diff options
Diffstat (limited to 'src/odr-audioenc.cpp')
| -rw-r--r-- | src/odr-audioenc.cpp | 45 |
1 files changed, 35 insertions, 10 deletions
diff --git a/src/odr-audioenc.cpp b/src/odr-audioenc.cpp index f4cf01e..45aacf2 100644 --- a/src/odr-audioenc.cpp +++ b/src/odr-audioenc.cpp @@ -34,6 +34,7 @@ * Interesting starting points for the encoder * - \ref odr-audioenc.cpp Main encoder file * - \ref VLCInput.h VLC Input + * - \ref GSTInput.h GST Input * - \ref AlsaInput.h Alsa Input * - \ref JackInput.h JACK Input * - \ref Outputs.h ZeroMQ, file and EDI outputs @@ -53,6 +54,7 @@ #include "FileInput.h" #include "JackInput.h" #include "VLCInput.h" +#include "GSTInput.h" #include "SampleQueue.h" #include "AACDecoder.h" #include "StatsPublish.h" @@ -65,6 +67,7 @@ extern "C" { #include "utils.h" } +#include <algorithm> #include <vector> #include <deque> #include <chrono> @@ -103,7 +106,7 @@ void usage(const char* name) "ODR-AudioEnc %s is an audio encoder for both DAB and DAB+.\n" "The encoder can read from JACK, ALSA or\n" "a file source and encode to a ZeroMQ output for ODR-DabMux.\n" - "It can also use libvlc as an input.\n" + "It can also use libvlc and GStreamer as input.\n" "\n" "The -D option enables sound card clock drift compensation.\n" "A consumer sound card has a clock that is always a bit imprecise, and\n" @@ -162,6 +165,12 @@ void usage(const char* name) #else " The VLC input was disabled at compile-time\n" #endif + " For the GStreamer input:\n" +#if HAVE_GST + " -G, --gst-uri=uri Enable GStreamer input and use the URI given as source\n" +#else + " The GStreamer input was disabled at compile-time\n" +#endif " Drift compensation\n" " -D, --drift-comp Enable ALSA/VLC sound card drift compensation.\n" " Encoder parameters:\n" @@ -414,6 +423,8 @@ public: vector<string> vlc_additional_opts; unsigned verbosity = 0; + string gst_uri; + string jack_name; bool drift_compensation = false; @@ -449,8 +460,8 @@ public: string dab_channel_mode; /* Keep track of peaks */ - int peak_left = 0; - int peak_right = 0; + int16_t peak_left = 0; + int16_t peak_right = 0; /* On silence, die after the silence_timeout expires */ bool die_on_silence = false; @@ -487,7 +498,7 @@ public: ~AudioEnc(); int run(); - bool send_frame(const uint8_t *buf, size_t len, int peak_left, int peak_right); + bool send_frame(const uint8_t *buf, size_t len, int16_t peak_left, int16_t peak_right); shared_ptr<InputInterface> initialise_input(); }; @@ -504,6 +515,9 @@ int AudioEnc::run() #if HAVE_VLC if (not vlc_uri.empty()) num_inputs++; #endif +#if HAVE_GST + if (not gst_uri.empty()) num_inputs++; +#endif if (num_inputs == 0) { fprintf(stderr, "No input defined!\n"); @@ -977,8 +991,8 @@ int AudioEnc::run() for (int i = 0; i < read_bytes; i+=4) { int16_t l = input_buf[i] | (input_buf[i+1] << 8); int16_t r = input_buf[i+2] | (input_buf[i+3] << 8); - peak_left = MAX(peak_left, l); - peak_right = MAX(peak_right, r); + peak_left = std::max(peak_left, l); + peak_right = std::max(peak_right, r); } if (stats_publisher) { @@ -992,7 +1006,7 @@ int AudioEnc::run() * threshold is 0, and not configurable. The rationale is that we want to * guard against connection issues, not source level issues. */ - if (die_on_silence && MAX(peak_left, peak_right) == 0) { + if (die_on_silence && std::max(peak_left, peak_right) == 0) { const unsigned int frame_time_msec = 1000ul * read_bytes / (BYTES_PER_SAMPLE * channels * sample_rate); @@ -1174,7 +1188,7 @@ int AudioEnc::run() if (show_level) { if (channels == 1) { fprintf(stderr, "\rIn: [%-6s] %1s %1s %1s", - level(1, MAX(peak_right, peak_left)), + level(1, std::max(peak_right, peak_left)), status & STATUS_PAD_INSERTED ? "P" : " ", status & STATUS_UNDERRUN ? "U" : " ", status & STATUS_OVERRUN ? "O" : " "); @@ -1215,7 +1229,7 @@ int AudioEnc::run() return retval; } -bool AudioEnc::send_frame(const uint8_t *buf, size_t len, int peak_left, int peak_right) +bool AudioEnc::send_frame(const uint8_t *buf, size_t len, int16_t peak_left, int16_t peak_right) { if (file_output) { file_output->update_audio_levels(peak_left, peak_right); @@ -1287,6 +1301,11 @@ shared_ptr<InputInterface> AudioEnc::initialise_input() queue); } #endif +#if HAVE_GST + else if (not gst_uri.empty()) { + input = make_shared<GSTInput>(gst_uri, sample_rate, channels, queue); + } +#endif #if HAVE_ALSA else if (drift_compensation) { input = make_shared<AlsaInputThreaded>(alsa_device, channels, @@ -1322,6 +1341,7 @@ int main(int argc, char *argv[]) {"timestamp-delay", required_argument, 0, 'T'}, {"decode", required_argument, 0, 6 }, {"format", required_argument, 0, 'f'}, + {"gst-uri", required_argument, 0, 'G'}, {"input", required_argument, 0, 'i'}, {"jack", required_argument, 0, 'j'}, {"output", required_argument, 0, 'o'}, @@ -1372,7 +1392,7 @@ int main(int argc, char *argv[]) int ch=0; int index; while(ch != -1) { - ch = getopt_long(argc, argv, "aAhDlRVb:B:c:e:f:i:j:k:L:o:r:d:p:P:s:S:T:v:w:Wg:C:", longopts, &index); + ch = getopt_long(argc, argv, "aAhDlRVb:B:c:e:f:G:i:j:k:L:o:r:d:p:P:s:S:T:v:w:Wg:C:", longopts, &index); switch (ch) { case 0: // AAC-LC audio_enc.aot = AOT_DABPLUS_AAC_LC; @@ -1442,6 +1462,11 @@ int main(int argc, char *argv[]) return 1; } break; +#ifdef HAVE_GST + case 'G': + audio_enc.gst_uri = optarg; + break; +#endif case 'i': audio_enc.infile = optarg; break; |