--- /dev/null
+/***
+ This file is part of PulseAudio.
+
+ Copyright (C) 2020 Asymptotic <sanchayan@asymptotic.io>
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as
+ published by the Free Software Foundation; either version 2.1 of the
+ License, or (at your option) any later version.
+
+ PulseAudio is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public
+ License along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <arpa/inet.h>
+
+#include <pulsecore/log.h>
+#include <pulsecore/macro.h>
+#include <pulsecore/once.h>
+#include <pulsecore/core-util.h>
+#include <pulse/sample.h>
+
+#include "a2dp-codecs.h"
+#include "a2dp-codec-api.h"
+#include "a2dp-codec-gst.h"
+
+/* Called from the GStreamer streaming thread */
+static void enc_sink_eos(GstAppSink *appsink, gpointer userdata) {
+ pa_log_debug("Encoder got EOS");
+}
+
+/* Called from the GStreamer streaming thread */
+static GstFlowReturn enc_sink_new_sample(GstAppSink *appsink, gpointer userdata) {
+ struct gst_info *info = (struct gst_info *) userdata;
+ GstSample *sample = NULL;
+ GstBuffer *buf;
+
+ sample = gst_app_sink_pull_sample(GST_APP_SINK(info->enc_sink));
+ if (!sample)
+ return GST_FLOW_OK;
+
+ buf = gst_sample_get_buffer(sample);
+ gst_buffer_ref(buf);
+ gst_adapter_push(info->enc_adapter, buf);
+ gst_sample_unref(sample);
+ pa_fdsem_post(info->enc_fdsem);
+
+ return GST_FLOW_OK;
+}
+
+/* Called from the GStreamer streaming thread */
+static void dec_sink_eos(GstAppSink *appsink, gpointer userdata) {
+ pa_log_debug("Decoder got EOS");
+}
+
+/* Called from the GStreamer streaming thread */
+static GstFlowReturn dec_sink_new_sample(GstAppSink *appsink, gpointer userdata) {
+ struct gst_info *info = (struct gst_info *) userdata;
+ GstSample *sample = NULL;
+ GstBuffer *buf;
+
+ sample = gst_app_sink_pull_sample(GST_APP_SINK(info->dec_sink));
+ if (!sample)
+ return GST_FLOW_OK;
+
+ buf = gst_sample_get_buffer(sample);
+ gst_buffer_ref(buf);
+ gst_adapter_push(info->dec_adapter, buf);
+ gst_sample_unref(sample);
+ pa_fdsem_post(info->dec_fdsem);
+
+ return GST_FLOW_OK;
+}
+
+static void gst_deinit_enc_common(struct gst_info *info) {
+ if (!info)
+ return;
+ if (info->enc_fdsem)
+ pa_fdsem_free(info->enc_fdsem);
+ if (info->enc_src)
+ gst_object_unref(info->enc_src);
+ if (info->gst_enc)
+ gst_object_unref(info->gst_enc);
+ if (info->enc_sink)
+ gst_object_unref(info->enc_sink);
+ if (info->enc_adapter)
+ g_object_unref(info->enc_adapter);
+ if (info->enc_pipeline)
+ gst_object_unref(info->enc_pipeline);
+}
+
+static void gst_deinit_dec_common(struct gst_info *info) {
+ if (!info)
+ return;
+ if (info->dec_fdsem)
+ pa_fdsem_free(info->dec_fdsem);
+ if (info->dec_src)
+ gst_object_unref(info->dec_src);
+ if (info->gst_dec)
+ gst_object_unref(info->gst_dec);
+ if (info->dec_sink)
+ gst_object_unref(info->dec_sink);
+ if (info->dec_adapter)
+ g_object_unref(info->dec_adapter);
+ if (info->dec_pipeline)
+ gst_object_unref(info->dec_pipeline);
+}
+
+static bool gst_init_ldac(struct gst_info *info, pa_sample_spec *ss) {
+ GstElement *rtpldacpay;
+ GstElement *enc;
+ GstCaps *caps;
+
+ ss->format = PA_SAMPLE_S32LE;
+
