diff options
author | 2019-01-16 19:44:28 -0500 | |
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committer | 2019-02-11 15:02:37 +0000 | |
commit | 2a950adb3229056313c8a8267dec67d2ceadf5ef (patch) | |
tree | bafe6ed8a99ea76e4e6b98d7fd223645eebb9186 /app/fileplayer.cpp | |
parent | 546f5137a2391984f96d65b318f74f1a918ba131 (diff) |
Replace QtMultimedia usage with 4A + gstreamer
Replace QtMultimedia usage for ringtone playing with a gstreamer
pipeline that uses the provided 4A role ALSA device for output.
For now, a "phone" role is assumed to be available, but it does
not exist in the current set of 4A policy and HALs, and needs to
be added. Testing was done by making the required role changes
locally and using some debug QML tweaks to allow triggering the
ringtone manually.
Bug-AGL: SPEC-1596
Change-Id: I55c2229de1bc5470ee818e5be382b64664fa2d29
Signed-off-by: Scott Murray <scott.murray@konsulko.com>
(cherry picked from commit 90b83db9213308a4451d9c86b380aba6d13313b5)
Diffstat (limited to 'app/fileplayer.cpp')
-rw-r--r-- | app/fileplayer.cpp | 290 |
1 files changed, 290 insertions, 0 deletions
diff --git a/app/fileplayer.cpp b/app/fileplayer.cpp new file mode 100644 index 0000000..e26a3e6 --- /dev/null +++ b/app/fileplayer.cpp @@ -0,0 +1,290 @@ +/* + * Copyright (C) 2018 Konsulko Group + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include "fileplayer.h" +#include <string> +#include <cstring> +#include <iostream> +#include <mutex> +#include <condition_variable> +#include <json-c/json.h> + +#undef DEBUG + +struct set_role_data +{ + bool state; + std::string output; + std::condition_variable cv; +}; + +static void on_hangup(void *closure, struct afb_wsj1 *wsj) +{ +} + +static void on_call(void *closure, const char *api, const char *verb, struct afb_wsj1_msg *msg) +{ +} + +static void on_event(void* closure, const char* event, struct afb_wsj1_msg *msg) +{ +} + +static void on_reply(void *closure, struct afb_wsj1_msg *msg) +{ + struct set_role_data *data = (struct set_role_data*) closure; + struct json_object* reply; + + if(!(data && data->state)) + goto reply_done; + + // We opened the role, return the output + reply = afb_wsj1_msg_object_j(msg); + if(reply) { +#ifdef DEBUG + std::cerr << __FUNCTION__ << ": reply = " << \ + json_object_to_json_string_ext(reply, JSON_C_TO_STRING_SPACED | JSON_C_TO_STRING_PRETTY) << \ + std::endl; +#endif + struct json_object* response; + int rc = json_object_object_get_ex(reply, "response", &response); + if(rc) { + struct json_object* val; + rc = json_object_object_get_ex(response, "device_uri", &val); + if (rc && json_object_get_string_len(val)) { + const char* jres_pcm = json_object_get_string(val); + data->output = jres_pcm; +#ifdef DEBUG + std::cerr << __FUNCTION__ << ": output = " << jres_pcm << std::endl; +#endif + } + } + } +reply_done: + // Signal reply is done + data->cv.notify_one(); +} + +static void *afb_loop_thread(struct sd_event* loop) +{ + for(;;) + sd_event_run(loop, 30000000); +} + +static void *gst_loop_thread(GMainLoop *loop) +{ + if(loop) + g_main_loop_run(loop); +} + +FilePlayer::FilePlayer(const int port, const std::string &token, const std::string &path, const std::string &role) : + m_path(path), + m_role(role) +{ + std::string uri; + + if(sd_event_new(&m_afb_loop) < 0) { + std::cerr << __FUNCTION__ << ": Failed to create event loop" << std::endl; + return; + } + + // Initialize interface for websocket + m_itf.on_hangup = on_hangup; + m_itf.on_call = on_call; + m_itf.on_event = on_event; + + uri = "ws://localhost:" + std::to_string(port) + "/api?token=" + token; +#ifdef DEBUG + std::cerr << "Using URI: " << uri << std::endl; +#endif + m_ws = afb_ws_client_connect_wsj1(m_afb_loop, uri.c_str(), &m_itf, NULL); + if(m_ws) { + m_afb_thread = std::thread(afb_loop_thread, m_afb_loop); + } else { + std::cerr << __FUNCTION__ << ": Failed to create websocket connection" << std::endl; + goto error; + } + + // Initialize GStreamer + gst_init(NULL, NULL); + + // Create elements we need + m_playbin = gst_element_factory_make("playbin", "play"); + m_alsa_sink = gst_element_factory_make("alsasink", NULL); + if(!