adts_find_frame(DecoderBuffer *buffer) } /* is it a frame? */ - size_t frame_length = adts_check_frame(data.data); + const size_t frame_length = adts_check_frame(data.data); if (frame_length == 0) { /* it's just some random 0xff byte; discard it and continue searching */ @@ -124,12 +122,18 @@ adts_find_frame(DecoderBuffer *buffer) static float adts_song_duration(DecoderBuffer *buffer) { + const InputStream &is = decoder_buffer_get_stream(buffer); + const bool estimate = !is.CheapSeeking(); + const auto file_size = is.GetSize(); + if (estimate && file_size <= 0) + return -1; + unsigned sample_rate = 0; /* Read all frames to ensure correct time and bitrate */ unsigned frames = 0; for (;; frames++) { - unsigned frame_length = adts_find_frame(buffer); + const unsigned frame_length = adts_find_frame(buffer); if (frame_length == 0) break; @@ -139,15 +143,35 @@ adts_song_duration(DecoderBuffer *buffer) assert(frame_length <= data.size); sample_rate = adts_sample_rates[(data.data[2] & 0x3c) >> 2]; + if (sample_rate == 0) + break; } decoder_buffer_consume(buffer, frame_length); + + if (estimate && frames == 128) { + /* if this is a remote file, don't slurp the + whole file just for checking the song + duration; instead, stop after some time and + extrapolate the song duration from what we + have until now */ + + const auto offset = is.GetOffset() + - decoder_buffer_available(buffer); + if (offset <= 0) + return -1; + + frames = (frames * file_size) / offset; + break; + } } - float frames_per_second = (float)sample_rate / 1024.0; - if (frames_per_second <= 0) + if (sample_rate == 0) return -1; + float frames_per_second = (float)sample_rate / 1024.0; + assert(frames_per_second > 0); + return (float)frames / frames_per_second; } @@ -171,7 +195,7 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is) tagsize += 10; - bool success = decoder_buffer_skip(buffer, tagsize) && + const bool success = decoder_buffer_skip(buffer, tagsize) && decoder_buffer_fill(buffer); if (!success) return -1; @@ -181,20 +205,21 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is) return -1; } - if (is.IsSeekable() && data.size >= 2 && - data.data[0] == 0xFF && ((data.data[1] & 0xF6) == 0xF0)) { + if (data.size >= 8 && adts_check_frame(data.data) > 0) { /* obtain the duration from the ADTS header */ + + if (!is.IsSeekable()) + return -1; + float song_length = adts_song_duration(buffer); is.LockSeek(tagsize, IgnoreError()); decoder_buffer_clear(buffer); - decoder_buffer_fill(buffer); return song_length; } else if (data.size >= 5 && memcmp(data.data, "ADIF", 4) == 0) { /* obtain the duration from the ADIF header */ - unsigned bit_rate; size_t skip_size = (data.data[4] & 0x80) ? 9 : 0; if (8 + skip_size > data.size) @@ -202,7 +227,7 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is) header */ return -1; - bit_rate = ((data.data[4 + skip_size] & 0x0F) << 19) | + unsigned bit_rate = ((data.data[4 + skip_size] & 0x0F) << 19) | (data.data[5 + skip_size] << 11) | (data.data[6 + skip_size] << 3) | (data.data[7 + skip_size] & 0xE0); @@ -215,6 +240,21 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is) return -1; } +static NeAACDecHandle +faad_decoder_new() +{ + const NeAACDecHandle decoder = NeAACDecOpen(); + + NeAACDecConfigurationPtr config = + NeAACDecGetCurrentConfiguration(decoder); + config->outputFormat = FAAD_FMT_16BIT; + config->downMatrix = 1; + config->dontUpSampleImplicitSBR = 0; + NeAACDecSetConfiguration(decoder, config); + + return decoder; +} + /** * Wrapper for NeAACDecInit() which works around some API * inconsistencies in libfaad. @@ -223,6 +263,13 @@ static bool faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer, AudioFormat &audio_format, Error &error) { + auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer)); + if (data.IsEmpty()) { + error.Set(faad_decoder_domain, "Empty file"); + return false; + } + + uint8_t channels; uint32_t sample_rate; #ifdef HAVE_FAAD_LONG /* neaacdec.h declares all arguments as "unsigned long", but @@ -232,19 +279,11 @@ faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer, #else uint32_t *sample_rate_p = &sample_rate; #endif - - auto data = ConstBuffer<uint8_t>::FromVoid(decoder_buffer_read(buffer)); - if (data.IsEmpty()) { - error.Set(faad_decoder_domain, "Empty file"); - return false; - } - - uint8_t channels; - int32_t nbytes = NeAACDecInit(decoder, - /* deconst hack, libfaad requires this */ - const_cast<uint8_t *>(data.data), - data.size, - sample_rate_p, &channels); + long nbytes = NeAACDecInit(decoder, + /* deconst hack, libfaad requires this */ + const_cast<unsigned char *>(data.data), + data.size, + sample_rate_p, &channels); if (nbytes < 0) { error.Set(faad_decoder_domain, "Not an AAC stream"); return false; @@ -284,16 +323,11 @@ faad_get_file_time_float(InputStream &is) { DecoderBuffer *buffer = decoder_buffer_new(nullptr, is, - FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS); + FAAD_MIN_STREAMSIZE * MAX_CHANNELS); float length = faad_song_duration(buffer, is); if (length < 0) { - NeAACDecHandle decoder = NeAACDecOpen(); - - NeAACDecConfigurationPtr config = - NeAACDecGetCurrentConfiguration(decoder); - config->outputFormat = FAAD_FMT_16BIT; - NeAACDecSetConfiguration(decoder, config); + NeAACDecHandle decoder = faad_decoder_new(); decoder_buffer_fill(buffer); @@ -318,13 +352,11 @@ faad_get_file_time_float(InputStream &is) static int faad_get_file_time(InputStream &is) { - int file_time = -1; - float length; - - if ((length = faad_get_file_time_float(is)) >= 0) - file_time = length + 0.5; + float length = faad_get_file_time_float(is); + if (length < 0) + return -1; - return file_time; + return int(length + 0.5); } static void @@ -332,19 +364,12 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is) { DecoderBuffer *buffer = decoder_buffer_new(&mpd_decoder, is, - FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS); + FAAD_MIN_STREAMSIZE * MAX_CHANNELS); const float total_time = faad_song_duration(buffer, is); /* create the libfaad decoder */ - NeAACDecHandle decoder = NeAACDecOpen(); - - NeAACDecConfigurationPtr config = - NeAACDecGetCurrentConfiguration(decoder); - config->outputFormat = FAAD_FMT_16BIT; - config->downMatrix = 1; - config->dontUpSampleImplicitSBR = 0; - NeAACDecSetConfiguration(decoder, config); + const NeAACDecHandle decoder = faad_decoder_new(); while (!decoder_buffer_is_full(buffer) && !is.LockIsEOF() && decoder_get_command(mpd_decoder) == DecoderCommand::NONE) { @@ -372,20 +397,18 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is) DecoderCommand cmd; unsigned bit_rate = 0; do { - size_t frame_size; - const void *decoded; - NeAACDecFrameInfo frame_info; - /* find the next frame */ - frame_size = adts_find_frame(buffer); + const size_t frame_size = adts_find_frame(buffer); if (frame_size == 0) /* end of file */ break; /* decode it */ - decoded = faad_decoder_decode(decoder, buffer, &frame_info); + NeAACDecFrameInfo frame_info; + const void *const decoded = + faad_decoder_decode(decoder, buffer, &frame_info); if (frame_info.error > 0) { FormatWarning(faad_decoder_domain, @@ -437,7 +460,6 @@ faad_scan_stream(InputStream &is, const struct tag_handler *handler, void *handler_ctx) { int file_time = faad_get_file_time(is); - if (file_time < 0) return false; |