+ switch (info->a2dp_codec_t.ldac_config->frequency) {
+ case LDAC_SAMPLING_FREQ_44100:
+ ss->rate = 44100u;
+ break;
+ case LDAC_SAMPLING_FREQ_48000:
+ ss->rate = 48000u;
+ break;
+ case LDAC_SAMPLING_FREQ_88200:
+ ss->rate = 88200;
+ break;
+ case LDAC_SAMPLING_FREQ_96000:
+ ss->rate = 96000;
+ break;
+ default:
+ pa_log_error("LDAC invalid frequency %d", info->a2dp_codec_t.ldac_config->frequency);
+ goto fail;
+ }
+
+ switch (info->a2dp_codec_t.ldac_config->channel_mode) {
+ case LDAC_CHANNEL_MODE_STEREO:
+ ss->channels = 2;
+ break;
+ case LDAC_CHANNEL_MODE_MONO:
+ ss->channels = 1;
+ break;
+ case LDAC_CHANNEL_MODE_DUAL:
+ ss->channels = 1;
+ break;
+ default:
+ pa_log_error("LDAC invalid channel mode %d", info->a2dp_codec_t.ldac_config->channel_mode);
+ goto fail;
+ }
+
+ enc = gst_element_factory_make("ldacenc", "ldac_enc");
+ if (!enc) {
+ pa_log_error("Could not create LDAC encoder element");
+ goto fail;
+ }
+
+ switch (info->codec_type) {
+ case LDAC_EQMID_HQ:
+ g_object_set(enc, "eqmid", 0, NULL);
+ break;
+ case LDAC_EQMID_SQ:
+ g_object_set(enc, "eqmid", 1, NULL);
+ break;
+ case LDAC_EQMID_MQ:
+ g_object_set(enc, "eqmid", 2, NULL);
+ break;
+ default:
+ goto fail;
+ }
+
+ caps = gst_caps_new_simple("audio/x-raw",
+ "format", G_TYPE_STRING, "S32LE",
+ "rate", G_TYPE_INT, (int) ss->rate,
+ "channels", G_TYPE_INT, (int) ss->channels,
+ "channel-mask", G_TYPE_INT, 0,
+ "layout", G_TYPE_STRING, "interleaved",
+ NULL);
+ g_object_set(info->enc_src, "caps", caps, NULL);
+ gst_caps_unref(caps);
+
+ rtpldacpay = gst_element_factory_make("rtpldacpay", "rtp_ldac_pay");
+ if (!rtpldacpay) {
+ pa_log_error("Could not create RTP LDAC payloader element");
+ goto fail;
+ }
+
+ gst_bin_add_many(GST_BIN(info->enc_pipeline), info->enc_src, enc, rtpldacpay, info->enc_sink, NULL);
+
+ if (!gst_element_link_many(info->enc_src, enc, rtpldacpay, info->enc_sink, NULL)) {
+ pa_log_error("Failed to link elements for LDAC encoder");
+ goto bin_remove;
+ }
+
+ if (gst_element_set_state(info->enc_pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
+ pa_log_error("Could not start LDAC encoder pipeline");
+ goto bin_remove;
+ }
+
+ info->gst_enc = enc;
+
+ return true;
+
+bin_remove:
+ gst_bin_remove_many(GST_BIN(info->enc_pipeline), info->enc_src, enc, rtpldacpay, info->enc_sink, NULL);
+fail:
+ pa_log_error("LDAC encoder initialisation failed");
+ return false;
+}
+
+static bool gst_init_aptx(struct gst_info *info, pa_sample_spec *ss) {
+ GstElement *enc, *dec;
+ GstCaps *caps;
+ const char *aptx_codec_media_type;
+
+ ss->format = PA_SAMPLE_S24LE;
+
+ if (info->codec_type == APTX_HD) {
+ switch (info->a2dp_codec_t.aptx_hd_config->aptx.frequency) {
+ case APTX_SAMPLING_FREQ_16000:
+ ss->rate = 16000u;
+ break;
+ case APTX_SAMPLING_FREQ_32000:
+ ss->rate = 32000u;
+ break;
+ case APTX_SAMPLING_FREQ_44100:
+ ss->rate = 44100u;
+ break;
+ case APTX_SAMPLING_FREQ_48000:
+ ss->rate = 48000u;
+ break;
+ default:
+ pa_log_error("aptX HD invalid frequency %d", info->a2dp_codec_t.aptx_hd_config->aptx.frequency);
+ goto fail;
+ }
+
+ switch (info->a2dp_codec_t.aptx_hd_config->aptx.channel_mode) {
+ case APTX_CHANNEL_MODE_STEREO:
+ ss->channels = 2;