(m_playbin && m_alsa_sink)) + goto error; + + // Set up bus callback + m_bus = gst_pipeline_get_bus(GST_PIPELINE(m_playbin)); + if(!m_bus) + goto error; + m_gst_loop = g_main_loop_new(NULL, FALSE); + if(!m_gst_loop) + goto error; + gst_bus_add_watch(m_bus, gstreamer_bus_callback, this); + + // Start thread to run glib main loop for gstreamer bus + m_gst_thread = std::thread(gst_loop_thread, m_gst_loop); + + m_valid = true; + return; +error: + gst_object_unref(m_playbin); + m_playbin = nullptr; + gst_object_unref(m_alsa_sink); + m_alsa_sink = nullptr; + gst_object_unref(m_bus); + m_bus = nullptr; + + if(m_gst_loop) { + g_main_loop_quit(m_gst_loop); + m_gst_loop = nullptr; + } + if(m_afb_loop) { + sd_event_unref(m_afb_loop); + m_afb_loop = nullptr; + } + return; +} + +FilePlayer::~FilePlayer(void) +{ + gst_element_set_state(m_playbin, GST_STATE_NULL); + gst_object_unref(m_playbin); + gst_object_unref(m_alsa_sink); + gst_object_unref(m_bus); + sd_event_unref(m_afb_loop); +} + +void FilePlayer::play(bool loop) +{ + std::string output; + + if(!m_valid || m_playing) + return; + + if(set_role_state(true, &output) != 0) + return; + + if(output.empty()) + return; + + m_playing = true; + m_looping = loop; + + g_object_set(m_alsa_sink, "device", output.c_str(), NULL); + g_object_set(m_playbin, "audio-sink", m_alsa_sink, NULL); + std::string uri = "file://" + m_path; + g_object_set(m_playbin, "uri", uri.c_str(), NULL); + + // Start playback + gst_element_set_state(m_playbin, GST_STATE_PLAYING); + + return; +} + +void FilePlayer::stop(void) +{ + if(!(m_valid && m_playing)) + return; + + // Stop playback + gst_element_set_state(m_playbin, GST_STATE_PAUSED); +} + +gboolean FilePlayer::gstreamer_bus_callback(GstBus *bus, GstMessage *msg, gpointer data) +{ + if(!data) + return TRUE; + + return static_cast<FilePlayer*>(data)->bus_callback(bus, msg); +} + +int FilePlayer::set_role_state(bool state, std::string *output) +{ + if(!m_valid) + return -1; + + set_role_data data; + data.state = state; + json_object *jsonData = json_object_new_object(); + json_object_object_add(jsonData, "action", json_object_new_string(state ? "open" : "close")); + int rc = afb_wsj1_call_j(m_ws, "ahl-4a", m_role.c_str(), jsonData, on_reply, (void*) &data); + if(rc >= 0) { + // Wait for response + std::mutex m; + std::unique_lock<std::mutex> lk(m); + data.cv.wait(lk); + if(state && output) + *output = data.output; + } else { + std::cerr << __FUNCTION__ << ": Failed to call ahl-4a/" << m_role.c_str() << std::endl; + } + return rc; +} + +gboolean FilePlayer::bus_callback(GstBus *bus, GstMessage *msg) +{ + switch (GST_MESSAGE_TYPE(msg)) { + case GST_MESSAGE_EOS: +#ifdef DEBUG + std::cerr << __FUNCTION__ << ": GST_MESSAGE_EOS" << std::endl; +#endif + if(m_looping) { + // restart playback if at end + if (!gst_element_seek(m_playbin, + 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, + GST_SEEK_TYPE_SET, 0, + GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE)) { + std::cerr << "Seek failed!" << std::endl; + } + } else { + // Move stream to paused state since we're done + gst_element_set_state(m_playbin, GST_STATE_PAUSED); + } + break; + case GST_MESSAGE_STATE_CHANGED: { + if((GstElement*) GST_MESSAGE_SRC(msg) != m_playbin) + break; + + GstState old_state, new_state; + gst_message_parse_state_changed(msg, &old_state, &new_state, NULL); +#ifdef DEBUG + std::cerr << __FUNCTION__ << ": GST_MESSAGE_STATE_CHANGE: " << (int) old_state << " to " << (int) new_state << std::endl; +#endif + if(old_state == GST_STATE_PLAYING && new_state == GST_STATE_PAUSED) { + set_role_state(false); + + // Seek back to beginning so any subsequent play starts there + if (!gst_element_seek(m_playbin, + 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, + GST_SEEK_TYPE_SET, 0, + GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE)) { + std::cerr << "Seek failed!" << std::endl; + } + + m_playing = false; + } else if(old_state == GST_STATE_READY && new_state == GST_STATE_NULL) { + // clean up + g_main_loop_quit(m_gst_loop); + } + break; + } + default: + break; + } + return TRUE; +} |