+ break;
+ default:
+ pa_log_error("aptX HD invalid channel mode %d", info->a2dp_codec_t.aptx_hd_config->aptx.frequency);
+ goto fail;
+ }
+ } else {
+ switch (info->a2dp_codec_t.aptx_config->frequency) {
+ case APTX_SAMPLING_FREQ_16000:
+ ss->rate = 16000u;
+ break;
+ case APTX_SAMPLING_FREQ_32000:
+ ss->rate = 32000u;
+ break;
+ case APTX_SAMPLING_FREQ_44100:
+ ss->rate = 44100u;
+ break;
+ case APTX_SAMPLING_FREQ_48000:
+ ss->rate = 48000u;
+ break;
+ default:
+ pa_log_error("aptX invalid frequency %d", info->a2dp_codec_t.aptx_config->frequency);
+ goto fail;
+ }
+
+ switch (info->a2dp_codec_t.aptx_config->channel_mode) {
+ case APTX_CHANNEL_MODE_STEREO:
+ ss->channels = 2;
+ break;
+ default:
+ pa_log_error("aptX invalid channel mode %d", info->a2dp_codec_t.aptx_config->frequency);
+ goto fail;
+ }
+ }
+
+ enc = gst_element_factory_make("openaptxenc", "aptx_encoder");
+
+ if (enc == NULL) {
+ pa_log_error("Could not create aptX encoder element");
+ goto fail;
+ }
+
+ dec = gst_element_factory_make("openaptxdec", "aptx_decoder");
+
+ if (dec == NULL) {
+ pa_log_error("Could not create aptX decoder element");
+ goto fail;
+ }
+
+ aptx_codec_media_type = info->codec_type == APTX_HD ? "audio/aptx-hd" : "audio/aptx";
+
+ caps = gst_caps_new_simple("audio/x-raw",
+ "format", G_TYPE_STRING, "S24LE",
+ "rate", G_TYPE_INT, (int) ss->rate,
+ "channels", G_TYPE_INT, (int) ss->channels,
+ "channel-mask", G_TYPE_INT, 0,
+ "layout", G_TYPE_STRING, "interleaved",
+ NULL);
+ g_object_set(info->enc_src, "caps", caps, NULL);
+ gst_caps_unref(caps);
+
+ caps = gst_caps_new_simple(aptx_codec_media_type,
+ "rate", G_TYPE_INT, (int) ss->rate,
+ "channels", G_TYPE_INT, (int) ss->channels,
+ NULL);
+ g_object_set(info->enc_sink, "caps", caps, NULL);
+ gst_caps_unref(caps);
+
+ caps = gst_caps_new_simple(aptx_codec_media_type,
+ "rate", G_TYPE_INT, (int) ss->rate,
+ "channels", G_TYPE_INT, (int) ss->channels,
+ NULL);
+ g_object_set(info->dec_src, "caps", caps, NULL);
+ gst_caps_unref(caps);
+
+ caps = gst_caps_new_simple("audio/x-raw",
+ "format", G_TYPE_STRING, "S24LE",
+ "rate", G_TYPE_INT, (int) ss->rate,
+ "channels", G_TYPE_INT, (int) ss->channels,
+ "layout", G_TYPE_STRING, "interleaved",
+ NULL);
+ g_object_set(info->dec_sink, "caps", caps, NULL);
+ gst_caps_unref(caps);
+
+ gst_bin_add_many(GST_BIN(info->enc_pipeline), info->enc_src, enc, info->enc_sink, NULL);
+ gst_bin_add_many(GST_BIN(info->dec_pipeline), info->dec_src, dec, info->dec_sink, NULL);
+
+ if (!gst_element_link_many(info->enc_src, enc, info->enc_sink, NULL)) {
+ pa_log_error("Failed to link elements for aptX encoder");
+ goto bin_remove;
+ }
+
+ if (!gst_element_link_many(info->dec_src, dec, info->dec_sink, NULL)) {
+ pa_log_error("Failed to link elements for aptX decoder");
+ goto bin_remove;
+ }
+
+ if (gst_element_set_state(info->enc_pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
+ pa_log_error("Could not start aptX encoder pipeline");
+ goto bin_remove;
+ }
+
+ if (gst_element_set_state(info->dec_pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
+ pa_log_error("Could not start aptX decoder pipeline");
+ goto bin_remove;
+ }
+
+ info->gst_enc = enc;
+ info->gst_dec = dec;
+
+ return true;
+
+bin_remove:
+ gst_bin_remove_many(GST_BIN(info->enc_pipeline), info->enc_src, enc, info->enc_sink, NULL);
+ gst_bin_remove_many(GST_BIN(info->dec_pipeline), info->dec_src, dec, info->dec_sink, NULL);
+fail:
+ pa_log_error("aptX initialisation failed");
+ return false;
+}
+
+static bool gst_init_enc_common(struct gst_info *info, pa_sample_spec *ss) {
+ GstElement *pipeline = NULL;
+ GstElement *appsrc = NULL, *appsink = NULL;
+ GstAdapter *adapter;
+ GstAppSinkCallbacks callbacks = { 0, };
+
+ appsrc = gst_element_factory_make("appsrc", "enc_source");
+ if (!appsrc) {
+ pa_log_error("Could not create appsrc element");
+ goto fail;
+ }
+ g_object_set(appsrc, "is-live", FALSE, "format", GST_FORMAT_TIME, "stream-type", 0, "max-bytes", 0, NULL);
+
+ appsink = gst_element_factory_make("appsink", "enc_sink");
+ if (!appsink) {
+ pa_log_error("Could not create appsink element");
+ goto fail;
+ }
+ g_object_set(appsink, "sync", FALSE, "async", FALSE, "enable-last-sample", FALSE, NULL);
+
+ callbacks.eos = enc_sink_eos;
+ callbacks.new_sample = enc_sink_new_sample;
+ gst_app_sink_set_callbacks(GST_APP_SINK(appsink), &callbacks, info, NULL);
+
+ adapter = gst_adapter_new();
+ pa_assert(adapter);
+
+ pipeline = gst_pipeline_new(NULL);
+ pa_assert(pipeline);
+
+ info->enc_src = appsrc;
+ info->enc_sink = appsink;
+ info->enc_adapter = adapter;
+ info->enc_pipeline = pipeline;
+ info->enc_fdsem = pa_fdsem_new();
+
+ return true;
+
+fail:
+ gst_deinit_enc_common(info);
+
+ return false;
+}
+
+static bool gst_init_dec_common(struct gst_info *info, pa_sample_spec *ss) {
+ GstElement *pipeline = NULL;
+ GstElement *appsrc = NULL, *appsink = NULL;
+ GstAdapter *adapter;
+ GstAppSinkCallbacks callbacks = { 0, };
+
+ appsrc = gst_element_factory_make("appsrc", "dec_source");
+ if (!appsrc) {
+ pa_log_error("Could not create decoder appsrc element");
+ goto fail;
+ }
+ g_object_set(appsrc, "is-live", FALSE, "format", GST_FORMAT_TIME, "stream-type", 0, "max-bytes", 0, NULL);
+
+ appsink = gst_element_factory_make("appsink", "dec_sink");
+ if (!appsink) {
+ pa_log_error("Could not create decoder appsink element");
+ goto fail;
+ }
+ g_object_set(appsink, "sync", FALSE, "async", FALSE, "enable-last-sample", FALSE, NULL);
+
+ callbacks.eos = dec_sink_eos;
+ callbacks.new_sample = dec_sink_new_sample;
+ gst_app_sink_set_callbacks(GST_APP_SINK(appsink), &callbacks, info, NULL);
+
+ adapter = gst_adapter_new();
+ pa_assert(adapter);
+
+ pipeline = gst_pipeline_new(NULL);
+ pa_assert(pipeline);
+
+ info->dec_src = appsrc;
+ info->dec_sink = appsink;
+ info->dec_adapter = adapter;
+ info->dec_pipeline = pipeline;
+ info->dec_fdsem = pa_fdsem_new();
+
+ return true;
+
+fail:
+ gst_deinit_dec_common(info);
+
+ return false;
+}
+
+/*
+ * The idea of using buffer probes is as follows. We set a buffer probe on the
+ * encoder sink pad. In the buffer probe, we set an idle probe on the upstream
+ * source pad. In encode_buffer, we wait on the fdsem. The fdsem gets posted
+ * when either new_sample or idle probe gets called. We do this, to make the
+ * appsink behave synchronously.
+ *
+ * For buffer probes, see
+ * https://gstreamer.freedesktop.org/documentation/additional/design/probes.html?gi-language=c
+ */
+static GstPadProbeReturn gst_enc_appsink_buffer_probe(GstPad *pad, GstPadProbeInfo *probe_info, gpointer userdata)
+{
+ struct gst_info *info = (struct gst_info *)userdata;
+
+ pa_assert(probe_info->type & GST_PAD_PROBE_TYPE_IDLE);
+
+ pa_fdsem_post(info->enc_fdsem);
+
+ return GST_PAD_PROBE_REMOVE;
+}
+
+static GstPadProbeReturn gst_encoder_buffer_probe(GstPad *pad, GstPadProbeInfo *probe_info, gpointer userdata)
+{
+ struct gst_info *info = (struct gst_info *)userdata;
+ GstPad *peer_pad;
+
+ pa_assert(probe_info->type & GST_PAD_PROBE_TYPE_BUFFER);
+
+ peer_pad = gst_pad_get_peer(pad);
+ gst_pad_add_probe(peer_pad, GST_PAD_PROBE_TYPE_IDLE, gst_enc_appsink_buffer_probe, info, NULL);
+ gst_object_unref(peer_pad);
+
+ return GST_PAD_PROBE_OK;
+}
+
+static GstPadProbeReturn gst_dec_appsink_buffer_probe(GstPad *pad, GstPadProbeInfo *probe_info, gpointer userdata)
+{
+ struct gst_info *info = (struct gst_info *)userdata;
+
+ pa_assert(probe_info->type & GST_PAD_PROBE_TYPE_IDLE);
+
+ pa_fdsem_post(info->dec_fdsem);
+
+ return GST_PAD_PROBE_REMOVE;
+}
+
+static GstPadProbeReturn gst_decoder_buffer_probe(GstPad *pad, GstPadProbeInfo *probe_info, gpointer userdata)
+{
+ struct gst_info *info = (struct gst_info *)userdata;
+ GstPad *peer_pad;
+
+ pa_assert(probe_info->type & GST_PAD_PROBE_TYPE_BUFFER);
+
+ peer_pad = gst_pad_get_peer(pad);
+ gst_pad_add_probe(peer_pad, GST_PAD_PROBE_TYPE_IDLE, gst_dec_appsink_buffer_probe, info, NULL);
+ gst_object_unref(peer_pad);
+
+ return GST_PAD_PROBE_OK;
+}
+
+static bool gst_init_common(struct gst_info *info, pa_sample_spec *ss) {
+ GstPad *pad;
+
+ info->seq_num = 0;
+
+ if (!gst_init_enc_common(info, ss))
+ goto fail;
+
+ switch (info->codec_type) {
+ case AAC:
+ goto fail;
+ break;
+ case APTX:
+ case APTX_HD:
+ if (!gst_init_dec_common(info, ss))
+ goto enc_fail;
+
+ if (!gst_init_aptx(info, ss))
+ goto dec_fail;
+ break;
+ case LDAC_EQMID_HQ:
+ case LDAC_EQMID_SQ:
+ case LDAC_EQMID_MQ:
+ if (!gst_init_ldac(info, ss))
+ goto dec_fail;
+ break;
+ default:
+ goto fail;
+ }
+
+ /* See the comment on buffer probe functions */
+ if (info->gst_enc) {
+ pad = gst_element_get_static_pad(info->gst_enc, "sink");
+ gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, gst_encoder_buffer_probe, info, NULL);
+ gst_object_unref(pad);
+ }
+
+ if (info->gst_dec) {
+ pad = gst_element_get_static_pad(info->gst_dec, "sink");
+ gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, gst_decoder_buffer_probe, info, NULL);
+ gst_object_unref(pad);
+ }
+
+ pa_log_info("Gstreamer pipeline initialisation succeeded");
+
+ return true;
+
+dec_fail:
+ if (info->dec_pipeline) {
+ gst_element_set_state(info->dec_pipeline, GST_STATE_NULL);
+ gst_object_unref(info->dec_pipeline);
+ }
+enc_fail:
+ if (info->enc_pipeline) {
+ gst_element_set_state(info->enc_pipeline, GST_STATE_NULL);
+ gst_object_unref(info->enc_pipeline);
+ }
+fail:
+ pa_log_error("Gstreamer pipeline initialisation failed");
+
+ return false;
+}
+
+void *gst_codec_init(enum a2dp_codec_type codec_type, const uint8_t *config_buffer, uint8_t config_size, pa_sample_spec *ss) {
+ struct gst_info *info = NULL;
+ GError *error = NULL;
+ bool ret;
+
+ if (!gst_init_check(NULL, NULL, &error)) {
+ pa_log_error("Could not initialise GStreamer: %s", error->message);
+ g_error_free(error);
+ goto fail;
+ }
+
+ info = pa_xnew0(struct gst_info, 1);
+ pa_assert(info);
+
+ switch (codec_type) {
+ case AAC:
+ info->codec_type = AAC;
+ info->a2dp_codec_t.aac_config = (const a2dp_aac_t *) config_buffer;
+ pa_assert(config_size == sizeof(*(info->a2dp_codec_t.aac_config)));
+ break;
+ case APTX:
+ info->codec_type = APTX;
+ info->a2dp_codec_t.aptx_config = (const a2dp_aptx_t *) config_buffer;
+ pa_assert(config_size == sizeof(*(info->a2dp_codec_t.aptx_config)));
+ break;
+ case APTX_HD:
+ info->codec_type = APTX_HD;
+ info->a2dp_codec_t.aptx_hd_config = (const a2dp_aptx_hd_t *) config_buffer;
+ pa_assert(config_size == sizeof(*(info->a2dp_codec_t.aptx_hd_config)));
+ break;
+ case LDAC_EQMID_HQ:
+ case LDAC_EQMID_SQ:
+ case LDAC_EQMID_MQ:
+ info->codec_type = codec_type;
+ info->a2dp_codec_t.ldac_config = (const a2dp_ldac_t *) config_buffer;
+ pa_assert(config_size == sizeof(*(info->a2dp_codec_t.ldac_config)));
+ break;
+ default:
+ pa_log_error("Unsupported bluetooth codec");
+ goto fail;
+ }
+
+ ret = gst_init_common(info, ss);
+ if (!ret)
+ goto fail;
+
+ info->ss = ss;
+
+ pa_log_info("Rate: %d Channels: %d Format: %d", ss->rate, ss->channels, ss->format);
+
+ return info;
+
+fail:
+ if (info)
+ pa_xfree(info);
+
+ return NULL;
+}
+
+size_t gst_encode_buffer(void *codec_info, uint32_t timestamp, const uint8_t *input_buffer, size_t input_size, uint8_t *output_buffer, size_t output_size, size_t *processed) {
+ struct gst_info *info = (struct gst_info *) codec_info;
+ gsize available, encoded;
+ GstBuffer *in_buf;
+ GstMapInfo map_info;
+ GstFlowReturn ret;
+ size_t written = 0;
+
+ in_buf = gst_buffer_new_allocate(NULL, input_size, NULL);
+ pa_assert(in_buf);
+
+ pa_assert_se(gst_buffer_map(in_buf, &map_info, GST_MAP_WRITE));
+ memcpy(map_info.data, input_buffer, input_size);
+ gst_buffer_unmap(in_buf, &map_info);
+
+ ret = gst_app_src_push_buffer(GST_APP_SRC(info->enc_src), in_buf);
+ if (ret != GST_FLOW_OK) {
+ pa_log_error("failed to push buffer for encoding %d", ret);
+ goto fail;
+ }
+
+ pa_fdsem_wait(info->enc_fdsem);
+
+ available = gst_adapter_available(info->enc_adapter);
+
+ if (available) {
+ encoded = PA_MIN(available, output_size);
+
+ gst_adapter_copy(info->enc_adapter, output_buffer, 0, encoded);
+ gst_adapter_flush(info->enc_adapter, encoded);
+
+ written += encoded;
+ } else
+ pa_log_debug("No encoded data available in adapter");
+
+ *processed = input_size;
+
+ return written;
+
+fail:
+ *processed = 0;
+
+ return written;
+}
+
+size_t gst_decode_buffer(void *codec_info, const uint8_t *input_buffer, size_t input_size, uint8_t *output_buffer, size_t output_size, size_t *processed) {
+ struct gst_info *info = (struct gst_info *) codec_info;
+ gsize available, decoded;
+ GstBuffer *in_buf;
+ GstMapInfo map_info;
+ GstFlowReturn ret;
+ size_t written = 0;
+
+ in_buf = gst_buffer_new_allocate(NULL, input_size, NULL);
+ pa_assert(in_buf);
+
+ pa_assert_se(gst_buffer_map(in_buf, &map_info, GST_MAP_WRITE));
+ memcpy(map_info.data, input_buffer, input_size);
+ gst_buffer_unmap(in_buf, &map_info);
+
+ ret = gst_app_src_push_buffer(GST_APP_SRC(info->dec_src), in_buf);
+ if (ret != GST_FLOW_OK) {
+ pa_log_error("failed to push buffer for decoding %d", ret);
+ goto fail;
+ }
+
+ pa_fdsem_wait(info->dec_fdsem);
+
+ available = gst_adapter_available(info->dec_adapter);
+
+ if (available) {
+ decoded = PA_MIN(available, output_size);
+
+ gst_adapter_copy(info->dec_adapter, output_buffer, 0, decoded);
+ gst_adapter_flush(info->dec_adapter, decoded);
+
+ written += decoded;
+ } else
+ pa_log_debug("No decoded data available in adapter");
+
+ *processed = input_size;
+
+ return written;
+
+fail:
+ *processed = 0;
+
+ return written;
+}
+
+void gst_codec_deinit(void *codec_info) {
+ struct gst_info *info = (struct gst_info *) codec_info;
+
+ if (info->enc_fdsem)
+ pa_fdsem_free(info->enc_fdsem);
+
+ if (info->dec_fdsem)
+ pa_fdsem_free(info->dec_fdsem);
+
+ if (info->enc_pipeline) {
+ gst_element_set_state(info->enc_pipeline, GST_STATE_NULL);
+ gst_object_unref(info->enc_pipeline);
+ }
+
+ if (info->dec_pipeline) {
+ gst_element_set_state(info->dec_pipeline, GST_STATE_NULL);
+ gst_object_unref(info->dec_pipeline);
+ }
+
+ if (info->enc_adapter)
+ g_object_unref(info->enc_adapter);
+
+ if (info->dec_adapter)
+ g_object_unref(info->dec_adapter);
+
+ pa_xfree(info);
+}
projects / platform / upstream / pulseaudio.git / commitdiff